summaryrefslogtreecommitdiff
path: root/sound
diff options
context:
space:
mode:
authorLinus Torvalds <torvalds@linux-foundation.org>2020-04-03 01:50:04 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2020-04-03 01:50:04 +0300
commit848960e576dafc8ed54c691b2f70b92e1fdea9ba (patch)
tree27ea80003da03b81f0b188d3712f0194745126d9 /sound
parentbc3b3f4bfbded031a11c4284106adddbfacd05bb (diff)
parent5c6cd7021a05a02fcf37f360592d7c18d4d807fb (diff)
downloadlinux-848960e576dafc8ed54c691b2f70b92e1fdea9ba.tar.xz
Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This became again a busy development cycle. There are few ALSA core updates (merely API cleanups and sparse fixes), with the majority of other changes are found in ASoC scene. Here are some highlights: ALSA core: - More helper macros for sparse warning fixes (e.g. bitwise types) - Slight optimization of PCM OSS locks - Make common handling for PCM / compress buffers (for SOF) ASoC: - Lots of code refactoring and modernization for (still ongoing) componentization works - Conversion of SND_SOC_ALL_CODECS to use imply - Continued refactoring and fixing of the Intel SOF/SST support, including the initial (but still incomplete) SoundWire support - SoundWire and more advanced clocking support for Realtek RT5682 - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and TLV320ADCX140 HD-audio: - Optimizations in HDMI jack handling - A few new quirks and fixups for Realtek codecs USB-audio: - Delayed registration support - New quirks for Motu, Kingston, Presonus" * tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits) ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h" ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - a fake key event is triggered by running shutup ALSA: hda: default enable CA0132 DSP support ASoC: amd: acp3x-pcm-dma: clean up two indentation issues ASoC: tlv320adcx140: Remove undocumented property ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: boards: add sof_sdw machine driver ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms ASoC: rt5682: move DAI clock registry to I2S mode ASoC: pxa: magician: convert to use i2c_new_client_device() ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers ...
Diffstat (limited to 'sound')
-rw-r--r--sound/arm/pxa2xx-pcm-lib.c8
-rw-r--r--sound/core/compress_offload.c42
-rw-r--r--sound/core/device.c21
-rw-r--r--sound/core/info.c2
-rw-r--r--sound/core/oss/pcm_oss.c23
-rw-r--r--sound/core/oss/pcm_plugin.c120
-rw-r--r--sound/core/oss/rate.c2
-rw-r--r--sound/core/pcm.c2
-rw-r--r--sound/core/pcm_dmaengine.c8
-rw-r--r--sound/core/pcm_misc.c35
-rw-r--r--sound/core/pcm_native.c47
-rw-r--r--sound/drivers/aloop.c6
-rw-r--r--sound/drivers/dummy.c6
-rw-r--r--sound/firewire/bebob/bebob.c2
-rw-r--r--sound/firewire/digi00x/digi00x.c2
-rw-r--r--sound/firewire/fireface/ff.c2
-rw-r--r--sound/firewire/fireworks/fireworks.c2
-rw-r--r--sound/firewire/tascam/tascam-hwdep.c2
-rw-r--r--sound/firewire/tascam/tascam.c2
-rw-r--r--sound/hda/hdac_device.c2
-rw-r--r--sound/isa/sb/emu8000_pcm.c4
-rw-r--r--sound/pci/ali5451/ali5451.c6
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c4
-rw-r--r--sound/pci/hda/Kconfig1
-rw-r--r--sound/pci/hda/hda_codec.c2
-rw-r--r--sound/pci/hda/hda_controller.c2
-rw-r--r--sound/pci/hda/patch_ca0132.c3
-rw-r--r--sound/pci/hda/patch_hdmi.c313
-rw-r--r--sound/pci/hda/patch_realtek.c231
-rw-r--r--sound/pci/korg1212/korg1212.c2
-rw-r--r--sound/pci/rme9652/hdsp.c3
-rw-r--r--sound/pci/via82xx.c6
-rw-r--r--sound/pci/via82xx_modem.c6
-rw-r--r--sound/ppc/keywest.c9
-rw-r--r--sound/soc/amd/Kconfig10
-rw-r--r--sound/soc/amd/Makefile2
-rw-r--r--sound/soc/amd/acp-da7219-max98357a.c2
-rw-r--r--sound/soc/amd/acp-rt5645.c4
-rw-r--r--sound/soc/amd/acp3x-rt5682-max9836.c376
-rw-r--r--sound/soc/amd/raven/acp3x-i2s.c44
-rw-r--r--sound/soc/amd/raven/acp3x-pcm-dma.c6
-rw-r--r--sound/soc/amd/raven/pci-acp3x.c7
-rw-r--r--sound/soc/atmel/atmel-pcm-dma.c4
-rw-r--r--sound/soc/atmel/atmel-pcm-pdc.c2
-rw-r--r--sound/soc/atmel/atmel_wm8904.c2
-rw-r--r--sound/soc/atmel/mchp-i2s-mcc.c8
-rw-r--r--sound/soc/atmel/mikroe-proto.c2
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c2
-rw-r--r--sound/soc/atmel/sam9x5_wm8731.c2
-rw-r--r--sound/soc/au1x/db1200.c2
-rw-r--r--sound/soc/au1x/dbdma2.c2
-rw-r--r--sound/soc/au1x/dma.c2
-rw-r--r--sound/soc/au1x/psc-ac97.c2
-rw-r--r--sound/soc/bcm/Kconfig9
-rw-r--r--sound/soc/bcm/Makefile4
-rw-r--r--sound/soc/bcm/bcm63xx-i2s-whistler.c317
-rw-r--r--sound/soc/bcm/bcm63xx-i2s.h90
-rw-r--r--sound/soc/bcm/bcm63xx-pcm-whistler.c485
-rw-r--r--sound/soc/bcm/cygnus-pcm.c22
-rw-r--r--sound/soc/cirrus/edb93xx.c4
-rw-r--r--sound/soc/cirrus/snappercl15.c4
-rw-r--r--sound/soc/codecs/Kconfig627
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/cros_ec_codec.c25
-rw-r--r--sound/soc/codecs/cs4271.c4
-rw-r--r--sound/soc/codecs/cs47l15.c4
-rw-r--r--sound/soc/codecs/cs47l24.c6
-rw-r--r--sound/soc/codecs/cs47l35.c6
-rw-r--r--sound/soc/codecs/cs47l85.c6
-rw-r--r--sound/soc/codecs/cs47l90.c6
-rw-r--r--sound/soc/codecs/cs47l92.c4
-rw-r--r--sound/soc/codecs/hdac_hdmi.c6
-rw-r--r--sound/soc/codecs/max98357a.c37
-rw-r--r--sound/soc/codecs/mt6660.c81
-rw-r--r--sound/soc/codecs/rk3328_codec.c31
-rw-r--r--sound/soc/codecs/rl6231.c1
-rw-r--r--sound/soc/codecs/rl6231.h2
-rw-r--r--sound/soc/codecs/rt1015.c10
-rw-r--r--sound/soc/codecs/rt1308-sdw.c38
-rw-r--r--sound/soc/codecs/rt1308-sdw.h2
-rw-r--r--sound/soc/codecs/rt5659.c2
-rw-r--r--sound/soc/codecs/rt5682-sdw.c333
-rw-r--r--sound/soc/codecs/rt5682-sdw.h20
-rw-r--r--sound/soc/codecs/rt5682.c1298
-rw-r--r--sound/soc/codecs/rt5682.h100
-rw-r--r--sound/soc/codecs/tas2562.c121
-rw-r--r--sound/soc/codecs/tas2562.h12
-rw-r--r--sound/soc/codecs/tlv320adcx140.c920
-rw-r--r--sound/soc/codecs/tlv320adcx140.h131
-rw-r--r--sound/soc/codecs/wcd9335.c18
-rw-r--r--sound/soc/codecs/wcd9335.h7
-rw-r--r--sound/soc/codecs/wcd934x.c37
-rw-r--r--sound/soc/codecs/wm0010.c2
-rw-r--r--sound/soc/codecs/wm5110.c6
-rw-r--r--sound/soc/codecs/wm8974.c8
-rw-r--r--sound/soc/codecs/wm_adsp.c14
-rw-r--r--sound/soc/codecs/wsa881x.c46
-rw-r--r--sound/soc/dwc/dwc-i2s.c8
-rw-r--r--sound/soc/dwc/dwc-pcm.c2
-rw-r--r--sound/soc/fsl/eukrea-tlv320.c4
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c10
-rw-r--r--sound/soc/fsl/fsl_asrc_dma.c10
-rw-r--r--sound/soc/fsl/fsl_spdif.c10
-rw-r--r--sound/soc/fsl/fsl_ssi.c8
-rw-r--r--sound/soc/fsl/imx-audmix.c8
-rw-r--r--sound/soc/fsl/imx-mc13783.c4
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c2
-rw-r--r--sound/soc/fsl/mpc5200_dma.c10
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c2
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c4
-rw-r--r--sound/soc/fsl/mx27vis-aic32x4.c4
-rw-r--r--sound/soc/fsl/p1022_ds.c4
-rw-r--r--sound/soc/fsl/p1022_rdk.c4
-rw-r--r--sound/soc/fsl/wm1133-ev1.c6
-rw-r--r--sound/soc/generic/simple-card-utils.c60
-rw-r--r--sound/soc/img/img-i2s-in.c2
-rw-r--r--sound/soc/img/img-i2s-out.c2
-rw-r--r--sound/soc/intel/atom/sst-atom-controls.c2
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c6
-rw-r--r--sound/soc/intel/atom/sst/sst_pci.c2
-rw-r--r--sound/soc/intel/boards/Kconfig57
-rw-r--r--sound/soc/intel/boards/Makefile12
-rw-r--r--sound/soc/intel/boards/bdw-rt5650.c15
-rw-r--r--sound/soc/intel/boards/bdw-rt5677.c15
-rw-r--r--sound/soc/intel/boards/broadwell.c13
-rw-r--r--sound/soc/intel/boards/bxt_da7219_max98357a.c10
-rw-r--r--sound/soc/intel/boards/bxt_rt298.c10
-rw-r--r--sound/soc/intel/boards/byt-max98090.c2
-rw-r--r--sound/soc/intel/boards/byt-rt5640.c4
-rw-r--r--sound/soc/intel/boards/bytcht_cx2072x.c10
-rw-r--r--sound/soc/intel/boards/bytcht_da7213.c10
-rw-r--r--sound/soc/intel/boards/bytcht_es8316.c8
-rw-r--r--sound/soc/intel/boards/bytcht_nocodec.c4
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c8
-rw-r--r--sound/soc/intel/boards/bytcr_rt5651.c8
-rw-r--r--sound/soc/intel/boards/cht_bsw_max98090_ti.c8
-rw-r--r--sound/soc/intel/boards/cht_bsw_nau8824.c6
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5645.c16
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5672.c8
-rw-r--r--sound/soc/intel/boards/cml_rt1011_rt5682.c13
-rw-r--r--sound/soc/intel/boards/glk_rt5682_max98357a.c13
-rw-r--r--sound/soc/intel/boards/haswell.c4
-rw-r--r--sound/soc/intel/boards/kbl_da7219_max98357a.c8
-rw-r--r--sound/soc/intel/boards/kbl_da7219_max98927.c14
-rw-r--r--sound/soc/intel/boards/kbl_rt5660.c6
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_max98927.c12
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c12
-rw-r--r--sound/soc/intel/boards/skl_hda_dsp_common.h4
-rw-r--r--sound/soc/intel/boards/skl_hda_dsp_generic.c27
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_max98357a.c14
-rw-r--r--sound/soc/intel/boards/skl_nau88l25_ssm4567.c19
-rw-r--r--sound/soc/intel/boards/skl_rt286.c8
-rw-r--r--sound/soc/intel/boards/sof_da7219_max98373.c83
-rw-r--r--sound/soc/intel/boards/sof_maxim_common.c80
-rw-r--r--sound/soc/intel/boards/sof_maxim_common.h24
-rw-r--r--sound/soc/intel/boards/sof_pcm512x.c448
-rw-r--r--sound/soc/intel/boards/sof_rt5682.c137
-rw-r--r--sound/soc/intel/boards/sof_sdw.c962
-rw-r--r--sound/soc/intel/boards/sof_sdw_common.h114
-rw-r--r--sound/soc/intel/boards/sof_sdw_dmic.c42
-rw-r--r--sound/soc/intel/boards/sof_sdw_hdmi.c97
-rw-r--r--sound/soc/intel/boards/sof_sdw_rt1308.c151
-rw-r--r--sound/soc/intel/boards/sof_sdw_rt5682.c126
-rw-r--r--sound/soc/intel/boards/sof_sdw_rt700.c125
-rw-r--r--sound/soc/intel/boards/sof_sdw_rt711.c156
-rw-r--r--sound/soc/intel/boards/sof_sdw_rt715.c42
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-bxt-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-cht-match.c7
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-cml-match.c111
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-icl-match.c103
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-jsl-match.c34
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-tgl-match.c92
-rw-r--r--sound/soc/intel/haswell/sst-haswell-pcm.c26
-rw-r--r--sound/soc/intel/skylake/bxt-sst.c3
-rw-r--r--sound/soc/intel/skylake/cnl-sst.c35
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c3
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c20
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.h2
-rw-r--r--sound/soc/intel/skylake/skl.c33
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c78
-rw-r--r--sound/soc/kirkwood/armada-370-db.c2
-rw-r--r--sound/soc/kirkwood/kirkwood-dma.c2
-rw-r--r--sound/soc/mediatek/common/mtk-afe-fe-dai.c10
-rw-r--r--sound/soc/mediatek/common/mtk-afe-platform-driver.c2
-rw-r--r--sound/soc/mediatek/mt2701/mt2701-afe-pcm.c2
-rw-r--r--sound/soc/mediatek/mt2701/mt2701-cs42448.c4
-rw-r--r--sound/soc/mediatek/mt2701/mt2701-wm8960.c4
-rw-r--r--sound/soc/mediatek/mt6797/mt6797-afe-pcm.c2
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-afe-pcm.c2
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-max98090.c4
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c4
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c6
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650.c23
-rw-r--r--sound/soc/mediatek/mt8183/mt8183-afe-pcm.c2
-rw-r--r--sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c117
-rw-r--r--sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c2
-rw-r--r--sound/soc/meson/Kconfig41
-rw-r--r--sound/soc/meson/Makefile19
-rw-r--r--sound/soc/meson/aiu-acodec-ctrl.c203
-rw-r--r--sound/soc/meson/aiu-codec-ctrl.c151
-rw-r--r--sound/soc/meson/aiu-encoder-i2s.c365
-rw-r--r--sound/soc/meson/aiu-encoder-spdif.c209
-rw-r--r--sound/soc/meson/aiu-fifo-i2s.c153
-rw-r--r--sound/soc/meson/aiu-fifo-spdif.c186
-rw-r--r--sound/soc/meson/aiu-fifo.c223
-rw-r--r--sound/soc/meson/aiu-fifo.h50
-rw-r--r--sound/soc/meson/aiu.c388
-rw-r--r--sound/soc/meson/aiu.h89
-rw-r--r--sound/soc/meson/axg-card.c414
-rw-r--r--sound/soc/meson/axg-fifo.c2
-rw-r--r--sound/soc/meson/g12a-toacodec.c252
-rw-r--r--sound/soc/meson/g12a-tohdmitx.c219
-rw-r--r--sound/soc/meson/gx-card.c141
-rw-r--r--sound/soc/meson/meson-card-utils.c385
-rw-r--r--sound/soc/meson/meson-card.h55
-rw-r--r--sound/soc/meson/meson-codec-glue.c149
-rw-r--r--sound/soc/meson/meson-codec-glue.h32
-rw-r--r--sound/soc/meson/t9015.c333
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c4
-rw-r--r--sound/soc/pxa/Kconfig22
-rw-r--r--sound/soc/pxa/brownstone.c4
-rw-r--r--sound/soc/pxa/corgi.c4
-rw-r--r--sound/soc/pxa/hx4700.c4
-rw-r--r--sound/soc/pxa/imote2.c4
-rw-r--r--sound/soc/pxa/magician.c14
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c4
-rw-r--r--sound/soc/pxa/mmp-pcm.c2
-rw-r--r--sound/soc/pxa/mmp-sspa.c2
-rw-r--r--sound/soc/pxa/poodle.c4
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c2
-rw-r--r--sound/soc/pxa/spitz.c4
-rw-r--r--sound/soc/pxa/ttc-dkb.c2
-rw-r--r--sound/soc/pxa/z2.c4
-rw-r--r--sound/soc/pxa/zylonite.c6
-rw-r--r--sound/soc/qcom/Kconfig2
-rw-r--r--sound/soc/qcom/apq8016_sbc.c9
-rw-r--r--sound/soc/qcom/apq8096.c6
-rw-r--r--sound/soc/qcom/lpass-platform.c4
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c173
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.c243
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.h51
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c21
-rw-r--r--sound/soc/qcom/sdm845.c105
-rw-r--r--sound/soc/qcom/storm.c2
-rw-r--r--sound/soc/rockchip/rk3288_hdmi_analog.c4
-rw-r--r--sound/soc/rockchip/rk3399_gru_sound.c16
-rw-r--r--sound/soc/rockchip/rockchip_max98090.c6
-rw-r--r--sound/soc/rockchip/rockchip_rt5645.c6
-rw-r--r--sound/soc/samsung/Kconfig4
-rw-r--r--sound/soc/samsung/arndale.c10
-rw-r--r--sound/soc/samsung/bells.c16
-rw-r--r--sound/soc/samsung/h1940_uda1380.c2
-rw-r--r--sound/soc/samsung/i2s.c2
-rw-r--r--sound/soc/samsung/jive_wm8750.c4
-rw-r--r--sound/soc/samsung/littlemill.c16
-rw-r--r--sound/soc/samsung/lowland.c6
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c10
-rw-r--r--sound/soc/samsung/odroid.c6
-rw-r--r--sound/soc/samsung/pcm.c4
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c2
-rw-r--r--sound/soc/samsung/s3c-i2s-v2.c2
-rw-r--r--sound/soc/samsung/s3c24xx_simtec.c4
-rw-r--r--sound/soc/samsung/s3c24xx_uda134x.c6
-rw-r--r--sound/soc/samsung/smartq_wm8987.c4
-rw-r--r--sound/soc/samsung/smdk_spdif.c2
-rw-r--r--sound/soc/samsung/smdk_wm8580.c2
-rw-r--r--sound/soc/samsung/smdk_wm8994.c4
-rw-r--r--sound/soc/samsung/smdk_wm8994pcm.c6
-rw-r--r--sound/soc/samsung/snow.c8
-rw-r--r--sound/soc/samsung/spdif.c8
-rw-r--r--sound/soc/samsung/speyside.c10
-rw-r--r--sound/soc/samsung/tm2_wm5110.c19
-rw-r--r--sound/soc/samsung/tobermory.c10
-rw-r--r--sound/soc/sh/dma-sh7760.c16
-rw-r--r--sound/soc/sh/fsi.c5
-rw-r--r--sound/soc/sh/migor.c6
-rw-r--r--sound/soc/sh/rcar/core.c2
-rw-r--r--sound/soc/soc-compress.c5
-rw-r--r--sound/soc/soc-core.c290
-rw-r--r--sound/soc/soc-dai.c18
-rw-r--r--sound/soc/soc-dapm.c220
-rw-r--r--sound/soc/soc-generic-dmaengine-pcm.c26
-rw-r--r--sound/soc/soc-pcm.c1619
-rw-r--r--sound/soc/soc-topology.c24
-rw-r--r--sound/soc/sof/Kconfig9
-rw-r--r--sound/soc/sof/Makefile1
-rw-r--r--sound/soc/sof/compress.c146
-rw-r--r--sound/soc/sof/compress.h31
-rw-r--r--sound/soc/sof/core.c10
-rw-r--r--sound/soc/sof/debug.c226
-rw-r--r--sound/soc/sof/imx/imx8.c57
-rw-r--r--sound/soc/sof/intel/Kconfig20
-rw-r--r--sound/soc/sof/intel/Makefile1
-rw-r--r--sound/soc/sof/intel/apl.c9
-rw-r--r--sound/soc/sof/intel/cnl.c51
-rw-r--r--sound/soc/sof/intel/hda-codec.c11
-rw-r--r--sound/soc/sof/intel/hda-compress.c114
-rw-r--r--sound/soc/sof/intel/hda-ctrl.c40
-rw-r--r--sound/soc/sof/intel/hda-dai.c130
-rw-r--r--sound/soc/sof/intel/hda-dsp.c331
-rw-r--r--sound/soc/sof/intel/hda-ipc.c24
-rw-r--r--sound/soc/sof/intel/hda-loader.c40
-rw-r--r--sound/soc/sof/intel/hda-pcm.c8
-rw-r--r--sound/soc/sof/intel/hda-stream.c27
-rw-r--r--sound/soc/sof/intel/hda.c433
-rw-r--r--sound/soc/sof/intel/hda.h120
-rw-r--r--sound/soc/sof/ipc.c41
-rw-r--r--sound/soc/sof/loader.c6
-rw-r--r--sound/soc/sof/ops.h59
-rw-r--r--sound/soc/sof/pcm.c19
-rw-r--r--sound/soc/sof/pm.c176
-rw-r--r--sound/soc/sof/probe.c290
-rw-r--r--sound/soc/sof/probe.h85
-rw-r--r--sound/soc/sof/sof-audio.c59
-rw-r--r--sound/soc/sof/sof-audio.h6
-rw-r--r--sound/soc/sof/sof-of-dev.c10
-rw-r--r--sound/soc/sof/sof-priv.h71
-rw-r--r--sound/soc/sof/topology.c25
-rw-r--r--sound/soc/sprd/Kconfig2
-rw-r--r--sound/soc/sprd/sprd-mcdt.h2
-rw-r--r--sound/soc/sprd/sprd-pcm-compress.c4
-rw-r--r--sound/soc/sprd/sprd-pcm-dma.c2
-rw-r--r--sound/soc/stm/stm32_adfsdm.c12
-rw-r--r--sound/soc/stm/stm32_i2s.c75
-rw-r--r--sound/soc/stm/stm32_sai.c26
-rw-r--r--sound/soc/stm/stm32_sai_sub.c13
-rw-r--r--sound/soc/stm/stm32_spdifrx.c89
-rw-r--r--sound/soc/sunxi/sun4i-spdif.c2
-rw-r--r--sound/soc/sunxi/sun8i-codec.c3
-rw-r--r--sound/soc/tegra/tegra_alc5632.c2
-rw-r--r--sound/soc/tegra/tegra_max98090.c2
-rw-r--r--sound/soc/tegra/tegra_rt5640.c2
-rw-r--r--sound/soc/tegra/tegra_rt5677.c2
-rw-r--r--sound/soc/tegra/tegra_sgtl5000.c2
-rw-r--r--sound/soc/tegra/tegra_wm8753.c2
-rw-r--r--sound/soc/tegra/tegra_wm8903.c24
-rw-r--r--sound/soc/tegra/trimslice.c2
-rw-r--r--sound/soc/ti/Kconfig8
-rw-r--r--sound/soc/ti/Makefile2
-rw-r--r--sound/soc/ti/ams-delta.c4
-rw-r--r--sound/soc/ti/davinci-evm.c4
-rw-r--r--sound/soc/ti/davinci-mcasp.c13
-rw-r--r--sound/soc/ti/davinci-vcif.c4
-rw-r--r--sound/soc/ti/n810.c2
-rw-r--r--sound/soc/ti/omap-abe-twl6040.c6
-rw-r--r--sound/soc/ti/omap-mcbsp-st.c2
-rw-r--r--sound/soc/ti/omap-mcbsp.c4
-rw-r--r--sound/soc/ti/omap-mcpdm.c2
-rw-r--r--sound/soc/ti/omap3pandora.c4
-rw-r--r--sound/soc/ti/osk5912.c2
-rw-r--r--sound/soc/ti/rx51.c2
-rw-r--r--sound/soc/ti/udma-pcm.c43
-rw-r--r--sound/soc/ti/udma-pcm.h18
-rw-r--r--sound/soc/txx9/txx9aclc.c2
-rw-r--r--sound/soc/uniphier/aio-compress.c22
-rw-r--r--sound/soc/uniphier/aio-dma.c6
-rw-r--r--sound/soc/ux500/mop500_ab8500.c6
-rw-r--r--sound/soc/ux500/ux500_pcm.c8
-rw-r--r--sound/soc/xtensa/xtfpga-i2s.c2
-rw-r--r--sound/soc/zte/zx-spdif.c1
-rw-r--r--sound/soc/zte/zx-tdm.c3
-rw-r--r--sound/usb/Makefile1
-rw-r--r--sound/usb/card.c38
-rw-r--r--sound/usb/clock.c59
-rw-r--r--sound/usb/format.c37
-rw-r--r--sound/usb/midi.c31
-rw-r--r--sound/usb/mixer.c33
-rw-r--r--sound/usb/mixer_quirks.c5
-rw-r--r--sound/usb/mixer_s1810c.c595
-rw-r--r--sound/usb/mixer_s1810c.h7
-rw-r--r--sound/usb/pcm.c7
-rw-r--r--sound/usb/proc.c2
-rw-r--r--sound/usb/quirks-table.h2
-rw-r--r--sound/usb/quirks.c88
-rw-r--r--sound/usb/quirks.h2
-rw-r--r--sound/usb/stream.c3
-rw-r--r--sound/usb/usbaudio.h1
-rw-r--r--sound/usb/usx2y/usbusx2yaudio.c9
378 files changed, 17505 insertions, 3925 deletions
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index a86c95d89824..e81083e1bc68 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -38,7 +38,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct dma_slave_config config;
int ret;
- dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
if (!dma_params)
return 0;
@@ -47,7 +47,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
return ret;
snd_dmaengine_pcm_set_config_from_dai_data(substream,
- snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+ snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream),
&config);
ret = dmaengine_slave_config(chan, &config);
@@ -95,7 +95,7 @@ int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
runtime->hw = pxa2xx_pcm_hardware;
- dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
if (!dma_params)
return 0;
@@ -120,7 +120,7 @@ int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
return ret;
return snd_dmaengine_pcm_open(
- substream, dma_request_slave_channel(rtd->cpu_dai->dev,
+ substream, dma_request_slave_channel(asoc_rtd_to_cpu(rtd, 0)->dev,
dma_params->chan_name));
}
EXPORT_SYMBOL(pxa2xx_pcm_open);
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 9de1c9a0173e..509290f2efa8 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -488,6 +488,48 @@ out:
}
#endif /* !COMPR_CODEC_CAPS_OVERFLOW */
+int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size)
+{
+ struct snd_dma_buffer *dmab;
+ int ret;
+
+ if (snd_BUG_ON(!(stream) || !(stream)->runtime))
+ return -EINVAL;
+ dmab = kzalloc(sizeof(*dmab), GFP_KERNEL);
+ if (!dmab)
+ return -ENOMEM;
+ dmab->dev = stream->dma_buffer.dev;
+ ret = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev, size, dmab);
+ if (ret < 0) {
+ kfree(dmab);
+ return ret;
+ }
+
+ snd_compr_set_runtime_buffer(stream, dmab);
+ stream->runtime->dma_bytes = size;
+ return 1;
+}
+EXPORT_SYMBOL(snd_compr_malloc_pages);
+
+int snd_compr_free_pages(struct snd_compr_stream *stream)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+
+ if (snd_BUG_ON(!(stream) || !(stream)->runtime))
+ return -EINVAL;
+ if (runtime->dma_area == NULL)
+ return 0;
+ if (runtime->dma_buffer_p != &stream->dma_buffer) {
+ /* It's a newly allocated buffer. Release it now. */
+ snd_dma_free_pages(runtime->dma_buffer_p);
+ kfree(runtime->dma_buffer_p);
+ }
+
+ snd_compr_set_runtime_buffer(stream, NULL);
+ return 0;
+}
+EXPORT_SYMBOL(snd_compr_free_pages);
+
/* revisit this with snd_pcm_preallocate_xxx */
static int snd_compr_allocate_buffer(struct snd_compr_stream *stream,
struct snd_compr_params *params)
diff --git a/sound/core/device.c b/sound/core/device.c
index cdc5af526739..bf0b04a7ee79 100644
--- a/sound/core/device.c
+++ b/sound/core/device.c
@@ -237,3 +237,24 @@ void snd_device_free_all(struct snd_card *card)
list_for_each_entry_safe_reverse(dev, next, &card->devices, list)
__snd_device_free(dev);
}
+
+/**
+ * snd_device_get_state - Get the current state of the given device
+ * @card: the card instance
+ * @device_data: the data pointer to release
+ *
+ * Returns the current state of the given device object. For the valid
+ * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or
+ * @SNDRV_DEV_DISCONNECTED is returned.
+ * Or for a non-existing device, -1 is returned as an error.
+ */
+int snd_device_get_state(struct snd_card *card, void *device_data)
+{
+ struct snd_device *dev;
+
+ dev = look_for_dev(card, device_data);
+ if (dev)
+ return dev->state;
+ return -1;
+}
+EXPORT_SYMBOL_GPL(snd_device_get_state);
diff --git a/sound/core/info.c b/sound/core/info.c
index ca87ae4c30ba..8c6bc5241df5 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -604,7 +604,7 @@ int snd_info_card_free(struct snd_card *card)
*/
int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
{
- int c = -1;
+ int c;
if (snd_BUG_ON(!buffer || !buffer->buffer))
return 1;
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 13db77771f0f..930def8201f4 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -884,20 +884,17 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
sformat = snd_pcm_plug_slave_format(format, sformat_mask);
if ((__force int)sformat < 0 ||
- !snd_mask_test(sformat_mask, (__force int)sformat)) {
- for (sformat = (__force snd_pcm_format_t)0;
- (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST;
- sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) {
- if (snd_mask_test(sformat_mask, (__force int)sformat) &&
+ !snd_mask_test_format(sformat_mask, sformat)) {
+ pcm_for_each_format(sformat) {
+ if (snd_mask_test_format(sformat_mask, sformat) &&
snd_pcm_oss_format_to(sformat) >= 0)
- break;
- }
- if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) {
- pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
- err = -EINVAL;
- goto failure;
+ goto format_found;
}
+ pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
+ err = -EINVAL;
+ goto failure;
}
+ format_found:
err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0);
if (err < 0)
goto failure;
@@ -1220,8 +1217,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const
if (ret < 0)
break;
}
+ mutex_unlock(&runtime->oss.params_lock);
ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
frames, in_kernel);
+ mutex_lock(&runtime->oss.params_lock);
if (ret != -EPIPE && ret != -ESTRPIPE)
break;
/* test, if we can't store new data, because the stream */
@@ -1257,8 +1256,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p
ret = snd_pcm_oss_capture_position_fixup(substream, &delay);
if (ret < 0)
break;
+ mutex_unlock(&runtime->oss.params_lock);
ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
frames, in_kernel);
+ mutex_lock(&runtime->oss.params_lock);
if (ret == -EPIPE) {
if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 752d078908e9..fbda4ebf38b3 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin)
return 0;
}
-snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
+static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug,
+ snd_pcm_sframes_t frames)
{
- struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
- int stream;
+ struct snd_pcm_plugin *plugin, *plugin_next;
- if (snd_BUG_ON(!plug))
- return -ENXIO;
- if (drv_frames == 0)
- return 0;
- stream = snd_pcm_plug_stream(plug);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- plugin = snd_pcm_plug_last(plug);
- while (plugin && drv_frames > 0) {
- if (drv_frames > plugin->buf_frames)
- drv_frames = plugin->buf_frames;
- plugin_prev = plugin->prev;
- if (plugin->src_frames)
- drv_frames = plugin->src_frames(plugin, drv_frames);
- plugin = plugin_prev;
+ plugin = snd_pcm_plug_first(plug);
+ while (plugin && frames > 0) {
+ plugin_next = plugin->next;
+ if (plugin->dst_frames) {
+ frames = plugin->dst_frames(plugin, frames);
+ if (frames < 0)
+ return frames;
}
- } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
- plugin = snd_pcm_plug_first(plug);
- while (plugin && drv_frames > 0) {
- plugin_next = plugin->next;
- if (plugin->dst_frames)
- drv_frames = plugin->dst_frames(plugin, drv_frames);
- if (drv_frames > plugin->buf_frames)
- drv_frames = plugin->buf_frames;
- plugin = plugin_next;
+ if (frames > plugin->buf_frames)
+ frames = plugin->buf_frames;
+ plugin = plugin_next;
+ }
+ return frames;
+}
+
+static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug,
+ snd_pcm_sframes_t frames)
+{
+ struct snd_pcm_plugin *plugin, *plugin_prev;
+
+ plugin = snd_pcm_plug_last(plug);
+ while (plugin && frames > 0) {
+ if (frames > plugin->buf_frames)
+ frames = plugin->buf_frames;
+ plugin_prev = plugin->prev;
+ if (plugin->src_frames) {
+ frames = plugin->src_frames(plugin, frames);
+ if (frames < 0)
+ return frames;
}
- } else
+ plugin = plugin_prev;
+ }
+ return frames;
+}
+
+snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
+{
+ if (snd_BUG_ON(!plug))
+ return -ENXIO;
+ switch (snd_pcm_plug_stream(plug)) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ return calc_src_frames(plug, drv_frames);
+ case SNDRV_PCM_STREAM_CAPTURE:
+ return calc_dst_frames(plug, drv_frames);
+ default:
snd_BUG();
- return drv_frames;
+ return -EINVAL;
+ }
}
snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames)
{
- struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
- snd_pcm_sframes_t frames;
- int stream;
-
if (snd_BUG_ON(!plug))
return -ENXIO;
- if (clt_frames == 0)
- return 0;
- frames = clt_frames;
- stream = snd_pcm_plug_stream(plug);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- plugin = snd_pcm_plug_first(plug);
- while (plugin && frames > 0) {
- plugin_next = plugin->next;
- if (plugin->dst_frames) {
- frames = plugin->dst_frames(plugin, frames);
- if (frames < 0)
- return frames;
- }
- if (frames > plugin->buf_frames)
- frames = plugin->buf_frames;
- plugin = plugin_next;
- }
- } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
- plugin = snd_pcm_plug_last(plug);
- while (plugin) {
- if (frames > plugin->buf_frames)
- frames = plugin->buf_frames;
- plugin_prev = plugin->prev;
- if (plugin->src_frames) {
- frames = plugin->src_frames(plugin, frames);
- if (frames < 0)
- return frames;
- }
- plugin = plugin_prev;
- }
- } else
+ switch (snd_pcm_plug_stream(plug)) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ return calc_dst_frames(plug, clt_frames);
+ case SNDRV_PCM_STREAM_CAPTURE:
+ return calc_src_frames(plug, clt_frames);
+ default:
snd_BUG();
- return frames;
+ return -EINVAL;
+ }
}
static int snd_pcm_plug_formats(const struct snd_mask *mask,
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index 7cd09cef6961..d381f4c967c9 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -47,7 +47,7 @@ struct rate_priv {
unsigned int pos;
rate_f func;
snd_pcm_sframes_t old_src_frames, old_dst_frames;
- struct rate_channel channels[0];
+ struct rate_channel channels[];
};
static void rate_init(struct snd_pcm_plugin *plugin)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index a141a301369f..b6d2331a82f7 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -1019,7 +1019,7 @@ static ssize_t show_pcm_class(struct device *dev,
str = "none";
else
str = strs[pcm->dev_class];
- return snprintf(buf, PAGE_SIZE, "%s\n", str);
+ return sprintf(buf, "%s\n", str);
}
static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL);
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 5749a8a49784..4d059ff2b2e4 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -240,6 +240,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue);
snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
{
struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct dma_tx_state state;
enum dma_status status;
unsigned int buf_size;
@@ -250,9 +251,12 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
buf_size = snd_pcm_lib_buffer_bytes(substream);
if (state.residue > 0 && state.residue <= buf_size)
pos = buf_size - state.residue;
+
+ runtime->delay = bytes_to_frames(runtime,
+ state.in_flight_bytes);
}
- return bytes_to_frames(substream->runtime, pos);
+ return bytes_to_frames(runtime, pos);
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer);
@@ -426,7 +430,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams(
* default assumption is that it supports 1, 2 and 4 bytes
* widths.
*/
- for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+ pcm_for_each_format(i) {
int bits = snd_pcm_format_physical_width(i);
/*
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index a6a541511534..257d412eac5d 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -42,6 +42,11 @@ struct pcm_format_data {
/* we do lots of calculations on snd_pcm_format_t; shut up sparse */
#define INT __force int
+static bool valid_format(snd_pcm_format_t format)
+{
+ return (INT)format >= 0 && (INT)format <= (INT)SNDRV_PCM_FORMAT_LAST;
+}
+
static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
[SNDRV_PCM_FORMAT_S8] = {
.width = 8, .phys = 8, .le = -1, .signd = 1,
@@ -259,7 +264,7 @@ static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] =
int snd_pcm_format_signed(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].signd) < 0)
return -EINVAL;
@@ -307,7 +312,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear);
int snd_pcm_format_little_endian(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].le) < 0)
return -EINVAL;
@@ -343,7 +348,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian);
int snd_pcm_format_width(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].width) == 0)
return -EINVAL;
@@ -361,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_width);
int snd_pcm_format_physical_width(snd_pcm_format_t format)
{
int val;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if ((val = pcm_formats[(INT)format].phys) == 0)
return -EINVAL;
@@ -394,7 +399,7 @@ EXPORT_SYMBOL(snd_pcm_format_size);
*/
const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format)
{
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return NULL;
if (! pcm_formats[(INT)format].phys)
return NULL;
@@ -418,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
unsigned char *dst;
const unsigned char *pat;
- if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+ if (!valid_format(format))
return -EINVAL;
if (samples == 0)
return 0;
@@ -474,32 +479,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
EXPORT_SYMBOL(snd_pcm_format_set_silence);
/**
- * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
- * @runtime: the runtime instance
+ * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields
+ * @hw: the pcm hw instance
*
* Determines the rate_min and rate_max fields from the rates bits of
- * the given runtime->hw.
+ * the given hw.
*
* Return: Zero if successful.
*/
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw)
{
int i;
for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
- if (runtime->hw.rates & (1 << i)) {
- runtime->hw.rate_min = snd_pcm_known_rates.list[i];
+ if (hw->rates & (1 << i)) {
+ hw->rate_min = snd_pcm_known_rates.list[i];
break;
}
}
for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
- if (runtime->hw.rates & (1 << i)) {
- runtime->hw.rate_max = snd_pcm_known_rates.list[i];
+ if (hw->rates & (1 << i)) {
+ hw->rate_max = snd_pcm_known_rates.list[i];
break;
}
}
return 0;
}
-EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
+EXPORT_SYMBOL(snd_pcm_hw_limit_rates);
/**
* snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index cbdf061667fa..aef860256278 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -228,6 +228,9 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream,
return err;
}
+/* macro for simplified cast */
+#define PARAM_MASK_BIT(b) (1U << (__force int)(b))
+
static bool hw_support_mmap(struct snd_pcm_substream *substream)
{
if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP))
@@ -257,7 +260,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream,
return -EINVAL;
/* This parameter is not requested to change by a caller. */
- if (!(params->rmask & (1 << k)))
+ if (!(params->rmask & PARAM_MASK_BIT(k)))
continue;
if (trace_hw_mask_param_enabled())
@@ -271,7 +274,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream,
/* Set corresponding flag so that the caller gets it. */
trace_hw_mask_param(substream, k, 0, &old_mask, m);
- params->cmask |= 1 << k;
+ params->cmask |= PARAM_MASK_BIT(k);
}
return 0;
@@ -293,7 +296,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream,
return -EINVAL;
/* This parameter is not requested to change by a caller. */
- if (!(params->rmask & (1 << k)))
+ if (!(params->rmask & PARAM_MASK_BIT(k)))
continue;
if (trace_hw_interval_param_enabled())
@@ -307,7 +310,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream,
/* Set corresponding flag so that the caller gets it. */
trace_hw_interval_param(substream, k, 0, &old_interval, i);
- params->cmask |= 1 << k;
+ params->cmask |= PARAM_MASK_BIT(k);
}
return 0;
@@ -349,7 +352,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream,
* have 0 so that the parameters are never changed anymore.
*/
for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++)
- vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0;
+ vstamps[k] = (params->rmask & PARAM_MASK_BIT(k)) ? 1 : 0;
/* Due to the above design, actual sequence number starts at 2. */
stamp = 2;
@@ -417,7 +420,7 @@ retry:
hw_param_interval(params, r->var));
}
- params->cmask |= (1 << r->var);
+ params->cmask |= PARAM_MASK_BIT(r->var);
vstamps[r->var] = stamp;
again = true;
}
@@ -486,9 +489,9 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
params->info = 0;
params->fifo_size = 0;
- if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
+ if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
params->msbits = 0;
- if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) {
+ if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_RATE)) {
params->rate_num = 0;
params->rate_den = 0;
}
@@ -2293,21 +2296,21 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params,
static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
- unsigned int k;
+ snd_pcm_format_t k;
const struct snd_interval *i =
hw_param_interval_c(params, rule->deps[0]);
struct snd_mask m;
struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
snd_mask_any(&m);
- for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+ pcm_for_each_format(k) {
int bits;
- if (! snd_mask_test(mask, k))
+ if (!snd_mask_test_format(mask, k))
continue;
bits = snd_pcm_format_physical_width(k);
if (bits <= 0)
continue; /* ignore invalid formats */
if ((unsigned)bits < i->min || (unsigned)bits > i->max)
- snd_mask_reset(&m, k);
+ snd_mask_reset(&m, (__force unsigned)k);
}
return snd_mask_refine(mask, &m);
}
@@ -2316,14 +2319,15 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval t;
- unsigned int k;
+ snd_pcm_format_t k;
+
t.min = UINT_MAX;
t.max = 0;
t.openmin = 0;
t.openmax = 0;
- for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+ pcm_for_each_format(k) {
int bits;
- if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
+ if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
continue;
bits = snd_pcm_format_physical_width(k);
if (bits <= 0)
@@ -2505,16 +2509,16 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
unsigned int mask = 0;
if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED);
if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED);
if (hw_support_mmap(substream)) {
if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_INTERLEAVED);
if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
- mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED);
if (hw->info & SNDRV_PCM_INFO_COMPLEX)
- mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX;
+ mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_COMPLEX);
}
err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask);
if (err < 0)
@@ -2524,7 +2528,8 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
if (err < 0)
return err;
- err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD);
+ err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT,
+ PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD));
if (err < 0)
return err;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index d78a27271d6d..251eaf1152e2 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -118,7 +118,7 @@ struct loopback_cable {
struct loopback_setup {
unsigned int notify: 1;
unsigned int rate_shift;
- unsigned int format;
+ snd_pcm_format_t format;
unsigned int rate;
unsigned int channels;
struct snd_ctl_elem_id active_id;
@@ -1432,7 +1432,7 @@ static int loopback_format_info(struct snd_kcontrol *kcontrol,
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST;
+ uinfo->value.integer.max = (__force int)SNDRV_PCM_FORMAT_LAST;
uinfo->value.integer.step = 1;
return 0;
}
@@ -1443,7 +1443,7 @@ static int loopback_format_get(struct snd_kcontrol *kcontrol,
struct loopback *loopback = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] =
- loopback->setup[kcontrol->id.subdevice]
+ (__force int)loopback->setup[kcontrol->id.subdevice]
[kcontrol->id.device].format;
return 0;
}
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 02ac3f4e0c02..b5486de08b97 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -901,10 +901,10 @@ static int snd_card_dummy_new_mixer(struct snd_dummy *dummy)
static void print_formats(struct snd_dummy *dummy,
struct snd_info_buffer *buffer)
{
- int i;
+ snd_pcm_format_t i;
- for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
- if (dummy->pcm_hw.formats & (1ULL << i))
+ pcm_for_each_format(i) {
+ if (dummy->pcm_hw.formats & pcm_format_to_bits(i))
snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
}
}
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 976d8cb9a34f..2c8e3392a490 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -509,7 +509,7 @@ MODULE_DEVICE_TABLE(ieee1394, bebob_id_table);
static struct fw_driver bebob_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-bebob",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = bebob_probe,
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index 1f5fc0e7c024..c84b913a9fe0 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -192,7 +192,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table);
static struct fw_driver dg00x_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-firewire-digi00x",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = snd_dg00x_probe,
diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c
index f5a016560eb8..b62a4fd22407 100644
--- a/sound/firewire/fireface/ff.c
+++ b/sound/firewire/fireface/ff.c
@@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_ff_id_table);
static struct fw_driver ff_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-fireface",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = snd_ff_probe,
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 134fc9ee26b9..b1cc013a3540 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -362,7 +362,7 @@ MODULE_DEVICE_TABLE(ieee1394, efw_id_table);
static struct fw_driver efw_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-fireworks",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = efw_probe,
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
index c29a97f6f638..6f38335fe10b 100644
--- a/sound/firewire/tascam/tascam-hwdep.c
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -17,6 +17,7 @@
static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
long count, loff_t *offset)
+ __releases(&tscm->lock)
{
struct snd_firewire_event_lock_status event = {
.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS,
@@ -36,6 +37,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf,
long remained, loff_t *offset)
+ __releases(&tscm->lock)
{
char __user *pos = buf;
unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL;
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index addc464503bc..5dac0d9fc58e 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
static struct fw_driver tscm_driver = {
.driver = {
.owner = THIS_MODULE,
- .name = "snd-firewire-tascam",
+ .name = KBUILD_MODNAME,
.bus = &fw_bus_type,
},
.probe = snd_tscm_probe,
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 9a526aeef8da..e3119f5cb0d5 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -204,7 +204,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name);
*/
int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size)
{
- return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
+ return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
codec->vendor_id, codec->revision_id, codec->type);
}
EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);
diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c
index e377ac93f37f..8e8257c574b0 100644
--- a/sound/isa/sb/emu8000_pcm.c
+++ b/sound/isa/sb/emu8000_pcm.c
@@ -435,7 +435,7 @@ enum {
#define LOOP_WRITE(rec, offset, _buf, count, mode) \
do { \
struct snd_emu8000 *emu = (rec)->emu; \
- unsigned short *buf = (unsigned short *)(_buf); \
+ unsigned short *buf = (__force unsigned short *)(_buf); \
snd_emu8000_write_wait(emu, 1); \
EMU8000_SMALW_WRITE(emu, offset); \
while (count > 0) { \
@@ -492,7 +492,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs,
#define LOOP_WRITE(rec, pos, _buf, count, mode) \
do { \
struct snd_emu8000 *emu = rec->emu; \
- unsigned short *buf = (unsigned short *)(_buf); \
+ unsigned short *buf = (__force unsigned short *)(_buf); \
snd_emu8000_write_wait(emu, 1); \
EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]); \
if (rec->voices > 1) \
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 4f524a9dbbca..4462375d2d82 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1070,7 +1070,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
{
struct snd_ali *codec = snd_pcm_substream_chip(substream);
struct snd_pcm_substream *s;
- unsigned int what, whati, capture_flag;
+ unsigned int what, whati;
struct snd_ali_voice *pvoice, *evoice;
unsigned int val;
int do_start;
@@ -1088,7 +1088,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
return -EINVAL;
}
- what = whati = capture_flag = 0;
+ what = whati = 0;
snd_pcm_group_for_each_entry(s, substream) {
if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) {
pvoice = s->runtime->private_data;
@@ -1110,8 +1110,6 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
evoice->running = 0;
}
snd_pcm_trigger_done(s, substream);
- if (pvoice->mode)
- capture_flag = 1;
}
}
spin_lock(&codec->reg_lock);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index a89a7e603ca8..6ff581733a19 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1789,6 +1789,7 @@ int snd_emu10k1_create(struct snd_card *card,
int idx, err;
int is_audigy;
size_t page_table_size;
+ __le32 *pgtbl;
unsigned int silent_page;
const struct snd_emu_chip_details *c;
static const struct snd_device_ops ops = {
@@ -2009,8 +2010,9 @@ int snd_emu10k1_create(struct snd_card *card,
/* Clear silent pages and set up pointers */
memset(emu->silent_page.area, 0, emu->silent_page.bytes);
silent_page = emu->silent_page.addr << emu->address_mode;
+ pgtbl = (__le32 *)emu->ptb_pages.area;
for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++)
- ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx);
+ pgtbl[idx] = cpu_to_le32(silent_page | idx);
/* set up voice indices */
for (idx = 0; idx < NUM_G; idx++) {
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index bd48335d09d7..e1d3082a4fe9 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver"
config SND_HDA_CODEC_CA0132_DSP
bool "Support new DSP code for CA0132 codec"
depends on SND_HDA_CODEC_CA0132
+ default y
select SND_HDA_DSP_LOADER
select FW_LOADER
help
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 53e7732ef752..a34a2c9f4bcf 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -88,7 +88,7 @@ struct hda_conn_list {
struct list_head list;
int len;
hda_nid_t nid;
- hda_nid_t conns[0];
+ hda_nid_t conns[];
};
/* look up the cached results */
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2609e391ce54..9765652a73d7 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -373,7 +373,7 @@ static int azx_get_sync_time(ktime_t *device,
u32 wallclk_ctr, wallclk_cycles;
bool direction;
u32 dma_select;
- u32 timeout = 200;
+ u32 timeout;
u32 retry_count = 0;
runtime = substream->runtime;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index ded8bc07d755..34fe753a46fb 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1180,6 +1180,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
+ SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
{}
@@ -2698,7 +2699,7 @@ struct dsp_image_seg {
u32 magic;
u32 chip_addr;
u32 count;
- u32 data[0];
+ u32 data[];
};
static const u32 g_magic_value = 0x4c46584d;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 5119a9ae3d8a..bb287a916dae 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -154,7 +154,6 @@ struct hdmi_spec {
struct hda_multi_out multiout;
struct hda_pcm_stream pcm_playback;
- bool use_jack_detect; /* jack detection enabled */
bool use_acomp_notifier; /* use eld_notify callback for hotplug */
bool acomp_registered; /* audio component registered in this driver */
struct drm_audio_component_audio_ops drm_audio_ops;
@@ -753,7 +752,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
* Unsolicited events
*/
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
int dev_id)
@@ -764,8 +763,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
if (pin_idx < 0)
return;
mutex_lock(&spec->pcm_lock);
- if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
- snd_hda_jack_report_sync(codec);
+ hdmi_present_sense(get_pin(spec, pin_idx), 1);
mutex_unlock(&spec->pcm_lock);
}
@@ -779,21 +777,9 @@ static void jack_callback(struct hda_codec *codec,
check_presence_and_report(codec, jack->nid, jack->dev_id);
}
-static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res,
+ struct hda_jack_tbl *jack)
{
- int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
- struct hda_jack_tbl *jack;
-
- if (codec->dp_mst) {
- int dev_entry =
- (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
-
- jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry);
- } else {
- jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0);
- }
- if (!jack)
- return;
jack->jack_dirty = 1;
codec_dbg(codec,
@@ -853,7 +839,7 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
}
if (subtag == 0)
- hdmi_intrinsic_event(codec, res);
+ hdmi_intrinsic_event(codec, res, jack);
else
hdmi_non_intrinsic_event(codec, res);
}
@@ -1480,21 +1466,60 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec,
per_pin->channels = 0;
}
+static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
+ struct hdmi_spec_per_pin *per_pin)
+{
+ struct hdmi_spec *spec = codec->spec;
+
+ if (per_pin->pcm_idx >= 0)
+ return spec->pcm_rec[per_pin->pcm_idx].jack;
+ else
+ return NULL;
+}
+
/* update per_pin ELD from the given new ELD;
* setup info frame and notification accordingly
+ * also notify ELD kctl and report jack status changes
*/
-static bool update_eld(struct hda_codec *codec,
+static void update_eld(struct hda_codec *codec,
struct hdmi_spec_per_pin *per_pin,
- struct hdmi_eld *eld)
+ struct hdmi_eld *eld,
+ int repoll)
{
struct hdmi_eld *pin_eld = &per_pin->sink_eld;
struct hdmi_spec *spec = codec->spec;
+ struct snd_jack *pcm_jack;
bool old_eld_valid = pin_eld->eld_valid;
bool eld_changed;
int pcm_idx;
+ if (eld->eld_valid) {
+ if (eld->eld_size <= 0 ||
+ snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
+ eld->eld_size) < 0) {
+ eld->eld_valid = false;
+ if (repoll) {
+ schedule_delayed_work(&per_pin->work,
+ msecs_to_jiffies(300));
+ return;
+ }
+ }
+ }
+
+ if (!eld->eld_valid || eld->eld_size <= 0) {
+ eld->eld_valid = false;
+ eld->eld_size = 0;
+ }
+
/* for monitor disconnection, save pcm_idx firstly */
pcm_idx = per_pin->pcm_idx;
+
+ /*
+ * pcm_idx >=0 before update_eld() means it is in monitor
+ * disconnected event. Jack must be fetched before update_eld().
+ */
+ pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
+
if (spec->dyn_pcm_assign) {
if (eld->eld_valid) {
hdmi_attach_hda_pcm(spec, per_pin);
@@ -1509,6 +1534,8 @@ static bool update_eld(struct hda_codec *codec,
*/
if (pcm_idx == -1)
pcm_idx = per_pin->pcm_idx;
+ if (!pcm_jack)
+ pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
if (eld->eld_valid)
snd_hdmi_show_eld(codec, &eld->info);
@@ -1547,42 +1574,17 @@ static bool update_eld(struct hda_codec *codec,
SNDRV_CTL_EVENT_MASK_VALUE |
SNDRV_CTL_EVENT_MASK_INFO,
&get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id);
- return eld_changed;
-}
-static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
- struct hdmi_spec_per_pin *per_pin)
-{
- struct hdmi_spec *spec = codec->spec;
- struct snd_jack *jack = NULL;
- struct hda_jack_tbl *jack_tbl;
-
- /* if !dyn_pcm_assign, get jack from hda_jack_tbl
- * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not
- * NULL even after snd_hda_jack_tbl_clear() is called to
- * free snd_jack. This may cause access invalid memory
- * when calling snd_jack_report
- */
- if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) {
- jack = spec->pcm_rec[per_pin->pcm_idx].jack;
- } else if (!spec->dyn_pcm_assign) {
- /*
- * jack tbl doesn't support DP MST
- * DP MST will use dyn_pcm_assign,
- * so DP MST will never come here
- */
- jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
- per_pin->dev_id);
- if (jack_tbl)
- jack = jack_tbl->jack;
- }
- return jack;
+ if (eld_changed && pcm_jack)
+ snd_jack_report(pcm_jack,
+ (eld->monitor_present && eld->eld_valid) ?
+ SND_JACK_AVOUT : 0);
}
+
/* update ELD and jack state via HD-audio verbs */
-static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
+static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
int repoll)
{
- struct hda_jack_tbl *jack;
struct hda_codec *codec = per_pin->codec;
struct hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->temp_eld;
@@ -1597,9 +1599,11 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
* the unsolicited response to avoid custom WARs.
*/
int present;
- bool ret;
- bool do_repoll = false;
- struct snd_jack *pcm_jack = NULL;
+ int ret;
+
+ ret = snd_hda_power_up_pm(codec);
+ if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec)))
+ goto out;
present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id);
@@ -1618,62 +1622,12 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
if (spec->ops.pin_get_eld(codec, pin_nid, dev_id,
eld->eld_buffer, &eld->eld_size) < 0)
eld->eld_valid = false;
- else {
- if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
- eld->eld_size) < 0)
- eld->eld_valid = false;
- }
- if (!eld->eld_valid && repoll)
- do_repoll = true;
}
- if (do_repoll) {
- schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300));
- } else {
- /*
- * pcm_idx >=0 before update_eld() means it is in monitor
- * disconnected event. Jack must be fetched before
- * update_eld().
- */
- pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
- update_eld(codec, per_pin, eld);
- if (!pcm_jack)
- pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
- }
-
- ret = !repoll || !eld->monitor_present || eld->eld_valid;
-
- jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id);
- if (jack) {
- jack->block_report = !ret;
- jack->pin_sense = (eld->monitor_present && eld->eld_valid) ?
- AC_PINSENSE_PRESENCE : 0;
-
- if (spec->dyn_pcm_assign && pcm_jack && !do_repoll) {
- int state = 0;
-
- if (jack->pin_sense & AC_PINSENSE_PRESENCE)
- state = SND_JACK_AVOUT;
- snd_jack_report(pcm_jack, state);
- }
-
- /*
- * snd_hda_jack_pin_sense() call at the beginning of this
- * function, updates jack->pins_sense and clears
- * jack->jack_dirty, therefore snd_hda_jack_report_sync() will
- * not override the jack->pin_sense.
- *
- * snd_hda_jack_report_sync() is superfluous for dyn_pcm_assign
- * case. The jack->pin_sense update was already performed, and
- * hda_jack->jack is NULL for dyn_pcm_assign.
- *
- * Don't call snd_hda_jack_report_sync() for
- * dyn_pcm_assign.
- */
- ret = ret && !spec->dyn_pcm_assign;
- }
+ update_eld(codec, per_pin, eld, repoll);
mutex_unlock(&per_pin->lock);
- return ret;
+ out:
+ snd_hda_power_down_pm(codec);
}
/* update ELD and jack state via audio component */
@@ -1682,64 +1636,25 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
{
struct hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->temp_eld;
- struct snd_jack *jack = NULL;
- bool changed;
- int size;
mutex_lock(&per_pin->lock);
eld->monitor_present = false;
- size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
+ eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
per_pin->dev_id, &eld->monitor_present,
eld->eld_buffer, ELD_MAX_SIZE);
- if (size > 0) {
- size = min(size, ELD_MAX_SIZE);
- if (snd_hdmi_parse_eld(codec, &eld->info,
- eld->eld_buffer, size) < 0)
- size = -EINVAL;
- }
-
- if (size > 0) {
- eld->eld_valid = true;
- eld->eld_size = size;
- } else {
- eld->eld_valid = false;
- eld->eld_size = 0;
- }
-
- /* pcm_idx >=0 before update_eld() means it is in monitor
- * disconnected event. Jack must be fetched before update_eld()
- */
- jack = pin_idx_to_pcm_jack(codec, per_pin);
- changed = update_eld(codec, per_pin, eld);
- if (jack == NULL)
- jack = pin_idx_to_pcm_jack(codec, per_pin);
- if (changed && jack)
- snd_jack_report(jack,
- (eld->monitor_present && eld->eld_valid) ?
- SND_JACK_AVOUT : 0);
+ eld->eld_valid = (eld->eld_size > 0);
+ update_eld(codec, per_pin, eld, 0);
mutex_unlock(&per_pin->lock);
}
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
{
struct hda_codec *codec = per_pin->codec;
- int ret;
- /* no temporary power up/down needed for component notifier */
- if (!codec_has_acomp(codec)) {
- ret = snd_hda_power_up_pm(codec);
- if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) {
- snd_hda_power_down_pm(codec);
- return false;
- }
- ret = hdmi_present_sense_via_verbs(per_pin, repoll);
- snd_hda_power_down_pm(codec);
- } else {
+ if (!codec_has_acomp(codec))
+ hdmi_present_sense_via_verbs(per_pin, repoll);
+ else
sync_eld_via_acomp(codec, per_pin);
- ret = false; /* don't call snd_hda_jack_report_sync() */
- }
-
- return ret;
}
static void hdmi_repoll_eld(struct work_struct *work)
@@ -1759,8 +1674,7 @@ static void hdmi_repoll_eld(struct work_struct *work)
per_pin->repoll_count = 0;
mutex_lock(&spec->pcm_lock);
- if (hdmi_present_sense(per_pin, per_pin->repoll_count))
- snd_hda_jack_report_sync(per_pin->codec);
+ hdmi_present_sense(per_pin, per_pin->repoll_count);
mutex_unlock(&spec->pcm_lock);
}
@@ -2206,15 +2120,23 @@ static void free_hdmi_jack_priv(struct snd_jack *jack)
pcm->jack = NULL;
}
-static int add_hdmi_jack_kctl(struct hda_codec *codec,
- struct hdmi_spec *spec,
- int pcm_idx,
- const char *name)
+static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
{
+ char hdmi_str[32] = "HDMI/DP";
+ struct hdmi_spec *spec = codec->spec;
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx);
struct snd_jack *jack;
+ int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
int err;
- err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack,
+ if (pcmdev > 0)
+ sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
+ if (!spec->dyn_pcm_assign &&
+ !is_jack_detectable(codec, per_pin->pin_nid))
+ strncat(hdmi_str, " Phantom",
+ sizeof(hdmi_str) - strlen(hdmi_str) - 1);
+
+ err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack,
true, false);
if (err < 0)
return err;
@@ -2225,48 +2147,6 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec,
return 0;
}
-static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
-{
- char hdmi_str[32] = "HDMI/DP";
- struct hdmi_spec *spec = codec->spec;
- struct hdmi_spec_per_pin *per_pin;
- struct hda_jack_tbl *jack;
- int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
- bool phantom_jack;
- int ret;
-
- if (pcmdev > 0)
- sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
-
- if (spec->dyn_pcm_assign)
- return add_hdmi_jack_kctl(codec, spec, pcm_idx, hdmi_str);
-
- /* for !dyn_pcm_assign, we still use hda_jack for compatibility */
- /* if !dyn_pcm_assign, it must be non-MST mode.
- * This means pcms and pins are statically mapped.
- * And pcm_idx is pin_idx.
- */
- per_pin = get_pin(spec, pcm_idx);
- phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid);
- if (phantom_jack)
- strncat(hdmi_str, " Phantom",
- sizeof(hdmi_str) - strlen(hdmi_str) - 1);
- ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid,
- per_pin->dev_id, hdmi_str, phantom_jack,
- 0, NULL);
- if (ret < 0)
- return ret;
- jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
- per_pin->dev_id);
- if (jack == NULL)
- return 0;
- /* assign jack->jack to pcm_rec[].jack to
- * align with dyn_pcm_assign mode
- */
- spec->pcm_rec[pcm_idx].jack = jack->jack;
- return 0;
-}
-
static int generic_hdmi_build_controls(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
@@ -2355,7 +2235,6 @@ static int generic_hdmi_init(struct hda_codec *codec)
int pin_idx;
mutex_lock(&spec->bind_lock);
- spec->use_jack_detect = !codec->jackpoll_interval;
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
@@ -2365,12 +2244,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
hdmi_init_pin(codec, pin_nid);
if (codec_has_acomp(codec))
continue;
- if (spec->use_jack_detect)
- snd_hda_jack_detect_enable(codec, pin_nid, dev_id);
- else
- snd_hda_jack_detect_enable_callback_mst(codec, pin_nid,
- dev_id,
- jack_callback);
+ snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, dev_id,
+ jack_callback);
}
mutex_unlock(&spec->bind_lock);
return 0;
@@ -2532,12 +2407,6 @@ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid,
unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag);
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE, val);
- } else {
- /* if no jack entry was defined beforehand, create a new one
- * at need (i.e. only when notifier is cleared)
- */
- if (!use_acomp)
- snd_hda_jack_detect_enable(codec, nid, dev_id);
}
}
@@ -2553,13 +2422,11 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp,
spec->use_acomp_notifier = use_acomp;
spec->codec->relaxed_resume = use_acomp;
/* reprogram each jack detection logic depending on the notifier */
- if (spec->use_jack_detect) {
- for (i = 0; i < spec->num_pins; i++)
- reprogram_jack_detect(spec->codec,
- get_pin(spec, i)->pin_nid,
- get_pin(spec, i)->dev_id,
- use_acomp);
- }
+ for (i = 0; i < spec->num_pins; i++)
+ reprogram_jack_detect(spec->codec,
+ get_pin(spec, i)->pin_nid,
+ get_pin(spec, i)->dev_id,
+ use_acomp);
mutex_unlock(&spec->bind_lock);
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 63e1a56f705b..f66a48154a57 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -107,6 +107,7 @@ struct alc_spec {
unsigned int done_hp_init:1;
unsigned int no_shutup_pins:1;
unsigned int ultra_low_power:1;
+ unsigned int has_hs_key:1;
/* for PLL fix */
hda_nid_t pll_nid;
@@ -367,7 +368,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0215:
case 0x10ec0233:
case 0x10ec0235:
+ case 0x10ec0236:
case 0x10ec0255:
+ case 0x10ec0256:
case 0x10ec0257:
case 0x10ec0282:
case 0x10ec0283:
@@ -379,11 +382,6 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0300:
alc_update_coef_idx(codec, 0x10, 1<<9, 0);
break;
- case 0x10ec0236:
- case 0x10ec0256:
- alc_write_coef_idx(codec, 0x36, 0x5757);
- alc_update_coef_idx(codec, 0x10, 1<<9, 0);
- break;
case 0x10ec0275:
alc_update_coef_idx(codec, 0xe, 0, 1<<0);
break;
@@ -2982,6 +2980,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
return alc_parse_auto_config(codec, alc269_ignore, ssids);
}
+static const struct hda_jack_keymap alc_headset_btn_keymap[] = {
+ { SND_JACK_BTN_0, KEY_PLAYPAUSE },
+ { SND_JACK_BTN_1, KEY_VOICECOMMAND },
+ { SND_JACK_BTN_2, KEY_VOLUMEUP },
+ { SND_JACK_BTN_3, KEY_VOLUMEDOWN },
+ {}
+};
+
+static void alc_headset_btn_callback(struct hda_codec *codec,
+ struct hda_jack_callback *jack)
+{
+ int report = 0;
+
+ if (jack->unsol_res & (7 << 13))
+ report |= SND_JACK_BTN_0;
+
+ if (jack->unsol_res & (1 << 16 | 3 << 8))
+ report |= SND_JACK_BTN_1;
+
+ /* Volume up key */
+ if (jack->unsol_res & (7 << 23))
+ report |= SND_JACK_BTN_2;
+
+ /* Volume down key */
+ if (jack->unsol_res & (7 << 10))
+ report |= SND_JACK_BTN_3;
+
+ jack->jack->button_state = report;
+}
+
+static void alc_disable_headset_jack_key(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (!spec->has_hs_key)
+ return;
+
+ switch (codec->core.vendor_id) {
+ case 0x10ec0215:
+ case 0x10ec0225:
+ case 0x10ec0285:
+ case 0x10ec0295:
+ case 0x10ec0289:
+ case 0x10ec0299:
+ alc_write_coef_idx(codec, 0x48, 0x0);
+ alc_update_coef_idx(codec, 0x49, 0x0045, 0x0);
+ alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0);
+ break;
+ case 0x10ec0236:
+ case 0x10ec0256:
+ alc_write_coef_idx(codec, 0x48, 0x0);
+ alc_update_coef_idx(codec, 0x49, 0x0045, 0x0);
+ break;
+ }
+}
+
+static void alc_enable_headset_jack_key(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (!spec->has_hs_key)
+ return;
+
+ switch (codec->core.vendor_id) {
+ case 0x10ec0215:
+ case 0x10ec0225:
+ case 0x10ec0285:
+ case 0x10ec0295:
+ case 0x10ec0289:
+ case 0x10ec0299:
+ alc_write_coef_idx(codec, 0x48, 0xd011);
+ alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
+ alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8);
+ break;
+ case 0x10ec0236:
+ case 0x10ec0256:
+ alc_write_coef_idx(codec, 0x48, 0xd011);
+ alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
+ break;
+ }
+}
+
+static void alc_fixup_headset_jack(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->has_hs_key = 1;
+ snd_hda_jack_detect_enable_callback(codec, 0x55,
+ alc_headset_btn_callback);
+ snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false,
+ SND_JACK_HEADSET, alc_headset_btn_keymap);
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ alc_enable_headset_jack_key(codec);
+ break;
+ }
+}
+
static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up)
{
alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0);
@@ -3269,7 +3368,13 @@ static void alc256_init(struct hda_codec *codec)
alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */
alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 1 << 15); /* Clear bit */
alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 0 << 15);
- alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
+ /*
+ * Expose headphone mic (or possibly Line In on some machines) instead
+ * of PC Beep on 1Ah, and disable 1Ah loopback for all outputs. See
+ * Documentation/sound/hd-audio/realtek-pc-beep.rst for details of
+ * this register.
+ */
+ alc_write_coef_idx(codec, 0x36, 0x5757);
}
static void alc256_shutup(struct hda_codec *codec)
@@ -3372,6 +3477,8 @@ static void alc225_shutup(struct hda_codec *codec)
if (!hp_pin)
hp_pin = 0x21;
+
+ alc_disable_headset_jack_key(codec);
/* 3k pull low control for Headset jack. */
alc_update_coef_idx(codec, 0x4a, 0, 3 << 10);
@@ -3411,6 +3518,9 @@ static void alc225_shutup(struct hda_codec *codec)
alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4);
msleep(30);
}
+
+ alc_update_coef_idx(codec, 0x4a, 3 << 10, 0);
+ alc_enable_headset_jack_key(codec);
}
static void alc_default_init(struct hda_codec *codec)
@@ -4008,6 +4118,12 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10);
}
+static void alc285_fixup_hp_gpio_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ alc_fixup_hp_gpio_led(codec, action, 0x04, 0x00);
+}
+
static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -5375,17 +5491,6 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
}
}
-static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec,
- const struct hda_fixup *fix,
- int action)
-{
- if (action != HDA_FIXUP_ACT_PRE_PROBE)
- return;
-
- snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1);
- snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP);
-}
-
static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -5662,69 +5767,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec,
snd_hda_override_wcaps(codec, 0x03, 0);
}
-static const struct hda_jack_keymap alc_headset_btn_keymap[] = {
- { SND_JACK_BTN_0, KEY_PLAYPAUSE },
- { SND_JACK_BTN_1, KEY_VOICECOMMAND },
- { SND_JACK_BTN_2, KEY_VOLUMEUP },
- { SND_JACK_BTN_3, KEY_VOLUMEDOWN },
- {}
-};
-
-static void alc_headset_btn_callback(struct hda_codec *codec,
- struct hda_jack_callback *jack)
-{
- int report = 0;
-
- if (jack->unsol_res & (7 << 13))
- report |= SND_JACK_BTN_0;
-
- if (jack->unsol_res & (1 << 16 | 3 << 8))
- report |= SND_JACK_BTN_1;
-
- /* Volume up key */
- if (jack->unsol_res & (7 << 23))
- report |= SND_JACK_BTN_2;
-
- /* Volume down key */
- if (jack->unsol_res & (7 << 10))
- report |= SND_JACK_BTN_3;
-
- jack->jack->button_state = report;
-}
-
-static void alc_fixup_headset_jack(struct hda_codec *codec,
- const struct hda_fixup *fix, int action)
-{
-
- switch (action) {
- case HDA_FIXUP_ACT_PRE_PROBE:
- snd_hda_jack_detect_enable_callback(codec, 0x55,
- alc_headset_btn_callback);
- snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false,
- SND_JACK_HEADSET, alc_headset_btn_keymap);
- break;
- case HDA_FIXUP_ACT_INIT:
- switch (codec->core.vendor_id) {
- case 0x10ec0215:
- case 0x10ec0225:
- case 0x10ec0285:
- case 0x10ec0295:
- case 0x10ec0289:
- case 0x10ec0299:
- alc_write_coef_idx(codec, 0x48, 0xd011);
- alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
- alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8);
- break;
- case 0x10ec0236:
- case 0x10ec0256:
- alc_write_coef_idx(codec, 0x48, 0xd011);
- alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
- break;
- }
- break;
- }
-}
-
static void alc295_fixup_chromebook(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -5863,8 +5905,6 @@ enum {
ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
ALC275_FIXUP_DELL_XPS,
- ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
- ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2,
ALC293_FIXUP_LENOVO_SPK_NOISE,
ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
ALC255_FIXUP_DELL_SPK_NOISE,
@@ -5923,6 +5963,7 @@ enum {
ALC294_FIXUP_ASUS_DUAL_SPK,
ALC285_FIXUP_THINKPAD_HEADSET_JACK,
ALC294_FIXUP_ASUS_HPE,
+ ALC285_FIXUP_HP_GPIO_LED,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -6604,23 +6645,6 @@ static const struct hda_fixup alc269_fixups[] = {
{}
}
},
- [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = {
- .type = HDA_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- /* Disable pass-through path for FRONT 14h */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x36},
- {0x20, AC_VERB_SET_PROC_COEF, 0x1737},
- {}
- },
- .chained = true,
- .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
- },
- [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = {
- .type = HDA_FIXUP_FUNC,
- .v.func = alc256_fixup_dell_xps_13_headphone_noise2,
- .chained = true,
- .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE
- },
[ALC293_FIXUP_LENOVO_SPK_NOISE] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_disable_aamix,
@@ -7061,6 +7085,10 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
},
+ [ALC285_FIXUP_HP_GPIO_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_hp_gpio_led,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7114,17 +7142,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
- SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE),
SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP),
- SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME),
SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME),
SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3),
SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC),
@@ -7208,6 +7233,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+ SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -7477,7 +7503,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"},
{.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"},
{.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"},
- {.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"},
{.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"},
{.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"},
{.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"},
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 21ab9cc50c71..65a887b217ee 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -30,7 +30,7 @@
#if K1212_DEBUG_LEVEL > 0
#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args)
#else
-#define K1212_DEBUG_PRINTK(fmt,...)
+#define K1212_DEBUG_PRINTK(fmt,...) do { } while (0)
#endif
#if K1212_DEBUG_LEVEL > 1
#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args)
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index cc06f0a1a7e4..227aece17e39 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -3353,7 +3353,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
return;
}
} else {
- int err = -EINVAL;
+ int err;
+
err = hdsp_request_fw_loader(hdsp);
if (err < 0) {
snd_iprintf(buffer,
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 799789c8eea9..8b03e2dc503f 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -414,6 +414,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
{
unsigned int i, idx, ofs, rest;
struct via82xx *chip = snd_pcm_substream_chip(substream);
+ __le32 *pgtbl;
if (dev->table.area == NULL) {
/* the start of each lists must be aligned to 8 bytes,
@@ -435,6 +436,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
/* fill the entries */
idx = 0;
ofs = 0;
+ pgtbl = (__le32 *)dev->table.area;
for (i = 0; i < periods; i++) {
rest = fragsize;
/* fill descriptors for a period.
@@ -451,7 +453,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
return -EINVAL;
}
addr = snd_pcm_sgbuf_get_addr(substream, ofs);
- ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+ pgtbl[idx << 1] = cpu_to_le32(addr);
r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest);
rest -= r;
if (! rest) {
@@ -466,7 +468,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
"tbl %d: at %d size %d (rest %d)\n",
idx, ofs, r, rest);
*/
- ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+ pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
dev->idx_table[idx].offset = ofs;
dev->idx_table[idx].size = r;
ofs += r;
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 84e589803e2e..607b7100db1c 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -267,6 +267,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
{
unsigned int i, idx, ofs, rest;
struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ __le32 *pgtbl;
if (dev->table.area == NULL) {
/* the start of each lists must be aligned to 8 bytes,
@@ -288,6 +289,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
/* fill the entries */
idx = 0;
ofs = 0;
+ pgtbl = (__le32 *)dev->table.area;
for (i = 0; i < periods; i++) {
rest = fragsize;
/* fill descriptors for a period.
@@ -304,7 +306,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
return -EINVAL;
}
addr = snd_pcm_sgbuf_get_addr(substream, ofs);
- ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+ pgtbl[idx << 1] = cpu_to_le32(addr);
r = PAGE_SIZE - (ofs % PAGE_SIZE);
if (rest < r)
r = rest;
@@ -321,7 +323,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
"tbl %d: at %d size %d (rest %d)\n",
idx, ofs, r, rest);
*/
- ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+ pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
dev->idx_table[idx].offset = ofs;
dev->idx_table[idx].size = r;
ofs += r;
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 093806d735c6..9554a0c506af 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -40,6 +40,7 @@ static int keywest_probe(struct i2c_client *client,
static int keywest_attach_adapter(struct i2c_adapter *adapter)
{
struct i2c_board_info info;
+ struct i2c_client *client;
if (! keywest_ctx)
return -EINVAL;
@@ -50,9 +51,11 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
memset(&info, 0, sizeof(struct i2c_board_info));
strlcpy(info.type, "keywest", I2C_NAME_SIZE);
info.addr = keywest_ctx->addr;
- keywest_ctx->client = i2c_new_device(adapter, &info);
- if (!keywest_ctx->client)
- return -ENODEV;
+ client = i2c_new_client_device(adapter, &info);
+ if (IS_ERR(client))
+ return PTR_ERR(client);
+ keywest_ctx->client = client;
+
/*
* We know the driver is already loaded, so the device should be
* already bound. If not it means binding failed, and then there
diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig
index 5f40517717c4..bce4cee5cb54 100644
--- a/sound/soc/amd/Kconfig
+++ b/sound/soc/amd/Kconfig
@@ -26,3 +26,13 @@ config SND_SOC_AMD_ACP3x
depends on X86 && PCI
help
This option enables ACP v3.x I2S support on AMD platform
+
+config SND_SOC_AMD_RV_RT5682_MACH
+ tristate "AMD RV support for RT5682"
+ select SND_SOC_RT5682
+ select SND_SOC_MAX98357A
+ select SND_SOC_CROS_EC_CODEC
+ select I2C_CROS_EC_TUNNEL
+ depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC
+ help
+ This option enables machine driver for RT5682 and MAX9835.
diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile
index c4ddc6adb6f0..e6f3d9b469f3 100644
--- a/sound/soc/amd/Makefile
+++ b/sound/soc/amd/Makefile
@@ -2,8 +2,10 @@
acp_audio_dma-objs := acp-pcm-dma.o
snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o
snd-soc-acp-rt5645-mach-objs := acp-rt5645.o
+snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o
obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o
obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o
obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o
obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/
+obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 7a5621e5e233..9414d7269c4f 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -54,7 +54,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
struct snd_soc_card *card = rtd->card;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_component *component = codec_dai->component;
dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name);
diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c
index 91abeb92b648..73b31f88a6b5 100644
--- a/sound/soc/amd/acp-rt5645.c
+++ b/sound/soc/amd/acp-rt5645.c
@@ -48,7 +48,7 @@ static int cz_aif1_hw_params(struct snd_pcm_substream *substream,
{
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
CZ_PLAT_CLK, params_rate(params) * 512);
@@ -73,7 +73,7 @@ static int cz_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_card *card;
struct snd_soc_component *codec;
- codec = rtd->codec_dai->component;
+ codec = asoc_rtd_to_codec(rtd, 0)->component;
card = rtd->card;
ret = snd_soc_card_jack_new(card, "Headset Jack",
diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c
new file mode 100644
index 000000000000..024a7ee54cd5
--- /dev/null
+++ b/sound/soc/amd/acp3x-rt5682-max9836.c
@@ -0,0 +1,376 @@
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec.
+//
+//Copyright 2016 Advanced Micro Devices, Inc.
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <linux/clk.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/io.h>
+#include <linux/acpi.h>
+
+#include "raven/acp3x.h"
+#include "../codecs/rt5682.h"
+
+#define PCO_PLAT_CLK 48000000
+#define RT5682_PLL_FREQ (48000 * 512)
+#define DUAL_CHANNEL 2
+
+static struct snd_soc_jack pco_jack;
+static struct clk *rt5682_dai_wclk;
+static struct clk *rt5682_dai_bclk;
+static struct gpio_desc *dmic_sel;
+
+static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_component *component = codec_dai->component;
+
+ dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name);
+
+ /* set rt5682 dai fmt */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ dev_err(rtd->card->dev,
+ "Failed to set rt5682 dai fmt: %d\n", ret);
+ return ret;
+ }
+
+ /* set codec PLL */
+ ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK,
+ PCO_PLAT_CLK, RT5682_PLL_FREQ);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set rt5682 PLL: %d\n", ret);
+ return ret;
+ }
+
+ /* Set codec sysclk */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2,
+ RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "Failed to set rt5682 SYSCLK: %d\n", ret);
+ return ret;
+ }
+
+ /* Set tdm/i2s1 master bclk ratio */
+ ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64);
+ if (ret < 0) {
+ dev_err(rtd->dev,
+ "Failed to set rt5682 tdm bclk ratio: %d\n", ret);
+ return ret;
+ }
+
+ rt5682_dai_wclk = clk_get(component->dev, "rt5682-dai-wclk");
+ rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk");
+
+ ret = snd_soc_card_jack_new(card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_LINEOUT |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3,
+ &pco_jack, NULL, 0);
+ if (ret) {
+ dev_err(card->dev, "HP jack creation failed %d\n", ret);
+ return ret;
+ }
+
+ snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+ snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+ snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+ snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+ ret = snd_soc_component_set_jack(component, &pco_jack, NULL);
+ if (ret) {
+ dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+static int rt5682_clk_enable(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* RT5682 will support only 48K output with 48M mclk */
+ clk_set_rate(rt5682_dai_wclk, 48000);
+ clk_set_rate(rt5682_dai_bclk, 48000 * 64);
+ ret = clk_prepare_enable(rt5682_dai_wclk);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't enable wclk %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+static void rt5682_clk_disable(void)
+{
+ clk_disable_unprepare(rt5682_dai_wclk);
+}
+
+static const unsigned int channels[] = {
+ DUAL_CHANNEL,
+};
+
+static const unsigned int rates[] = {
+ 48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+};
+
+static int acp3x_5682_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+ machine->play_i2s_instance = I2S_SP_INSTANCE;
+ machine->cap_i2s_instance = I2S_SP_INSTANCE;
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+ return rt5682_clk_enable(substream);
+}
+
+static int acp3x_max_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+ machine->play_i2s_instance = I2S_BT_INSTANCE;
+
+ runtime->hw.channels_max = DUAL_CHANNEL;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+ return rt5682_clk_enable(substream);
+}
+
+static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+ machine->cap_i2s_instance = I2S_BT_INSTANCE;
+ snd_soc_dai_set_bclk_ratio(codec_dai, 64);
+ if (dmic_sel)
+ gpiod_set_value(dmic_sel, 0);
+
+ return rt5682_clk_enable(substream);
+}
+
+static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+ machine->cap_i2s_instance = I2S_BT_INSTANCE;
+ snd_soc_dai_set_bclk_ratio(codec_dai, 64);
+ if (dmic_sel)
+ gpiod_set_value(dmic_sel, 1);
+
+ return rt5682_clk_enable(substream);
+}
+
+static void rt5682_shutdown(struct snd_pcm_substream *substream)
+{
+ rt5682_clk_disable();
+}
+
+static const struct snd_soc_ops acp3x_5682_ops = {
+ .startup = acp3x_5682_startup,
+ .shutdown = rt5682_shutdown,
+};
+
+static const struct snd_soc_ops acp3x_max_play_ops = {
+ .startup = acp3x_max_startup,
+ .shutdown = rt5682_shutdown,
+};
+
+static const struct snd_soc_ops acp3x_ec_cap0_ops = {
+ .startup = acp3x_ec_dmic0_startup,
+ .shutdown = rt5682_shutdown,
+};
+
+static const struct snd_soc_ops acp3x_ec_cap1_ops = {
+ .startup = acp3x_ec_dmic1_startup,
+ .shutdown = rt5682_shutdown,
+};
+
+SND_SOC_DAILINK_DEF(acp3x_i2s,
+ DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0")));
+SND_SOC_DAILINK_DEF(acp3x_bt,
+ DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.2")));
+
+SND_SOC_DAILINK_DEF(rt5682,
+ DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1")));
+SND_SOC_DAILINK_DEF(max,
+ DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi")));
+SND_SOC_DAILINK_DEF(cros_ec,
+ DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX")));
+
+SND_SOC_DAILINK_DEF(platform,
+ DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0")));
+
+static struct snd_soc_dai_link acp3x_dai_5682_98357[] = {
+ {
+ .name = "acp3x-5682-play",
+ .stream_name = "Playback",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .init = acp3x_5682_init,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &acp3x_5682_ops,
+ SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform),
+ },
+ {
+ .name = "acp3x-max98357-play",
+ .stream_name = "HiFi Playback",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .dpcm_playback = 1,
+ .ops = &acp3x_max_play_ops,
+ SND_SOC_DAILINK_REG(acp3x_bt, max, platform),
+ },
+ {
+ .name = "acp3x-ec-dmic0-capture",
+ .stream_name = "Capture DMIC0",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .dpcm_capture = 1,
+ .ops = &acp3x_ec_cap0_ops,
+ SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform),
+ },
+ {
+ .name = "acp3x-ec-dmic1-capture",
+ .stream_name = "Capture DMIC1",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .dpcm_capture = 1,
+ .ops = &acp3x_ec_cap1_ops,
+ SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform),
+ },
+};
+
+static const struct snd_soc_dapm_widget acp3x_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Spk", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route acp3x_audio_route[] = {
+ {"Headphone Jack", NULL, "HPOL"},
+ {"Headphone Jack", NULL, "HPOR"},
+ {"IN1P", NULL, "Headset Mic"},
+ {"Spk", NULL, "Speaker"},
+};
+
+static const struct snd_kcontrol_new acp3x_mc_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Spk"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static struct snd_soc_card acp3x_card = {
+ .name = "acp3xalc5682m98357",
+ .owner = THIS_MODULE,
+ .dai_link = acp3x_dai_5682_98357,
+ .num_links = ARRAY_SIZE(acp3x_dai_5682_98357),
+ .dapm_widgets = acp3x_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets),
+ .dapm_routes = acp3x_audio_route,
+ .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route),
+ .controls = acp3x_mc_controls,
+ .num_controls = ARRAY_SIZE(acp3x_mc_controls),
+};
+
+static int acp3x_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct snd_soc_card *card;
+ struct acp3x_platform_info *machine;
+
+ machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL);
+ if (!machine)
+ return -ENOMEM;
+
+ card = &acp3x_card;
+ acp3x_card.dev = &pdev->dev;
+ platform_set_drvdata(pdev, card);
+ snd_soc_card_set_drvdata(card, machine);
+
+ dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW);
+ if (IS_ERR(dmic_sel)) {
+ dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n",
+ PTR_ERR(dmic_sel));
+ return PTR_ERR(dmic_sel);
+ }
+
+ ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "devm_snd_soc_register_card(%s) failed: %d\n",
+ acp3x_card.name, ret);
+ return ret;
+ }
+ return 0;
+}
+
+static const struct acpi_device_id acp3x_audio_acpi_match[] = {
+ { "AMDI5682", 0 },
+ {},
+};
+MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match);
+
+static struct platform_driver acp3x_audio = {
+ .driver = {
+ .name = "acp3x-alc5682-max98357",
+ .acpi_match_table = ACPI_PTR(acp3x_audio_acpi_match),
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = acp3x_probe,
+};
+
+module_platform_driver(acp3x_audio);
+
+MODULE_AUTHOR("akshu.agrawal@amd.com");
+MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c
index 91a388184e52..3a3c47e820ab 100644
--- a/sound/soc/amd/raven/acp3x-i2s.c
+++ b/sound/soc/amd/raven/acp3x-i2s.c
@@ -42,7 +42,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai,
u32 tx_mask, u32 rx_mask, int slots, int slot_width)
{
struct i2s_dev_data *adata;
- u32 val, reg_val, frmt_reg, frm_len;
+ u32 frm_len;
u16 slot_len;
adata = snd_soc_dai_get_drvdata(cpu_dai);
@@ -64,36 +64,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai,
default:
return -EINVAL;
}
-
- /* Enable I2S/BT channels TDM, respective TX/RX frame lengths.*/
-
frm_len = FRM_LEN | (slots << 15) | (slot_len << 18);
- if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) {
- switch (adata->i2s_instance) {
- case I2S_BT_INSTANCE:
- reg_val = mmACP_BTTDM_ITER;
- frmt_reg = mmACP_BTTDM_TXFRMT;
- break;
- case I2S_SP_INSTANCE:
- default:
- reg_val = mmACP_I2STDM_ITER;
- frmt_reg = mmACP_I2STDM_TXFRMT;
- }
- } else {
- switch (adata->i2s_instance) {
- case I2S_BT_INSTANCE:
- reg_val = mmACP_BTTDM_IRER;
- frmt_reg = mmACP_BTTDM_RXFRMT;
- break;
- case I2S_SP_INSTANCE:
- default:
- reg_val = mmACP_I2STDM_IRER;
- frmt_reg = mmACP_I2STDM_RXFRMT;
- }
- }
- val = rv_readl(adata->acp3x_base + reg_val);
- rv_writel(val | 0x2, adata->acp3x_base + reg_val);
- rv_writel(frm_len, adata->acp3x_base + frmt_reg);
adata->tdm_fmt = frm_len;
return 0;
}
@@ -105,12 +76,14 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *prtd;
struct snd_soc_card *card;
struct acp3x_platform_info *pinfo;
+ struct i2s_dev_data *adata;
u32 val;
- u32 reg_val;
+ u32 reg_val, frmt_reg;
prtd = substream->private_data;
rtd = substream->runtime->private_data;
card = prtd->card;
+ adata = snd_soc_dai_get_drvdata(dai);
pinfo = snd_soc_card_get_drvdata(card);
if (pinfo) {
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -141,21 +114,30 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream,
switch (rtd->i2s_instance) {
case I2S_BT_INSTANCE:
reg_val = mmACP_BTTDM_ITER;
+ frmt_reg = mmACP_BTTDM_TXFRMT;
break;
case I2S_SP_INSTANCE:
default:
reg_val = mmACP_I2STDM_ITER;
+ frmt_reg = mmACP_I2STDM_TXFRMT;
}
} else {
switch (rtd->i2s_instance) {
case I2S_BT_INSTANCE:
reg_val = mmACP_BTTDM_IRER;
+ frmt_reg = mmACP_BTTDM_RXFRMT;
break;
case I2S_SP_INSTANCE:
default:
reg_val = mmACP_I2STDM_IRER;
+ frmt_reg = mmACP_I2STDM_RXFRMT;
}
}
+ if (adata->tdm_mode) {
+ val = rv_readl(rtd->acp3x_base + reg_val);
+ rv_writel(val | 0x2, rtd->acp3x_base + reg_val);
+ rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg);
+ }
val = rv_readl(rtd->acp3x_base + reg_val);
val = val | (rtd->xfer_resolution << 3);
rv_writel(val, rtd->acp3x_base + reg_val);
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index d62c0d90c41e..e362f0bc9e46 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -458,7 +458,8 @@ static int acp3x_resume(struct device *dev)
reg_val = mmACP_I2STDM_ITER;
frmt_val = mmACP_I2STDM_TXFRMT;
}
- rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val);
+ rv_writel((rtd->xfer_resolution << 3),
+ rtd->acp3x_base + reg_val);
}
if (adata->capture_stream && adata->capture_stream->runtime) {
struct i2s_stream_instance *rtd =
@@ -474,7 +475,8 @@ static int acp3x_resume(struct device *dev)
reg_val = mmACP_I2STDM_IRER;
frmt_val = mmACP_I2STDM_RXFRMT;
}
- rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val);
+ rv_writel((rtd->xfer_resolution << 3),
+ rtd->acp3x_base + reg_val);
}
if (adata->tdm_mode == TDM_ENABLE) {
rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val);
diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c
index da60e2ec5535..f25ce50f1a90 100644
--- a/sound/soc/amd/raven/pci-acp3x.c
+++ b/sound/soc/amd/raven/pci-acp3x.c
@@ -38,8 +38,13 @@ static int acp3x_power_on(void __iomem *acp3x_base)
timeout = 0;
while (++timeout < 500) {
val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS);
- if (!val)
+ if (!val) {
+ /* Set PME_EN as after ACP power On,
+ * PME_EN gets cleared
+ */
+ rv_writel(0x1, acp3x_base + mmACP_PME_EN);
return 0;
+ }
udelay(1);
}
return -ETIMEDOUT;
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index db67f5ba1e9a..cb03c4f7324c 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -56,7 +56,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct atmel_pcm_dma_params *prtd;
- prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
if (ssc_sr & prtd->mask->ssc_error) {
if (snd_pcm_running(substream))
@@ -83,7 +83,7 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
struct ssc_device *ssc;
int ret;
- prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
ssc = prtd->ssc;
ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c
index 59c1331a6984..a8daebcbf6c8 100644
--- a/sound/soc/atmel/atmel-pcm-pdc.c
+++ b/sound/soc/atmel/atmel-pcm-pdc.c
@@ -213,7 +213,7 @@ static int atmel_pcm_hw_params(struct snd_soc_component *component,
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
- prtd->params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ prtd->params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
prtd->dma_buffer = runtime->dma_addr;
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index 776b27d3686e..148c943cb538 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -27,7 +27,7 @@ static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK,
diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index befc2a3a05b0..3cb63886195f 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -239,10 +239,10 @@ struct mchp_i2s_mcc_dev {
unsigned int frame_length;
int tdm_slots;
int channels;
- int gclk_use:1;
- int gclk_running:1;
- int tx_rdy:1;
- int rx_rdy:1;
+ unsigned int gclk_use:1;
+ unsigned int gclk_running:1;
+ unsigned int tx_rdy:1;
+ unsigned int rx_rdy:1;
};
static irqreturn_t mchp_i2s_mcc_interrupt(int irq, void *dev_id)
diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c
index aa6d0d78566f..f9a85fd01b79 100644
--- a/sound/soc/atmel/mikroe-proto.c
+++ b/sound/soc/atmel/mikroe-proto.c
@@ -21,7 +21,7 @@
static int snd_proto_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/* Set proto sysclk */
int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index b1bef2bf142d..ed1f69b57024 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -96,7 +96,7 @@ static const struct snd_soc_dapm_route intercon[] = {
*/
static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct device *dev = rtd->dev;
int ret;
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 7822425d5e61..9fbc3c1113cc 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -40,7 +40,7 @@ struct sam9x5_drvdata {
*/
static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct device *dev = rtd->dev;
int ret;
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index d6b692fff29a..d649037bda9b 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -95,7 +95,7 @@ static struct snd_soc_card db1550_ac97_machine = {
static int db1200_i2s_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/* WM8731 has its own 12MHz crystal */
snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 8f855644c6b4..e82bbf2d1eea 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -281,7 +281,7 @@ static int au1xpsc_pcm_open(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int stype = substream->stream, *dmaids;
- dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
if (!dmaids)
return -ENODEV; /* whoa, has ordering changed? */
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
index c9a038a5e2d3..4e246c7e78f2 100644
--- a/sound/soc/au1x/dma.c
+++ b/sound/soc/au1x/dma.c
@@ -195,7 +195,7 @@ static int alchemy_pcm_open(struct snd_soc_component *component,
int *dmaids, s = substream->stream;
char *name;
- dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
if (!dmaids)
return -ENODEV; /* whoa, has ordering changed? */
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 0227993c5da8..05eb36991f14 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -58,7 +58,7 @@ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x)
{
struct snd_soc_card *c = x->bus->card->private_data;
- return snd_soc_dai_get_drvdata(c->rtd->cpu_dai);
+ return snd_soc_dai_get_drvdata(c->asoc_rtd_to_cpu(rtd, 0));
}
#else
diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig
index 0037e96aa228..4218057b0874 100644
--- a/sound/soc/bcm/Kconfig
+++ b/sound/soc/bcm/Kconfig
@@ -17,3 +17,12 @@ config SND_SOC_CYGNUS
Cygnus chips (bcm958300, bcm958305, bcm911360)
If you don't know what to do here, say N.
+
+config SND_BCM63XX_I2S_WHISTLER
+ tristate "SoC Audio support for the Broadcom BCM63XX I2S module"
+ select REGMAP_MMIO
+ help
+ Say Y if you want to add support for ASoC audio on Broadcom
+ DSL/PON chips (bcm63158, bcm63178)
+
+ If you don't know what to do here, say N
diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile
index b81fa421ec27..7c2d7899603b 100644
--- a/sound/soc/bcm/Makefile
+++ b/sound/soc/bcm/Makefile
@@ -9,3 +9,7 @@ snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o
obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o
+# BCM63XX Platform Support
+snd-soc-63xx-objs := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o
+
+obj-$(CONFIG_SND_BCM63XX_I2S_WHISTLER) += snd-soc-63xx.o \ No newline at end of file
diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c
new file mode 100644
index 000000000000..246a57ac6679
--- /dev/null
+++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c
@@ -0,0 +1,317 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+// linux/sound/bcm/bcm63xx-i2s-whistler.c
+// BCM63xx whistler i2s driver
+// Copyright (c) 2020 Broadcom Corporation
+// Author: Kevin-Ke Li <kevin-ke.li@broadcom.com>
+
+#include <linux/clk.h>
+#include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "bcm63xx-i2s.h"
+
+#define DRV_NAME "brcm-i2s"
+
+static bool brcm_i2s_wr_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case I2S_TX_CFG ... I2S_TX_DESC_IFF_LEN:
+ case I2S_TX_CFG_2 ... I2S_RX_DESC_IFF_LEN:
+ case I2S_RX_CFG_2 ... I2S_REG_MAX:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool brcm_i2s_rd_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case I2S_TX_CFG ... I2S_REG_MAX:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool brcm_i2s_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case I2S_TX_CFG:
+ case I2S_TX_IRQ_CTL:
+ case I2S_TX_DESC_IFF_ADDR:
+ case I2S_TX_DESC_IFF_LEN:
+ case I2S_TX_DESC_OFF_ADDR:
+ case I2S_TX_DESC_OFF_LEN:
+ case I2S_TX_CFG_2:
+ case I2S_RX_CFG:
+ case I2S_RX_IRQ_CTL:
+ case I2S_RX_DESC_OFF_ADDR:
+ case I2S_RX_DESC_OFF_LEN:
+ case I2S_RX_DESC_IFF_LEN:
+ case I2S_RX_DESC_IFF_ADDR:
+ case I2S_RX_CFG_2:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config brcm_i2s_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = I2S_REG_MAX,
+ .writeable_reg = brcm_i2s_wr_reg,
+ .readable_reg = brcm_i2s_rd_reg,
+ .volatile_reg = brcm_i2s_volatile_reg,
+ .cache_type = REGCACHE_FLAT,
+};
+
+static int bcm63xx_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int ret = 0;
+ struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai);
+
+ ret = clk_set_rate(i2s_priv->i2s_clk, params_rate(params));
+ if (ret < 0)
+ dev_err(i2s_priv->dev,
+ "Can't set sample rate, err: %d\n", ret);
+
+ return ret;
+}
+
+static int bcm63xx_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ unsigned int slavemode;
+ struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai);
+ struct regmap *regmap_i2s = i2s_priv->regmap_i2s;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ regmap_update_bits(regmap_i2s, I2S_TX_CFG,
+ I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT |
+ I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE,
+ I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT |
+ I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE);
+ regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 0);
+ regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 0);
+ regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 1);
+
+ /* TX and RX block each have an independent bit to indicate
+ * if it is generating the clock for the I2S bus. The bus
+ * clocks need to be generated from either the TX or RX block,
+ * but not both
+ */
+ regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode);
+ if (slavemode & I2S_RX_SLAVE_MODE_MASK)
+ regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+ I2S_TX_SLAVE_MODE_MASK,
+ I2S_TX_MASTER_MODE);
+ else
+ regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+ I2S_TX_SLAVE_MODE_MASK,
+ I2S_TX_SLAVE_MODE);
+ } else {
+ regmap_update_bits(regmap_i2s, I2S_RX_CFG,
+ I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT |
+ I2S_RX_CLOCK_ENABLE,
+ I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT |
+ I2S_RX_CLOCK_ENABLE);
+ regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 0);
+ regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 0);
+ regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 1);
+
+ regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode);
+ if (slavemode & I2S_TX_SLAVE_MODE_MASK)
+ regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+ I2S_RX_SLAVE_MODE_MASK, 0);
+ else
+ regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+ I2S_RX_SLAVE_MODE_MASK,
+ I2S_RX_SLAVE_MODE);
+ }
+ return 0;
+}
+
+static void bcm63xx_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ unsigned int enabled, slavemode;
+ struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai);
+ struct regmap *regmap_i2s = i2s_priv->regmap_i2s;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ regmap_update_bits(regmap_i2s, I2S_TX_CFG,
+ I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT |
+ I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, 0);
+ regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 1);
+ regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 4);
+ regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 4);
+
+ regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode);
+ slavemode = slavemode & I2S_TX_SLAVE_MODE_MASK;
+ if (!slavemode) {
+ regmap_read(regmap_i2s, I2S_RX_CFG, &enabled);
+ enabled = enabled & I2S_RX_ENABLE_MASK;
+ if (enabled)
+ regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+ I2S_RX_SLAVE_MODE_MASK,
+ I2S_RX_MASTER_MODE);
+ }
+ regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+ I2S_TX_SLAVE_MODE_MASK,
+ I2S_TX_SLAVE_MODE);
+ } else {
+ regmap_update_bits(regmap_i2s, I2S_RX_CFG,
+ I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT |
+ I2S_RX_CLOCK_ENABLE, 0);
+ regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 1);
+ regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 4);
+ regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 4);
+
+ regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode);
+ slavemode = slavemode & I2S_RX_SLAVE_MODE_MASK;
+ if (!slavemode) {
+ regmap_read(regmap_i2s, I2S_TX_CFG, &enabled);
+ enabled = enabled & I2S_TX_ENABLE_MASK;
+ if (enabled)
+ regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+ I2S_TX_SLAVE_MODE_MASK,
+ I2S_TX_MASTER_MODE);
+ }
+
+ regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+ I2S_RX_SLAVE_MODE_MASK, I2S_RX_SLAVE_MODE);
+ }
+}
+
+static const struct snd_soc_dai_ops bcm63xx_i2s_dai_ops = {
+ .startup = bcm63xx_i2s_startup,
+ .shutdown = bcm63xx_i2s_shutdown,
+ .hw_params = bcm63xx_i2s_hw_params,
+};
+
+static struct snd_soc_dai_driver bcm63xx_i2s_dai = {
+ .name = DRV_NAME,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &bcm63xx_i2s_dai_ops,
+ .symmetric_rates = 1,
+ .symmetric_channels = 1,
+};
+
+static const struct snd_soc_component_driver bcm63xx_i2s_component = {
+ .name = "bcm63xx",
+};
+
+static int bcm63xx_i2s_dev_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+ void __iomem *regs;
+ struct resource *r_mem, *region;
+ struct bcm_i2s_priv *i2s_priv;
+ struct regmap *regmap_i2s;
+ struct clk *i2s_clk;
+
+ i2s_priv = devm_kzalloc(&pdev->dev, sizeof(*i2s_priv), GFP_KERNEL);
+ if (!i2s_priv)
+ return -ENOMEM;
+
+ i2s_clk = devm_clk_get(&pdev->dev, "i2sclk");
+ if (IS_ERR(i2s_clk)) {
+ dev_err(&pdev->dev, "%s: cannot get a brcm clock: %ld\n",
+ __func__, PTR_ERR(i2s_clk));
+ return PTR_ERR(i2s_clk);
+ }
+
+ r_mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r_mem) {
+ dev_err(&pdev->dev, "Unable to get register resource.\n");
+ return -ENODEV;
+ }
+
+ region = devm_request_mem_region(&pdev->dev, r_mem->start,
+ resource_size(r_mem), DRV_NAME);
+ if (!region) {
+ dev_err(&pdev->dev, "Memory region already claimed\n");
+ return -EBUSY;
+ }
+
+ regs = devm_ioremap_resource(&pdev->dev, r_mem);
+ if (IS_ERR(regs)) {
+ ret = PTR_ERR(regs);
+ return ret;
+ }
+
+ regmap_i2s = devm_regmap_init_mmio(&pdev->dev,
+ regs, &brcm_i2s_regmap_config);
+ if (IS_ERR(regmap_i2s))
+ return PTR_ERR(regmap_i2s);
+
+ regmap_update_bits(regmap_i2s, I2S_MISC_CFG,
+ I2S_PAD_LVL_LOOP_DIS_MASK,
+ I2S_PAD_LVL_LOOP_DIS_ENABLE);
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &bcm63xx_i2s_component,
+ &bcm63xx_i2s_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to register the dai\n");
+ return ret;
+ }
+
+ i2s_priv->dev = &pdev->dev;
+ i2s_priv->i2s_clk = i2s_clk;
+ i2s_priv->regmap_i2s = regmap_i2s;
+ dev_set_drvdata(&pdev->dev, i2s_priv);
+
+ ret = bcm63xx_soc_platform_probe(pdev, i2s_priv);
+ if (ret)
+ dev_err(&pdev->dev, "failed to register the pcm\n");
+
+ return ret;
+}
+
+static int bcm63xx_i2s_dev_remove(struct platform_device *pdev)
+{
+ bcm63xx_soc_platform_remove(pdev);
+ return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id snd_soc_bcm_audio_match[] = {
+ {.compatible = "brcm,bcm63xx-i2s"},
+ { }
+};
+#endif
+
+static struct platform_driver bcm63xx_i2s_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .of_match_table = of_match_ptr(snd_soc_bcm_audio_match),
+ },
+ .probe = bcm63xx_i2s_dev_probe,
+ .remove = bcm63xx_i2s_dev_remove,
+};
+
+module_platform_driver(bcm63xx_i2s_driver);
+
+MODULE_AUTHOR("Kevin,Li <kevin-ke.li@broadcom.com>");
+MODULE_DESCRIPTION("Broadcom DSL XPON ASOC I2S Interface");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/bcm/bcm63xx-i2s.h b/sound/soc/bcm/bcm63xx-i2s.h
new file mode 100644
index 000000000000..edc328ba53d3
--- /dev/null
+++ b/sound/soc/bcm/bcm63xx-i2s.h
@@ -0,0 +1,90 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+// linux/sound/soc/bcm/bcm63xx-i2s.h
+// Copyright (c) 2020 Broadcom Corporation
+// Author: Kevin-Ke Li <kevin-ke.li@broadcom.com>
+
+#ifndef __BCM63XX_I2S_H
+#define __BCM63XX_I2S_H
+
+#define I2S_DESC_FIFO_DEPTH 8
+#define I2S_MISC_CFG (0x003C)
+#define I2S_PAD_LVL_LOOP_DIS_MASK (1 << 2)
+#define I2S_PAD_LVL_LOOP_DIS_ENABLE I2S_PAD_LVL_LOOP_DIS_MASK
+
+#define I2S_TX_ENABLE_MASK (1 << 31)
+#define I2S_TX_ENABLE I2S_TX_ENABLE_MASK
+#define I2S_TX_OUT_R (1 << 19)
+#define I2S_TX_DATA_ALIGNMENT (1 << 2)
+#define I2S_TX_DATA_ENABLE (1 << 1)
+#define I2S_TX_CLOCK_ENABLE (1 << 0)
+
+#define I2S_TX_DESC_OFF_LEVEL_SHIFT 12
+#define I2S_TX_DESC_OFF_LEVEL_MASK (0x0F << I2S_TX_DESC_OFF_LEVEL_SHIFT)
+#define I2S_TX_DESC_IFF_LEVEL_SHIFT 8
+#define I2S_TX_DESC_IFF_LEVEL_MASK (0x0F << I2S_TX_DESC_IFF_LEVEL_SHIFT)
+#define I2S_TX_DESC_OFF_INTR_EN_MSK (1 << 1)
+#define I2S_TX_DESC_OFF_INTR_EN I2S_TX_DESC_OFF_INTR_EN_MSK
+
+#define I2S_TX_CFG (0x0000)
+#define I2S_TX_IRQ_CTL (0x0004)
+#define I2S_TX_IRQ_EN (0x0008)
+#define I2S_TX_IRQ_IFF_THLD (0x000c)
+#define I2S_TX_IRQ_OFF_THLD (0x0010)
+#define I2S_TX_DESC_IFF_ADDR (0x0014)
+#define I2S_TX_DESC_IFF_LEN (0x0018)
+#define I2S_TX_DESC_OFF_ADDR (0x001C)
+#define I2S_TX_DESC_OFF_LEN (0x0020)
+#define I2S_TX_CFG_2 (0x0024)
+#define I2S_TX_SLAVE_MODE_SHIFT 13
+#define I2S_TX_SLAVE_MODE_MASK (1 << I2S_TX_SLAVE_MODE_SHIFT)
+#define I2S_TX_SLAVE_MODE I2S_TX_SLAVE_MODE_MASK
+#define I2S_TX_MASTER_MODE 0
+#define I2S_TX_INTR_MASK 0x0F
+
+#define I2S_RX_ENABLE_MASK (1 << 31)
+#define I2S_RX_ENABLE I2S_RX_ENABLE_MASK
+#define I2S_RX_IN_R (1 << 19)
+#define I2S_RX_DATA_ALIGNMENT (1 << 2)
+#define I2S_RX_CLOCK_ENABLE (1 << 0)
+
+#define I2S_RX_DESC_OFF_LEVEL_SHIFT 12
+#define I2S_RX_DESC_OFF_LEVEL_MASK (0x0F << I2S_RX_DESC_OFF_LEVEL_SHIFT)
+#define I2S_RX_DESC_IFF_LEVEL_SHIFT 8
+#define I2S_RX_DESC_IFF_LEVEL_MASK (0x0F << I2S_RX_DESC_IFF_LEVEL_SHIFT)
+#define I2S_RX_DESC_OFF_INTR_EN_MSK (1 << 1)
+#define I2S_RX_DESC_OFF_INTR_EN I2S_RX_DESC_OFF_INTR_EN_MSK
+
+#define I2S_RX_CFG (0x0040) /* 20c0 */
+#define I2S_RX_IRQ_CTL (0x0044)
+#define I2S_RX_IRQ_EN (0x0048)
+#define I2S_RX_IRQ_IFF_THLD (0x004C)
+#define I2S_RX_IRQ_OFF_THLD (0x0050)
+#define I2S_RX_DESC_IFF_ADDR (0x0054)
+#define I2S_RX_DESC_IFF_LEN (0x0058)
+#define I2S_RX_DESC_OFF_ADDR (0x005C)
+#define I2S_RX_DESC_OFF_LEN (0x0060)
+#define I2S_RX_CFG_2 (0x0064)
+#define I2S_RX_SLAVE_MODE_SHIFT 13
+#define I2S_RX_SLAVE_MODE_MASK (1 << I2S_RX_SLAVE_MODE_SHIFT)
+#define I2S_RX_SLAVE_MODE I2S_RX_SLAVE_MODE_MASK
+#define I2S_RX_MASTER_MODE 0
+#define I2S_RX_INTR_MASK 0x0F
+
+#define I2S_REG_MAX 0x007C
+
+struct bcm_i2s_priv {
+ struct device *dev;
+ struct resource *r_irq;
+ struct regmap *regmap_i2s;
+ struct clk *i2s_clk;
+ struct snd_pcm_substream *play_substream;
+ struct snd_pcm_substream *capture_substream;
+ struct i2s_dma_desc *play_dma_desc;
+ struct i2s_dma_desc *capture_dma_desc;
+};
+
+extern int bcm63xx_soc_platform_probe(struct platform_device *pdev,
+ struct bcm_i2s_priv *i2s_priv);
+extern int bcm63xx_soc_platform_remove(struct platform_device *pdev);
+
+#endif
diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c
new file mode 100644
index 000000000000..e46c390683e7
--- /dev/null
+++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c
@@ -0,0 +1,485 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+// linux/sound/bcm/bcm63xx-pcm-whistler.c
+// BCM63xx whistler pcm interface
+// Copyright (c) 2020 Broadcom Corporation
+// Author: Kevin-Ke Li <kevin-ke.li@broadcom.com>
+
+#include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <linux/regmap.h>
+#include <linux/of_device.h>
+#include <sound/soc.h>
+#include "bcm63xx-i2s.h"
+
+
+struct i2s_dma_desc {
+ unsigned char *dma_area;
+ dma_addr_t dma_addr;
+ unsigned int dma_len;
+};
+
+struct bcm63xx_runtime_data {
+ int dma_len;
+ dma_addr_t dma_addr;
+ dma_addr_t dma_addr_next;
+};
+
+static const struct snd_pcm_hardware bcm63xx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE, /* support S32 only */
+ .period_bytes_max = 8192 - 32,
+ .periods_min = 1,
+ .periods_max = PAGE_SIZE/sizeof(struct i2s_dma_desc),
+ .buffer_bytes_max = 128 * 1024,
+ .fifo_size = 32,
+};
+
+static int bcm63xx_pcm_hw_params(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct i2s_dma_desc *dma_desc;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ dma_desc = kzalloc(sizeof(*dma_desc), GFP_NOWAIT);
+ if (!dma_desc)
+ return -ENOMEM;
+
+ snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_desc);
+
+ return 0;
+}
+
+static int bcm63xx_pcm_hw_free(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct i2s_dma_desc *dma_desc;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+ kfree(dma_desc);
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ return 0;
+}
+
+static int bcm63xx_pcm_trigger(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_soc_pcm_runtime *rtd;
+ struct bcm_i2s_priv *i2s_priv;
+ struct regmap *regmap_i2s;
+
+ rtd = substream->private_data;
+ i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev);
+ regmap_i2s = i2s_priv->regmap_i2s;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ regmap_update_bits(regmap_i2s,
+ I2S_TX_IRQ_EN,
+ I2S_TX_DESC_OFF_INTR_EN,
+ I2S_TX_DESC_OFF_INTR_EN);
+ regmap_update_bits(regmap_i2s,
+ I2S_TX_CFG,
+ I2S_TX_ENABLE_MASK,
+ I2S_TX_ENABLE);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ regmap_write(regmap_i2s,
+ I2S_TX_IRQ_EN,
+ 0);
+ regmap_update_bits(regmap_i2s,
+ I2S_TX_CFG,
+ I2S_TX_ENABLE_MASK,
+ 0);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ } else {
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ regmap_update_bits(regmap_i2s,
+ I2S_RX_IRQ_EN,
+ I2S_RX_DESC_OFF_INTR_EN_MSK,
+ I2S_RX_DESC_OFF_INTR_EN);
+ regmap_update_bits(regmap_i2s,
+ I2S_RX_CFG,
+ I2S_RX_ENABLE_MASK,
+ I2S_RX_ENABLE);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ regmap_update_bits(regmap_i2s,
+ I2S_RX_IRQ_EN,
+ I2S_RX_DESC_OFF_INTR_EN_MSK,
+ 0);
+ regmap_update_bits(regmap_i2s,
+ I2S_RX_CFG,
+ I2S_RX_ENABLE_MASK,
+ 0);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ }
+ return ret;
+}
+
+static int bcm63xx_pcm_prepare(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct i2s_dma_desc *dma_desc;
+ struct regmap *regmap_i2s;
+ struct bcm_i2s_priv *i2s_priv;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ uint32_t regaddr_desclen, regaddr_descaddr;
+
+ dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+ dma_desc->dma_len = snd_pcm_lib_period_bytes(substream);
+ dma_desc->dma_addr = runtime->dma_addr;
+ dma_desc->dma_area = runtime->dma_area;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ regaddr_desclen = I2S_TX_DESC_IFF_LEN;
+ regaddr_descaddr = I2S_TX_DESC_IFF_ADDR;
+ } else {
+ regaddr_desclen = I2S_RX_DESC_IFF_LEN;
+ regaddr_descaddr = I2S_RX_DESC_IFF_ADDR;
+ }
+
+ i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev);
+ regmap_i2s = i2s_priv->regmap_i2s;
+
+ regmap_write(regmap_i2s, regaddr_desclen, dma_desc->dma_len);
+ regmap_write(regmap_i2s, regaddr_descaddr, dma_desc->dma_addr);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+bcm63xx_pcm_pointer(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ snd_pcm_uframes_t x;
+ struct bcm63xx_runtime_data *prtd = substream->runtime->private_data;
+
+ if ((void *)prtd->dma_addr_next == NULL)
+ prtd->dma_addr_next = substream->runtime->dma_addr;
+
+ x = bytes_to_frames(substream->runtime,
+ prtd->dma_addr_next - substream->runtime->dma_addr);
+
+ return x == substream->runtime->buffer_size ? 0 : x;
+}
+
+static int bcm63xx_pcm_mmap(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_wc(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+
+}
+
+static int bcm63xx_pcm_open(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct bcm63xx_runtime_data *prtd;
+
+ runtime->hw = bcm63xx_pcm_hardware;
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret)
+ goto out;
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret)
+ goto out;
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ ret = -ENOMEM;
+ prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+ if (!prtd)
+ goto out;
+
+ runtime->private_data = prtd;
+ return 0;
+out:
+ return ret;
+}
+
+static int bcm63xx_pcm_close(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct bcm63xx_runtime_data *prtd = runtime->private_data;
+
+ kfree(prtd);
+ return 0;
+}
+
+static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv)
+{
+ unsigned int availdepth, ifflevel, offlevel, int_status, val_1, val_2;
+ struct bcm63xx_runtime_data *prtd;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ struct regmap *regmap_i2s;
+ struct i2s_dma_desc *dma_desc;
+ struct snd_soc_pcm_runtime *rtd;
+ struct bcm_i2s_priv *i2s_priv;
+
+ i2s_priv = (struct bcm_i2s_priv *)bcm_i2s_priv;
+ regmap_i2s = i2s_priv->regmap_i2s;
+
+ /* rx */
+ regmap_read(regmap_i2s, I2S_RX_IRQ_CTL, &int_status);
+
+ if (int_status & I2S_RX_DESC_OFF_INTR_EN_MSK) {
+ substream = i2s_priv->capture_substream;
+ runtime = substream->runtime;
+ rtd = substream->private_data;
+ prtd = runtime->private_data;
+ dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+
+ offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >>
+ I2S_RX_DESC_OFF_LEVEL_SHIFT;
+ while (offlevel) {
+ regmap_read(regmap_i2s, I2S_RX_DESC_OFF_ADDR, &val_1);
+ regmap_read(regmap_i2s, I2S_RX_DESC_OFF_LEN, &val_2);
+ offlevel--;
+ }
+ prtd->dma_addr_next = val_1 + val_2;
+ ifflevel = (int_status & I2S_RX_DESC_IFF_LEVEL_MASK) >>
+ I2S_RX_DESC_IFF_LEVEL_SHIFT;
+
+ availdepth = I2S_DESC_FIFO_DEPTH - ifflevel;
+ while (availdepth) {
+ dma_desc->dma_addr +=
+ snd_pcm_lib_period_bytes(substream);
+ dma_desc->dma_area +=
+ snd_pcm_lib_period_bytes(substream);
+ if (dma_desc->dma_addr - runtime->dma_addr >=
+ runtime->dma_bytes) {
+ dma_desc->dma_addr = runtime->dma_addr;
+ dma_desc->dma_area = runtime->dma_area;
+ }
+
+ prtd->dma_addr = dma_desc->dma_addr;
+ regmap_write(regmap_i2s, I2S_RX_DESC_IFF_LEN,
+ snd_pcm_lib_period_bytes(substream));
+ regmap_write(regmap_i2s, I2S_RX_DESC_IFF_ADDR,
+ dma_desc->dma_addr);
+ availdepth--;
+ }
+
+ snd_pcm_period_elapsed(substream);
+
+ /* Clear interrupt by writing 0 */
+ regmap_update_bits(regmap_i2s, I2S_RX_IRQ_CTL,
+ I2S_RX_INTR_MASK, 0);
+ }
+
+ /* tx */
+ regmap_read(regmap_i2s, I2S_TX_IRQ_CTL, &int_status);
+
+ if (int_status & I2S_TX_DESC_OFF_INTR_EN_MSK) {
+ substream = i2s_priv->play_substream;
+ runtime = substream->runtime;
+ rtd = substream->private_data;
+ prtd = runtime->private_data;
+ dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+
+ offlevel = (int_status & I2S_TX_DESC_OFF_LEVEL_MASK) >>
+ I2S_TX_DESC_OFF_LEVEL_SHIFT;
+ while (offlevel) {
+ regmap_read(regmap_i2s, I2S_TX_DESC_OFF_ADDR, &val_1);
+ regmap_read(regmap_i2s, I2S_TX_DESC_OFF_LEN, &val_2);
+ prtd->dma_addr_next = val_1 + val_2;
+ offlevel--;
+ }
+
+ ifflevel = (int_status & I2S_TX_DESC_IFF_LEVEL_MASK) >>
+ I2S_TX_DESC_IFF_LEVEL_SHIFT;
+ availdepth = I2S_DESC_FIFO_DEPTH - ifflevel;
+
+ while (availdepth) {
+ dma_desc->dma_addr +=
+ snd_pcm_lib_period_bytes(substream);
+ dma_desc->dma_area +=
+ snd_pcm_lib_period_bytes(substream);
+
+ if (dma_desc->dma_addr - runtime->dma_addr >=
+ runtime->dma_bytes) {
+ dma_desc->dma_addr = runtime->dma_addr;
+ dma_desc->dma_area = runtime->dma_area;
+ }
+
+ prtd->dma_addr = dma_desc->dma_addr;
+ regmap_write(regmap_i2s, I2S_TX_DESC_IFF_LEN,
+ snd_pcm_lib_period_bytes(substream));
+ regmap_write(regmap_i2s, I2S_TX_DESC_IFF_ADDR,
+ dma_desc->dma_addr);
+ availdepth--;
+ }
+
+ snd_pcm_period_elapsed(substream);
+
+ /* Clear interrupt by writing 0 */
+ regmap_update_bits(regmap_i2s, I2S_TX_IRQ_CTL,
+ I2S_TX_INTR_MASK, 0);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static int bcm63xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = bcm63xx_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+
+ buf->area = dma_alloc_wc(pcm->card->dev,
+ size, &buf->addr,
+ GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+ return 0;
+}
+
+static int bcm63xx_soc_pcm_new(struct snd_soc_component *component,
+ struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ struct bcm_i2s_priv *i2s_priv;
+ int ret;
+
+ i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev);
+
+ of_dma_configure(pcm->card->dev, pcm->card->dev->of_node, 1);
+
+ ret = dma_coerce_mask_and_coherent(pcm->card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ goto out;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = bcm63xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+
+ i2s_priv->play_substream =
+ pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = bcm63xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ i2s_priv->capture_substream =
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ }
+
+out:
+ return ret;
+}
+
+static void bcm63xx_pcm_free_dma_buffers(struct snd_soc_component *component,
+ struct snd_pcm *pcm)
+{
+ int stream;
+ struct snd_dma_buffer *buf;
+ struct snd_pcm_substream *substream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ dma_free_wc(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static const struct snd_soc_component_driver bcm63xx_soc_platform = {
+ .open = bcm63xx_pcm_open,
+ .close = bcm63xx_pcm_close,
+ .hw_params = bcm63xx_pcm_hw_params,
+ .hw_free = bcm63xx_pcm_hw_free,
+ .prepare = bcm63xx_pcm_prepare,
+ .trigger = bcm63xx_pcm_trigger,
+ .pointer = bcm63xx_pcm_pointer,
+ .mmap = bcm63xx_pcm_mmap,
+ .pcm_construct = bcm63xx_soc_pcm_new,
+ .pcm_destruct = bcm63xx_pcm_free_dma_buffers,
+};
+
+int bcm63xx_soc_platform_probe(struct platform_device *pdev,
+ struct bcm_i2s_priv *i2s_priv)
+{
+ int ret;
+
+ i2s_priv->r_irq = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+ if (!i2s_priv->r_irq) {
+ dev_err(&pdev->dev, "Unable to get register irq resource.\n");
+ return -ENODEV;
+ }
+
+ ret = devm_request_irq(&pdev->dev, i2s_priv->r_irq->start, i2s_dma_isr,
+ i2s_priv->r_irq->flags, "i2s_dma", (void *)i2s_priv);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "i2s_init: failed to request interrupt.ret=%d\n", ret);
+ return ret;
+ }
+
+ return devm_snd_soc_register_component(&pdev->dev,
+ &bcm63xx_soc_platform, NULL, 0);
+}
+
+int bcm63xx_soc_platform_remove(struct platform_device *pdev)
+{
+ return 0;
+}
+
+MODULE_AUTHOR("Kevin,Li <kevin-ke.li@broadcom.com>");
+MODULE_DESCRIPTION("Broadcom DSL XPON ASOC PCM Interface");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c
index 3a80c613bc3f..f96d27c8b301 100644
--- a/sound/soc/bcm/cygnus-pcm.c
+++ b/sound/soc/bcm/cygnus-pcm.c
@@ -209,7 +209,7 @@ static struct cygnus_aio_port *cygnus_dai_get_dma_data(
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream);
+ return snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(soc_runtime, 0), substream);
}
static void ringbuf_set_initial(void __iomem *audio_io,
@@ -359,7 +359,7 @@ static void disable_intr(struct snd_pcm_substream *substream)
aio = cygnus_dai_get_dma_data(substream);
- dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum);
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s on port %d\n", __func__, aio->portnum);
/* The port number maps to the bit position to be set */
set_mask = BIT(aio->portnum);
@@ -590,7 +590,7 @@ static int cygnus_pcm_open(struct snd_soc_component *component,
if (!aio)
return -ENODEV;
- dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw);
@@ -623,7 +623,7 @@ static int cygnus_pcm_close(struct snd_soc_component *component,
aio = cygnus_dai_get_dma_data(substream);
- dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
aio->play_stream = NULL;
@@ -631,7 +631,7 @@ static int cygnus_pcm_close(struct snd_soc_component *component,
aio->capture_stream = NULL;
if (!aio->play_stream && !aio->capture_stream)
- dev_dbg(rtd->cpu_dai->dev, "freed port %d\n", aio->portnum);
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "freed port %d\n", aio->portnum);
return 0;
}
@@ -645,7 +645,7 @@ static int cygnus_pcm_hw_params(struct snd_soc_component *component,
struct cygnus_aio_port *aio;
aio = cygnus_dai_get_dma_data(substream);
- dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
@@ -660,7 +660,7 @@ static int cygnus_pcm_hw_free(struct snd_soc_component *component,
struct cygnus_aio_port *aio;
aio = cygnus_dai_get_dma_data(substream);
- dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
@@ -678,12 +678,12 @@ static int cygnus_pcm_prepare(struct snd_soc_component *component,
struct ringbuf_regs *p_rbuf = NULL;
aio = cygnus_dai_get_dma_data(substream);
- dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
bufsize = snd_pcm_lib_buffer_bytes(substream);
periodsize = snd_pcm_lib_period_bytes(substream);
- dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n",
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s (buf_size %lu) (period_size %lu)\n",
__func__, bufsize, periodsize);
configure_ringbuf_regs(substream);
@@ -745,11 +745,11 @@ static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
buf->area = dma_alloc_coherent(pcm->card->dev, size,
&buf->addr, GFP_KERNEL);
- dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%zx @ %pK\n",
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: size 0x%zx @ %pK\n",
__func__, size, buf->area);
if (!buf->area) {
- dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__);
+ dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: dma_alloc failed\n", __func__);
return -ENOMEM;
}
buf->bytes = size;
diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c
index 10961190068e..ccf65f087ea6 100644
--- a/sound/soc/cirrus/edb93xx.c
+++ b/sound/soc/cirrus/edb93xx.c
@@ -23,8 +23,8 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int err;
unsigned int mclk_rate;
unsigned int rate = params_rate(params);
diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c
index 70c2f3e08d6d..cb133e80b7c3 100644
--- a/sound/soc/cirrus/snappercl15.c
+++ b/sound/soc/cirrus/snappercl15.c
@@ -23,8 +23,8 @@ static int snappercl15_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int err;
err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK,
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index ea912439e446..e6a0c5d05fa5 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -14,262 +14,264 @@ menu "CODEC drivers"
config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
depends on COMPILE_TEST
- select SND_SOC_88PM860X if MFD_88PM860X
- select SND_SOC_L3
- select SND_SOC_AB8500_CODEC if ABX500_CORE
- select SND_SOC_AC97_CODEC
- select SND_SOC_AD1836 if SPI_MASTER
- select SND_SOC_AD193X_SPI if SPI_MASTER
- select SND_SOC_AD193X_I2C if I2C
- select SND_SOC_AD1980 if SND_SOC_AC97_BUS
- select SND_SOC_AD73311
- select SND_SOC_ADAU1373 if I2C
- select SND_SOC_ADAU1761_I2C if I2C
- select SND_SOC_ADAU1761_SPI if SPI
- select SND_SOC_ADAU1781_I2C if I2C
- select SND_SOC_ADAU1781_SPI if SPI
- select SND_SOC_ADAV801 if SPI_MASTER
- select SND_SOC_ADAV803 if I2C
- select SND_SOC_ADAU1977_SPI if SPI_MASTER
- select SND_SOC_ADAU1977_I2C if I2C
- select SND_SOC_ADAU1701 if I2C
- select SND_SOC_ADAU7002
- select SND_SOC_ADAU7118_I2C if I2C
- select SND_SOC_ADAU7118_HW
- select SND_SOC_ADS117X
- select SND_SOC_AK4104 if SPI_MASTER
- select SND_SOC_AK4118 if I2C
- select SND_SOC_AK4458 if I2C
- select SND_SOC_AK4535 if I2C
- select SND_SOC_AK4554
- select SND_SOC_AK4613 if I2C
- select SND_SOC_AK4641 if I2C
- select SND_SOC_AK4642 if I2C
- select SND_SOC_AK4671 if I2C
- select SND_SOC_AK5386
- select SND_SOC_AK5558 if I2C
- select SND_SOC_ALC5623 if I2C
- select SND_SOC_ALC5632 if I2C
- select SND_SOC_BT_SCO
- select SND_SOC_BD28623
- select SND_SOC_CQ0093VC
- select SND_SOC_CROS_EC_CODEC if CROS_EC
- select SND_SOC_CS35L32 if I2C
- select SND_SOC_CS35L33 if I2C
- select SND_SOC_CS35L34 if I2C
- select SND_SOC_CS35L35 if I2C
- select SND_SOC_CS35L36 if I2C
- select SND_SOC_CS42L42 if I2C
- select SND_SOC_CS42L51_I2C if I2C
- select SND_SOC_CS42L52 if I2C && INPUT
- select SND_SOC_CS42L56 if I2C && INPUT
- select SND_SOC_CS42L73 if I2C
- select SND_SOC_CS4265 if I2C
- select SND_SOC_CS4270 if I2C
- select SND_SOC_CS4271_I2C if I2C
- select SND_SOC_CS4271_SPI if SPI_MASTER
- select SND_SOC_CS42XX8_I2C if I2C
- select SND_SOC_CS43130 if I2C
- select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI
- select SND_SOC_CS4349 if I2C
- select SND_SOC_CS47L15 if MFD_CS47L15
- select SND_SOC_CS47L24 if MFD_CS47L24
- select SND_SOC_CS47L35 if MFD_CS47L35
- select SND_SOC_CS47L85 if MFD_CS47L85
- select SND_SOC_CS47L90 if MFD_CS47L90
- select SND_SOC_CS47L92 if MFD_CS47L92
- select SND_SOC_CS53L30 if I2C
- select SND_SOC_CX20442 if TTY
- select SND_SOC_CX2072X if I2C
- select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
- select SND_SOC_DA7213 if I2C
- select SND_SOC_DA7218 if I2C
- select SND_SOC_DA7219 if I2C
- select SND_SOC_DA732X if I2C
- select SND_SOC_DA9055 if I2C
- select SND_SOC_DMIC if GPIOLIB
- select SND_SOC_ES8316 if I2C
- select SND_SOC_ES8328_SPI if SPI_MASTER
- select SND_SOC_ES8328_I2C if I2C
- select SND_SOC_ES7134
- select SND_SOC_ES7241
- select SND_SOC_GTM601
- select SND_SOC_HDAC_HDMI
- select SND_SOC_HDAC_HDA
- select SND_SOC_ICS43432
- select SND_SOC_INNO_RK3036
- select SND_SOC_ISABELLE if I2C
- select SND_SOC_JZ4740_CODEC
- select SND_SOC_JZ4725B_CODEC
- select SND_SOC_JZ4770_CODEC
- select SND_SOC_LM4857 if I2C
- select SND_SOC_LM49453 if I2C
- select SND_SOC_LOCHNAGAR_SC if MFD_LOCHNAGAR
- select SND_SOC_MAX98088 if I2C
- select SND_SOC_MAX98090 if I2C
- select SND_SOC_MAX98095 if I2C
- select SND_SOC_MAX98357A if GPIOLIB
- select SND_SOC_MAX98371 if I2C
- select SND_SOC_MAX98504 if I2C
- select SND_SOC_MAX9867 if I2C
- select SND_SOC_MAX98925 if I2C
- select SND_SOC_MAX98926 if I2C
- select SND_SOC_MAX98927 if I2C
- select SND_SOC_MAX98373 if I2C
- select SND_SOC_MAX9850 if I2C
- select SND_SOC_MAX9860 if I2C
- select SND_SOC_MAX9759
- select SND_SOC_MAX9768 if I2C
- select SND_SOC_MAX9877 if I2C
- select SND_SOC_MC13783 if MFD_MC13XXX
- select SND_SOC_ML26124 if I2C
- select SND_SOC_MT6351 if MTK_PMIC_WRAP
- select SND_SOC_MT6358 if MTK_PMIC_WRAP
- select SND_SOC_MT6660 if I2C
- select SND_SOC_NAU8540 if I2C
- select SND_SOC_NAU8810 if I2C
- select SND_SOC_NAU8822 if I2C
- select SND_SOC_NAU8824 if I2C
- select SND_SOC_NAU8825 if I2C
- select SND_SOC_HDMI_CODEC
- select SND_SOC_PCM1681 if I2C
- select SND_SOC_PCM1789_I2C if I2C
- select SND_SOC_PCM179X_I2C if I2C
- select SND_SOC_PCM179X_SPI if SPI_MASTER
- select SND_SOC_PCM186X_I2C if I2C
- select SND_SOC_PCM186X_SPI if SPI_MASTER
- select SND_SOC_PCM3008
- select SND_SOC_PCM3060_I2C if I2C
- select SND_SOC_PCM3060_SPI if SPI_MASTER
- select SND_SOC_PCM3168A_I2C if I2C
- select SND_SOC_PCM3168A_SPI if SPI_MASTER
- select SND_SOC_PCM5102A
- select SND_SOC_PCM512x_I2C if I2C
- select SND_SOC_PCM512x_SPI if SPI_MASTER
- select SND_SOC_RK3328
- select SND_SOC_RT274 if I2C
- select SND_SOC_RT286 if I2C
- select SND_SOC_RT298 if I2C
- select SND_SOC_RT1011 if I2C
- select SND_SOC_RT1015 if I2C
- select SND_SOC_RT1305 if I2C
- select SND_SOC_RT1308 if I2C
- select SND_SOC_RT5514 if I2C
- select SND_SOC_RT5616 if I2C
- select SND_SOC_RT5631 if I2C
- select SND_SOC_RT5640 if I2C
- select SND_SOC_RT5645 if I2C
- select SND_SOC_RT5651 if I2C
- select SND_SOC_RT5659 if I2C
- select SND_SOC_RT5660 if I2C
- select SND_SOC_RT5663 if I2C
- select SND_SOC_RT5665 if I2C
- select SND_SOC_RT5668 if I2C
- select SND_SOC_RT5670 if I2C
- select SND_SOC_RT5677 if I2C && SPI_MASTER
- select SND_SOC_RT5682 if I2C
- select SND_SOC_RT700_SDW if SOUNDWIRE
- select SND_SOC_RT711_SDW if SOUNDWIRE
- select SND_SOC_RT715_SDW if SOUNDWIRE
- select SND_SOC_RT1308_SDW if SOUNDWIRE
- select SND_SOC_SGTL5000 if I2C
- select SND_SOC_SI476X if MFD_SI476X_CORE
- select SND_SOC_SIMPLE_AMPLIFIER
- select SND_SOC_SIRF_AUDIO_CODEC
- select SND_SOC_SPDIF
- select SND_SOC_SSM2305
- select SND_SOC_SSM2518 if I2C
- select SND_SOC_SSM2602_SPI if SPI_MASTER
- select SND_SOC_SSM2602_I2C if I2C
- select SND_SOC_SSM4567 if I2C
- select SND_SOC_STA32X if I2C
- select SND_SOC_STA350 if I2C
- select SND_SOC_STA529 if I2C
- select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
- select SND_SOC_STI_SAS
- select SND_SOC_TAS2552 if I2C
- select SND_SOC_TAS2562 if I2C
- select SND_SOC_TAS2770 if I2C
- select SND_SOC_TAS5086 if I2C
- select SND_SOC_TAS571X if I2C
- select SND_SOC_TAS5720 if I2C
- select SND_SOC_TAS6424 if I2C
- select SND_SOC_TDA7419 if I2C
- select SND_SOC_TFA9879 if I2C
- select SND_SOC_TLV320AIC23_I2C if I2C
- select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
- select SND_SOC_TLV320AIC26 if SPI_MASTER
- select SND_SOC_TLV320AIC31XX if I2C
- select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK
- select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK
- select SND_SOC_TLV320AIC3X if I2C
- select SND_SOC_TPA6130A2 if I2C
- select SND_SOC_TLV320DAC33 if I2C
- select SND_SOC_TSCS42XX if I2C
- select SND_SOC_TSCS454 if I2C
- select SND_SOC_TS3A227E if I2C
- select SND_SOC_TWL4030 if TWL4030_CORE
- select SND_SOC_TWL6040 if TWL6040_CORE
- select SND_SOC_UDA1334 if GPIOLIB
- select SND_SOC_UDA134X
- select SND_SOC_UDA1380 if I2C
- select SND_SOC_WCD9335 if SLIMBUS
- select SND_SOC_WCD934X if MFD_WCD934X && COMMON_CLK
- select SND_SOC_WL1273 if MFD_WL1273_CORE
- select SND_SOC_WM0010 if SPI_MASTER
- select SND_SOC_WM1250_EV1 if I2C
- select SND_SOC_WM2000 if I2C
- select SND_SOC_WM2200 if I2C
- select SND_SOC_WM5100 if I2C
- select SND_SOC_WM5102 if MFD_WM5102
- select SND_SOC_WM5110 if MFD_WM5110
- select SND_SOC_WM8350 if MFD_WM8350
- select SND_SOC_WM8400 if MFD_WM8400
- select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8523 if I2C
- select SND_SOC_WM8524 if GPIOLIB
- select SND_SOC_WM8580 if I2C
- select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8727
- select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8737 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8770 if SPI_MASTER
- select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8782
- select SND_SOC_WM8804_I2C if I2C
- select SND_SOC_WM8804_SPI if SPI_MASTER
- select SND_SOC_WM8900 if I2C
- select SND_SOC_WM8903 if I2C
- select SND_SOC_WM8904 if I2C
- select SND_SOC_WM8940 if I2C
- select SND_SOC_WM8955 if I2C
- select SND_SOC_WM8960 if I2C
- select SND_SOC_WM8961 if I2C
- select SND_SOC_WM8962 if I2C && INPUT
- select SND_SOC_WM8971 if I2C
- select SND_SOC_WM8974 if I2C
- select SND_SOC_WM8978 if I2C
- select SND_SOC_WM8983 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8990 if I2C
- select SND_SOC_WM8991 if I2C
- select SND_SOC_WM8993 if I2C
- select SND_SOC_WM8994 if MFD_WM8994
- select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
- select SND_SOC_WM8996 if I2C
- select SND_SOC_WM8997 if MFD_WM8997
- select SND_SOC_WM8998 if MFD_WM8998
- select SND_SOC_WM9081 if I2C
- select SND_SOC_WM9090 if I2C
- select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW)
- select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW)
- select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW)
- select SND_SOC_WSA881X if SOUNDWIRE
+ imply SND_SOC_88PM860X
+ imply SND_SOC_L3
+ imply SND_SOC_AB8500_CODEC
+ imply SND_SOC_AC97_CODEC
+ imply SND_SOC_AD1836
+ imply SND_SOC_AD193X_SPI
+ imply SND_SOC_AD193X_I2C
+ imply SND_SOC_AD1980
+ imply SND_SOC_AD73311
+ imply SND_SOC_ADAU1373
+ imply SND_SOC_ADAU1761_I2C
+ imply SND_SOC_ADAU1761_SPI
+ imply SND_SOC_ADAU1781_I2C
+ imply SND_SOC_ADAU1781_SPI
+ imply SND_SOC_ADAV801
+ imply SND_SOC_ADAV803
+ imply SND_SOC_ADAU1977_SPI
+ imply SND_SOC_ADAU1977_I2C
+ imply SND_SOC_ADAU1701
+ imply SND_SOC_ADAU7002
+ imply SND_SOC_ADAU7118_I2C
+ imply SND_SOC_ADAU7118_HW
+ imply SND_SOC_ADS117X
+ imply SND_SOC_AK4104
+ imply SND_SOC_AK4118
+ imply SND_SOC_AK4458
+ imply SND_SOC_AK4535
+ imply SND_SOC_AK4554
+ imply SND_SOC_AK4613
+ imply SND_SOC_AK4641
+ imply SND_SOC_AK4642
+ imply SND_SOC_AK4671
+ imply SND_SOC_AK5386
+ imply SND_SOC_AK5558
+ imply SND_SOC_ALC5623
+ imply SND_SOC_ALC5632
+ imply SND_SOC_BT_SCO
+ imply SND_SOC_BD28623
+ imply SND_SOC_CQ0093VC
+ imply SND_SOC_CROS_EC_CODEC
+ imply SND_SOC_CS35L32
+ imply SND_SOC_CS35L33
+ imply SND_SOC_CS35L34
+ imply SND_SOC_CS35L35
+ imply SND_SOC_CS35L36
+ imply SND_SOC_CS42L42
+ imply SND_SOC_CS42L51_I2C
+ imply SND_SOC_CS42L52
+ imply SND_SOC_CS42L56
+ imply SND_SOC_CS42L73
+ imply SND_SOC_CS4265
+ imply SND_SOC_CS4270
+ imply SND_SOC_CS4271_I2C
+ imply SND_SOC_CS4271_SPI
+ imply SND_SOC_CS42XX8_I2C
+ imply SND_SOC_CS43130
+ imply SND_SOC_CS4341
+ imply SND_SOC_CS4349
+ imply SND_SOC_CS47L15
+ imply SND_SOC_CS47L24
+ imply SND_SOC_CS47L35
+ imply SND_SOC_CS47L85
+ imply SND_SOC_CS47L90
+ imply SND_SOC_CS47L92
+ imply SND_SOC_CS53L30
+ imply SND_SOC_CX20442
+ imply SND_SOC_CX2072X
+ imply SND_SOC_DA7210
+ imply SND_SOC_DA7213
+ imply SND_SOC_DA7218
+ imply SND_SOC_DA7219
+ imply SND_SOC_DA732X
+ imply SND_SOC_DA9055
+ imply SND_SOC_DMIC
+ imply SND_SOC_ES8316
+ imply SND_SOC_ES8328_SPI
+ imply SND_SOC_ES8328_I2C
+ imply SND_SOC_ES7134
+ imply SND_SOC_ES7241
+ imply SND_SOC_GTM601
+ imply SND_SOC_HDAC_HDMI
+ imply SND_SOC_HDAC_HDA
+ imply SND_SOC_ICS43432
+ imply SND_SOC_INNO_RK3036
+ imply SND_SOC_ISABELLE
+ imply SND_SOC_JZ4740_CODEC
+ imply SND_SOC_JZ4725B_CODEC
+ imply SND_SOC_JZ4770_CODEC
+ imply SND_SOC_LM4857
+ imply SND_SOC_LM49453
+ imply SND_SOC_LOCHNAGAR_SC
+ imply SND_SOC_MAX98088
+ imply SND_SOC_MAX98090
+ imply SND_SOC_MAX98095
+ imply SND_SOC_MAX98357A
+ imply SND_SOC_MAX98371
+ imply SND_SOC_MAX98504
+ imply SND_SOC_MAX9867
+ imply SND_SOC_MAX98925
+ imply SND_SOC_MAX98926
+ imply SND_SOC_MAX98927
+ imply SND_SOC_MAX98373
+ imply SND_SOC_MAX9850
+ imply SND_SOC_MAX9860
+ imply SND_SOC_MAX9759
+ imply SND_SOC_MAX9768
+ imply SND_SOC_MAX9877
+ imply SND_SOC_MC13783
+ imply SND_SOC_ML26124
+ imply SND_SOC_MT6351
+ imply SND_SOC_MT6358
+ imply SND_SOC_MT6660
+ imply SND_SOC_NAU8540
+ imply SND_SOC_NAU8810
+ imply SND_SOC_NAU8822
+ imply SND_SOC_NAU8824
+ imply SND_SOC_NAU8825
+ imply SND_SOC_HDMI_CODEC
+ imply SND_SOC_PCM1681
+ imply SND_SOC_PCM1789_I2C
+ imply SND_SOC_PCM179X_I2C
+ imply SND_SOC_PCM179X_SPI
+ imply SND_SOC_PCM186X_I2C
+ imply SND_SOC_PCM186X_SPI
+ imply SND_SOC_PCM3008
+ imply SND_SOC_PCM3060_I2C
+ imply SND_SOC_PCM3060_SPI
+ imply SND_SOC_PCM3168A_I2C
+ imply SND_SOC_PCM3168A_SPI
+ imply SND_SOC_PCM5102A
+ imply SND_SOC_PCM512x_I2C
+ imply SND_SOC_PCM512x_SPI
+ imply SND_SOC_RK3328
+ imply SND_SOC_RT274
+ imply SND_SOC_RT286
+ imply SND_SOC_RT298
+ imply SND_SOC_RT1011
+ imply SND_SOC_RT1015
+ imply SND_SOC_RT1305
+ imply SND_SOC_RT1308
+ imply SND_SOC_RT5514
+ imply SND_SOC_RT5616
+ imply SND_SOC_RT5631
+ imply SND_SOC_RT5640
+ imply SND_SOC_RT5645
+ imply SND_SOC_RT5651
+ imply SND_SOC_RT5659
+ imply SND_SOC_RT5660
+ imply SND_SOC_RT5663
+ imply SND_SOC_RT5665
+ imply SND_SOC_RT5668
+ imply SND_SOC_RT5670
+ imply SND_SOC_RT5677
+ imply SND_SOC_RT5682
+ imply SND_SOC_RT5682_SDW
+ imply SND_SOC_RT700_SDW
+ imply SND_SOC_RT711_SDW
+ imply SND_SOC_RT715_SDW
+ imply SND_SOC_RT1308_SDW
+ imply SND_SOC_SGTL5000
+ imply SND_SOC_SI476X
+ imply SND_SOC_SIMPLE_AMPLIFIER
+ imply SND_SOC_SIRF_AUDIO_CODEC
+ imply SND_SOC_SPDIF
+ imply SND_SOC_SSM2305
+ imply SND_SOC_SSM2518
+ imply SND_SOC_SSM2602_SPI
+ imply SND_SOC_SSM2602_I2C
+ imply SND_SOC_SSM4567
+ imply SND_SOC_STA32X
+ imply SND_SOC_STA350
+ imply SND_SOC_STA529
+ imply SND_SOC_STAC9766
+ imply SND_SOC_STI_SAS
+ imply SND_SOC_TAS2552
+ imply SND_SOC_TAS2562
+ imply SND_SOC_TAS2770
+ imply SND_SOC_TAS5086
+ imply SND_SOC_TAS571X
+ imply SND_SOC_TAS5720
+ imply SND_SOC_TAS6424
+ imply SND_SOC_TDA7419
+ imply SND_SOC_TFA9879
+ imply SND_SOC_TLV320ADCX140
+ imply SND_SOC_TLV320AIC23_I2C
+ imply SND_SOC_TLV320AIC23_SPI
+ imply SND_SOC_TLV320AIC26
+ imply SND_SOC_TLV320AIC31XX
+ imply SND_SOC_TLV320AIC32X4_I2C
+ imply SND_SOC_TLV320AIC32X4_SPI
+ imply SND_SOC_TLV320AIC3X
+ imply SND_SOC_TPA6130A2
+ imply SND_SOC_TLV320DAC33
+ imply SND_SOC_TSCS42XX
+ imply SND_SOC_TSCS454
+ imply SND_SOC_TS3A227E
+ imply SND_SOC_TWL4030
+ imply SND_SOC_TWL6040
+ imply SND_SOC_UDA1334
+ imply SND_SOC_UDA134X
+ imply SND_SOC_UDA1380
+ imply SND_SOC_WCD9335
+ imply SND_SOC_WCD934X
+ imply SND_SOC_WL1273
+ imply SND_SOC_WM0010
+ imply SND_SOC_WM1250_EV1
+ imply SND_SOC_WM2000
+ imply SND_SOC_WM2200
+ imply SND_SOC_WM5100
+ imply SND_SOC_WM5102
+ imply SND_SOC_WM5110
+ imply SND_SOC_WM8350
+ imply SND_SOC_WM8400
+ imply SND_SOC_WM8510
+ imply SND_SOC_WM8523
+ imply SND_SOC_WM8524
+ imply SND_SOC_WM8580
+ imply SND_SOC_WM8711
+ imply SND_SOC_WM8727
+ imply SND_SOC_WM8728
+ imply SND_SOC_WM8731
+ imply SND_SOC_WM8737
+ imply SND_SOC_WM8741
+ imply SND_SOC_WM8750
+ imply SND_SOC_WM8753
+ imply SND_SOC_WM8770
+ imply SND_SOC_WM8776
+ imply SND_SOC_WM8782
+ imply SND_SOC_WM8804_I2C
+ imply SND_SOC_WM8804_SPI
+ imply SND_SOC_WM8900
+ imply SND_SOC_WM8903
+ imply SND_SOC_WM8904
+ imply SND_SOC_WM8940
+ imply SND_SOC_WM8955
+ imply SND_SOC_WM8960
+ imply SND_SOC_WM8961
+ imply SND_SOC_WM8962
+ imply SND_SOC_WM8971
+ imply SND_SOC_WM8974
+ imply SND_SOC_WM8978
+ imply SND_SOC_WM8983
+ imply SND_SOC_WM8985
+ imply SND_SOC_WM8988
+ imply SND_SOC_WM8990
+ imply SND_SOC_WM8991
+ imply SND_SOC_WM8993
+ imply SND_SOC_WM8994
+ imply SND_SOC_WM8995
+ imply SND_SOC_WM8996
+ imply SND_SOC_WM8997
+ imply SND_SOC_WM8998
+ imply SND_SOC_WM9081
+ imply SND_SOC_WM9090
+ imply SND_SOC_WM9705
+ imply SND_SOC_WM9712
+ imply SND_SOC_WM9713
+ imply SND_SOC_WSA881X
help
Normally ASoC codec drivers are only built if a machine driver which
uses them is also built since they are only usable with a machine
@@ -283,6 +285,7 @@ config SND_SOC_ALL_CODECS
config SND_SOC_88PM860X
tristate
+ depends on MFD_88PM860X
config SND_SOC_ARIZONA
tristate
@@ -318,6 +321,7 @@ config SND_SOC_WM_ADSP
config SND_SOC_AB8500_CODEC
tristate
+ depends on ABX500_CORE
config SND_SOC_AC97_CODEC
tristate "Build generic ASoC AC97 CODEC driver"
@@ -326,21 +330,25 @@ config SND_SOC_AC97_CODEC
config SND_SOC_AD1836
tristate
+ depends on SPI_MASTER
config SND_SOC_AD193X
tristate
config SND_SOC_AD193X_SPI
tristate
+ depends on SPI_MASTER
select SND_SOC_AD193X
config SND_SOC_AD193X_I2C
tristate
+ depends on I2C
select SND_SOC_AD193X
config SND_SOC_AD1980
- select REGMAP_AC97
tristate
+ depends on SND_SOC_AC97_BUS
+ select REGMAP_AC97
config SND_SOC_AD73311
tristate
@@ -350,6 +358,7 @@ config SND_SOC_ADAU_UTILS
config SND_SOC_ADAU1373
tristate
+ depends on I2C
select SND_SOC_ADAU_UTILS
config SND_SOC_ADAU1701
@@ -384,11 +393,13 @@ config SND_SOC_ADAU1781
config SND_SOC_ADAU1781_I2C
tristate
+ depends on I2C
select SND_SOC_ADAU1781
select REGMAP_I2C
config SND_SOC_ADAU1781_SPI
tristate
+ depends on SPI_MASTER
select SND_SOC_ADAU1781
select REGMAP_SPI
@@ -397,11 +408,13 @@ config SND_SOC_ADAU1977
config SND_SOC_ADAU1977_SPI
tristate
+ depends on SPI_MASTER
select SND_SOC_ADAU1977
select REGMAP_SPI
config SND_SOC_ADAU1977_I2C
tristate
+ depends on I2C
select SND_SOC_ADAU1977
select REGMAP_I2C
@@ -440,10 +453,12 @@ config SND_SOC_ADAV80X
config SND_SOC_ADAV801
tristate
+ depends on SPI_MASTER
select SND_SOC_ADAV80X
config SND_SOC_ADAV803
tristate
+ depends on I2C
select SND_SOC_ADAV80X
config SND_SOC_ADS117X
@@ -465,6 +480,7 @@ config SND_SOC_AK4458
config SND_SOC_AK4535
tristate
+ depends on I2C
config SND_SOC_AK4554
tristate "AKM AK4554 CODEC"
@@ -475,6 +491,7 @@ config SND_SOC_AK4613
config SND_SOC_AK4641
tristate
+ depends on I2C
config SND_SOC_AK4642
tristate "AKM AK4642 CODEC"
@@ -482,6 +499,7 @@ config SND_SOC_AK4642
config SND_SOC_AK4671
tristate
+ depends on I2C
config SND_SOC_AK5386
tristate "AKM AK5638 CODEC"
@@ -497,6 +515,7 @@ config SND_SOC_ALC5623
config SND_SOC_ALC5632
tristate
+ depends on I2C
config SND_SOC_BD28623
tristate "ROHM BD28623 CODEC"
@@ -631,6 +650,7 @@ config SND_SOC_CS47L15
config SND_SOC_CS47L24
tristate
+ depends on MFD_CS47L24
config SND_SOC_CS47L35
tristate
@@ -697,6 +717,7 @@ config SND_SOC_L3
config SND_SOC_DA7210
tristate
+ depends on I2C
config SND_SOC_DA7213
tristate "Dialog DA7213 CODEC"
@@ -704,15 +725,19 @@ config SND_SOC_DA7213
config SND_SOC_DA7218
tristate
+ depends on I2C
config SND_SOC_DA7219
tristate
+ depends on I2C
config SND_SOC_DA732X
tristate
+ depends on I2C
config SND_SOC_DA9055
tristate
+ depends on I2C
config SND_SOC_DMIC
tristate "Generic Digital Microphone CODEC"
@@ -772,9 +797,11 @@ config SND_SOC_INNO_RK3036
config SND_SOC_ISABELLE
tristate
+ depends on I2C
config SND_SOC_LM49453
tristate
+ depends on I2C
config SND_SOC_LOCHNAGAR_SC
tristate "Lochnagar Sound Card"
@@ -801,17 +828,20 @@ config SND_SOC_MAX98088
depends on I2C
config SND_SOC_MAX98090
- tristate
+ tristate
+ depends on I2C
config SND_SOC_MAX98095
- tristate
+ tristate
+ depends on I2C
config SND_SOC_MAX98357A
tristate "Maxim MAX98357A CODEC"
depends on GPIOLIB
config SND_SOC_MAX98371
- tristate
+ tristate
+ depends on I2C
config SND_SOC_MAX98504
tristate "Maxim MAX98504 speaker amplifier"
@@ -822,10 +852,12 @@ config SND_SOC_MAX9867
depends on I2C
config SND_SOC_MAX98925
- tristate
+ tristate
+ depends on I2C
config SND_SOC_MAX98926
tristate
+ depends on I2C
config SND_SOC_MAX98927
tristate "Maxim Integrated MAX98927 Speaker Amplifier"
@@ -837,6 +869,7 @@ config SND_SOC_MAX98373
config SND_SOC_MAX9850
tristate
+ depends on I2C
config SND_SOC_MAX9860
tristate "Maxim MAX9860 Mono Audio Voice Codec"
@@ -1015,26 +1048,32 @@ config SND_SOC_RT298
config SND_SOC_RT1011
tristate
+ depends on I2C
config SND_SOC_RT1015
tristate
+ depends on I2C
config SND_SOC_RT1305
tristate
+ depends on I2C
config SND_SOC_RT1308
tristate
+ depends on I2C
config SND_SOC_RT1308_SDW
tristate "Realtek RT1308 Codec - SDW"
- depends on SOUNDWIRE
+ depends on I2C && SOUNDWIRE
select REGMAP_SOUNDWIRE
config SND_SOC_RT5514
tristate
+ depends on I2C
config SND_SOC_RT5514_SPI
tristate
+ depends on SPI_MASTER
config SND_SOC_RT5514_SPI_BUILTIN
bool # force RT5514_SPI to be built-in to avoid link errors
@@ -1050,33 +1089,43 @@ config SND_SOC_RT5631
config SND_SOC_RT5640
tristate
+ depends on I2C
config SND_SOC_RT5645
tristate
+ depends on I2C
config SND_SOC_RT5651
tristate
+ depends on I2C
config SND_SOC_RT5659
tristate
+ depends on I2C
config SND_SOC_RT5660
tristate
+ depends on I2C
config SND_SOC_RT5663
tristate
+ depends on I2C
config SND_SOC_RT5665
tristate
+ depends on I2C
config SND_SOC_RT5668
tristate
+ depends on I2C
config SND_SOC_RT5670
tristate
+ depends on I2C
config SND_SOC_RT5677
tristate
+ depends on I2C
select REGMAP_I2C
select REGMAP_IRQ
@@ -1086,6 +1135,13 @@ config SND_SOC_RT5677_SPI
config SND_SOC_RT5682
tristate
+ depends on I2C || SOUNDWIRE
+
+config SND_SOC_RT5682_SDW
+ tristate "Realtek RT5682 Codec - SDW"
+ depends on SOUNDWIRE
+ select SND_SOC_RT5682
+ select REGMAP_SOUNDWIRE
config SND_SOC_RT700
tristate
@@ -1153,6 +1209,7 @@ config SND_SOC_SSM2305
config SND_SOC_SSM2518
tristate
+ depends on I2C
config SND_SOC_SSM2602
tristate
@@ -1184,9 +1241,11 @@ config SND_SOC_STA350
config SND_SOC_STA529
tristate
+ depends on I2C
config SND_SOC_STAC9766
tristate
+ depends on SND_SOC_AC97_BUS
config SND_SOC_STI_SAS
tristate "codec Audio support for STI SAS codec"
@@ -1281,6 +1340,15 @@ config SND_SOC_TLV320AIC3X
config SND_SOC_TLV320DAC33
tristate
+ depends on I2C
+
+config SND_SOC_TLV320ADCX140
+ tristate "Texas Instruments TLV320ADCX140 CODEC family"
+ depends on I2C
+ select REGMAP_I2C
+ help
+ Add support for Texas Instruments tlv320adc3140, tlv320adc5140 and
+ tlv320adc6140 quad channel ADCs.
config SND_SOC_TS3A227E
tristate "TI Headset/Mic detect and keypress chip"
@@ -1301,11 +1369,13 @@ config SND_SOC_TSCS454
Add support for Tempo Semiconductor's TSCS454 audio CODEC.
config SND_SOC_TWL4030
- select MFD_TWL4030_AUDIO
tristate
+ depends on TWL4030_CORE
+ select MFD_TWL4030_AUDIO
config SND_SOC_TWL6040
tristate
+ depends on TWL6040_CORE
config SND_SOC_UDA1334
tristate "NXP UDA1334 DAC"
@@ -1345,30 +1415,40 @@ config SND_SOC_WL1273
config SND_SOC_WM0010
tristate
+ depends on SPI_MASTER
config SND_SOC_WM1250_EV1
tristate
+ depends on I2C
config SND_SOC_WM2000
tristate
+ depends on I2C
config SND_SOC_WM2200
tristate
+ depends on I2C
config SND_SOC_WM5100
tristate
+ depends on I2C
config SND_SOC_WM5102
tristate
+ depends on MFD_WM5102
config SND_SOC_WM5110
tristate
+ depends on MFD_WM5110
config SND_SOC_WM8350
tristate
+ depends on MFD_WM8350
config SND_SOC_WM8400
tristate
+ # FIXME nothing selects SND_SOC_WM8400??
+ depends on MFD_WM8400
config SND_SOC_WM8510
tristate "Wolfson Microelectronics WM8510 CODEC"
@@ -1456,9 +1536,11 @@ config SND_SOC_WM8904
config SND_SOC_WM8940
tristate
+ depends on I2C
config SND_SOC_WM8955
tristate
+ depends on I2C
config SND_SOC_WM8960
tristate "Wolfson Microelectronics WM8960 CODEC"
@@ -1466,6 +1548,7 @@ config SND_SOC_WM8960
config SND_SOC_WM8961
tristate
+ depends on I2C
config SND_SOC_WM8962
tristate "Wolfson Microelectronics WM8962 CODEC"
@@ -1473,6 +1556,7 @@ config SND_SOC_WM8962
config SND_SOC_WM8971
tristate
+ depends on I2C
config SND_SOC_WM8974
tristate "Wolfson Microelectronics WM8974 codec"
@@ -1484,6 +1568,7 @@ config SND_SOC_WM8978
config SND_SOC_WM8983
tristate
+ depends on I2C
config SND_SOC_WM8985
tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver"
@@ -1494,12 +1579,15 @@ config SND_SOC_WM8988
config SND_SOC_WM8990
tristate
+ depends on I2C
config SND_SOC_WM8991
tristate
+ depends on I2C
config SND_SOC_WM8993
tristate
+ depends on I2C
config SND_SOC_WM8994
tristate
@@ -1509,12 +1597,15 @@ config SND_SOC_WM8995
config SND_SOC_WM8996
tristate
+ depends on I2C
config SND_SOC_WM8997
tristate
+ depends on MFD_WM8997
config SND_SOC_WM8998
tristate
+ depends on MFD_WM8998
config SND_SOC_WM9081
tristate
@@ -1522,19 +1613,23 @@ config SND_SOC_WM9081
config SND_SOC_WM9090
tristate
+ depends on I2C
config SND_SOC_WM9705
tristate
+ depends on SND_SOC_AC97_BUS
select REGMAP_AC97
select AC97_BUS_COMPAT if AC97_BUS_NEW
config SND_SOC_WM9712
tristate
+ depends on SND_SOC_AC97_BUS
select REGMAP_AC97
select AC97_BUS_COMPAT if AC97_BUS_NEW
config SND_SOC_WM9713
tristate
+ depends on SND_SOC_AC97_BUS
select REGMAP_AC97
select AC97_BUS_COMPAT if AC97_BUS_NEW
@@ -1555,6 +1650,7 @@ config SND_SOC_ZX_AUD96P22
# Amp
config SND_SOC_LM4857
tristate
+ depends on I2C
config SND_SOC_MAX9759
tristate "Maxim MAX9759 speaker Amplifier"
@@ -1562,15 +1658,19 @@ config SND_SOC_MAX9759
config SND_SOC_MAX9768
tristate
+ depends on I2C
config SND_SOC_MAX9877
tristate
+ depends on I2C
config SND_SOC_MC13783
tristate
+ depends on MFD_MC13XXX
config SND_SOC_ML26124
tristate
+ depends on I2C
config SND_SOC_MT6351
tristate "MediaTek MT6351 Codec"
@@ -1608,6 +1708,7 @@ config SND_SOC_NAU8824
config SND_SOC_NAU8825
tristate
+ depends on I2C
config SND_SOC_TPA6130A2
tristate "Texas Instruments TPA6130A2 headphone amplifier"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index ba1b4b3fa2da..03533157cda6 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -177,6 +177,7 @@ snd-soc-rt5670-objs := rt5670.o
snd-soc-rt5677-objs := rt5677.o
snd-soc-rt5677-spi-objs := rt5677-spi.o
snd-soc-rt5682-objs := rt5682.o
+snd-soc-rt5682-sdw-objs := rt5682-sdw.o
snd-soc-rt700-objs := rt700.o rt700-sdw.o
snd-soc-rt711-objs := rt711.o rt711-sdw.o
snd-soc-rt715-objs := rt715.o rt715-sdw.o
@@ -218,6 +219,7 @@ snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o
snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-tlv320dac33-objs := tlv320dac33.o
+snd-soc-tlv320adcx140-objs := tlv320adcx140.o
snd-soc-tscs42xx-objs := tscs42xx.o
snd-soc-tscs454-objs := tscs454.o
snd-soc-ts3a227e-objs := ts3a227e.o
@@ -476,6 +478,7 @@ obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o
obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o
obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o
obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o
+obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o
obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o
obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o
obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o
@@ -516,6 +519,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o
obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
+obj-$(CONFIG_SND_SOC_TLV320ADCX140) += snd-soc-tlv320adcx140.o
obj-$(CONFIG_SND_SOC_TSCS42XX) += snd-soc-tscs42xx.o
obj-$(CONFIG_SND_SOC_TSCS454) += snd-soc-tscs454.o
obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o
diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c
index 6a24f570c5e8..d3dc42aa6825 100644
--- a/sound/soc/codecs/cros_ec_codec.c
+++ b/sound/soc/codecs/cros_ec_codec.c
@@ -45,6 +45,9 @@ struct cros_ec_codec_priv {
/* DMIC */
atomic_t dmic_probed;
+ /* I2S_RX */
+ uint32_t i2s_rx_bclk_ratio;
+
/* WoV */
bool wov_enabled;
uint8_t *wov_audio_shm_p;
@@ -259,6 +262,7 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream,
snd_soc_component_get_drvdata(component);
struct ec_param_ec_codec_i2s_rx p;
enum ec_codec_i2s_rx_sample_depth depth;
+ uint32_t bclk;
int ret;
if (params_rate(params) != 48000)
@@ -284,15 +288,29 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- dev_dbg(component->dev, "set bclk to %u\n",
- snd_soc_params_to_bclk(params));
+ if (priv->i2s_rx_bclk_ratio)
+ bclk = params_rate(params) * priv->i2s_rx_bclk_ratio;
+ else
+ bclk = snd_soc_params_to_bclk(params);
+
+ dev_dbg(component->dev, "set bclk to %u\n", bclk);
p.cmd = EC_CODEC_I2S_RX_SET_BCLK;
- p.set_bclk_param.bclk = snd_soc_params_to_bclk(params);
+ p.set_bclk_param.bclk = bclk;
return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX,
(uint8_t *)&p, sizeof(p), NULL, 0);
}
+static int i2s_rx_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+ struct snd_soc_component *component = dai->component;
+ struct cros_ec_codec_priv *priv =
+ snd_soc_component_get_drvdata(component);
+
+ priv->i2s_rx_bclk_ratio = ratio;
+ return 0;
+}
+
static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_component *component = dai->component;
@@ -340,6 +358,7 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
static const struct snd_soc_dai_ops i2s_rx_dai_ops = {
.hw_params = i2s_rx_hw_params,
.set_fmt = i2s_rx_set_fmt,
+ .set_bclk_ratio = i2s_rx_set_bclk_ratio,
};
static int i2s_rx_event(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 04b86a51e055..62f412d6f9f2 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -356,9 +356,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
*/
if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
- !dai->capture_active) ||
+ !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) ||
(substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
- !dai->playback_active)) {
+ !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) {
ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
CS4271_MODE2_PDN,
CS4271_MODE2_PDN);
diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c
index e8840dc142ef..8d1869bf7f9c 100644
--- a/sound/soc/codecs/cs47l15.c
+++ b/sound/soc/codecs/cs47l15.c
@@ -1239,12 +1239,12 @@ static int cs47l15_open(struct snd_compr_stream *stream)
struct madera *madera = priv->madera;
int n_adsp;
- if (strcmp(rtd->codec_dai->name, "cs47l15-dsp-trace") == 0) {
+ if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l15-dsp-trace") == 0) {
n_adsp = 0;
} else {
dev_err(madera->dev,
"No suitable compressed stream for DAI '%s'\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
return -EINVAL;
}
diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c
index 25bffc2968f0..6b0570f59630 100644
--- a/sound/soc/codecs/cs47l24.c
+++ b/sound/soc/codecs/cs47l24.c
@@ -1076,14 +1076,14 @@ static int cs47l24_open(struct snd_compr_stream *stream)
struct arizona *arizona = priv->core.arizona;
int n_adsp;
- if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) {
+ if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-voicectrl") == 0) {
n_adsp = 2;
- } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) {
+ } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-trace") == 0) {
n_adsp = 1;
} else {
dev_err(arizona->dev,
"No suitable compressed stream for DAI '%s'\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
return -EINVAL;
}
diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c
index 3d48a0d9ecc5..18839807c9d1 100644
--- a/sound/soc/codecs/cs47l35.c
+++ b/sound/soc/codecs/cs47l35.c
@@ -1514,14 +1514,14 @@ static int cs47l35_open(struct snd_compr_stream *stream)
struct madera *madera = priv->madera;
int n_adsp;
- if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-voicectrl") == 0) {
+ if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-voicectrl") == 0) {
n_adsp = 2;
- } else if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-trace") == 0) {
+ } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-trace") == 0) {
n_adsp = 0;
} else {
dev_err(madera->dev,
"No suitable compressed stream for DAI '%s'\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
return -EINVAL;
}
diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c
index bef3471f482d..a575113207f0 100644
--- a/sound/soc/codecs/cs47l85.c
+++ b/sound/soc/codecs/cs47l85.c
@@ -2457,14 +2457,14 @@ static int cs47l85_open(struct snd_compr_stream *stream)
struct madera *madera = priv->madera;
int n_adsp;
- if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-voicectrl") == 0) {
+ if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-voicectrl") == 0) {
n_adsp = 5;
- } else if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-trace") == 0) {
+ } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-trace") == 0) {
n_adsp = 0;
} else {
dev_err(madera->dev,
"No suitable compressed stream for DAI '%s'\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
return -EINVAL;
}
diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c
index 266eade82764..81a1311b14e6 100644
--- a/sound/soc/codecs/cs47l90.c
+++ b/sound/soc/codecs/cs47l90.c
@@ -2368,14 +2368,14 @@ static int cs47l90_open(struct snd_compr_stream *stream)
struct madera *madera = priv->madera;
int n_adsp;
- if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-voicectrl") == 0) {
+ if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-voicectrl") == 0) {
n_adsp = 5;
- } else if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-trace") == 0) {
+ } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-trace") == 0) {
n_adsp = 0;
} else {
dev_err(madera->dev,
"No suitable compressed stream for DAI '%s'\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
return -EINVAL;
}
diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c
index 942040fd354f..15fc213d178d 100644
--- a/sound/soc/codecs/cs47l92.c
+++ b/sound/soc/codecs/cs47l92.c
@@ -1840,12 +1840,12 @@ static int cs47l92_open(struct snd_compr_stream *stream)
struct madera *madera = priv->madera;
int n_adsp;
- if (strcmp(rtd->codec_dai->name, "cs47l92-dsp-trace") == 0) {
+ if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l92-dsp-trace") == 0) {
n_adsp = 0;
} else {
dev_err(madera->dev,
"No suitable compressed stream for DAI '%s'\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
return -EINVAL;
}
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index e6558475e006..fba9b749839d 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1998,11 +1998,11 @@ static struct hdac_hdmi_drv_data intel_drv_data = {
static int hdac_hdmi_dev_probe(struct hdac_device *hdev)
{
- struct hdac_hdmi_priv *hdmi_priv = NULL;
+ struct hdac_hdmi_priv *hdmi_priv;
struct snd_soc_dai_driver *hdmi_dais = NULL;
- struct hdac_ext_link *hlink = NULL;
+ struct hdac_ext_link *hlink;
int num_dais = 0;
- int ret = 0;
+ int ret;
struct hdac_driver *hdrv = drv_to_hdac_driver(hdev->dev.driver);
const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv);
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 16313b973eaa..a8bd793a7867 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -5,6 +5,7 @@
*/
#include <linux/acpi.h>
+#include <linux/delay.h>
#include <linux/device.h>
#include <linux/err.h>
#include <linux/gpio.h>
@@ -24,26 +25,24 @@ struct max98357a_priv {
unsigned int sdmode_delay;
};
-static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
- int cmd, struct snd_soc_dai *dai)
+static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
{
- struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai);
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+ struct max98357a_priv *max98357a =
+ snd_soc_component_get_drvdata(component);
if (!max98357a->sdmode)
return 0;
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- mdelay(max98357a->sdmode_delay);
+ if (event & SND_SOC_DAPM_POST_PMU) {
+ msleep(max98357a->sdmode_delay);
gpiod_set_value(max98357a->sdmode, 1);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dev_dbg(component->dev, "set sdmode to 1");
+ } else if (event & SND_SOC_DAPM_PRE_PMD) {
gpiod_set_value(max98357a->sdmode, 0);
- break;
+ dev_dbg(component->dev, "set sdmode to 0");
}
return 0;
@@ -51,10 +50,14 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("Speaker"),
+ SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0,
+ max98357a_sdmode_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
};
static const struct snd_soc_dapm_route max98357a_dapm_routes[] = {
- {"Speaker", NULL, "HiFi Playback"},
+ {"SD_MODE", NULL, "HiFi Playback"},
+ {"Speaker", NULL, "SD_MODE"},
};
static const struct snd_soc_component_driver max98357a_component_driver = {
@@ -68,10 +71,6 @@ static const struct snd_soc_component_driver max98357a_component_driver = {
.non_legacy_dai_naming = 1,
};
-static const struct snd_soc_dai_ops max98357a_dai_ops = {
- .trigger = max98357a_daiops_trigger,
-};
-
static struct snd_soc_dai_driver max98357a_dai_driver = {
.name = "HiFi",
.playback = {
@@ -91,7 +90,6 @@ static struct snd_soc_dai_driver max98357a_dai_driver = {
.channels_min = 1,
.channels_max = 2,
},
- .ops = &max98357a_dai_ops,
};
static int max98357a_platform_probe(struct platform_device *pdev)
@@ -135,6 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id);
#ifdef CONFIG_ACPI
static const struct acpi_device_id max98357a_acpi_match[] = {
{ "MX98357A", 0 },
+ { "MX98360A", 0 },
{},
};
MODULE_DEVICE_TABLE(acpi, max98357a_acpi_match);
diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c
index a36c416caad4..d1797003c83d 100644
--- a/sound/soc/codecs/mt6660.c
+++ b/sound/soc/codecs/mt6660.c
@@ -1,15 +1,13 @@
-// SPDX-License-Identifier: GPL-2.0 //
+// SPDX-License-Identifier: GPL-2.0
// Copyright (c) 2019 MediaTek Inc.
#include <linux/module.h>
#include <linux/kernel.h>
-#include <linux/version.h>
#include <linux/err.h>
#include <linux/i2c.h>
#include <linux/pm_runtime.h>
#include <linux/delay.h>
-#include <linux/debugfs.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/pcm_params.h>
@@ -225,14 +223,87 @@ static int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off)
0x01, on_off ? 0x00 : 0x01);
}
+struct reg_table {
+ uint32_t addr;
+ uint32_t mask;
+ uint32_t val;
+};
+
+static const struct reg_table mt6660_setting_table[] = {
+ { 0x20, 0x80, 0x00 },
+ { 0x30, 0x01, 0x00 },
+ { 0x50, 0x1c, 0x04 },
+ { 0xB1, 0x0c, 0x00 },
+ { 0xD3, 0x03, 0x03 },
+ { 0xE0, 0x01, 0x00 },
+ { 0x98, 0x44, 0x04 },
+ { 0xB9, 0xff, 0x82 },
+ { 0xB7, 0x7777, 0x7273 },
+ { 0xB6, 0x07, 0x03 },
+ { 0x6B, 0xe0, 0x20 },
+ { 0x07, 0xff, 0x70 },
+ { 0xBB, 0xff, 0x20 },
+ { 0x69, 0xff, 0x40 },
+ { 0xBD, 0xffff, 0x17f8 },
+ { 0x70, 0xff, 0x15 },
+ { 0x7C, 0xff, 0x00 },
+ { 0x46, 0xff, 0x1d },
+ { 0x1A, 0xffffffff, 0x7fdb7ffe },
+ { 0x1B, 0xffffffff, 0x7fdb7ffe },
+ { 0x51, 0xff, 0x58 },
+ { 0xA2, 0xff, 0xce },
+ { 0x33, 0xffff, 0x7fff },
+ { 0x4C, 0xffff, 0x0116 },
+ { 0x16, 0x1800, 0x0800 },
+ { 0x68, 0x1f, 0x07 },
+};
+
+static int mt6660_component_setting(struct snd_soc_component *component)
+{
+ struct mt6660_chip *chip = snd_soc_component_get_drvdata(component);
+ int ret = 0;
+ size_t i = 0;
+
+ ret = _mt6660_chip_power_on(chip, 1);
+ if (ret < 0) {
+ dev_err(component->dev, "%s chip power on failed\n", __func__);
+ return ret;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(mt6660_setting_table); i++) {
+ ret = snd_soc_component_update_bits(component,
+ mt6660_setting_table[i].addr,
+ mt6660_setting_table[i].mask,
+ mt6660_setting_table[i].val);
+ if (ret < 0) {
+ dev_err(component->dev, "%s update 0x%02x failed\n",
+ __func__, mt6660_setting_table[i].addr);
+ return ret;
+ }
+ }
+
+ ret = _mt6660_chip_power_on(chip, 0);
+ if (ret < 0) {
+ dev_err(component->dev, "%s chip power off failed\n", __func__);
+ return ret;
+ }
+
+ return 0;
+}
+
static int mt6660_component_probe(struct snd_soc_component *component)
{
struct mt6660_chip *chip = snd_soc_component_get_drvdata(component);
+ int ret;
dev_dbg(component->dev, "%s\n", __func__);
snd_soc_component_init_regmap(component, chip->regmap);
- return 0;
+ ret = mt6660_component_setting(component);
+ if (ret < 0)
+ dev_err(chip->dev, "mt6660 component setting failed\n");
+
+ return ret;
}
static void mt6660_component_remove(struct snd_soc_component *component)
@@ -506,4 +577,4 @@ module_i2c_driver(mt6660_i2c_driver);
MODULE_AUTHOR("Jeff Chang <jeff_chang@richtek.com>");
MODULE_DESCRIPTION("MT6660 SPKAMP Driver");
MODULE_LICENSE("GPL");
-MODULE_VERSION("1.0.7_G");
+MODULE_VERSION("1.0.8_G");
diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c
index 287c962ba00d..115706a55577 100644
--- a/sound/soc/codecs/rk3328_codec.c
+++ b/sound/soc/codecs/rk3328_codec.c
@@ -7,6 +7,7 @@
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/device.h>
+#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <linux/of.h>
#include <linux/platform_device.h>
@@ -31,7 +32,7 @@
struct rk3328_codec_priv {
struct regmap *regmap;
- struct regmap *grf;
+ struct gpio_desc *mute;
struct clk *mclk;
struct clk *pclk;
unsigned int sclk;
@@ -106,16 +107,6 @@ static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
-static void rk3328_analog_output(struct rk3328_codec_priv *rk3328, int mute)
-{
- unsigned int val = BIT(17);
-
- if (mute)
- val |= BIT(1);
-
- regmap_write(rk3328->grf, RK3328_GRF_SOC_CON10, val);
-}
-
static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute)
{
struct rk3328_codec_priv *rk3328 =
@@ -205,7 +196,7 @@ static int rk3328_codec_open_playback(struct rk3328_codec_priv *rk3328)
}
msleep(rk3328->spk_depop_time);
- rk3328_analog_output(rk3328, 1);
+ gpiod_set_value(rk3328->mute, 0);
regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL,
HPOUTL_GAIN_MASK, OUT_VOLUME);
@@ -246,7 +237,7 @@ static int rk3328_codec_close_playback(struct rk3328_codec_priv *rk3328)
{
size_t i;
- rk3328_analog_output(rk3328, 0);
+ gpiod_set_value(rk3328->mute, 1);
regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL,
HPOUTL_GAIN_MASK, 0);
@@ -446,7 +437,6 @@ static int rk3328_platform_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "missing 'rockchip,grf'\n");
return PTR_ERR(grf);
}
- rk3328->grf = grf;
/* enable i2s_acodec_en */
regmap_write(grf, RK3328_GRF_SOC_CON2,
(BIT(14) << 16 | BIT(14)));
@@ -458,7 +448,18 @@ static int rk3328_platform_probe(struct platform_device *pdev)
rk3328->spk_depop_time = 200;
}
- rk3328_analog_output(rk3328, 0);
+ rk3328->mute = gpiod_get_optional(&pdev->dev, "mute", GPIOD_OUT_HIGH);
+ if (IS_ERR(rk3328->mute))
+ return PTR_ERR(rk3328->mute);
+ /*
+ * Rock64 is the only supported platform to have widely relied on
+ * this; if we do happen to come across an old DTB, just leave the
+ * external mute forced off.
+ */
+ if (!rk3328->mute && of_machine_is_compatible("pine64,rock64")) {
+ dev_warn(&pdev->dev, "assuming implicit control of GPIO_MUTE; update devicetree if possible\n");
+ regmap_write(grf, RK3328_GRF_SOC_CON10, BIT(17) | BIT(1));
+ }
rk3328->mclk = devm_clk_get(&pdev->dev, "mclk");
if (IS_ERR(rk3328->mclk))
diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c
index a887d5ccb10d..d181c217d835 100644
--- a/sound/soc/codecs/rl6231.c
+++ b/sound/soc/codecs/rl6231.c
@@ -102,6 +102,7 @@ struct pll_calc_map {
static const struct pll_calc_map pll_preset_table[] = {
{19200000, 4096000, 23, 14, 1, false},
{19200000, 24576000, 3, 30, 3, false},
+ {3840000, 24576000, 3, 30, 0, true},
};
static unsigned int find_best_div(unsigned int in,
diff --git a/sound/soc/codecs/rl6231.h b/sound/soc/codecs/rl6231.h
index 31a9643b0afd..6d8ed0377296 100644
--- a/sound/soc/codecs/rl6231.h
+++ b/sound/soc/codecs/rl6231.h
@@ -10,7 +10,7 @@
#ifndef __RL6231_H__
#define __RL6231_H__
-#define RL6231_PLL_INP_MAX 40000000
+#define RL6231_PLL_INP_MAX 50000000
#define RL6231_PLL_INP_MIN 256000
#define RL6231_PLL_N_MAX 0x1ff
#define RL6231_PLL_K_MAX 0x1f
diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 66eb55b4ffd4..bb310bc7febd 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -444,7 +444,7 @@ static int rt1015_boost_mode_put(struct snd_kcontrol *kcontrol,
return 0;
}
-static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol,
+static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component =
@@ -457,7 +457,7 @@ static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol,
return 0;
}
-static int rt5518_bypass_boost_put(struct snd_kcontrol *kcontrol,
+static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component =
@@ -497,7 +497,7 @@ static const struct snd_kcontrol_new rt1015_snd_controls[] = {
rt1015_boost_mode_get, rt1015_boost_mode_put),
SOC_ENUM("Mono LR Select", rt1015_mono_lr_sel),
SOC_SINGLE_EXT("Bypass Boost", SND_SOC_NOPM, 0, 1, 0,
- rt5518_bypass_boost_get, rt5518_bypass_boost_put),
+ rt1015_bypass_boost_get, rt1015_bypass_boost_put),
};
static int rt1015_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
@@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component)
#define RT1015_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
-struct snd_soc_dai_ops rt1015_aif_dai_ops = {
+static struct snd_soc_dai_ops rt1015_aif_dai_ops = {
.hw_params = rt1015_hw_params,
.set_fmt = rt1015_set_dai_fmt,
};
-struct snd_soc_dai_driver rt1015_dai[] = {
+static struct snd_soc_dai_driver rt1015_dai[] = {
{
.name = "rt1015-aif",
.id = 0,
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
index d930f60cb797..a5a7e46de246 100644
--- a/sound/soc/codecs/rt1308-sdw.c
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -507,6 +507,28 @@ static void rt1308_sdw_shutdown(struct snd_pcm_substream *substream,
kfree(stream);
}
+static int rt1308_sdw_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask,
+ unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt1308_sdw_priv *rt1308 =
+ snd_soc_component_get_drvdata(component);
+
+ if (tx_mask)
+ return -EINVAL;
+
+ if (slots > 2)
+ return -EINVAL;
+
+ rt1308->rx_mask = rx_mask;
+ rt1308->slots = slots;
+ /* slot_width is not used since it's irrelevant for SoundWire */
+
+ return 0;
+}
+
static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
@@ -517,7 +539,7 @@ static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream,
struct sdw_port_config port_config;
enum sdw_data_direction direction;
struct sdw_stream_data *stream;
- int retval, port, num_channels;
+ int retval, port, num_channels, ch_mask;
dev_dbg(dai->dev, "%s %s", __func__, dai->name);
stream = snd_soc_dai_get_dma_data(dai, substream);
@@ -537,13 +559,20 @@ static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
+ if (rt1308->slots) {
+ num_channels = rt1308->slots;
+ ch_mask = rt1308->rx_mask;
+ } else {
+ num_channels = params_channels(params);
+ ch_mask = (1 << num_channels) - 1;
+ }
+
stream_config.frame_rate = params_rate(params);
- stream_config.ch_count = params_channels(params);
+ stream_config.ch_count = num_channels;
stream_config.bps = snd_pcm_format_width(params_format(params));
stream_config.direction = direction;
- num_channels = params_channels(params);
- port_config.ch_mask = (1 << (num_channels)) - 1;
+ port_config.ch_mask = ch_mask;
port_config.num = port;
retval = sdw_stream_add_slave(rt1308->sdw_slave, &stream_config,
@@ -597,6 +626,7 @@ static const struct snd_soc_dai_ops rt1308_aif_dai_ops = {
.hw_free = rt1308_sdw_pcm_hw_free,
.set_sdw_stream = rt1308_set_sdw_stream,
.shutdown = rt1308_sdw_shutdown,
+ .set_tdm_slot = rt1308_sdw_set_tdm_slot,
};
#define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000
diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h
index c9341e70d6cf..c5ce75666dcc 100644
--- a/sound/soc/codecs/rt1308-sdw.h
+++ b/sound/soc/codecs/rt1308-sdw.h
@@ -160,6 +160,8 @@ struct rt1308_sdw_priv {
struct sdw_bus_params params;
bool hw_init;
bool first_hw_init;
+ int rx_mask;
+ int slots;
};
struct sdw_stream_data {
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index e66d08398f74..89e0f58512fa 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -1604,7 +1604,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
- int pd, idx = -EINVAL;
+ int pd, idx;
pd = rl6231_get_pre_div(rt5659->regmap,
RT5659_ADDA_CLK_1, RT5659_I2S_PD1_SFT);
diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
new file mode 100644
index 000000000000..a2d1d3ae1e31
--- /dev/null
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -0,0 +1,333 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// rt5682-sdw.c -- RT5682 ALSA SoC audio component driver
+//
+// Copyright 2019 Realtek Semiconductor Corp.
+// Author: Oder Chiou <oder_chiou@realtek.com>
+//
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/acpi.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mutex.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "rt5682.h"
+#include "rt5682-sdw.h"
+
+static bool rt5682_sdw_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case 0x00e0:
+ case 0x00f0:
+ case 0x3000:
+ case 0x3001:
+ case 0x3004:
+ case 0x3005:
+ case 0x3008:
+ return true;
+ default:
+ return false;
+ }
+}
+
+const struct regmap_config rt5682_sdw_regmap = {
+ .name = "sdw",
+ .reg_bits = 32,
+ .val_bits = 8,
+ .max_register = RT5682_I2C_MODE,
+ .readable_reg = rt5682_sdw_readable_register,
+ .cache_type = REGCACHE_NONE,
+ .use_single_read = true,
+ .use_single_write = true,
+};
+
+static int rt5682_update_status(struct sdw_slave *slave,
+ enum sdw_slave_status status)
+{
+ struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+
+ /* Update the status */
+ rt5682->status = status;
+
+ if (status == SDW_SLAVE_UNATTACHED)
+ rt5682->hw_init = false;
+
+ /*
+ * Perform initialization only if slave status is present and
+ * hw_init flag is false
+ */
+ if (rt5682->hw_init || rt5682->status != SDW_SLAVE_ATTACHED)
+ return 0;
+
+ /* perform I/O transfers required for Slave initialization */
+ return rt5682_io_init(&slave->dev, slave);
+}
+
+static int rt5682_read_prop(struct sdw_slave *slave)
+{
+ struct sdw_slave_prop *prop = &slave->prop;
+ int nval, i, num_of_ports = 1;
+ u32 bit;
+ unsigned long addr;
+ struct sdw_dpn_prop *dpn;
+
+ prop->paging_support = false;
+
+ /* first we need to allocate memory for set bits in port lists */
+ prop->source_ports = 0x4; /* BITMAP: 00000100 */
+ prop->sink_ports = 0x2; /* BITMAP: 00000010 */
+
+ nval = hweight32(prop->source_ports);
+ num_of_ports += nval;
+ prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval,
+ sizeof(*prop->src_dpn_prop),
+ GFP_KERNEL);
+ if (!prop->src_dpn_prop)
+ return -ENOMEM;
+
+ i = 0;
+ dpn = prop->src_dpn_prop;
+ addr = prop->source_ports;
+ for_each_set_bit(bit, &addr, 32) {
+ dpn[i].num = bit;
+ dpn[i].type = SDW_DPN_FULL;
+ dpn[i].simple_ch_prep_sm = true;
+ dpn[i].ch_prep_timeout = 10;
+ i++;
+ }
+
+ /* do this again for sink now */
+ nval = hweight32(prop->sink_ports);
+ num_of_ports += nval;
+ prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval,
+ sizeof(*prop->sink_dpn_prop),
+ GFP_KERNEL);
+ if (!prop->sink_dpn_prop)
+ return -ENOMEM;
+
+ i = 0;
+ dpn = prop->sink_dpn_prop;
+ addr = prop->sink_ports;
+ for_each_set_bit(bit, &addr, 32) {
+ dpn[i].num = bit;
+ dpn[i].type = SDW_DPN_FULL;
+ dpn[i].simple_ch_prep_sm = true;
+ dpn[i].ch_prep_timeout = 10;
+ i++;
+ }
+
+ /* Allocate port_ready based on num_of_ports */
+ slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports,
+ sizeof(*slave->port_ready),
+ GFP_KERNEL);
+ if (!slave->port_ready)
+ return -ENOMEM;
+
+ /* Initialize completion */
+ for (i = 0; i < num_of_ports; i++)
+ init_completion(&slave->port_ready[i]);
+
+ /* set the timeout values */
+ prop->clk_stop_timeout = 20;
+
+ /* wake-up event */
+ prop->wake_capable = 1;
+
+ return 0;
+}
+
+/* Bus clock frequency */
+#define RT5682_CLK_FREQ_9600000HZ 9600000
+#define RT5682_CLK_FREQ_12000000HZ 12000000
+#define RT5682_CLK_FREQ_6000000HZ 6000000
+#define RT5682_CLK_FREQ_4800000HZ 4800000
+#define RT5682_CLK_FREQ_2400000HZ 2400000
+#define RT5682_CLK_FREQ_12288000HZ 12288000
+
+static int rt5682_clock_config(struct device *dev)
+{
+ struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+ unsigned int clk_freq, value;
+
+ clk_freq = (rt5682->params.curr_dr_freq >> 1);
+
+ switch (clk_freq) {
+ case RT5682_CLK_FREQ_12000000HZ:
+ value = 0x0;
+ break;
+ case RT5682_CLK_FREQ_6000000HZ:
+ value = 0x1;
+ break;
+ case RT5682_CLK_FREQ_9600000HZ:
+ value = 0x2;
+ break;
+ case RT5682_CLK_FREQ_4800000HZ:
+ value = 0x3;
+ break;
+ case RT5682_CLK_FREQ_2400000HZ:
+ value = 0x4;
+ break;
+ case RT5682_CLK_FREQ_12288000HZ:
+ value = 0x5;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_write(rt5682->sdw_regmap, 0xe0, value);
+ regmap_write(rt5682->sdw_regmap, 0xf0, value);
+
+ dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq);
+
+ return 0;
+}
+
+static int rt5682_bus_config(struct sdw_slave *slave,
+ struct sdw_bus_params *params)
+{
+ struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+ int ret;
+
+ memcpy(&rt5682->params, params, sizeof(*params));
+
+ ret = rt5682_clock_config(&slave->dev);
+ if (ret < 0)
+ dev_err(&slave->dev, "Invalid clk config");
+
+ return ret;
+}
+
+static int rt5682_interrupt_callback(struct sdw_slave *slave,
+ struct sdw_slave_intr_status *status)
+{
+ struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+
+ dev_dbg(&slave->dev,
+ "%s control_port_stat=%x", __func__, status->control_port);
+
+ if (status->control_port & 0x4) {
+ mod_delayed_work(system_power_efficient_wq,
+ &rt5682->jack_detect_work, msecs_to_jiffies(250));
+ }
+
+ return 0;
+}
+
+static struct sdw_slave_ops rt5682_slave_ops = {
+ .read_prop = rt5682_read_prop,
+ .interrupt_callback = rt5682_interrupt_callback,
+ .update_status = rt5682_update_status,
+ .bus_config = rt5682_bus_config,
+};
+
+static int rt5682_sdw_probe(struct sdw_slave *slave,
+ const struct sdw_device_id *id)
+{
+ struct regmap *regmap;
+
+ /* Assign ops */
+ slave->ops = &rt5682_slave_ops;
+
+ /* Regmap Initialization */
+ regmap = devm_regmap_init_sdw(slave, &rt5682_sdw_regmap);
+ if (IS_ERR(regmap))
+ return -EINVAL;
+
+ rt5682_sdw_init(&slave->dev, regmap, slave);
+
+ return 0;
+}
+
+static int rt5682_sdw_remove(struct sdw_slave *slave)
+{
+ struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+
+ if (rt5682 && rt5682->hw_init)
+ cancel_delayed_work(&rt5682->jack_detect_work);
+
+ return 0;
+}
+
+static const struct sdw_device_id rt5682_id[] = {
+ SDW_SLAVE_ENTRY(0x025d, 0x5682, 0),
+ {},
+};
+MODULE_DEVICE_TABLE(sdw, rt5682_id);
+
+static int __maybe_unused rt5682_dev_suspend(struct device *dev)
+{
+ struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+
+ if (!rt5682->hw_init)
+ return 0;
+
+ regcache_cache_only(rt5682->regmap, true);
+ regcache_mark_dirty(rt5682->regmap);
+
+ return 0;
+}
+
+static int __maybe_unused rt5682_dev_resume(struct device *dev)
+{
+ struct sdw_slave *slave = dev_to_sdw_dev(dev);
+ struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+ unsigned long time;
+
+ if (!rt5682->hw_init)
+ return 0;
+
+ if (!slave->unattach_request)
+ goto regmap_sync;
+
+ time = wait_for_completion_timeout(&slave->initialization_complete,
+ msecs_to_jiffies(RT5682_PROBE_TIMEOUT));
+ if (!time) {
+ dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ return -ETIMEDOUT;
+ }
+
+regmap_sync:
+ slave->unattach_request = 0;
+ regcache_cache_only(rt5682->regmap, false);
+ regcache_sync(rt5682->regmap);
+
+ return 0;
+}
+
+static const struct dev_pm_ops rt5682_pm = {
+ SET_SYSTEM_SLEEP_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume)
+ SET_RUNTIME_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume, NULL)
+};
+
+static struct sdw_driver rt5682_sdw_driver = {
+ .driver = {
+ .name = "rt5682",
+ .owner = THIS_MODULE,
+ .pm = &rt5682_pm,
+ },
+ .probe = rt5682_sdw_probe,
+ .remove = rt5682_sdw_remove,
+ .ops = &rt5682_slave_ops,
+ .id_table = rt5682_id,
+};
+module_sdw_driver(rt5682_sdw_driver);
+
+MODULE_DESCRIPTION("ASoC RT5682 driver SDW");
+MODULE_AUTHOR("Oder Chiou <oder_chiou@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt5682-sdw.h b/sound/soc/codecs/rt5682-sdw.h
new file mode 100644
index 000000000000..76e6f607066e
--- /dev/null
+++ b/sound/soc/codecs/rt5682-sdw.h
@@ -0,0 +1,20 @@
+/* SPDX-License-Identifier: GPL-2.0-only
+ *
+ * rt5682-sdw.h -- RT5682 SDW ALSA SoC audio driver
+ *
+ * Copyright 2019 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <oder_chiou@realtek.com>
+ */
+
+#ifndef __RT5682_SDW_H__
+#define __RT5682_SDW_H__
+
+#define RT5682_SDW_ADDR_L 0x3000
+#define RT5682_SDW_ADDR_H 0x3001
+#define RT5682_SDW_DATA_L 0x3004
+#define RT5682_SDW_DATA_H 0x3005
+#define RT5682_SDW_CMD 0x3008
+
+#define RT5682_PROBE_TIMEOUT 2000
+
+#endif /* __RT5682_SDW_H__ */
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index ae6f6121bc1b..c9268a230daa 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -11,13 +11,13 @@
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
+#include <linux/pm_runtime.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <linux/acpi.h>
#include <linux/gpio.h>
#include <linux/of_gpio.h>
-#include <linux/regulator/consumer.h>
#include <linux/mutex.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -31,8 +31,7 @@
#include "rl6231.h"
#include "rt5682.h"
-
-#define RT5682_NUM_SUPPLIES 3
+#include "rt5682-sdw.h"
static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = {
"AVDD",
@@ -45,35 +44,15 @@ static const struct rt5682_platform_data i2s_default_platform_data = {
.dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3,
.jd_src = RT5682_JD1,
.btndet_delay = 16,
-};
-
-struct rt5682_priv {
- struct snd_soc_component *component;
- struct rt5682_platform_data pdata;
- struct regmap *regmap;
- struct snd_soc_jack *hs_jack;
- struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES];
- struct delayed_work jack_detect_work;
- struct delayed_work jd_check_work;
- struct mutex calibrate_mutex;
-
- int sysclk;
- int sysclk_src;
- int lrck[RT5682_AIFS];
- int bclk[RT5682_AIFS];
- int master[RT5682_AIFS];
-
- int pll_src;
- int pll_in;
- int pll_out;
-
- int jack_type;
+ .dai_clk_names[RT5682_DAI_WCLK_IDX] = "rt5682-dai-wclk",
+ .dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk",
};
static const struct reg_sequence patch_list[] = {
{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
{RT5682_I2C_CTRL, 0x000f},
+ {RT5682_PLL2_INTERNAL, 0x8266},
};
static const struct reg_default rt5682_reg[] = {
@@ -221,7 +200,7 @@ static const struct reg_default rt5682_reg[] = {
{0x0148, 0x0000},
{0x0149, 0x0000},
{0x0150, 0x79a1},
- {0x0151, 0x0000},
+ {0x0156, 0xaaaa},
{0x0160, 0x4ec0},
{0x0161, 0x0080},
{0x0162, 0x0200},
@@ -805,10 +784,27 @@ static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux =
static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux =
SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum);
-static void rt5682_reset(struct regmap *regmap)
+static const char * const rt5682_dac_select[] = {
+ "IF1", "SOUND"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5682_dacl_enum,
+ RT5682_AD_DA_MIXER, RT5682_DAC1_L_SEL_SFT, rt5682_dac_select);
+
+static const struct snd_kcontrol_new rt5682_dac_l_mux =
+ SOC_DAPM_ENUM("DAC L Mux", rt5682_dacl_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_dacr_enum,
+ RT5682_AD_DA_MIXER, RT5682_DAC1_R_SEL_SFT, rt5682_dac_select);
+
+static const struct snd_kcontrol_new rt5682_dac_r_mux =
+ SOC_DAPM_ENUM("DAC R Mux", rt5682_dacr_enum);
+
+static void rt5682_reset(struct rt5682_priv *rt5682)
{
- regmap_write(regmap, RT5682_RESET, 0);
- regmap_write(regmap, RT5682_I2C_MODE, 1);
+ regmap_write(rt5682->regmap, RT5682_RESET, 0);
+ if (!rt5682->is_sdw)
+ regmap_write(rt5682->regmap, RT5682_I2C_MODE, 1);
}
/**
* rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters
@@ -871,6 +867,8 @@ static int rt5682_button_detect(struct snd_soc_component *component)
static void rt5682_enable_push_button_irq(struct snd_soc_component *component,
bool enable)
{
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
if (enable) {
snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN);
@@ -880,8 +878,15 @@ static void rt5682_enable_push_button_irq(struct snd_soc_component *component,
snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK,
RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR);
- snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
- RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN);
+ if (rt5682->is_sdw)
+ snd_soc_component_update_bits(component,
+ RT5682_IRQ_CTRL_3,
+ RT5682_IL_IRQ_MASK | RT5682_IL_IRQ_TYPE_MASK,
+ RT5682_IL_IRQ_EN | RT5682_IL_IRQ_PUL);
+ else
+ snd_soc_component_update_bits(component,
+ RT5682_IRQ_CTRL_3, RT5682_IL_IRQ_MASK,
+ RT5682_IL_IRQ_EN);
} else {
snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS);
@@ -909,6 +914,7 @@ static int rt5682_headset_detect(struct snd_soc_component *component,
int jack_insert)
{
struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ struct snd_soc_dapm_context *dapm = &component->dapm;
unsigned int val, count;
if (jack_insert) {
@@ -917,10 +923,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component,
RT5682_PWR_VREF2 | RT5682_PWR_MB,
RT5682_PWR_VREF2 | RT5682_PWR_MB);
snd_soc_component_update_bits(component,
- RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0);
+ RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0);
usleep_range(15000, 20000);
snd_soc_component_update_bits(component,
- RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2);
+ RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
RT5682_PWR_CBJ, RT5682_PWR_CBJ);
@@ -951,8 +957,13 @@ static int rt5682_headset_detect(struct snd_soc_component *component,
rt5682_enable_push_button_irq(component, false);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
- snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
- RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
+ if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+ snd_soc_component_update_bits(component,
+ RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
+ else
+ snd_soc_component_update_bits(component,
+ RT5682_PWR_ANLG_1,
+ RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
RT5682_PWR_CBJ, 0);
@@ -999,62 +1010,69 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
rt5682->hs_jack = hs_jack;
- if (!hs_jack) {
- regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
- RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
- regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
- RT5682_POW_JDH | RT5682_POW_JDL, 0);
- cancel_delayed_work_sync(&rt5682->jack_detect_work);
- return 0;
- }
+ if (!rt5682->is_sdw) {
+ if (!hs_jack) {
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ RT5682_POW_JDH | RT5682_POW_JDL, 0);
+ cancel_delayed_work_sync(&rt5682->jack_detect_work);
+ return 0;
+ }
- switch (rt5682->pdata.jd_src) {
- case RT5682_JD1:
- snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
- RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
- snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042);
- snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3,
- RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
- snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
- RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN);
- regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
- RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ);
- regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ switch (rt5682->pdata.jd_src) {
+ case RT5682_JD1:
+ snd_soc_component_update_bits(component,
+ RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC,
+ RT5682_EXT_JD_SRC_MANUAL);
+ snd_soc_component_write(component, RT5682_CBJ_CTRL_1,
+ 0xd042);
+ snd_soc_component_update_bits(component,
+ RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN,
+ RT5682_CBJ_IN_BUF_EN);
+ snd_soc_component_update_bits(component,
+ RT5682_SAR_IL_CMD_1, RT5682_SAR_POW_MASK,
+ RT5682_SAR_POW_EN);
+ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+ RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
RT5682_POW_IRQ | RT5682_POW_JDH |
RT5682_POW_ANA, RT5682_POW_IRQ |
RT5682_POW_JDH | RT5682_POW_ANA);
- regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
- RT5682_PWR_JDH | RT5682_PWR_JDL,
- RT5682_PWR_JDH | RT5682_PWR_JDL);
- regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
- RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK,
- RT5682_JD1_EN | RT5682_JD1_POL_NOR);
- regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4,
- 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
- rt5682->pdata.btndet_delay));
- regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5,
- 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
- rt5682->pdata.btndet_delay));
- regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6,
- 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
- rt5682->pdata.btndet_delay));
- regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7,
- 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
- rt5682->pdata.btndet_delay));
- mod_delayed_work(system_power_efficient_wq,
- &rt5682->jack_detect_work, msecs_to_jiffies(250));
- break;
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
+ RT5682_PWR_JDH | RT5682_PWR_JDL,
+ RT5682_PWR_JDH | RT5682_PWR_JDL);
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK,
+ RT5682_JD1_EN | RT5682_JD1_POL_NOR);
+ regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4,
+ 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+ rt5682->pdata.btndet_delay));
+ regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5,
+ 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+ rt5682->pdata.btndet_delay));
+ regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6,
+ 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+ rt5682->pdata.btndet_delay));
+ regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7,
+ 0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+ rt5682->pdata.btndet_delay));
+ mod_delayed_work(system_power_efficient_wq,
+ &rt5682->jack_detect_work,
+ msecs_to_jiffies(250));
+ break;
- case RT5682_JD_NULL:
- regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
- RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
- regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
- RT5682_POW_JDH | RT5682_POW_JDL, 0);
- break;
+ case RT5682_JD_NULL:
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ RT5682_POW_JDH | RT5682_POW_JDL, 0);
+ break;
- default:
- dev_warn(component->dev, "Wrong JD source\n");
- break;
+ default:
+ dev_warn(component->dev, "Wrong JD source\n");
+ break;
+ }
}
return 0;
@@ -1134,11 +1152,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work)
SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3);
- if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
- SND_JACK_BTN_2 | SND_JACK_BTN_3))
- schedule_delayed_work(&rt5682->jd_check_work, 0);
- else
- cancel_delayed_work_sync(&rt5682->jd_check_work);
+ if (!rt5682->is_sdw) {
+ if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3))
+ schedule_delayed_work(&rt5682->jd_check_work, 0);
+ else
+ cancel_delayed_work_sync(&rt5682->jd_check_work);
+ }
mutex_unlock(&rt5682->calibrate_mutex);
}
@@ -1146,7 +1166,7 @@ static void rt5682_jack_detect_handler(struct work_struct *work)
static const struct snd_kcontrol_new rt5682_snd_controls[] = {
/* DAC Digital Volume */
SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL,
- RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv),
+ RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 87, 0, dac_vol_tlv),
/* IN Boost Volume */
SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL,
@@ -1177,11 +1197,11 @@ static int rt5682_div_sel(struct rt5682_priv *rt5682,
}
for (i = 0; i < size - 1; i++) {
- pr_info("div[%d]=%d\n", i, div[i]);
+ dev_dbg(rt5682->component->dev, "div[%d]=%d\n", i, div[i]);
if (target * div[i] == rt5682->sysclk)
return i;
if (target * div[i + 1] > rt5682->sysclk) {
- pr_err("can't find div for sysclk %d\n",
+ dev_dbg(rt5682->component->dev, "can't find div for sysclk %d\n",
rt5682->sysclk);
return i;
}
@@ -1211,10 +1231,13 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
struct snd_soc_component *component =
snd_soc_dapm_to_component(w->dapm);
struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
- int idx = -EINVAL;
+ int idx = -EINVAL, dmic_clk_rate = 3072000;
static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128};
- idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div));
+ if (rt5682->pdata.dmic_clk_rate)
+ dmic_clk_rate = rt5682->pdata.dmic_clk_rate;
+
+ idx = rt5682_div_sel(rt5682, dmic_clk_rate, div, ARRAY_SIZE(div));
snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1,
RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT);
@@ -1232,6 +1255,9 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w,
static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48};
static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48};
+ if (rt5682->is_sdw)
+ return 0;
+
val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) &
RT5682_GP4_PIN_MASK;
if (w->shift == RT5682_PWR_ADC_S1F_BIT &&
@@ -1278,6 +1304,21 @@ static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w,
return 0;
}
+static int is_sys_clk_from_pll2(struct snd_soc_dapm_widget *w,
+ struct snd_soc_dapm_widget *sink)
+{
+ unsigned int val;
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ val = snd_soc_component_read32(component, RT5682_GLB_CLK);
+ val &= RT5682_SCLK_SRC_MASK;
+ if (val == RT5682_SCLK_SRC_PLL2)
+ return 1;
+ else
+ return 0;
+}
+
static int is_using_asrc(struct snd_soc_dapm_widget *w,
struct snd_soc_dapm_widget *sink)
{
@@ -1503,10 +1544,18 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
static int set_dmic_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ unsigned int delay = 50;
+
+ if (rt5682->pdata.dmic_delay)
+ delay = rt5682->pdata.dmic_delay;
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/*Add delay to avoid pop noise*/
- msleep(150);
+ msleep(delay);
break;
default:
@@ -1516,7 +1565,7 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w,
return 0;
}
-static int rt5655_set_verf(struct snd_soc_dapm_widget *w,
+static int rt5682_set_verf(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component =
@@ -1592,9 +1641,12 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT,
0, NULL, 0),
SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT,
- 0, NULL, 0),
+ 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
- rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0),
/* ASRC */
SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
@@ -1686,6 +1738,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("SOUND DAC L", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("SOUND DAC R", SND_SOC_NOPM, 0, 0, NULL, 0),
/* Digital Interface Select */
SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
@@ -1702,12 +1756,19 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0,
&rt5682_adcdat_pin_ctrl),
+ SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_dac_l_mux),
+ SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0,
+ &rt5682_dac_r_mux),
+
/* Audio Interface */
SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0,
RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1),
SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0,
RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1),
SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SDWRX", "SDW Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SDWTX", "SDW Capture", 0, SND_SOC_NOPM, 0, 0),
/* Output Side */
/* DAC mixer before sound effect */
@@ -1776,7 +1837,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
/*PLL*/
{"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+ {"ADC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2},
+ {"ADC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2},
{"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+ {"DAC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2},
+ {"DAC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2},
/*ASRC*/
{"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc},
@@ -1860,8 +1925,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
{"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"},
{"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"},
{"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"},
- {"IF1_ADC Mux", NULL, "I2S1"},
{"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"},
+ {"AIF1TX", NULL, "I2S1"},
{"AIF1TX", NULL, "ADCDAT Mux"},
{"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
{"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
@@ -1870,6 +1935,10 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
{"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"},
{"AIF2TX", NULL, "ADCDAT Mux"},
+ {"SDWTX", NULL, "PLL2B"},
+ {"SDWTX", NULL, "PLL2F"},
+ {"SDWTX", NULL, "ADCDAT Mux"},
+
{"IF1 DAC1 L", NULL, "AIF1RX"},
{"IF1 DAC1 L", NULL, "I2S1"},
{"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"},
@@ -1877,10 +1946,24 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
{"IF1 DAC1 R", NULL, "I2S1"},
{"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"},
+ {"SOUND DAC L", NULL, "SDWRX"},
+ {"SOUND DAC L", NULL, "DAC Stereo1 Filter"},
+ {"SOUND DAC L", NULL, "PLL2B"},
+ {"SOUND DAC L", NULL, "PLL2F"},
+ {"SOUND DAC R", NULL, "SDWRX"},
+ {"SOUND DAC R", NULL, "DAC Stereo1 Filter"},
+ {"SOUND DAC R", NULL, "PLL2B"},
+ {"SOUND DAC R", NULL, "PLL2F"},
+
+ {"DAC L Mux", "IF1", "IF1 DAC1 L"},
+ {"DAC L Mux", "SOUND", "SOUND DAC L"},
+ {"DAC R Mux", "IF1", "IF1 DAC1 R"},
+ {"DAC R Mux", "SOUND", "SOUND DAC R"},
+
{"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"},
- {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"},
+ {"DAC1 MIXL", "DAC1 Switch", "DAC L Mux"},
{"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"},
- {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"},
+ {"DAC1 MIXR", "DAC1 Switch", "DAC R Mux"},
{"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"},
{"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"},
@@ -2033,8 +2116,10 @@ static int rt5682_hw_params(struct snd_pcm_substream *substream,
RT5682_I2S1_DL_MASK, len_1);
if (rt5682->master[RT5682_AIF1]) {
snd_soc_component_update_bits(component,
- RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK,
- pre_div << RT5682_I2S_M_DIV_SFT);
+ RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK |
+ RT5682_I2S_CLK_SRC_MASK,
+ pre_div << RT5682_I2S_M_DIV_SFT |
+ (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT);
}
if (params_channels(params) == 1) /* mono mode */
snd_soc_component_update_bits(component,
@@ -2207,61 +2292,157 @@ static int rt5682_set_component_pll(struct snd_soc_component *component,
unsigned int freq_out)
{
struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
- struct rl6231_pll_code pll_code;
+ struct rl6231_pll_code pll_code, pll2f_code, pll2b_code;
+ unsigned int pll2_fout1;
int ret;
- if (source == rt5682->pll_src && freq_in == rt5682->pll_in &&
- freq_out == rt5682->pll_out)
+ if (source == rt5682->pll_src[pll_id] &&
+ freq_in == rt5682->pll_in[pll_id] &&
+ freq_out == rt5682->pll_out[pll_id])
return 0;
if (!freq_in || !freq_out) {
dev_dbg(component->dev, "PLL disabled\n");
- rt5682->pll_in = 0;
- rt5682->pll_out = 0;
+ rt5682->pll_in[pll_id] = 0;
+ rt5682->pll_out[pll_id] = 0;
snd_soc_component_update_bits(component, RT5682_GLB_CLK,
RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK);
return 0;
}
- switch (source) {
- case RT5682_PLL1_S_MCLK:
- snd_soc_component_update_bits(component, RT5682_GLB_CLK,
- RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK);
- break;
- case RT5682_PLL1_S_BCLK1:
- snd_soc_component_update_bits(component, RT5682_GLB_CLK,
- RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1);
- break;
- default:
- dev_err(component->dev, "Unknown PLL Source %d\n", source);
- return -EINVAL;
- }
+ if (pll_id == RT5682_PLL2) {
+ switch (source) {
+ case RT5682_PLL2_S_MCLK:
+ snd_soc_component_update_bits(component,
+ RT5682_GLB_CLK, RT5682_PLL2_SRC_MASK,
+ RT5682_PLL2_SRC_MCLK);
+ break;
+ default:
+ dev_err(component->dev, "Unknown PLL2 Source %d\n",
+ source);
+ return -EINVAL;
+ }
- ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
- if (ret < 0) {
- dev_err(component->dev, "Unsupport input clock %d\n", freq_in);
- return ret;
+ /**
+ * PLL2 concatenates 2 PLL units.
+ * We suggest the Fout of the front PLL is 3.84MHz.
+ */
+ pll2_fout1 = 3840000;
+ ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code);
+ if (ret < 0) {
+ dev_err(component->dev, "Unsupport input clock %d\n",
+ freq_in);
+ return ret;
+ }
+ dev_dbg(component->dev, "PLL2F: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n",
+ freq_in, pll2_fout1,
+ pll2f_code.m_bp,
+ (pll2f_code.m_bp ? 0 : pll2f_code.m_code),
+ pll2f_code.n_code, pll2f_code.k_code);
+
+ ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code);
+ if (ret < 0) {
+ dev_err(component->dev, "Unsupport input clock %d\n",
+ pll2_fout1);
+ return ret;
+ }
+ dev_dbg(component->dev, "PLL2B: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n",
+ pll2_fout1, freq_out,
+ pll2b_code.m_bp,
+ (pll2b_code.m_bp ? 0 : pll2b_code.m_code),
+ pll2b_code.n_code, pll2b_code.k_code);
+
+ snd_soc_component_write(component, RT5682_PLL2_CTRL_1,
+ pll2f_code.k_code << RT5682_PLL2F_K_SFT |
+ pll2b_code.k_code << RT5682_PLL2B_K_SFT |
+ pll2b_code.m_code);
+ snd_soc_component_write(component, RT5682_PLL2_CTRL_2,
+ pll2f_code.m_code << RT5682_PLL2F_M_SFT |
+ pll2b_code.n_code);
+ snd_soc_component_write(component, RT5682_PLL2_CTRL_3,
+ pll2f_code.n_code << RT5682_PLL2F_N_SFT);
+ snd_soc_component_update_bits(component, RT5682_PLL2_CTRL_4,
+ RT5682_PLL2B_M_BP_MASK | RT5682_PLL2F_M_BP_MASK | 0xf,
+ (pll2b_code.m_bp ? 1 : 0) << RT5682_PLL2B_M_BP_SFT |
+ (pll2f_code.m_bp ? 1 : 0) << RT5682_PLL2F_M_BP_SFT |
+ 0xf);
+ } else {
+ switch (source) {
+ case RT5682_PLL1_S_MCLK:
+ snd_soc_component_update_bits(component,
+ RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK,
+ RT5682_PLL1_SRC_MCLK);
+ break;
+ case RT5682_PLL1_S_BCLK1:
+ snd_soc_component_update_bits(component,
+ RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK,
+ RT5682_PLL1_SRC_BCLK1);
+ break;
+ default:
+ dev_err(component->dev, "Unknown PLL1 Source %d\n",
+ source);
+ return -EINVAL;
+ }
+
+ ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
+ if (ret < 0) {
+ dev_err(component->dev, "Unsupport input clock %d\n",
+ freq_in);
+ return ret;
+ }
+
+ dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n",
+ pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
+ pll_code.n_code, pll_code.k_code);
+
+ snd_soc_component_write(component, RT5682_PLL_CTRL_1,
+ pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code);
+ snd_soc_component_write(component, RT5682_PLL_CTRL_2,
+ (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT |
+ pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST);
}
- dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n",
- pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
- pll_code.n_code, pll_code.k_code);
+ rt5682->pll_in[pll_id] = freq_in;
+ rt5682->pll_out[pll_id] = freq_out;
+ rt5682->pll_src[pll_id] = source;
- snd_soc_component_write(component, RT5682_PLL_CTRL_1,
- pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code);
- snd_soc_component_write(component, RT5682_PLL_CTRL_2,
- (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT |
- pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST);
+ return 0;
+}
- rt5682->pll_in = freq_in;
- rt5682->pll_out = freq_out;
- rt5682->pll_src = source;
+static int rt5682_set_bclk1_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+ rt5682->bclk[dai->id] = ratio;
+
+ switch (ratio) {
+ case 256:
+ snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+ RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_256);
+ break;
+ case 128:
+ snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+ RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_128);
+ break;
+ case 64:
+ snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+ RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_64);
+ break;
+ case 32:
+ snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+ RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_32);
+ break;
+ default:
+ dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio);
+ return -EINVAL;
+ }
return 0;
}
-static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+static int rt5682_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio)
{
struct snd_soc_component *component = dai->component;
struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
@@ -2280,7 +2461,7 @@ static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
RT5682_I2S2_BCLK_MS2_32);
break;
default:
- dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio);
+ dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio);
return -EINVAL;
}
@@ -2319,12 +2500,392 @@ static int rt5682_set_bias_level(struct snd_soc_component *component,
return 0;
}
+#ifdef CONFIG_COMMON_CLK
+#define CLK_PLL2_FIN 48000000
+#define CLK_PLL2_FOUT 24576000
+#define CLK_48 48000
+
+static bool rt5682_clk_check(struct rt5682_priv *rt5682)
+{
+ if (!rt5682->master[RT5682_AIF1]) {
+ dev_err(rt5682->component->dev, "sysclk/dai not set correctly\n");
+ return false;
+ }
+ return true;
+}
+
+static int rt5682_wclk_prepare(struct clk_hw *hw)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+ struct snd_soc_component *component = rt5682->component;
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(component);
+
+ if (!rt5682_clk_check(rt5682))
+ return -EINVAL;
+
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
+ snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_MB, RT5682_PWR_MB);
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B");
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+
+ return 0;
+}
+
+static void rt5682_wclk_unprepare(struct clk_hw *hw)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+ struct snd_soc_component *component = rt5682->component;
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(component);
+
+ if (!rt5682_clk_check(rt5682))
+ return;
+
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
+ if (!rt5682->jack_type)
+ snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_MB, 0);
+ snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B");
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw,
+ unsigned long parent_rate)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+
+ if (!rt5682_clk_check(rt5682))
+ return 0;
+ /*
+ * Only accept to set wclk rate to 48kHz temporarily.
+ */
+ return CLK_48;
+}
+
+static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate,
+ unsigned long *parent_rate)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+
+ if (!rt5682_clk_check(rt5682))
+ return -EINVAL;
+ /*
+ * Only accept to set wclk rate to 48kHz temporarily.
+ */
+ return CLK_48;
+}
+
+static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate,
+ unsigned long parent_rate)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+ struct snd_soc_component *component = rt5682->component;
+ struct clk *parent_clk;
+ const char * const clk_name = __clk_get_name(hw->clk);
+ int pre_div;
+
+ if (!rt5682_clk_check(rt5682))
+ return -EINVAL;
+
+ /*
+ * Whether the wclk's parent clk (mclk) exists or not, please ensure
+ * it is fixed or set to 48MHz before setting wclk rate. It's a
+ * temporary limitation. Only accept 48MHz clk as the clk provider.
+ *
+ * It will set the codec anyway by assuming mclk is 48MHz.
+ */
+ parent_clk = clk_get_parent(hw->clk);
+ if (!parent_clk)
+ dev_warn(component->dev,
+ "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n",
+ CLK_PLL2_FIN);
+
+ if (parent_rate != CLK_PLL2_FIN)
+ dev_warn(component->dev, "clk %s only support %d Hz input\n",
+ clk_name, CLK_PLL2_FIN);
+
+ /*
+ * It's a temporary limitation. Only accept to set wclk rate to 48kHz.
+ * It will force wclk to 48kHz even it's not.
+ */
+ if (rate != CLK_48) {
+ dev_warn(component->dev, "clk %s only support %d Hz output\n",
+ clk_name, CLK_48);
+ rate = CLK_48;
+ }
+
+ /*
+ * To achieve the rate conversion from 48MHz to 48kHz, PLL2 is needed.
+ */
+ rt5682_set_component_pll(component, RT5682_PLL2, RT5682_PLL2_S_MCLK,
+ CLK_PLL2_FIN, CLK_PLL2_FOUT);
+
+ rt5682_set_component_sysclk(component, RT5682_SCLK_S_PLL2, 0,
+ CLK_PLL2_FOUT, SND_SOC_CLOCK_IN);
+
+ pre_div = rl6231_get_clk_info(rt5682->sysclk, rate);
+
+ snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1,
+ RT5682_I2S_M_DIV_MASK | RT5682_I2S_CLK_SRC_MASK,
+ pre_div << RT5682_I2S_M_DIV_SFT |
+ (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT);
+
+ return 0;
+}
+
+static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw,
+ unsigned long parent_rate)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_BCLK_IDX]);
+ struct snd_soc_component *component = rt5682->component;
+ unsigned int bclks_per_wclk;
+
+ snd_soc_component_read(component, RT5682_TDM_TCON_CTRL,
+ &bclks_per_wclk);
+
+ switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) {
+ case RT5682_TDM_BCLK_MS1_256:
+ return parent_rate * 256;
+ case RT5682_TDM_BCLK_MS1_128:
+ return parent_rate * 128;
+ case RT5682_TDM_BCLK_MS1_64:
+ return parent_rate * 64;
+ case RT5682_TDM_BCLK_MS1_32:
+ return parent_rate * 32;
+ default:
+ return 0;
+ }
+}
+
+static unsigned long rt5682_bclk_get_factor(unsigned long rate,
+ unsigned long parent_rate)
+{
+ unsigned long factor;
+
+ factor = rate / parent_rate;
+ if (factor < 64)
+ return 32;
+ else if (factor < 128)
+ return 64;
+ else if (factor < 256)
+ return 128;
+ else
+ return 256;
+}
+
+static long rt5682_bclk_round_rate(struct clk_hw *hw, unsigned long rate,
+ unsigned long *parent_rate)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_BCLK_IDX]);
+ unsigned long factor;
+
+ if (!*parent_rate || !rt5682_clk_check(rt5682))
+ return -EINVAL;
+
+ /*
+ * BCLK rates are set as a multiplier of WCLK in HW.
+ * We don't allow changing the parent WCLK. We just do
+ * some rounding down based on the parent WCLK rate
+ * and find the appropriate multiplier of BCLK to
+ * get the rounded down BCLK value.
+ */
+ factor = rt5682_bclk_get_factor(rate, *parent_rate);
+
+ return *parent_rate * factor;
+}
+
+static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate,
+ unsigned long parent_rate)
+{
+ struct rt5682_priv *rt5682 =
+ container_of(hw, struct rt5682_priv,
+ dai_clks_hw[RT5682_DAI_BCLK_IDX]);
+ struct snd_soc_component *component = rt5682->component;
+ struct snd_soc_dai *dai = NULL;
+ unsigned long factor;
+
+ if (!rt5682_clk_check(rt5682))
+ return -EINVAL;
+
+ factor = rt5682_bclk_get_factor(rate, parent_rate);
+
+ for_each_component_dais(component, dai)
+ if (dai->id == RT5682_AIF1)
+ break;
+ if (!dai) {
+ dev_err(component->dev, "dai %d not found in component\n",
+ RT5682_AIF1);
+ return -ENODEV;
+ }
+
+ return rt5682_set_bclk1_ratio(dai, factor);
+}
+
+static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = {
+ [RT5682_DAI_WCLK_IDX] = {
+ .prepare = rt5682_wclk_prepare,
+ .unprepare = rt5682_wclk_unprepare,
+ .recalc_rate = rt5682_wclk_recalc_rate,
+ .round_rate = rt5682_wclk_round_rate,
+ .set_rate = rt5682_wclk_set_rate,
+ },
+ [RT5682_DAI_BCLK_IDX] = {
+ .recalc_rate = rt5682_bclk_recalc_rate,
+ .round_rate = rt5682_bclk_round_rate,
+ .set_rate = rt5682_bclk_set_rate,
+ },
+};
+
+static int rt5682_register_dai_clks(struct snd_soc_component *component)
+{
+ struct device *dev = component->dev;
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ struct rt5682_platform_data *pdata = &rt5682->pdata;
+ struct clk_init_data init;
+ struct clk *dai_clk;
+ struct clk_lookup *dai_clk_lookup;
+ struct clk_hw *dai_clk_hw;
+ const char *parent_name;
+ int i, ret;
+
+ for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) {
+ dai_clk_hw = &rt5682->dai_clks_hw[i];
+
+ switch (i) {
+ case RT5682_DAI_WCLK_IDX:
+ /* Make MCLK the parent of WCLK */
+ if (rt5682->mclk) {
+ parent_name = __clk_get_name(rt5682->mclk);
+ init.parent_names = &parent_name;
+ init.num_parents = 1;
+ } else {
+ init.parent_names = NULL;
+ init.num_parents = 0;
+ }
+ break;
+ case RT5682_DAI_BCLK_IDX:
+ /* Make WCLK the parent of BCLK */
+ parent_name = __clk_get_name(
+ rt5682->dai_clks[RT5682_DAI_WCLK_IDX]);
+ init.parent_names = &parent_name;
+ init.num_parents = 1;
+ break;
+ default:
+ dev_err(dev, "Invalid clock index\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ init.name = pdata->dai_clk_names[i];
+ init.ops = &rt5682_dai_clk_ops[i];
+ init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE;
+ dai_clk_hw->init = &init;
+
+ dai_clk = devm_clk_register(dev, dai_clk_hw);
+ if (IS_ERR(dai_clk)) {
+ dev_warn(dev, "Failed to register %s: %ld\n",
+ init.name, PTR_ERR(dai_clk));
+ ret = PTR_ERR(dai_clk);
+ goto err;
+ }
+ rt5682->dai_clks[i] = dai_clk;
+
+ if (dev->of_node) {
+ devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get,
+ dai_clk_hw);
+ } else {
+ dai_clk_lookup = clkdev_create(dai_clk, init.name,
+ "%s", dev_name(dev));
+ if (!dai_clk_lookup) {
+ ret = -ENOMEM;
+ goto err;
+ } else {
+ rt5682->dai_clks_lookup[i] = dai_clk_lookup;
+ }
+ }
+ }
+
+ return 0;
+
+err:
+ do {
+ if (rt5682->dai_clks_lookup[i])
+ clkdev_drop(rt5682->dai_clks_lookup[i]);
+ } while (i-- > 0);
+
+ return ret;
+}
+#endif /* CONFIG_COMMON_CLK */
+
static int rt5682_probe(struct snd_soc_component *component)
{
struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ struct sdw_slave *slave;
+ unsigned long time;
+#ifdef CONFIG_COMMON_CLK
+ int ret;
+#endif
rt5682->component = component;
+ if (rt5682->is_sdw) {
+ slave = rt5682->slave;
+ time = wait_for_completion_timeout(
+ &slave->initialization_complete,
+ msecs_to_jiffies(RT5682_PROBE_TIMEOUT));
+ if (!time) {
+ dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ return -ETIMEDOUT;
+ }
+ } else {
+#ifdef CONFIG_COMMON_CLK
+ /* Check if MCLK provided */
+ rt5682->mclk = devm_clk_get(component->dev, "mclk");
+ if (IS_ERR(rt5682->mclk)) {
+ if (PTR_ERR(rt5682->mclk) != -ENOENT) {
+ ret = PTR_ERR(rt5682->mclk);
+ return ret;
+ }
+ rt5682->mclk = NULL;
+ } else {
+ /* Register CCF DAI clock control */
+ ret = rt5682_register_dai_clks(component);
+ if (ret)
+ return ret;
+ }
+ /* Initial setup for CCF */
+ rt5682->lrck[RT5682_AIF1] = CLK_48;
+#endif
+ }
+
return 0;
}
@@ -2332,7 +2893,16 @@ static void rt5682_remove(struct snd_soc_component *component)
{
struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
- rt5682_reset(rt5682->regmap);
+#ifdef CONFIG_COMMON_CLK
+ int i;
+
+ for (i = RT5682_DAI_NUM_CLKS - 1; i >= 0; --i) {
+ if (rt5682->dai_clks_lookup[i])
+ clkdev_drop(rt5682->dai_clks_lookup[i]);
+ }
+#endif
+
+ rt5682_reset(rt5682);
}
#ifdef CONFIG_PM
@@ -2369,14 +2939,203 @@ static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = {
.hw_params = rt5682_hw_params,
.set_fmt = rt5682_set_dai_fmt,
.set_tdm_slot = rt5682_set_tdm_slot,
+ .set_bclk_ratio = rt5682_set_bclk1_ratio,
};
static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = {
.hw_params = rt5682_hw_params,
.set_fmt = rt5682_set_dai_fmt,
- .set_bclk_ratio = rt5682_set_bclk_ratio,
+ .set_bclk_ratio = rt5682_set_bclk2_ratio,
};
+#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW)
+struct sdw_stream_data {
+ struct sdw_stream_runtime *sdw_stream;
+};
+
+static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
+ int direction)
+{
+ struct sdw_stream_data *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (!stream)
+ return -ENOMEM;
+
+ stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream;
+
+ /* Use tx_mask or rx_mask to configure stream tag and set dma_data */
+ if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback_dma_data = stream;
+ else
+ dai->capture_dma_data = stream;
+
+ return 0;
+}
+
+static void rt5682_sdw_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sdw_stream_data *stream;
+
+ stream = snd_soc_dai_get_dma_data(dai, substream);
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+ kfree(stream);
+}
+
+static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ struct sdw_stream_config stream_config;
+ struct sdw_port_config port_config;
+ enum sdw_data_direction direction;
+ struct sdw_stream_data *stream;
+ int retval, port, num_channels;
+ unsigned int val_p = 0, val_c = 0, osr_p = 0, osr_c = 0;
+
+ dev_dbg(dai->dev, "%s %s", __func__, dai->name);
+ stream = snd_soc_dai_get_dma_data(dai, substream);
+
+ if (!stream)
+ return -ENOMEM;
+
+ if (!rt5682->slave)
+ return -EINVAL;
+
+ /* SoundWire specific configuration */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ direction = SDW_DATA_DIR_RX;
+ port = 1;
+ } else {
+ direction = SDW_DATA_DIR_TX;
+ port = 2;
+ }
+
+ stream_config.frame_rate = params_rate(params);
+ stream_config.ch_count = params_channels(params);
+ stream_config.bps = snd_pcm_format_width(params_format(params));
+ stream_config.direction = direction;
+
+ num_channels = params_channels(params);
+ port_config.ch_mask = (1 << (num_channels)) - 1;
+ port_config.num = port;
+
+ retval = sdw_stream_add_slave(rt5682->slave, &stream_config,
+ &port_config, 1, stream->sdw_stream);
+ if (retval) {
+ dev_err(dai->dev, "Unable to configure port\n");
+ return retval;
+ }
+
+ switch (params_rate(params)) {
+ case 48000:
+ val_p = RT5682_SDW_REF_1_48K;
+ val_c = RT5682_SDW_REF_2_48K;
+ break;
+ case 96000:
+ val_p = RT5682_SDW_REF_1_96K;
+ val_c = RT5682_SDW_REF_2_96K;
+ break;
+ case 192000:
+ val_p = RT5682_SDW_REF_1_192K;
+ val_c = RT5682_SDW_REF_2_192K;
+ break;
+ case 32000:
+ val_p = RT5682_SDW_REF_1_32K;
+ val_c = RT5682_SDW_REF_2_32K;
+ break;
+ case 24000:
+ val_p = RT5682_SDW_REF_1_24K;
+ val_c = RT5682_SDW_REF_2_24K;
+ break;
+ case 16000:
+ val_p = RT5682_SDW_REF_1_16K;
+ val_c = RT5682_SDW_REF_2_16K;
+ break;
+ case 12000:
+ val_p = RT5682_SDW_REF_1_12K;
+ val_c = RT5682_SDW_REF_2_12K;
+ break;
+ case 8000:
+ val_p = RT5682_SDW_REF_1_8K;
+ val_c = RT5682_SDW_REF_2_8K;
+ break;
+ case 44100:
+ val_p = RT5682_SDW_REF_1_44K;
+ val_c = RT5682_SDW_REF_2_44K;
+ break;
+ case 88200:
+ val_p = RT5682_SDW_REF_1_88K;
+ val_c = RT5682_SDW_REF_2_88K;
+ break;
+ case 176400:
+ val_p = RT5682_SDW_REF_1_176K;
+ val_c = RT5682_SDW_REF_2_176K;
+ break;
+ case 22050:
+ val_p = RT5682_SDW_REF_1_22K;
+ val_c = RT5682_SDW_REF_2_22K;
+ break;
+ case 11025:
+ val_p = RT5682_SDW_REF_1_11K;
+ val_c = RT5682_SDW_REF_2_11K;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (params_rate(params) <= 48000) {
+ osr_p = RT5682_DAC_OSR_D_8;
+ osr_c = RT5682_ADC_OSR_D_8;
+ } else if (params_rate(params) <= 96000) {
+ osr_p = RT5682_DAC_OSR_D_4;
+ osr_c = RT5682_ADC_OSR_D_4;
+ } else {
+ osr_p = RT5682_DAC_OSR_D_2;
+ osr_c = RT5682_ADC_OSR_D_2;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK,
+ RT5682_SDW_REF_1_MASK, val_p);
+ regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1,
+ RT5682_DAC_OSR_MASK, osr_p);
+ } else {
+ regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK,
+ RT5682_SDW_REF_2_MASK, val_c);
+ regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1,
+ RT5682_ADC_OSR_MASK, osr_c);
+ }
+
+ return retval;
+}
+
+static int rt5682_sdw_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+ struct sdw_stream_data *stream =
+ snd_soc_dai_get_dma_data(dai, substream);
+
+ if (!rt5682->slave)
+ return -EINVAL;
+
+ sdw_stream_remove_slave(rt5682->slave, stream->sdw_stream);
+ return 0;
+}
+
+static struct snd_soc_dai_ops rt5682_sdw_ops = {
+ .hw_params = rt5682_sdw_hw_params,
+ .hw_free = rt5682_sdw_hw_free,
+ .set_sdw_stream = rt5682_set_sdw_stream,
+ .shutdown = rt5682_sdw_shutdown,
+};
+#endif
+
static struct snd_soc_dai_driver rt5682_dai[] = {
{
.name = "rt5682-aif1",
@@ -2409,6 +3168,27 @@ static struct snd_soc_dai_driver rt5682_dai[] = {
},
.ops = &rt5682_aif2_dai_ops,
},
+#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW)
+ {
+ .name = "rt5682-sdw",
+ .id = RT5682_SDW,
+ .playback = {
+ .stream_name = "SDW Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5682_STEREO_RATES,
+ .formats = RT5682_FORMATS,
+ },
+ .capture = {
+ .stream_name = "SDW Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT5682_STEREO_RATES,
+ .formats = RT5682_FORMATS,
+ },
+ .ops = &rt5682_sdw_ops,
+ },
+#endif
};
static const struct snd_soc_component_driver soc_component_dev_rt5682 = {
@@ -2461,10 +3241,21 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev)
&rt5682->pdata.jd_src);
device_property_read_u32(dev, "realtek,btndet-delay",
&rt5682->pdata.btndet_delay);
+ device_property_read_u32(dev, "realtek,dmic-clk-rate-hz",
+ &rt5682->pdata.dmic_clk_rate);
+ device_property_read_u32(dev, "realtek,dmic-delay-ms",
+ &rt5682->pdata.dmic_delay);
rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node,
"realtek,ldo1-en-gpios", 0);
+ if (device_property_read_string_array(dev, "clock-output-names",
+ rt5682->pdata.dai_clk_names,
+ RT5682_DAI_NUM_CLKS) < 0)
+ dev_warn(dev, "Using default DAI clk names: %s, %s\n",
+ rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX],
+ rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]);
+
return 0;
}
@@ -2474,7 +3265,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
mutex_lock(&rt5682->calibrate_mutex);
- rt5682_reset(rt5682->regmap);
+ rt5682_reset(rt5682);
regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f);
regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
usleep_range(15000, 20000);
@@ -2520,6 +3311,221 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
}
+#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW)
+static int rt5682_sdw_read(void *context, unsigned int reg, unsigned int *val)
+{
+ struct device *dev = context;
+ struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+ unsigned int data_l, data_h;
+
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 0);
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff);
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff));
+ regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_H, &data_h);
+ regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_L, &data_l);
+
+ *val = (data_h << 8) | data_l;
+
+ dev_vdbg(dev, "[%s] %04x => %04x\n", __func__, reg, *val);
+
+ return 0;
+}
+
+static int rt5682_sdw_write(void *context, unsigned int reg, unsigned int val)
+{
+ struct device *dev = context;
+ struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 1);
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff);
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff));
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_H, (val >> 8) & 0xff);
+ regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_L, (val & 0xff));
+
+ dev_vdbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val);
+
+ return 0;
+}
+
+static const struct regmap_config rt5682_sdw_regmap = {
+ .reg_bits = 16,
+ .val_bits = 16,
+ .max_register = RT5682_I2C_MODE,
+ .volatile_reg = rt5682_volatile_register,
+ .readable_reg = rt5682_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = rt5682_reg,
+ .num_reg_defaults = ARRAY_SIZE(rt5682_reg),
+ .use_single_read = true,
+ .use_single_write = true,
+ .reg_read = rt5682_sdw_read,
+ .reg_write = rt5682_sdw_write,
+};
+
+int rt5682_sdw_init(struct device *dev, struct regmap *regmap,
+ struct sdw_slave *slave)
+{
+ struct rt5682_priv *rt5682;
+ int ret;
+
+ rt5682 = devm_kzalloc(dev, sizeof(*rt5682), GFP_KERNEL);
+ if (!rt5682)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, rt5682);
+ rt5682->slave = slave;
+ rt5682->sdw_regmap = regmap;
+ rt5682->is_sdw = true;
+
+ rt5682->regmap = devm_regmap_init(dev, NULL, dev, &rt5682_sdw_regmap);
+ if (IS_ERR(rt5682->regmap)) {
+ ret = PTR_ERR(rt5682->regmap);
+ dev_err(dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+
+ /*
+ * Mark hw_init to false
+ * HW init will be performed when device reports present
+ */
+ rt5682->hw_init = false;
+ rt5682->first_hw_init = false;
+
+ mutex_init(&rt5682->calibrate_mutex);
+ INIT_DELAYED_WORK(&rt5682->jack_detect_work,
+ rt5682_jack_detect_handler);
+
+ ret = devm_snd_soc_register_component(dev, &soc_component_dev_rt5682,
+ rt5682_dai, ARRAY_SIZE(rt5682_dai));
+
+ dev_dbg(&slave->dev, "%s\n", __func__);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(rt5682_sdw_init);
+
+int rt5682_io_init(struct device *dev, struct sdw_slave *slave)
+{
+ struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+ int ret = 0;
+ unsigned int val;
+
+ if (rt5682->hw_init)
+ return 0;
+
+ regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val);
+ if (val != DEVICE_ID) {
+ pr_err("Device with ID register %x is not rt5682\n", val);
+ return -ENODEV;
+ }
+
+ /*
+ * PM runtime is only enabled when a Slave reports as Attached
+ */
+ if (!rt5682->first_hw_init) {
+ /* set autosuspend parameters */
+ pm_runtime_set_autosuspend_delay(&slave->dev, 3000);
+ pm_runtime_use_autosuspend(&slave->dev);
+
+ /* update count of parent 'active' children */
+ pm_runtime_set_active(&slave->dev);
+
+ /* make sure the device does not suspend immediately */
+ pm_runtime_mark_last_busy(&slave->dev);
+
+ pm_runtime_enable(&slave->dev);
+ }
+
+ pm_runtime_get_noresume(&slave->dev);
+
+ rt5682_reset(rt5682);
+
+ if (rt5682->first_hw_init) {
+ regcache_cache_only(rt5682->regmap, false);
+ regcache_cache_bypass(rt5682->regmap, true);
+ }
+
+ rt5682_calibrate(rt5682);
+
+ if (rt5682->first_hw_init) {
+ regcache_cache_bypass(rt5682->regmap, false);
+ regcache_mark_dirty(rt5682->regmap);
+ regcache_sync(rt5682->regmap);
+
+ /* volatile registers */
+ regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2,
+ RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
+
+ goto reinit;
+ }
+
+ ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
+ ARRAY_SIZE(patch_list));
+ if (ret != 0)
+ dev_warn(dev, "Failed to apply regmap patch: %d\n", ret);
+
+ regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000);
+
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+ RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK,
+ RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X);
+ regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
+ regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+ regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8,
+ RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA);
+ regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1,
+ RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
+ regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1,
+ RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+
+ /* Soundwire */
+ regmap_write(rt5682->regmap, RT5682_PLL2_INTERNAL, 0xa266);
+ regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_1, 0x1700);
+ regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_2, 0x0006);
+ regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_3, 0x2600);
+ regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_4, 0x0c8f);
+ regmap_write(rt5682->regmap, RT5682_PLL_TRACK_2, 0x3000);
+ regmap_write(rt5682->regmap, RT5682_PLL_TRACK_3, 0x4000);
+ regmap_update_bits(rt5682->regmap, RT5682_GLB_CLK,
+ RT5682_SCLK_SRC_MASK | RT5682_PLL2_SRC_MASK,
+ RT5682_SCLK_SRC_PLL2 | RT5682_PLL2_SRC_SDW);
+
+ regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2,
+ RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
+ regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042);
+ regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3,
+ RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
+ regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1,
+ RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ RT5682_POW_IRQ | RT5682_POW_JDH |
+ RT5682_POW_ANA, RT5682_POW_IRQ |
+ RT5682_POW_JDH | RT5682_POW_ANA);
+ regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
+ RT5682_PWR_JDH, RT5682_PWR_JDH);
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK | RT5682_JD1_IRQ_MASK,
+ RT5682_JD1_EN | RT5682_JD1_IRQ_PUL);
+
+reinit:
+ mod_delayed_work(system_power_efficient_wq,
+ &rt5682->jack_detect_work, msecs_to_jiffies(250));
+
+ /* Mark Slave initialization complete */
+ rt5682->hw_init = true;
+ rt5682->first_hw_init = true;
+
+ pm_runtime_mark_last_busy(&slave->dev);
+ pm_runtime_put_autosuspend(&slave->dev);
+
+ dev_dbg(&slave->dev, "%s hw_init complete\n", __func__);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(rt5682_io_init);
+#endif
+
static int rt5682_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -2586,7 +3592,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
return -ENODEV;
}
- rt5682_reset(rt5682->regmap);
+ rt5682_reset(rt5682);
mutex_init(&rt5682->calibrate_mutex);
rt5682_calibrate(rt5682);
@@ -2651,6 +3657,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1,
RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+ regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+ RT5682_FIFO_CLK_DIV_MASK, RT5682_FIFO_CLK_DIV_2);
INIT_DELAYED_WORK(&rt5682->jack_detect_work,
rt5682_jack_detect_handler);
@@ -2676,7 +3684,7 @@ static void rt5682_i2c_shutdown(struct i2c_client *client)
{
struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
- rt5682_reset(rt5682->regmap);
+ rt5682_reset(rt5682);
}
#ifdef CONFIG_OF
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index 18faaa2a49a0..0baeece84ec4 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -10,6 +10,12 @@
#define __RT5682_H__
#include <sound/rt5682.h>
+#include <linux/regulator/consumer.h>
+#include <linux/clk.h>
+#include <linux/clkdev.h>
+#include <linux/clk-provider.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
#define DEVICE_ID 0x6530
@@ -177,7 +183,7 @@
#define RT5682_TEST_MODE_CTRL_4 0x0148
#define RT5682_TEST_MODE_CTRL_5 0x0149
#define RT5682_PLL1_INTERNAL 0x0150
-#define RT5682_PLL2_INTERNAL 0x0151
+#define RT5682_PLL2_INTERNAL 0x0156
#define RT5682_STO_NG2_CTRL_1 0x0160
#define RT5682_STO_NG2_CTRL_2 0x0161
#define RT5682_STO_NG2_CTRL_3 0x0162
@@ -651,6 +657,8 @@
#define RT5682_DMIC_1_EN_SFT 15
#define RT5682_DMIC_1_DIS (0x0 << 15)
#define RT5682_DMIC_1_EN (0x1 << 15)
+#define RT5682_FIFO_CLK_DIV_MASK (0x7 << 12)
+#define RT5682_FIFO_CLK_DIV_2 (0x1 << 12)
#define RT5682_DMIC_1_DP_MASK (0x3 << 4)
#define RT5682_DMIC_1_DP_SFT 4
#define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4)
@@ -738,7 +746,7 @@
#define RT5682_ADC_OSR_D_24 (0x7 << 12)
#define RT5682_ADC_OSR_D_32 (0x8 << 12)
#define RT5682_ADC_OSR_D_48 (0x9 << 12)
-#define RT5682_I2S_M_DIV_MASK (0xf << 12)
+#define RT5682_I2S_M_DIV_MASK (0xf << 8)
#define RT5682_I2S_M_DIV_SFT 8
#define RT5682_I2S_M_D_1 (0x0 << 8)
#define RT5682_I2S_M_D_2 (0x1 << 8)
@@ -820,6 +828,12 @@
#define RT5682_TDM_DF_PCM_B (0x3 << 11)
#define RT5682_TDM_DF_PCM_A_N (0x6 << 11)
#define RT5682_TDM_DF_PCM_B_N (0x7 << 11)
+#define RT5682_TDM_BCLK_MS1_MASK (0x3 << 9)
+#define RT5682_TDM_BCLK_MS1_SFT 9
+#define RT5682_TDM_BCLK_MS1_32 (0x0 << 9)
+#define RT5682_TDM_BCLK_MS1_64 (0x1 << 9)
+#define RT5682_TDM_BCLK_MS1_128 (0x2 << 9)
+#define RT5682_TDM_BCLK_MS1_256 (0x3 << 9)
#define RT5682_TDM_CL_MASK (0x3 << 4)
#define RT5682_TDM_CL_16 (0x0 << 4)
#define RT5682_TDM_CL_20 (0x1 << 4)
@@ -835,8 +849,8 @@
#define RT5682_TDM_M_LP_INV (0x1 << 1)
#define RT5682_TDM_MS_MASK (0x1 << 0)
#define RT5682_TDM_MS_SFT 0
-#define RT5682_TDM_MS_M (0x0 << 0)
-#define RT5682_TDM_MS_S (0x1 << 0)
+#define RT5682_TDM_MS_S (0x0 << 0)
+#define RT5682_TDM_MS_M (0x1 << 0)
/* Global Clock Control (0x0080) */
#define RT5682_SCLK_SRC_MASK (0x7 << 13)
@@ -1049,6 +1063,28 @@
#define RT5682_PWR_CLK1M_PD (0x0 << 8)
#define RT5682_PWR_CLK1M_PU (0x1 << 8)
+/* PLL2 M/N/K Code Control 1 (0x009b) */
+#define RT5682_PLL2F_K_MASK (0x1f << 8)
+#define RT5682_PLL2F_K_SFT 8
+#define RT5682_PLL2B_K_MASK (0xf << 4)
+#define RT5682_PLL2B_K_SFT 4
+#define RT5682_PLL2B_M_MASK (0xf << 0)
+
+/* PLL2 M/N/K Code Control 2 (0x009c) */
+#define RT5682_PLL2F_M_MASK (0x3f << 8)
+#define RT5682_PLL2F_M_SFT 8
+#define RT5682_PLL2B_N_MASK (0x3f << 0)
+
+/* PLL2 M/N/K Code Control 2 (0x009d) */
+#define RT5682_PLL2F_N_MASK (0x7f << 8)
+#define RT5682_PLL2F_N_SFT 8
+
+/* PLL2 M/N/K Code Control 2 (0x009e) */
+#define RT5682_PLL2B_M_BP_MASK (0x1 << 11)
+#define RT5682_PLL2B_M_BP_SFT 11
+#define RT5682_PLL2F_M_BP_MASK (0x1 << 7)
+#define RT5682_PLL2F_M_BP_SFT 7
+
/* RC Clock Control (0x009f) */
#define RT5682_POW_IRQ (0x1 << 15)
#define RT5682_POW_JDH (0x1 << 14)
@@ -1091,11 +1127,17 @@
#define RT5682_JD1_POL_MASK (0x1 << 13)
#define RT5682_JD1_POL_NOR (0x0 << 13)
#define RT5682_JD1_POL_INV (0x1 << 13)
+#define RT5682_JD1_IRQ_MASK (0x1 << 10)
+#define RT5682_JD1_IRQ_LEV (0x0 << 10)
+#define RT5682_JD1_IRQ_PUL (0x1 << 10)
/* IRQ Control 3 (0x00b8) */
#define RT5682_IL_IRQ_MASK (0x1 << 7)
#define RT5682_IL_IRQ_DIS (0x0 << 7)
#define RT5682_IL_IRQ_EN (0x1 << 7)
+#define RT5682_IL_IRQ_TYPE_MASK (0x1 << 4)
+#define RT5682_IL_IRQ_LEV (0x0 << 4)
+#define RT5682_IL_IRQ_PUL (0x1 << 4)
/* GPIO Control 1 (0x00c0) */
#define RT5682_GP1_PIN_MASK (0x3 << 14)
@@ -1309,11 +1351,19 @@ enum {
RT5682_PLL1_S_MCLK,
RT5682_PLL1_S_BCLK1,
RT5682_PLL1_S_RCCLK,
+ RT5682_PLL2_S_MCLK,
+};
+
+enum {
+ RT5682_PLL1,
+ RT5682_PLL2,
+ RT5682_PLLS,
};
enum {
RT5682_AIF1,
RT5682_AIF2,
+ RT5682_SDW,
RT5682_AIFS
};
@@ -1329,7 +1379,49 @@ enum {
RT5682_CLK_SEL_I2S2_ASRC,
};
+#define RT5682_NUM_SUPPLIES 3
+
+struct rt5682_priv {
+ struct snd_soc_component *component;
+ struct rt5682_platform_data pdata;
+ struct regmap *regmap;
+ struct regmap *sdw_regmap;
+ struct snd_soc_jack *hs_jack;
+ struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES];
+ struct delayed_work jack_detect_work;
+ struct delayed_work jd_check_work;
+ struct mutex calibrate_mutex;
+ struct sdw_slave *slave;
+ enum sdw_slave_status status;
+ struct sdw_bus_params params;
+ bool hw_init;
+ bool first_hw_init;
+ bool is_sdw;
+
+#ifdef CONFIG_COMMON_CLK
+ struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS];
+ struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS];
+ struct clk *dai_clks[RT5682_DAI_NUM_CLKS];
+ struct clk *mclk;
+#endif
+
+ int sysclk;
+ int sysclk_src;
+ int lrck[RT5682_AIFS];
+ int bclk[RT5682_AIFS];
+ int master[RT5682_AIFS];
+
+ int pll_src[RT5682_PLLS];
+ int pll_in[RT5682_PLLS];
+ int pll_out[RT5682_PLLS];
+
+ int jack_type;
+};
+
int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
unsigned int filter_mask, unsigned int clk_src);
+int rt5682_sdw_init(struct device *dev, struct regmap *regmap,
+ struct sdw_slave *slave);
+int rt5682_io_init(struct device *dev, struct sdw_slave *slave);
#endif /* __RT5682_H__ */
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index be52886a5edb..7fae88655a0f 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -26,6 +26,24 @@
#define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\
SNDRV_PCM_FORMAT_S32_LE)
+/* DVC equation involves floating point math
+ * round(10^(volume in dB/20)*2^30)
+ * so create a lookup table for 2dB step
+ */
+static const unsigned int float_vol_db_lookup[] = {
+0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151,
+0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b,
+0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a,
+0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f,
+0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7,
+0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d,
+0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a,
+0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27,
+0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68,
+0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362,
+0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000
+};
+
struct tas2562_data {
struct snd_soc_component *component;
struct gpio_desc *sdz_gpio;
@@ -34,6 +52,12 @@ struct tas2562_data {
struct i2c_client *client;
int v_sense_slot;
int i_sense_slot;
+ int volume_lvl;
+};
+
+enum tas256x_model {
+ TAS2562,
+ TAS2563,
};
static int tas2562_set_bias_level(struct snd_soc_component *component,
@@ -383,21 +407,81 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w,
struct snd_soc_component *component =
snd_soc_dapm_to_component(w->dapm);
struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component);
+ int ret;
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n");
+ ret = snd_soc_component_update_bits(component,
+ TAS2562_PWR_CTRL,
+ TAS2562_MODE_MASK,
+ TAS2562_MUTE);
+ if (ret)
+ goto end;
break;
case SND_SOC_DAPM_PRE_PMD:
- dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n");
+ ret = snd_soc_component_update_bits(component,
+ TAS2562_PWR_CTRL,
+ TAS2562_MODE_MASK,
+ TAS2562_SHUTDOWN);
+ if (ret)
+ goto end;
break;
default:
- break;
+ dev_err(tas2562->dev, "Not supported evevt\n");
+ return -EINVAL;
}
+end:
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+ struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component);
+
+ ucontrol->value.integer.value[0] = tas2562->volume_lvl;
return 0;
}
+static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+ struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component);
+ int ret;
+ u32 reg_val;
+
+ reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2];
+ ret = snd_soc_component_write(component, TAS2562_DVC_CFG4,
+ (reg_val & 0xff));
+ if (ret)
+ return ret;
+ ret = snd_soc_component_write(component, TAS2562_DVC_CFG3,
+ ((reg_val >> 8) & 0xff));
+ if (ret)
+ return ret;
+ ret = snd_soc_component_write(component, TAS2562_DVC_CFG2,
+ ((reg_val >> 16) & 0xff));
+ if (ret)
+ return ret;
+ ret = snd_soc_component_write(component, TAS2562_DVC_CFG1,
+ ((reg_val >> 24) & 0xff));
+ if (ret)
+ return ret;
+
+ tas2562->volume_lvl = ucontrol->value.integer.value[0];
+
+ return ret;
+}
+
+/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0);
+
static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0);
static const struct snd_kcontrol_new isense_switch =
@@ -409,14 +493,24 @@ static const struct snd_kcontrol_new vsense_switch =
1, 1);
static const struct snd_kcontrol_new tas2562_snd_controls[] = {
- SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0,
+ SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 1, 0x1c, 0,
tas2562_dac_tlv),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Volume Control",
+ .index = 0,
+ .tlv.p = dvc_tlv,
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_soc_info_volsw,
+ .get = tas2562_volume_control_get,
+ .put = tas2562_volume_control_put,
+ .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) ,
+ },
};
static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux),
- SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch),
@@ -431,7 +525,7 @@ static const struct snd_soc_dapm_route tas2562_audio_map[] = {
{"ASI1 Sel", "Left", "ASI1"},
{"ASI1 Sel", "Right", "ASI1"},
{"ASI1 Sel", "LeftRightDiv2", "ASI1"},
- { "DAC", NULL, "DAC IN" },
+ { "DAC", NULL, "ASI1 Sel" },
{ "OUT", NULL, "DAC" },
{"ISENSE", "Switch", "IMON"},
{"VSENSE", "Switch", "VMON"},
@@ -472,6 +566,13 @@ static struct snd_soc_dai_driver tas2562_dai[] = {
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = TAS2562_FORMATS,
},
+ .capture = {
+ .stream_name = "ASI1 Capture",
+ .channels_min = 0,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = TAS2562_FORMATS,
+ },
.ops = &tas2562_speaker_dai_ops,
},
};
@@ -495,6 +596,10 @@ static const struct reg_default tas2562_reg_defaults[] = {
{ TAS2562_PB_CFG1, 0x20 },
{ TAS2562_TDM_CFG0, 0x09 },
{ TAS2562_TDM_CFG1, 0x02 },
+ { TAS2562_DVC_CFG1, 0x40 },
+ { TAS2562_DVC_CFG2, 0x40 },
+ { TAS2562_DVC_CFG3, 0x00 },
+ { TAS2562_DVC_CFG4, 0x00 },
};
static const struct regmap_config tas2562_regmap_config = {
@@ -564,13 +669,15 @@ static int tas2562_probe(struct i2c_client *client,
}
static const struct i2c_device_id tas2562_id[] = {
- { "tas2562", 0 },
+ { "tas2562", TAS2562 },
+ { "tas2563", TAS2563 },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas2562_id);
static const struct of_device_id tas2562_of_match[] = {
{ .compatible = "ti,tas2562", },
+ { .compatible = "ti,tas2563", },
{ },
};
MODULE_DEVICE_TABLE(of, tas2562_of_match);
diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h
index 62e659ab786d..28e75fc431d0 100644
--- a/sound/soc/codecs/tas2562.h
+++ b/sound/soc/codecs/tas2562.h
@@ -35,12 +35,14 @@
#define TAS2562_REV_ID TAS2562_REG(0, 0x7d)
/* Page 2 */
-#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01)
-#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02)
+#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c)
+#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d)
+#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e)
+#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f)
#define TAS2562_RESET BIT(0)
-#define TAS2562_MODE_MASK 0x3
+#define TAS2562_MODE_MASK GENMASK(1,0)
#define TAS2562_ACTIVE 0x0
#define TAS2562_MUTE 0x1
#define TAS2562_SHUTDOWN 0x2
@@ -73,8 +75,8 @@
#define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3)
#define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3))
-#define TAS2562_VSENSE_POWER_EN BIT(2)
-#define TAS2562_ISENSE_POWER_EN BIT(3)
+#define TAS2562_VSENSE_POWER_EN 2
+#define TAS2562_ISENSE_POWER_EN 3
#define TAS2562_TDM_CFG5_VSNS_EN BIT(6)
#define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0)
diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c
new file mode 100644
index 000000000000..38897568ee96
--- /dev/null
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -0,0 +1,920 @@
+// SPDX-License-Identifier: GPL-2.0
+// TLV320ADCX140 Sound driver
+// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/gpio/consumer.h>
+#include <linux/regulator/consumer.h>
+#include <linux/acpi.h>
+#include <linux/of.h>
+#include <linux/of_gpio.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "tlv320adcx140.h"
+
+struct adcx140_priv {
+ struct snd_soc_component *component;
+ struct regulator *supply_areg;
+ struct gpio_desc *gpio_reset;
+ struct regmap *regmap;
+ struct device *dev;
+
+ int micbias_vg;
+
+ unsigned int dai_fmt;
+ unsigned int tdm_delay;
+ unsigned int slot_width;
+};
+
+static const struct reg_default adcx140_reg_defaults[] = {
+ { ADCX140_PAGE_SELECT, 0x00 },
+ { ADCX140_SW_RESET, 0x00 },
+ { ADCX140_SLEEP_CFG, 0x00 },
+ { ADCX140_SHDN_CFG, 0x05 },
+ { ADCX140_ASI_CFG0, 0x30 },
+ { ADCX140_ASI_CFG1, 0x00 },
+ { ADCX140_ASI_CFG2, 0x00 },
+ { ADCX140_ASI_CH1, 0x00 },
+ { ADCX140_ASI_CH2, 0x01 },
+ { ADCX140_ASI_CH3, 0x02 },
+ { ADCX140_ASI_CH4, 0x03 },
+ { ADCX140_ASI_CH5, 0x04 },
+ { ADCX140_ASI_CH6, 0x05 },
+ { ADCX140_ASI_CH7, 0x06 },
+ { ADCX140_ASI_CH8, 0x07 },
+ { ADCX140_MST_CFG0, 0x02 },
+ { ADCX140_MST_CFG1, 0x48 },
+ { ADCX140_ASI_STS, 0xff },
+ { ADCX140_CLK_SRC, 0x10 },
+ { ADCX140_PDMCLK_CFG, 0x40 },
+ { ADCX140_PDM_CFG, 0x00 },
+ { ADCX140_GPIO_CFG0, 0x22 },
+ { ADCX140_GPO_CFG1, 0x00 },
+ { ADCX140_GPO_CFG2, 0x00 },
+ { ADCX140_GPO_CFG3, 0x00 },
+ { ADCX140_GPO_CFG4, 0x00 },
+ { ADCX140_GPO_VAL, 0x00 },
+ { ADCX140_GPIO_MON, 0x00 },
+ { ADCX140_GPI_CFG0, 0x00 },
+ { ADCX140_GPI_CFG1, 0x00 },
+ { ADCX140_GPI_MON, 0x00 },
+ { ADCX140_INT_CFG, 0x00 },
+ { ADCX140_INT_MASK0, 0xff },
+ { ADCX140_INT_LTCH0, 0x00 },
+ { ADCX140_BIAS_CFG, 0x00 },
+ { ADCX140_CH1_CFG0, 0x00 },
+ { ADCX140_CH1_CFG1, 0x00 },
+ { ADCX140_CH1_CFG2, 0xc9 },
+ { ADCX140_CH1_CFG3, 0x80 },
+ { ADCX140_CH1_CFG4, 0x00 },
+ { ADCX140_CH2_CFG0, 0x00 },
+ { ADCX140_CH2_CFG1, 0x00 },
+ { ADCX140_CH2_CFG2, 0xc9 },
+ { ADCX140_CH2_CFG3, 0x80 },
+ { ADCX140_CH2_CFG4, 0x00 },
+ { ADCX140_CH3_CFG0, 0x00 },
+ { ADCX140_CH3_CFG1, 0x00 },
+ { ADCX140_CH3_CFG2, 0xc9 },
+ { ADCX140_CH3_CFG3, 0x80 },
+ { ADCX140_CH3_CFG4, 0x00 },
+ { ADCX140_CH4_CFG0, 0x00 },
+ { ADCX140_CH4_CFG1, 0x00 },
+ { ADCX140_CH4_CFG2, 0xc9 },
+ { ADCX140_CH4_CFG3, 0x80 },
+ { ADCX140_CH4_CFG4, 0x00 },
+ { ADCX140_CH5_CFG2, 0xc9 },
+ { ADCX140_CH5_CFG3, 0x80 },
+ { ADCX140_CH5_CFG4, 0x00 },
+ { ADCX140_CH6_CFG2, 0xc9 },
+ { ADCX140_CH6_CFG3, 0x80 },
+ { ADCX140_CH6_CFG4, 0x00 },
+ { ADCX140_CH7_CFG2, 0xc9 },
+ { ADCX140_CH7_CFG3, 0x80 },
+ { ADCX140_CH7_CFG4, 0x00 },
+ { ADCX140_CH8_CFG2, 0xc9 },
+ { ADCX140_CH8_CFG3, 0x80 },
+ { ADCX140_CH8_CFG4, 0x00 },
+ { ADCX140_DSP_CFG0, 0x01 },
+ { ADCX140_DSP_CFG1, 0x40 },
+ { ADCX140_DRE_CFG0, 0x7b },
+ { ADCX140_AGC_CFG0, 0xe7 },
+ { ADCX140_IN_CH_EN, 0xf0 },
+ { ADCX140_ASI_OUT_CH_EN, 0x00 },
+ { ADCX140_PWR_CFG, 0x00 },
+ { ADCX140_DEV_STS0, 0x00 },
+ { ADCX140_DEV_STS1, 0x80 },
+};
+
+static const struct regmap_range_cfg adcx140_ranges[] = {
+ {
+ .range_min = 0,
+ .range_max = 12 * 128,
+ .selector_reg = ADCX140_PAGE_SELECT,
+ .selector_mask = 0xff,
+ .selector_shift = 0,
+ .window_start = 0,
+ .window_len = 128,
+ },
+};
+
+static bool adcx140_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case ADCX140_SW_RESET:
+ case ADCX140_DEV_STS0:
+ case ADCX140_DEV_STS1:
+ case ADCX140_ASI_STS:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config adcx140_i2c_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .reg_defaults = adcx140_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(adcx140_reg_defaults),
+ .cache_type = REGCACHE_FLAT,
+ .ranges = adcx140_ranges,
+ .num_ranges = ARRAY_SIZE(adcx140_ranges),
+ .max_register = 12 * 128,
+ .volatile_reg = adcx140_volatile,
+};
+
+/* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */
+static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0);
+
+/* ADC gain. From 0 to 42 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0);
+
+/* DRE Level. From -12 dB to -66 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(dre_thresh_tlv, -6600, 100, 0);
+/* DRE Max Gain. From 2 dB to 26 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(dre_gain_tlv, 200, 200, 0);
+
+/* AGC Level. From -6 dB to -36 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0);
+/* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */
+static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0);
+
+static const char * const decimation_filter_text[] = {
+ "Linear Phase", "Low Latency", "Ultra-low Latency"
+};
+
+static SOC_ENUM_SINGLE_DECL(decimation_filter_enum, ADCX140_DSP_CFG0, 4,
+ decimation_filter_text);
+
+static const struct snd_kcontrol_new decimation_filter_controls[] = {
+ SOC_DAPM_ENUM("Decimation Filter", decimation_filter_enum),
+};
+
+static const char * const resistor_text[] = {
+ "2.5 kOhm", "10 kOhm", "20 kOhm"
+};
+
+static SOC_ENUM_SINGLE_DECL(in1_resistor_enum, ADCX140_CH1_CFG0, 2,
+ resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2_resistor_enum, ADCX140_CH2_CFG0, 2,
+ resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3_resistor_enum, ADCX140_CH3_CFG0, 2,
+ resistor_text);
+static SOC_ENUM_SINGLE_DECL(in4_resistor_enum, ADCX140_CH4_CFG0, 2,
+ resistor_text);
+
+static const struct snd_kcontrol_new in1_resistor_controls[] = {
+ SOC_DAPM_ENUM("CH1 Resistor Select", in1_resistor_enum),
+};
+static const struct snd_kcontrol_new in2_resistor_controls[] = {
+ SOC_DAPM_ENUM("CH2 Resistor Select", in2_resistor_enum),
+};
+static const struct snd_kcontrol_new in3_resistor_controls[] = {
+ SOC_DAPM_ENUM("CH3 Resistor Select", in3_resistor_enum),
+};
+static const struct snd_kcontrol_new in4_resistor_controls[] = {
+ SOC_DAPM_ENUM("CH4 Resistor Select", in4_resistor_enum),
+};
+
+/* Analog/Digital Selection */
+static const char *adcx140_mic_sel_text[] = {"Analog", "Line In", "Digital"};
+static const char *adcx140_analog_sel_text[] = {"Analog", "Line In"};
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic1p_enum,
+ ADCX140_CH1_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic1p_control =
+SOC_DAPM_ENUM("MIC1P MUX", adcx140_mic1p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic1_analog_enum,
+ ADCX140_CH1_CFG0, 7,
+ adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic1_analog_control =
+SOC_DAPM_ENUM("MIC1 Analog MUX", adcx140_mic1_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic1m_enum,
+ ADCX140_CH1_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic1m_control =
+SOC_DAPM_ENUM("MIC1M MUX", adcx140_mic1m_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic2p_enum,
+ ADCX140_CH2_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic2p_control =
+SOC_DAPM_ENUM("MIC2P MUX", adcx140_mic2p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic2_analog_enum,
+ ADCX140_CH2_CFG0, 7,
+ adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic2_analog_control =
+SOC_DAPM_ENUM("MIC2 Analog MUX", adcx140_mic2_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic2m_enum,
+ ADCX140_CH2_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic2m_control =
+SOC_DAPM_ENUM("MIC2M MUX", adcx140_mic2m_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic3p_enum,
+ ADCX140_CH3_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic3p_control =
+SOC_DAPM_ENUM("MIC3P MUX", adcx140_mic3p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic3_analog_enum,
+ ADCX140_CH3_CFG0, 7,
+ adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic3_analog_control =
+SOC_DAPM_ENUM("MIC3 Analog MUX", adcx140_mic3_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic3m_enum,
+ ADCX140_CH3_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic3m_control =
+SOC_DAPM_ENUM("MIC3M MUX", adcx140_mic3m_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic4p_enum,
+ ADCX140_CH4_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic4p_control =
+SOC_DAPM_ENUM("MIC4P MUX", adcx140_mic4p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic4_analog_enum,
+ ADCX140_CH4_CFG0, 7,
+ adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic4_analog_control =
+SOC_DAPM_ENUM("MIC4 Analog MUX", adcx140_mic4_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic4m_enum,
+ ADCX140_CH4_CFG0, 5,
+ adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic4m_control =
+SOC_DAPM_ENUM("MIC4M MUX", adcx140_mic4m_enum);
+
+static const struct snd_kcontrol_new adcx140_dapm_ch1_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 7, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch2_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 6, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 5, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0);
+
+static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch2_dre_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_CH2_CFG0, 0, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch3_dre_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_CH3_CFG0, 0, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0);
+
+static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch =
+ SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0);
+
+/* Output Mixer */
+static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0),
+ SOC_DAPM_SINGLE("Digital CH2 Switch", 0, 0, 0, 0),
+ SOC_DAPM_SINGLE("Digital CH3 Switch", 0, 0, 0, 0),
+ SOC_DAPM_SINGLE("Digital CH4 Switch", 0, 0, 0, 0),
+};
+
+static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = {
+ /* Analog Differential Inputs */
+ SND_SOC_DAPM_INPUT("MIC1P"),
+ SND_SOC_DAPM_INPUT("MIC1M"),
+ SND_SOC_DAPM_INPUT("MIC2P"),
+ SND_SOC_DAPM_INPUT("MIC2M"),
+ SND_SOC_DAPM_INPUT("MIC3P"),
+ SND_SOC_DAPM_INPUT("MIC3M"),
+ SND_SOC_DAPM_INPUT("MIC4P"),
+ SND_SOC_DAPM_INPUT("MIC4M"),
+
+ SND_SOC_DAPM_OUTPUT("CH1_OUT"),
+ SND_SOC_DAPM_OUTPUT("CH2_OUT"),
+ SND_SOC_DAPM_OUTPUT("CH3_OUT"),
+ SND_SOC_DAPM_OUTPUT("CH4_OUT"),
+ SND_SOC_DAPM_OUTPUT("CH5_OUT"),
+ SND_SOC_DAPM_OUTPUT("CH6_OUT"),
+ SND_SOC_DAPM_OUTPUT("CH7_OUT"),
+ SND_SOC_DAPM_OUTPUT("CH8_OUT"),
+
+ SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
+ &adcx140_output_mixer_controls[0],
+ ARRAY_SIZE(adcx140_output_mixer_controls)),
+
+ /* Input Selection to MIC_PGA */
+ SND_SOC_DAPM_MUX("MIC1P Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic1p_control),
+ SND_SOC_DAPM_MUX("MIC2P Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic2p_control),
+ SND_SOC_DAPM_MUX("MIC3P Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic3p_control),
+ SND_SOC_DAPM_MUX("MIC4P Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic4p_control),
+
+ /* Input Selection to MIC_PGA */
+ SND_SOC_DAPM_MUX("MIC1 Analog Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic1_analog_control),
+ SND_SOC_DAPM_MUX("MIC2 Analog Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic2_analog_control),
+ SND_SOC_DAPM_MUX("MIC3 Analog Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic3_analog_control),
+ SND_SOC_DAPM_MUX("MIC4 Analog Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic4_analog_control),
+
+ SND_SOC_DAPM_MUX("MIC1M Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic1m_control),
+ SND_SOC_DAPM_MUX("MIC2M Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic2m_control),
+ SND_SOC_DAPM_MUX("MIC3M Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic3m_control),
+ SND_SOC_DAPM_MUX("MIC4M Input Mux", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_mic4m_control),
+
+ SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH4", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_ADC("CH1_ADC", "CH1 Capture", ADCX140_IN_CH_EN, 7, 0),
+ SND_SOC_DAPM_ADC("CH2_ADC", "CH2 Capture", ADCX140_IN_CH_EN, 6, 0),
+ SND_SOC_DAPM_ADC("CH3_ADC", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0),
+ SND_SOC_DAPM_ADC("CH4_ADC", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0),
+
+ SND_SOC_DAPM_SWITCH("CH1_ASI_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch1_en_switch),
+ SND_SOC_DAPM_SWITCH("CH2_ASI_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch2_en_switch),
+ SND_SOC_DAPM_SWITCH("CH3_ASI_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch3_en_switch),
+ SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch4_en_switch),
+
+ SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_dre_en_switch),
+
+ SND_SOC_DAPM_SWITCH("CH1_DRE_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch1_dre_en_switch),
+ SND_SOC_DAPM_SWITCH("CH2_DRE_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch2_dre_en_switch),
+ SND_SOC_DAPM_SWITCH("CH3_DRE_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch3_dre_en_switch),
+ SND_SOC_DAPM_SWITCH("CH4_DRE_EN", SND_SOC_NOPM, 0, 0,
+ &adcx140_dapm_ch4_dre_en_switch),
+
+ SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+ in1_resistor_controls),
+ SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+ in2_resistor_controls),
+ SND_SOC_DAPM_MUX("IN3 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+ in3_resistor_controls),
+ SND_SOC_DAPM_MUX("IN4 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+ in4_resistor_controls),
+
+ SND_SOC_DAPM_MUX("Decimation Filter", SND_SOC_NOPM, 0, 0,
+ decimation_filter_controls),
+};
+
+static const struct snd_soc_dapm_route adcx140_audio_map[] = {
+ /* Outputs */
+ {"CH1_OUT", NULL, "Output Mixer"},
+ {"CH2_OUT", NULL, "Output Mixer"},
+ {"CH3_OUT", NULL, "Output Mixer"},
+ {"CH4_OUT", NULL, "Output Mixer"},
+
+ {"CH1_ASI_EN", "Switch", "CH1_ADC"},
+ {"CH2_ASI_EN", "Switch", "CH2_ADC"},
+ {"CH3_ASI_EN", "Switch", "CH3_ADC"},
+ {"CH4_ASI_EN", "Switch", "CH4_ADC"},
+
+ {"Decimation Filter", "Linear Phase", "DRE_ENABLE"},
+ {"Decimation Filter", "Low Latency", "DRE_ENABLE"},
+ {"Decimation Filter", "Ultra-low Latency", "DRE_ENABLE"},
+
+ {"DRE_ENABLE", "Switch", "CH1_DRE_EN"},
+ {"DRE_ENABLE", "Switch", "CH2_DRE_EN"},
+ {"DRE_ENABLE", "Switch", "CH3_DRE_EN"},
+ {"DRE_ENABLE", "Switch", "CH4_DRE_EN"},
+
+ {"CH1_DRE_EN", "Switch", "CH1_ADC"},
+ {"CH2_DRE_EN", "Switch", "CH2_ADC"},
+ {"CH3_DRE_EN", "Switch", "CH3_ADC"},
+ {"CH4_DRE_EN", "Switch", "CH4_ADC"},
+
+ /* Mic input */
+ {"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"},
+ {"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"},
+ {"CH3_ADC", NULL, "MIC_GAIN_CTL_CH3"},
+ {"CH4_ADC", NULL, "MIC_GAIN_CTL_CH4"},
+
+ {"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"},
+ {"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"},
+ {"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"},
+ {"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"},
+ {"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"},
+ {"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"},
+ {"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"},
+ {"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"},
+
+ {"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1P Input Mux"},
+ {"IN1 Analog Mic Resistor", "10 kOhm", "MIC1P Input Mux"},
+ {"IN1 Analog Mic Resistor", "20 kOhm", "MIC1P Input Mux"},
+
+ {"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1M Input Mux"},
+ {"IN1 Analog Mic Resistor", "10 kOhm", "MIC1M Input Mux"},
+ {"IN1 Analog Mic Resistor", "20 kOhm", "MIC1M Input Mux"},
+
+ {"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2P Input Mux"},
+ {"IN2 Analog Mic Resistor", "10 kOhm", "MIC2P Input Mux"},
+ {"IN2 Analog Mic Resistor", "20 kOhm", "MIC2P Input Mux"},
+
+ {"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2M Input Mux"},
+ {"IN2 Analog Mic Resistor", "10 kOhm", "MIC2M Input Mux"},
+ {"IN2 Analog Mic Resistor", "20 kOhm", "MIC2M Input Mux"},
+
+ {"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3P Input Mux"},
+ {"IN3 Analog Mic Resistor", "10 kOhm", "MIC3P Input Mux"},
+ {"IN3 Analog Mic Resistor", "20 kOhm", "MIC3P Input Mux"},
+
+ {"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3M Input Mux"},
+ {"IN3 Analog Mic Resistor", "10 kOhm", "MIC3M Input Mux"},
+ {"IN3 Analog Mic Resistor", "20 kOhm", "MIC3M Input Mux"},
+
+ {"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4P Input Mux"},
+ {"IN4 Analog Mic Resistor", "10 kOhm", "MIC4P Input Mux"},
+ {"IN4 Analog Mic Resistor", "20 kOhm", "MIC4P Input Mux"},
+
+ {"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4M Input Mux"},
+ {"IN4 Analog Mic Resistor", "10 kOhm", "MIC4M Input Mux"},
+ {"IN4 Analog Mic Resistor", "20 kOhm", "MIC4M Input Mux"},
+
+ {"MIC1 Analog Mux", "Line In", "MIC1P"},
+ {"MIC2 Analog Mux", "Line In", "MIC2P"},
+ {"MIC3 Analog Mux", "Line In", "MIC3P"},
+ {"MIC4 Analog Mux", "Line In", "MIC4P"},
+
+ {"MIC1P Input Mux", "Analog", "MIC1P"},
+ {"MIC1M Input Mux", "Analog", "MIC1M"},
+ {"MIC2P Input Mux", "Analog", "MIC2P"},
+ {"MIC2M Input Mux", "Analog", "MIC2M"},
+ {"MIC3P Input Mux", "Analog", "MIC3P"},
+ {"MIC3M Input Mux", "Analog", "MIC3M"},
+ {"MIC4P Input Mux", "Analog", "MIC4P"},
+ {"MIC4M Input Mux", "Analog", "MIC4M"},
+};
+
+static const struct snd_kcontrol_new adcx140_snd_controls[] = {
+ SOC_SINGLE_TLV("Analog CH1 Mic Gain Volume", ADCX140_CH1_CFG1, 2, 42, 0,
+ adc_tlv),
+ SOC_SINGLE_TLV("Analog CH2 Mic Gain Volume", ADCX140_CH1_CFG2, 2, 42, 0,
+ adc_tlv),
+ SOC_SINGLE_TLV("Analog CH3 Mic Gain Volume", ADCX140_CH1_CFG3, 2, 42, 0,
+ adc_tlv),
+ SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0,
+ adc_tlv),
+
+ SOC_SINGLE_TLV("DRE Threshold", ADCX140_DRE_CFG0, 4, 9, 0,
+ dre_thresh_tlv),
+ SOC_SINGLE_TLV("DRE Max Gain", ADCX140_DRE_CFG0, 0, 12, 0,
+ dre_gain_tlv),
+
+ SOC_SINGLE_TLV("AGC Threshold", ADCX140_AGC_CFG0, 4, 15, 0,
+ agc_thresh_tlv),
+ SOC_SINGLE_TLV("AGC Max Gain", ADCX140_AGC_CFG0, 0, 13, 0,
+ agc_gain_tlv),
+
+ SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+ SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+ SOC_SINGLE_TLV("Digital CH3 Out Volume", ADCX140_CH3_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+ SOC_SINGLE_TLV("Digital CH4 Out Volume", ADCX140_CH4_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+ SOC_SINGLE_TLV("Digital CH5 Out Volume", ADCX140_CH5_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+ SOC_SINGLE_TLV("Digital CH6 Out Volume", ADCX140_CH6_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+ SOC_SINGLE_TLV("Digital CH7 Out Volume", ADCX140_CH7_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+ SOC_SINGLE_TLV("Digital CH8 Out Volume", ADCX140_CH8_CFG2,
+ 0, 0xff, 0, dig_vol_tlv),
+};
+
+static int adcx140_reset(struct adcx140_priv *adcx140)
+{
+ int ret = 0;
+
+ if (adcx140->gpio_reset) {
+ gpiod_direction_output(adcx140->gpio_reset, 0);
+ /* 8.4.1: wait for hw shutdown (25ms) + >= 1ms */
+ usleep_range(30000, 100000);
+ gpiod_direction_output(adcx140->gpio_reset, 1);
+ } else {
+ ret = regmap_write(adcx140->regmap, ADCX140_SW_RESET,
+ ADCX140_RESET);
+ }
+
+ /* 8.4.2: wait >= 10 ms after entering sleep mode. */
+ usleep_range(10000, 100000);
+
+ return 0;
+}
+
+static int adcx140_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ u8 data = 0;
+
+ switch (params_width(params)) {
+ case 16:
+ data = ADCX140_16_BIT_WORD;
+ break;
+ case 20:
+ data = ADCX140_20_BIT_WORD;
+ break;
+ case 24:
+ data = ADCX140_24_BIT_WORD;
+ break;
+ case 32:
+ data = ADCX140_32_BIT_WORD;
+ break;
+ default:
+ dev_err(component->dev, "%s: Unsupported width %d\n",
+ __func__, params_width(params));
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, ADCX140_ASI_CFG0,
+ ADCX140_WORD_LEN_MSK, data);
+
+ return 0;
+}
+
+static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_component *component = codec_dai->component;
+ struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+ u8 iface_reg1 = 0;
+ u8 iface_reg2 = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface_reg2 |= ADCX140_BCLK_FSYNC_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ default:
+ dev_err(component->dev, "Invalid DAI master/slave interface\n");
+ return -EINVAL;
+ }
+
+ /* signal polarity */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_IF:
+ iface_reg1 |= ADCX140_FSYNCINV_BIT;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface_reg1 |= ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface_reg1 |= ADCX140_BCLKINV_BIT;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ dev_err(component->dev, "Invalid DAI clock signal polarity\n");
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface_reg1 |= ADCX140_I2S_MODE_BIT;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface_reg1 |= ADCX140_LEFT_JUST_BIT;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ break;
+ default:
+ dev_err(component->dev, "Invalid DAI interface format\n");
+ return -EINVAL;
+ }
+
+ adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+ snd_soc_component_update_bits(component, ADCX140_ASI_CFG0,
+ ADCX140_FSYNCINV_BIT |
+ ADCX140_BCLKINV_BIT |
+ ADCX140_ASI_FORMAT_MSK,
+ iface_reg1);
+ snd_soc_component_update_bits(component, ADCX140_MST_CFG0,
+ ADCX140_BCLK_FSYNC_MASTER, iface_reg2);
+
+ return 0;
+}
+
+static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_component *component = codec_dai->component;
+ struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+ unsigned int lsb;
+
+ if (tx_mask != rx_mask) {
+ dev_err(component->dev, "tx and rx masks must be symmetric\n");
+ return -EINVAL;
+ }
+
+ /* TDM based on DSP mode requires slots to be adjacent */
+ lsb = __ffs(tx_mask);
+ if ((lsb + 1) != __fls(tx_mask)) {
+ dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
+ return -EINVAL;
+ }
+
+ switch (slot_width) {
+ case 16:
+ case 20:
+ case 24:
+ case 32:
+ break;
+ default:
+ dev_err(component->dev, "Unsupported slot width %d\n", slot_width);
+ return -EINVAL;
+ }
+
+ adcx140->tdm_delay = lsb;
+ adcx140->slot_width = slot_width;
+
+ return 0;
+}
+
+static int adcx140_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+ int offset = 0;
+ int width = adcx140->slot_width;
+
+ if (!width)
+ width = substream->runtime->sample_bits;
+
+ /* TDM slot selection only valid in DSP_A/_B mode */
+ if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_A)
+ offset += (adcx140->tdm_delay * width + 1);
+ else if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_B)
+ offset += adcx140->tdm_delay * width;
+
+ /* Configure data offset */
+ snd_soc_component_update_bits(component, ADCX140_ASI_CFG1,
+ ADCX140_TX_OFFSET_MASK, offset);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops adcx140_dai_ops = {
+ .hw_params = adcx140_hw_params,
+ .set_fmt = adcx140_set_dai_fmt,
+ .prepare = adcx140_prepare,
+ .set_tdm_slot = adcx140_set_dai_tdm_slot,
+};
+
+static int adcx140_codec_probe(struct snd_soc_component *component)
+{
+ struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+ int sleep_cfg_val = ADCX140_WAKE_DEV;
+ u8 bias_source;
+ u8 vref_source;
+ int ret;
+
+ ret = device_property_read_u8(adcx140->dev, "ti,mic-bias-source",
+ &bias_source);
+ if (ret)
+ bias_source = ADCX140_MIC_BIAS_VAL_VREF;
+
+ if (bias_source < ADCX140_MIC_BIAS_VAL_VREF ||
+ bias_source > ADCX140_MIC_BIAS_VAL_AVDD) {
+ dev_err(adcx140->dev, "Mic Bias source value is invalid\n");
+ return -EINVAL;
+ }
+
+ ret = device_property_read_u8(adcx140->dev, "ti,vref-source",
+ &vref_source);
+ if (ret)
+ vref_source = ADCX140_MIC_BIAS_VREF_275V;
+
+ if (vref_source < ADCX140_MIC_BIAS_VREF_275V ||
+ vref_source > ADCX140_MIC_BIAS_VREF_1375V) {
+ dev_err(adcx140->dev, "Mic Bias source value is invalid\n");
+ return -EINVAL;
+ }
+
+ bias_source |= vref_source;
+
+ ret = adcx140_reset(adcx140);
+ if (ret)
+ goto out;
+
+ if(adcx140->supply_areg == NULL)
+ sleep_cfg_val |= ADCX140_AREG_INTERNAL;
+
+ ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val);
+ if (ret) {
+ dev_err(adcx140->dev, "setting sleep config failed %d\n", ret);
+ goto out;
+ }
+
+ /* 8.4.3: Wait >= 1ms after entering active mode. */
+ usleep_range(1000, 100000);
+
+ ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG,
+ ADCX140_MIC_BIAS_VAL_MSK |
+ ADCX140_MIC_BIAS_VREF_MSK, bias_source);
+ if (ret)
+ dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret);
+out:
+ return ret;
+}
+
+static int adcx140_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
+{
+ struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+ int pwr_cfg = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ |
+ ADCX140_PWR_CFG_ADC_PDZ;
+ break;
+ case SND_SOC_BIAS_OFF:
+ pwr_cfg = 0x0;
+ break;
+ }
+
+ return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg);
+}
+
+static const struct snd_soc_component_driver soc_codec_driver_adcx140 = {
+ .probe = adcx140_codec_probe,
+ .set_bias_level = adcx140_set_bias_level,
+ .controls = adcx140_snd_controls,
+ .num_controls = ARRAY_SIZE(adcx140_snd_controls),
+ .dapm_widgets = adcx140_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adcx140_dapm_widgets),
+ .dapm_routes = adcx140_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(adcx140_audio_map),
+ .suspend_bias_off = 1,
+ .idle_bias_on = 0,
+ .use_pmdown_time = 1,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+static struct snd_soc_dai_driver adcx140_dai_driver[] = {
+ {
+ .name = "tlv320adcx140-codec",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = ADCX140_MAX_CHANNELS,
+ .rates = ADCX140_RATES,
+ .formats = ADCX140_FORMATS,
+ },
+ .ops = &adcx140_dai_ops,
+ .symmetric_rates = 1,
+ }
+};
+
+static const struct of_device_id tlv320adcx140_of_match[] = {
+ { .compatible = "ti,tlv320adc3140" },
+ { .compatible = "ti,tlv320adc5140" },
+ { .compatible = "ti,tlv320adc6140" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, tlv320adcx140_of_match);
+
+static int adcx140_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct adcx140_priv *adcx140;
+ int ret;
+
+ adcx140 = devm_kzalloc(&i2c->dev, sizeof(*adcx140), GFP_KERNEL);
+ if (!adcx140)
+ return -ENOMEM;
+
+ adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev,
+ "reset", GPIOD_OUT_LOW);
+ if (IS_ERR(adcx140->gpio_reset))
+ dev_info(&i2c->dev, "Reset GPIO not defined\n");
+
+ adcx140->supply_areg = devm_regulator_get_optional(adcx140->dev,
+ "areg");
+ if (IS_ERR(adcx140->supply_areg)) {
+ if (PTR_ERR(adcx140->supply_areg) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+ else
+ adcx140->supply_areg = NULL;
+ } else {
+ ret = regulator_enable(adcx140->supply_areg);
+ if (ret) {
+ dev_err(adcx140->dev, "Failed to enable areg\n");
+ return ret;
+ }
+ }
+
+ adcx140->regmap = devm_regmap_init_i2c(i2c, &adcx140_i2c_regmap);
+ if (IS_ERR(adcx140->regmap)) {
+ ret = PTR_ERR(adcx140->regmap);
+ dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+ ret);
+ return ret;
+ }
+ adcx140->dev = &i2c->dev;
+ i2c_set_clientdata(i2c, adcx140);
+
+ return devm_snd_soc_register_component(&i2c->dev,
+ &soc_codec_driver_adcx140,
+ adcx140_dai_driver, 1);
+}
+
+static const struct i2c_device_id adcx140_i2c_id[] = {
+ { "tlv320adc3140", 0 },
+ { "tlv320adc5140", 1 },
+ { "tlv320adc6140", 2 },
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, adcx140_i2c_id);
+
+static struct i2c_driver adcx140_i2c_driver = {
+ .driver = {
+ .name = "tlv320adcx140-codec",
+ .of_match_table = of_match_ptr(tlv320adcx140_of_match),
+ },
+ .probe = adcx140_i2c_probe,
+ .id_table = adcx140_i2c_id,
+};
+module_i2c_driver(adcx140_i2c_driver);
+
+MODULE_AUTHOR("Dan Murphy <dmurphy@ti.com>");
+MODULE_DESCRIPTION("ASoC TLV320ADCX140 CODEC Driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h
new file mode 100644
index 000000000000..6d055e55909e
--- /dev/null
+++ b/sound/soc/codecs/tlv320adcx140.h
@@ -0,0 +1,131 @@
+// SPDX-License-Identifier: GPL-2.0
+// TLV320ADCX104 Sound driver
+// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/
+
+#ifndef _TLV320ADCX140_H
+#define _TLV320ADCX140_H
+
+#define ADCX140_RATES (SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define ADCX140_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+#define ADCX140_PAGE_SELECT 0x00
+#define ADCX140_SW_RESET 0x01
+#define ADCX140_SLEEP_CFG 0x02
+#define ADCX140_SHDN_CFG 0x05
+#define ADCX140_ASI_CFG0 0x07
+#define ADCX140_ASI_CFG1 0x08
+#define ADCX140_ASI_CFG2 0x09
+#define ADCX140_ASI_CH1 0x0b
+#define ADCX140_ASI_CH2 0x0c
+#define ADCX140_ASI_CH3 0x0d
+#define ADCX140_ASI_CH4 0x0e
+#define ADCX140_ASI_CH5 0x0f
+#define ADCX140_ASI_CH6 0x10
+#define ADCX140_ASI_CH7 0x11
+#define ADCX140_ASI_CH8 0x12
+#define ADCX140_MST_CFG0 0x13
+#define ADCX140_MST_CFG1 0x14
+#define ADCX140_ASI_STS 0x15
+#define ADCX140_CLK_SRC 0x16
+#define ADCX140_PDMCLK_CFG 0x1f
+#define ADCX140_PDM_CFG 0x20
+#define ADCX140_GPIO_CFG0 0x21
+#define ADCX140_GPO_CFG1 0x22
+#define ADCX140_GPO_CFG2 0x23
+#define ADCX140_GPO_CFG3 0x24
+#define ADCX140_GPO_CFG4 0x25
+#define ADCX140_GPO_VAL 0x29
+#define ADCX140_GPIO_MON 0x2a
+#define ADCX140_GPI_CFG0 0x2b
+#define ADCX140_GPI_CFG1 0x2c
+#define ADCX140_GPI_MON 0x2f
+#define ADCX140_INT_CFG 0x32
+#define ADCX140_INT_MASK0 0x33
+#define ADCX140_INT_LTCH0 0x36
+#define ADCX140_BIAS_CFG 0x3b
+#define ADCX140_CH1_CFG0 0x3c
+#define ADCX140_CH1_CFG1 0x3d
+#define ADCX140_CH1_CFG2 0x3e
+#define ADCX140_CH1_CFG3 0x3f
+#define ADCX140_CH1_CFG4 0x40
+#define ADCX140_CH2_CFG0 0x41
+#define ADCX140_CH2_CFG1 0x42
+#define ADCX140_CH2_CFG2 0x43
+#define ADCX140_CH2_CFG3 0x44
+#define ADCX140_CH2_CFG4 0x45
+#define ADCX140_CH3_CFG0 0x46
+#define ADCX140_CH3_CFG1 0x47
+#define ADCX140_CH3_CFG2 0x48
+#define ADCX140_CH3_CFG3 0x49
+#define ADCX140_CH3_CFG4 0x4a
+#define ADCX140_CH4_CFG0 0x4b
+#define ADCX140_CH4_CFG1 0x4c
+#define ADCX140_CH4_CFG2 0x4d
+#define ADCX140_CH4_CFG3 0x4e
+#define ADCX140_CH4_CFG4 0x4f
+#define ADCX140_CH5_CFG2 0x52
+#define ADCX140_CH5_CFG3 0x53
+#define ADCX140_CH5_CFG4 0x54
+#define ADCX140_CH6_CFG2 0x57
+#define ADCX140_CH6_CFG3 0x58
+#define ADCX140_CH6_CFG4 0x59
+#define ADCX140_CH7_CFG2 0x5c
+#define ADCX140_CH7_CFG3 0x5d
+#define ADCX140_CH7_CFG4 0x5e
+#define ADCX140_CH8_CFG2 0x61
+#define ADCX140_CH8_CFG3 0x62
+#define ADCX140_CH8_CFG4 0x63
+#define ADCX140_DSP_CFG0 0x6b
+#define ADCX140_DSP_CFG1 0x6c
+#define ADCX140_DRE_CFG0 0x6d
+#define ADCX140_AGC_CFG0 0x70
+#define ADCX140_IN_CH_EN 0x73
+#define ADCX140_ASI_OUT_CH_EN 0x74
+#define ADCX140_PWR_CFG 0x75
+#define ADCX140_DEV_STS0 0x76
+#define ADCX140_DEV_STS1 0x77
+
+#define ADCX140_RESET BIT(0)
+
+#define ADCX140_WAKE_DEV BIT(0)
+#define ADCX140_AREG_INTERNAL BIT(7)
+
+#define ADCX140_BCLKINV_BIT BIT(2)
+#define ADCX140_FSYNCINV_BIT BIT(3)
+#define ADCX140_INV_MSK (ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT)
+#define ADCX140_BCLK_FSYNC_MASTER BIT(7)
+#define ADCX140_I2S_MODE_BIT BIT(6)
+#define ADCX140_LEFT_JUST_BIT BIT(7)
+#define ADCX140_ASI_FORMAT_MSK (ADCX140_I2S_MODE_BIT | ADCX140_LEFT_JUST_BIT)
+
+#define ADCX140_16_BIT_WORD 0x0
+#define ADCX140_20_BIT_WORD BIT(4)
+#define ADCX140_24_BIT_WORD BIT(5)
+#define ADCX140_32_BIT_WORD (BIT(4) | BIT(5))
+#define ADCX140_WORD_LEN_MSK 0x30
+
+#define ADCX140_MAX_CHANNELS 8
+
+#define ADCX140_MIC_BIAS_VAL_VREF 0
+#define ADCX140_MIC_BIAS_VAL_VREF_1096 1
+#define ADCX140_MIC_BIAS_VAL_AVDD 6
+#define ADCX140_MIC_BIAS_VAL_MSK GENMASK(6, 4)
+
+#define ADCX140_MIC_BIAS_VREF_275V 0
+#define ADCX140_MIC_BIAS_VREF_25V 1
+#define ADCX140_MIC_BIAS_VREF_1375V 2
+#define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0)
+
+#define ADCX140_PWR_CFG_BIAS_PDZ BIT(7)
+#define ADCX140_PWR_CFG_ADC_PDZ BIT(6)
+#define ADCX140_PWR_CFG_PLL_PDZ BIT(5)
+
+#define ADCX140_TX_OFFSET_MASK GENMASK(4, 0)
+
+#endif /* _TLV320ADCX140_ */
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index f11ffa28683b..700cc1212770 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -4926,11 +4926,11 @@ static const struct regmap_range_cfg wcd9335_ranges[] = {
.name = "WCD9335",
.range_min = 0x0,
.range_max = WCD9335_MAX_REGISTER,
- .selector_reg = WCD9335_REG(0x0, 0),
+ .selector_reg = WCD9335_SEL_REGISTER,
.selector_mask = 0xff,
.selector_shift = 0,
- .window_start = 0x0,
- .window_len = 0x1000,
+ .window_start = 0x800,
+ .window_len = 0x100,
},
};
@@ -4968,12 +4968,12 @@ static const struct regmap_range_cfg wcd9335_ifc_ranges[] = {
{
.name = "WCD9335-IFC-DEV",
.range_min = 0x0,
- .range_max = WCD9335_REG(0, 0x7ff),
- .selector_reg = WCD9335_REG(0, 0x0),
- .selector_mask = 0xff,
+ .range_max = WCD9335_MAX_REGISTER,
+ .selector_reg = WCD9335_SEL_REGISTER,
+ .selector_mask = 0xfff,
.selector_shift = 0,
- .window_start = 0x0,
- .window_len = 0x1000,
+ .window_start = 0x800,
+ .window_len = 0x400,
},
};
@@ -4981,7 +4981,7 @@ static struct regmap_config wcd9335_ifc_regmap_config = {
.reg_bits = 16,
.val_bits = 8,
.can_multi_write = true,
- .max_register = WCD9335_REG(0, 0x7FF),
+ .max_register = WCD9335_MAX_REGISTER,
.ranges = wcd9335_ifc_ranges,
.num_ranges = ARRAY_SIZE(wcd9335_ifc_ranges),
};
diff --git a/sound/soc/codecs/wcd9335.h b/sound/soc/codecs/wcd9335.h
index 4d9be2496c30..72060824c743 100644
--- a/sound/soc/codecs/wcd9335.h
+++ b/sound/soc/codecs/wcd9335.h
@@ -8,9 +8,9 @@
* in slimbus mode the reg base starts from 0x800
* in i2s/i2c mode the reg base is 0x0
*/
-#define WCD9335_REG(pg, r) ((pg << 12) | (r) | 0x800)
+#define WCD9335_REG(pg, r) ((pg << 8) | (r))
#define WCD9335_REG_OFFSET(r) (r & 0xFF)
-#define WCD9335_PAGE_OFFSET(r) ((r >> 12) & 0xFF)
+#define WCD9335_PAGE_OFFSET(r) ((r >> 8) & 0xFF)
/* Page-0 Registers */
#define WCD9335_PAGE0_PAGE_REGISTER WCD9335_REG(0x00, 0x000)
@@ -600,7 +600,8 @@
#define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE BIT(0)
#define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_DISABLE 0
#define WCD9335_CDC_TOP_TOP_CFG1 WCD9335_REG(0x0d, 0x082)
-#define WCD9335_MAX_REGISTER WCD9335_REG(0x80, 0x0FF)
+#define WCD9335_MAX_REGISTER 0xffff
+#define WCD9335_SEL_REGISTER 0x800
/* SLIMBUS Slave Registers */
#define WCD9335_SLIM_PGD_PORT_INT_EN0 WCD9335_REG(0, 0x30)
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 158e878abd6c..5269857e2746 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -3,7 +3,6 @@
#include <linux/clk.h>
#include <linux/clk-provider.h>
-#include <linux/gpio.h>
#include <linux/interrupt.h>
#include <linux/kernel.h>
#include <linux/mfd/wcd934x/registers.h>
@@ -11,10 +10,7 @@
#include <linux/module.h>
#include <linux/mutex.h>
#include <linux/of_clk.h>
-#include <linux/of_device.h>
-#include <linux/of_gpio.h>
#include <linux/of.h>
-#include <linux/of_irq.h>
#include <linux/platform_device.h>
#include <linux/regmap.h>
#include <linux/regulator/consumer.h>
@@ -1202,11 +1198,6 @@ static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src)
regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO,
WCD934X_ANA_RCO_BG_EN_MASK, 0);
usleep_range(100, 110);
- } else if (sido_src == SIDO_SOURCE_RCO_BG) {
- regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO,
- WCD934X_ANA_RCO_BG_EN_MASK,
- WCD934X_ANA_RCO_BG_ENABLE);
- usleep_range(100, 110);
regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL,
WCD934X_ANA_BUCK_PRE_EN1_MASK,
WCD934X_ANA_BUCK_PRE_EN1_ENABLE);
@@ -1219,6 +1210,11 @@ static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src)
WCD934X_ANA_BUCK_HI_ACCU_EN_MASK,
WCD934X_ANA_BUCK_HI_ACCU_ENABLE);
usleep_range(100, 110);
+ } else if (sido_src == SIDO_SOURCE_RCO_BG) {
+ regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO,
+ WCD934X_ANA_RCO_BG_EN_MASK,
+ WCD934X_ANA_RCO_BG_ENABLE);
+ usleep_range(100, 110);
}
wcd->sido_input_src = sido_src;
@@ -1883,20 +1879,16 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai,
return -EINVAL;
}
- if (wcd->rx_chs) {
- wcd->num_rx_port = rx_num;
- for (i = 0; i < rx_num; i++) {
- wcd->rx_chs[i].ch_num = rx_slot[i];
- INIT_LIST_HEAD(&wcd->rx_chs[i].list);
- }
+ wcd->num_rx_port = rx_num;
+ for (i = 0; i < rx_num; i++) {
+ wcd->rx_chs[i].ch_num = rx_slot[i];
+ INIT_LIST_HEAD(&wcd->rx_chs[i].list);
}
- if (wcd->tx_chs) {
- wcd->num_tx_port = tx_num;
- for (i = 0; i < tx_num; i++) {
- wcd->tx_chs[i].ch_num = tx_slot[i];
- INIT_LIST_HEAD(&wcd->tx_chs[i].list);
- }
+ wcd->num_tx_port = tx_num;
+ for (i = 0; i < tx_num; i++) {
+ wcd->tx_chs[i].ch_num = tx_slot[i];
+ INIT_LIST_HEAD(&wcd->tx_chs[i].list);
}
return 0;
@@ -3392,18 +3384,15 @@ static void wcd934x_codec_hphdelay_lutbypass(struct snd_soc_component *comp,
{
u8 hph_dly_mask;
u16 hph_lut_bypass_reg = 0;
- u16 hph_comp_ctrl7 = 0;
switch (interp_idx) {
case INTERP_HPHL:
hph_dly_mask = 1;
hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHL_COMP_LUT;
- hph_comp_ctrl7 = WCD934X_CDC_COMPANDER1_CTL7;
break;
case INTERP_HPHR:
hph_dly_mask = 2;
hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHR_COMP_LUT;
- hph_comp_ctrl7 = WCD934X_CDC_COMPANDER2_CTL7;
break;
default:
return;
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 727d6703c905..fbcee21736e8 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -43,7 +43,7 @@ struct dfw_binrec {
u8 command;
u32 length:24;
u32 address;
- uint8_t data[0];
+ uint8_t data[];
} __packed;
struct dfw_inforec {
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 9dc215b5c504..499e87d1dfcc 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2245,14 +2245,14 @@ static int wm5110_open(struct snd_compr_stream *stream)
struct arizona *arizona = priv->core.arizona;
int n_adsp;
- if (strcmp(rtd->codec_dai->name, "wm5110-dsp-voicectrl") == 0) {
+ if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-voicectrl") == 0) {
n_adsp = 2;
- } else if (strcmp(rtd->codec_dai->name, "wm5110-dsp-trace") == 0) {
+ } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-trace") == 0) {
n_adsp = 0;
} else {
dev_err(arizona->dev,
"No suitable compressed stream for DAI '%s'\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index dc4fe4f5239d..06ba36595ddd 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -196,14 +196,6 @@ SOC_DAPM_SINGLE("MicN Switch", WM8974_INPUT, 1, 1, 0),
SOC_DAPM_SINGLE("MicP Switch", WM8974_INPUT, 0, 1, 0),
};
-/* AUX Input boost vol */
-static const struct snd_kcontrol_new wm8974_aux_boost_controls =
-SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0);
-
-/* Mic Input boost vol */
-static const struct snd_kcontrol_new wm8974_mic_boost_controls =
-SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0);
-
static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0,
&wm8974_speaker_mixer_controls[0],
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index d3d32b501aca..1ef69409ccd1 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1436,12 +1436,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp,
subname = NULL; /* don't append subname */
break;
case 2:
- ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+ ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
"%s%c %.12s %x", dsp->name, *region_name,
wm_adsp_fw_text[dsp->fw], alg_region->alg);
break;
default:
- ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+ ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
"%s %.12s %x", dsp->name,
wm_adsp_fw_text[dsp->fw], alg_region->alg);
break;
@@ -3467,22 +3467,22 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream)
if (wm_adsp_fw[dsp->fw].num_caps == 0) {
adsp_err(dsp, "%s: Firmware does not support compressed API\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
ret = -ENXIO;
goto out;
}
if (wm_adsp_fw[dsp->fw].compr_direction != stream->direction) {
adsp_err(dsp, "%s: Firmware does not support stream direction\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
ret = -EINVAL;
goto out;
}
list_for_each_entry(tmp, &dsp->compr_list, list) {
- if (!strcmp(tmp->name, rtd->codec_dai->name)) {
+ if (!strcmp(tmp->name, asoc_rtd_to_codec(rtd, 0)->name)) {
adsp_err(dsp, "%s: Only a single stream supported per dai\n",
- rtd->codec_dai->name);
+ asoc_rtd_to_codec(rtd, 0)->name);
ret = -EBUSY;
goto out;
}
@@ -3496,7 +3496,7 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream)
compr->dsp = dsp;
compr->stream = stream;
- compr->name = rtd->codec_dai->name;
+ compr->name = asoc_rtd_to_codec(rtd, 0)->name;
list_add_tail(&compr->list, &dsp->compr_list);
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index b59f1d0e7f84..f2d6f2f81f14 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -676,7 +676,6 @@ struct wsa881x_priv {
int active_ports;
bool port_prepared[WSA881X_MAX_SWR_PORTS];
bool port_enable[WSA881X_MAX_SWR_PORTS];
- bool stream_prepared;
};
static void wsa881x_init(struct wsa881x_priv *wsa881x)
@@ -954,41 +953,6 @@ static const struct snd_soc_dapm_widget wsa881x_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("SPKR"),
};
-static int wsa881x_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev);
- int ret;
-
- if (wsa881x->stream_prepared) {
- sdw_disable_stream(wsa881x->sruntime);
- sdw_deprepare_stream(wsa881x->sruntime);
- wsa881x->stream_prepared = false;
- }
-
-
- ret = sdw_prepare_stream(wsa881x->sruntime);
- if (ret)
- return ret;
-
- /**
- * NOTE: there is a strict hw requirement about the ordering of port
- * enables and actual PA enable. PA enable should only happen after
- * soundwire ports are enabled if not DC on the line is accumulated
- * resulting in Click/Pop Noise
- * PA enable/mute are handled as part of DAPM and digital mute.
- */
-
- ret = sdw_enable_stream(wsa881x->sruntime);
- if (ret) {
- sdw_deprepare_stream(wsa881x->sruntime);
- return ret;
- }
- wsa881x->stream_prepared = true;
-
- return ret;
-}
-
static int wsa881x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1016,12 +980,7 @@ static int wsa881x_hw_free(struct snd_pcm_substream *substream,
{
struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev);
- if (wsa881x->stream_prepared) {
- sdw_disable_stream(wsa881x->sruntime);
- sdw_deprepare_stream(wsa881x->sruntime);
- sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime);
- wsa881x->stream_prepared = false;
- }
+ sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime);
return 0;
}
@@ -1052,7 +1011,6 @@ static int wsa881x_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
static struct snd_soc_dai_ops wsa881x_dai_ops = {
.hw_params = wsa881x_hw_params,
- .prepare = wsa881x_prepare,
.hw_free = wsa881x_hw_free,
.mute_stream = wsa881x_digital_mute,
.set_sdw_stream = wsa881x_set_sdw_stream,
@@ -1150,7 +1108,7 @@ static int wsa881x_probe(struct sdw_slave *pdev,
wsa881x->sconfig.type = SDW_STREAM_PDM;
pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0);
pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop;
- gpiod_set_value(wsa881x->sd_n, 1);
+ gpiod_direction_output(wsa881x->sd_n, 1);
wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config);
if (IS_ERR(wsa881x->regmap)) {
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index 7eeca2150b2d..515f88456dbd 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -422,15 +422,15 @@ static int dw_i2s_resume(struct snd_soc_component *component)
{
struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component);
struct snd_soc_dai *dai;
+ int stream;
if (dev->capability & DW_I2S_MASTER)
clk_enable(dev->clk);
for_each_component_dais(component, dai) {
- if (dai->playback_active)
- dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK);
- if (dai->capture_active)
- dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE);
+ for_each_pcm_streams(stream)
+ if (dai->stream_active[stream])
+ dw_i2s_config(dev, stream);
}
return 0;
diff --git a/sound/soc/dwc/dwc-pcm.c b/sound/soc/dwc/dwc-pcm.c
index 4b25aca3804f..9868e7373d36 100644
--- a/sound/soc/dwc/dwc-pcm.c
+++ b/sound/soc/dwc/dwc-pcm.c
@@ -140,7 +140,7 @@ static int dw_pcm_open(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
snd_soc_set_runtime_hwparams(substream, &dw_pcm_hardware);
snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 6f3b768489f6..4ff2d21bb32f 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -31,8 +31,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 9ce55feaac22..bb33601fab84 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -159,7 +159,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
return 0;
/* Specific configurations of DAIs starts from here */
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
cpu_priv->sysclk_freq[tx],
cpu_priv->sysclk_dir[tx]);
if (ret && ret != -ENOTSUPP) {
@@ -168,7 +168,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
}
if (cpu_priv->slot_width) {
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2,
cpu_priv->slot_width);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set TDM slot for cpu dai\n");
@@ -257,7 +257,7 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
@@ -446,14 +446,14 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_pcm_runtime *rtd = list_first_entry(
&card->rtd_list, struct snd_soc_pcm_runtime, list);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = card->dev;
int ret;
if (fsl_asoc_card_is_ac97(priv)) {
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
/*
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index ece130f59d15..e7178817d7a7 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -152,7 +152,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
for_each_dpcm_be(rtd, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
struct snd_pcm_substream *substream_be;
- struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(be, 0);
if (dpcm->fe != rtd)
continue;
@@ -169,7 +169,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
}
/* Override dma_data of the Front-End and config its dmaengine */
- dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_params_fe = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index);
dma_params_fe->maxburst = dma_params_be->maxburst;
@@ -328,7 +328,7 @@ static int fsl_asrc_dma_startup(struct snd_soc_component *component,
goto dma_chan_err;
}
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
/* Refine the snd_imx_hardware according to caps of DMA. */
ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream,
@@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component,
return ret;
}
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ for_each_pcm_streams(i) {
substream = pcm->streams[i].substream;
if (!substream)
continue;
@@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component,
struct snd_pcm_substream *substream;
int i;
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ for_each_pcm_streams(i) {
substream = pcm->streams[i].substream;
if (!substream)
continue;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 7858a5499ac5..c711d2d93280 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -370,7 +370,7 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
int sample_rate)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
struct regmap *regmap = spdif_priv->regmap;
struct platform_device *pdev = spdif_priv->pdev;
@@ -458,7 +458,7 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct platform_device *pdev = spdif_priv->pdev;
struct regmap *regmap = spdif_priv->regmap;
u32 scr, mask;
@@ -534,7 +534,7 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct regmap *regmap = spdif_priv->regmap;
u32 scr, mask, i;
@@ -569,7 +569,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
struct platform_device *pdev = spdif_priv->pdev;
u32 sample_rate = params_rate(params);
@@ -597,7 +597,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct regmap *regmap = spdif_priv->regmap;
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
u32 intr = SIE_INTR_FOR(tx);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 5c97269be346..bad89b0d129e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -631,7 +631,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
int ret;
ret = clk_prepare_enable(ssi->clk);
@@ -655,7 +655,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
clk_disable_unprepare(ssi->clk);
}
@@ -854,7 +854,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
if (fsl_ssi_is_i2s_master(ssi) &&
ssi->baudclk_streams & BIT(substream->stream)) {
@@ -1059,7 +1059,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
switch (cmd) {
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 5ef6881395e0..e09b45de0efd 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -85,13 +85,13 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream,
dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN;
/* set DAI configuration */
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt);
if (ret) {
dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
return ret;
}
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir);
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), FSL_SAI_CLK_MAST1, 0, dir);
if (ret) {
dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
return ret;
@@ -101,7 +101,7 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream,
* Per datasheet, AUDMIX expects 8 slots and 32 bits
* for every slot in TDM mode.
*/
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1,
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), BIT(channels) - 1,
BIT(channels) - 1, 8, 32);
if (ret)
dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret);
@@ -125,7 +125,7 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream,
fmt |= SND_SOC_DAIFMT_CBM_CFM;
/* set AUDMIX DAI configuration */
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt);
if (ret)
dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret);
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 2b679680c93f..fab2d6c56653 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -27,8 +27,8 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16);
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 15e8b9343c35..f45cb4bbb6c4 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -30,7 +30,7 @@ static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
struct device *dev = rtd->card->dev;
int ret;
- ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), SGTL5000_SYSCLK,
data->clk_frequency, SND_SOC_CLOCK_IN);
if (ret) {
dev_err(dev, "could not set codec driver clock params\n");
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index ed7211d744b3..3b8c796d7829 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -115,7 +115,7 @@ static int psc_dma_trigger(struct snd_soc_component *component,
struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct snd_pcm_runtime *runtime = substream->runtime;
struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
@@ -217,7 +217,7 @@ static int psc_dma_open(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct psc_dma_stream *s;
int rc;
@@ -245,7 +245,7 @@ static int psc_dma_close(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct psc_dma_stream *s;
dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
@@ -271,7 +271,7 @@ psc_dma_pointer(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct psc_dma_stream *s;
dma_addr_t count;
@@ -298,7 +298,7 @@ static int psc_dma_new(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
- struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_pcm *pcm = rtd->pcm;
size_t size = psc_dma_hardware.buffer_bytes_max;
int rc;
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9bc01f374b39..1ab4fbda08cb 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -39,7 +39,7 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
u32 mode;
dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 23617eb09ba1..f7bd90051ce7 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -105,7 +105,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
int ret = 0;
/* Tell the codec driver what the serial protocol is. */
- ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format);
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), machine_data->dai_format);
if (ret < 0) {
dev_err(dev, "could not set codec driver audio format\n");
return ret;
@@ -115,7 +115,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
* Tell the codec driver what the MCLK frequency is, and whether it's
* a slave or master.
*/
- ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0,
machine_data->clk_frequency,
machine_data->codec_clk_direction);
if (ret < 0) {
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index 38ac4a397742..a36d4e8cd55c 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -37,8 +37,8 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 6114b01b90f7..fe3091590f20 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -128,7 +128,7 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream)
int ret = 0;
/* Tell the codec driver what the serial protocol is. */
- ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format);
if (ret < 0) {
dev_err(dev, "could not set codec driver audio format\n");
return ret;
@@ -138,7 +138,7 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream)
* Tell the codec driver what the MCLK frequency is, and whether it's
* a slave or master.
*/
- ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0, mdata->clk_frequency,
mdata->codec_clk_direction);
if (ret < 0) {
dev_err(dev, "could not set codec driver clock params\n");
diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c
index 72687235c0ae..f5374fe354ab 100644
--- a/sound/soc/fsl/p1022_rdk.c
+++ b/sound/soc/fsl/p1022_rdk.c
@@ -134,14 +134,14 @@ static int p1022_rdk_startup(struct snd_pcm_substream *substream)
int ret = 0;
/* Tell the codec driver what the serial protocol is. */
- ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format);
if (ret < 0) {
dev_err(dev, "could not set codec driver audio format (ret=%i)\n",
ret);
return ret;
}
- ret = snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, mdata->clk_frequency,
+ ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, mdata->clk_frequency,
mdata->clk_frequency);
if (ret < 0) {
dev_err(dev, "could not set codec PLL frequency (ret=%i)\n",
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index 52d321bede9c..8b1551c55452 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -76,8 +76,8 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int i, found = 0;
snd_pcm_format_t format = params_format(params);
unsigned int rate = params_rate(params);
@@ -196,7 +196,7 @@ static struct snd_soc_jack_pin mic_jack_pins[] = {
static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
/* Headphone jack detection */
snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index 9b794775df53..8c54dc6710fe 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -213,8 +213,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup);
void asoc_simple_shutdown(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct simple_dai_props *dai_props =
simple_priv_to_props(priv, rtd->num);
@@ -249,8 +249,8 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct simple_dai_props *dai_props =
simple_priv_to_props(priv, rtd->num);
@@ -331,22 +331,70 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai,
return 0;
}
+static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd,
+ struct simple_dai_props *dai_props)
+{
+ struct snd_soc_dai_link *dai_link = rtd->dai_link;
+ struct snd_soc_component *component;
+ struct snd_soc_pcm_stream *params;
+ struct snd_pcm_hardware hw;
+ int i, ret, stream;
+
+ /* Only codecs should have non_legacy_dai_naming set. */
+ for_each_rtd_components(rtd, i, component) {
+ if (!component->driver->non_legacy_dai_naming)
+ return 0;
+ }
+
+ /* Assumes the capabilities are the same for all supported streams */
+ for_each_pcm_streams(stream) {
+ ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
+ if (ret == 0)
+ break;
+ }
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
+ return ret;
+ }
+
+ params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
+ if (!params)
+ return -ENOMEM;
+
+ params->formats = hw.formats;
+ params->rates = hw.rates;
+ params->rate_min = hw.rate_min;
+ params->rate_max = hw.rate_max;
+ params->channels_min = hw.channels_min;
+ params->channels_max = hw.channels_max;
+
+ dai_link->params = params;
+ dai_link->num_params = 1;
+
+ return 0;
+}
+
int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num);
int ret;
- ret = asoc_simple_init_dai(rtd->codec_dai,
+ ret = asoc_simple_init_dai(asoc_rtd_to_codec(rtd, 0),
dai_props->codec_dai);
if (ret < 0)
return ret;
- ret = asoc_simple_init_dai(rtd->cpu_dai,
+ ret = asoc_simple_init_dai(asoc_rtd_to_cpu(rtd, 0),
dai_props->cpu_dai);
if (ret < 0)
return ret;
+ ret = asoc_simple_init_dai_link_params(rtd, dai_props);
+ if (ret < 0)
+ return ret;
+
return 0;
}
EXPORT_SYMBOL_GPL(asoc_simple_dai_init);
diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c
index fdd2c73fd2fa..a495d1050d49 100644
--- a/sound/soc/img/img-i2s-in.c
+++ b/sound/soc/img/img-i2s-in.c
@@ -397,7 +397,7 @@ static int img_i2s_in_dma_prepare_slave_config(struct snd_pcm_substream *st,
struct snd_dmaengine_dai_dma_data *dma_data;
int ret;
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st);
+ dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st);
ret = snd_hwparams_to_dma_slave_config(st, params, sc);
if (ret)
diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c
index 4b1853409633..db052ec17d5d 100644
--- a/sound/soc/img/img-i2s-out.c
+++ b/sound/soc/img/img-i2s-out.c
@@ -403,7 +403,7 @@ static int img_i2s_out_dma_prepare_slave_config(struct snd_pcm_substream *st,
struct snd_dmaengine_dai_dma_data *dma_data;
int ret;
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st);
+ dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st);
ret = snd_hwparams_to_dma_slave_config(st, params, sc);
if (ret)
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index baef461a99f1..f883c9340eee 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -1333,7 +1333,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute)
dai->capture_widget->name);
w = dai->capture_widget;
snd_soc_dapm_widget_for_each_source_path(w, p) {
- if (p->connected && !p->connected(w, p->sink))
+ if (p->connected && !p->connected(w, p->source))
continue;
if (p->connect && p->source->power &&
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 340bd2be39a7..82f2b6357778 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -649,7 +649,7 @@ static snd_pcm_uframes_t sst_soc_pointer(struct snd_soc_component *component,
static int sst_soc_pcm_new(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_pcm *pcm = rtd->pcm;
if (dai->driver->playback.channels_min ||
@@ -741,7 +741,7 @@ static int sst_soc_prepare(struct device *dev)
/* set the SSPs to idle */
for_each_card_rtds(drv->soc_card, rtd) {
- struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
if (dai->active) {
send_ssp_cmd(dai, dai->name, 0);
@@ -762,7 +762,7 @@ static void sst_soc_complete(struct device *dev)
/* restart SSPs */
for_each_card_rtds(drv->soc_card, rtd) {
- struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
if (dai->active) {
sst_handle_vb_timer(dai, true);
diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c
index d952719bc098..5862fe968083 100644
--- a/sound/soc/intel/atom/sst/sst_pci.c
+++ b/sound/soc/intel/atom/sst/sst_pci.c
@@ -99,7 +99,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx)
dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram);
do_release_regions:
pci_release_regions(pci);
- return 0;
+ return ret;
}
/*
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 9ca2567d0059..556c3104e641 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -289,7 +289,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
select SND_SOC_DA7219
select SND_SOC_MAX98357A
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
@@ -302,6 +301,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
tristate "Broxton with DA7219 and MAX98357A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
help
This adds support for ASoC machine driver for Broxton-P platforms
@@ -402,6 +402,7 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH
tristate "GLK with DA7219 and MAX98357A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
help
This adds support for ASoC machine driver for Geminilake platforms
@@ -413,10 +414,10 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
tristate "GLK with RT5682 and MAX98357A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_RT5682
select SND_SOC_MAX98357A
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
help
This adds support for ASoC machine driver for Geminilake platforms
@@ -430,7 +431,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC
config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
tristate "SKL/KBL/BXT/APL with HDA Codecs"
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_HDAC_HDMI
select SND_SOC_DMIC
# SND_SOC_HDAC_HDA is already selected
@@ -448,15 +449,31 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
depends on I2C && ACPI
depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\
(SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
+ depends on SND_HDA_CODEC_HDMI
+ select SND_SOC_MAX98373
+ select SND_SOC_RT1015
select SND_SOC_RT5682
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
help
This adds support for ASoC machine driver for SOF platforms
with rt5682 codec.
Say Y if you have such a device.
If unsure select "N".
+
+config SND_SOC_INTEL_SOF_PCM512x_MACH
+ tristate "SOF with TI PCM512x codec"
+ depends on I2C && ACPI
+ depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\
+ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
+ depends on SND_HDA_CODEC_HDMI
+ select SND_SOC_PCM512x_I2C
+ help
+ This adds support for ASoC machine driver for SOF platforms
+ with TI PCM512x I2S audio codec.
+ Say Y or m if you have such a device.
+ If unsure select "N".
+
endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL
if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK)
@@ -476,11 +493,11 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
tristate "CML with RT1011 and RT5682 in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_RT1011
select SND_SOC_RT5682
select SND_SOC_DMIC
select SND_SOC_HDAC_HDMI
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
help
This adds support for ASoC machine driver for SOF platform with
RT1011 + RT5682 I2S codec.
@@ -492,19 +509,43 @@ endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK
if SND_SOC_SOF_JASPERLAKE
config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH
- tristate "SOF with DA7219 and MAX98373 in I2S Mode"
+ tristate "SOF with DA7219 and MAX98373/MAX98360A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_DA7219
select SND_SOC_MAX98373
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
help
This adds support for ASoC machine driver for SOF platforms
- with DA7219 + MAX98373 I2S audio codec.
+ with DA7219 + MAX98373/MAX98360A I2S audio codec.
Say Y if you have such a device.
If unsure select "N".
endif ## SND_SOC_SOF_JASPERLAKE
+if SND_SOC_SOF_INTEL_SOUNDWIRE
+
+config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH
+ tristate "SoundWire generic machine driver"
+ depends on I2C && ACPI
+ depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST
+ depends on SOUNDWIRE
+ depends on SND_HDA_CODEC_HDMI
+ select SND_SOC_RT700_SDW
+ select SND_SOC_RT711_SDW
+ select SND_SOC_RT1308_SDW
+ select SND_SOC_RT1308
+ select SND_SOC_RT715_SDW
+ select SND_SOC_RT5682_SDW
+ select SND_SOC_DMIC
+ help
+ Add support for Intel SoundWire-based platforms connected to
+ RT700, RT711, RT1308 and RT715
+ If unsure select "N".
+
+endif
+
+
endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index b74ddd49bd39..1ef6e60bc2a0 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -7,6 +7,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o
snd-soc-sst-broadwell-objs := broadwell.o
snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o
snd-soc-sst-bxt-rt298-objs := bxt_rt298.o hda_dsp_common.o
+snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o hda_dsp_common.o
snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o hda_dsp_common.o
snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
@@ -18,7 +19,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o
snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o
snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
-snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o
+snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o
snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o
snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
@@ -30,13 +31,18 @@ snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o hda_dsp_c
snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o
-
+snd-soc-sof-sdw-objs += sof_sdw.o \
+ sof_sdw_rt711.o sof_sdw_rt700.o \
+ sof_sdw_rt1308.o sof_sdw_rt715.o \
+ sof_sdw_rt5682.o \
+ sof_sdw_dmic.o sof_sdw_hdmi.o hda_dsp_common.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
+obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o
obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o
@@ -62,4 +68,4 @@ obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max9
obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH) += snd-soc-sof_da7219_max98373.o
-
+obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o
diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c
index 1a302436d450..6c2fdb5659ed 100644
--- a/sound/soc/intel/boards/bdw-rt5650.c
+++ b/sound/soc/intel/boards/bdw-rt5650.c
@@ -107,7 +107,7 @@ static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
/* Workaround: set codec PLL to 19.2MHz that PLL source is
@@ -166,8 +166,8 @@ static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd)
{
struct bdw_rt5650_priv *bdw_rt5650 =
snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_component *component = codec_dai->component;
int ret;
/* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1.
@@ -226,9 +226,6 @@ SND_SOC_DAILINK_DEF(be,
#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
-#else
-SND_SOC_DAILINK_DEF(ssp0_port,
- DAILINK_COMP_ARRAY(COMP_DUMMY()));
#endif
static struct snd_soc_dai_link bdw_rt5650_dais[] = {
@@ -264,7 +261,11 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = {
.dpcm_playback = 1,
.dpcm_capture = 1,
.init = bdw_rt5650_init,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+ SND_SOC_DAILINK_REG(dummy, be, dummy),
+#else
SND_SOC_DAILINK_REG(ssp0_port, be, platform),
+#endif
},
};
@@ -298,7 +299,7 @@ static int bdw_rt5650_probe(struct platform_device *pdev)
return -ENOMEM;
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card,
mach->mach_params.platform);
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index bb643c99069d..6b4b64098d36 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -157,7 +157,7 @@ static int bdw_rt5677_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, 24576000,
@@ -174,7 +174,7 @@ static int bdw_rt5677_dsp_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_PLL1, 24576000,
@@ -226,7 +226,7 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd)
{
struct bdw_rt5677_priv *bdw_rt5677 =
snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
int ret;
@@ -298,9 +298,6 @@ SND_SOC_DAILINK_DEF(be,
#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
-#else
-SND_SOC_DAILINK_DEF(ssp0_port,
- DAILINK_COMP_ARRAY(COMP_DUMMY()));
#endif
/* Wake on voice interface */
@@ -350,7 +347,11 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
.dpcm_playback = 1,
.dpcm_capture = 1,
.init = bdw_rt5677_init,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+ SND_SOC_DAILINK_REG(dummy, be, dummy),
+#else
SND_SOC_DAILINK_REG(ssp0_port, be, platform),
+#endif
},
};
@@ -412,7 +413,7 @@ static int bdw_rt5677_probe(struct platform_device *pdev)
}
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5677_card,
mach->mach_params.platform);
if (ret)
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index b9c12e24c70b..acb4e36682cb 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -70,7 +70,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
int ret = 0;
ret = snd_soc_card_jack_new(rtd->card, "Headset",
SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset,
@@ -104,7 +104,7 @@ static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
@@ -167,9 +167,6 @@ SND_SOC_DAILINK_DEF(codec,
#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
-#else
-SND_SOC_DAILINK_DEF(ssp0_port,
- DAILINK_COMP_ARRAY(COMP_DUMMY()));
#endif
/* broadwell digital audio interface glue - connects codec <--> CPU */
@@ -226,7 +223,11 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.ops = &broadwell_rt286_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+ SND_SOC_DAILINK_REG(dummy, codec, dummy),
+#else
SND_SOC_DAILINK_REG(ssp0_port, codec, platform),
+#endif
},
};
@@ -283,7 +284,7 @@ static int broadwell_audio_probe(struct platform_device *pdev)
broadwell_rt286.dev = &pdev->dev;
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286,
mach->mach_params.platform);
if (ret)
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 9177401c37a5..44016c16f25e 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -179,8 +179,8 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
int clk_freq;
/* Configure sysclk for codec */
@@ -226,7 +226,7 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct bxt_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct bxt_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -244,7 +244,7 @@ static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
static int broxton_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
@@ -721,7 +721,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
}
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret = snd_soc_fixup_dai_links_platform_name(&broxton_audio_card,
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index 4b67f261377c..7a4decf34191 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -155,7 +155,7 @@ static const struct snd_soc_dapm_route geminilake_rt298_map[] = {
static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
@@ -165,7 +165,7 @@ static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd)
static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
int ret = 0;
ret = snd_soc_card_jack_new(rtd->card, "Headset",
@@ -186,7 +186,7 @@ static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct bxt_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -225,7 +225,7 @@ static int broxton_rt298_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL,
@@ -627,7 +627,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(card, ctx);
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret = snd_soc_fixup_dai_links_platform_name(card,
diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c
index 01739ad75b12..f5097da28828 100644
--- a/sound/soc/intel/boards/byt-max98090.c
+++ b/sound/soc/intel/boards/byt-max98090.c
@@ -89,7 +89,7 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
card->dapm.idle_bias_off = true;
- ret = snd_soc_dai_set_sysclk(runtime->codec_dai,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0),
M98090_REG_SYSTEM_CLOCK,
25000000, SND_SOC_CLOCK_IN);
if (ret < 0) {
diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c
index 0c76dafdd572..ace232f8aed6 100644
--- a/sound/soc/intel/boards/byt-rt5640.c
+++ b/sound/soc/intel/boards/byt-rt5640.c
@@ -73,7 +73,7 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
@@ -123,7 +123,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
- struct snd_soc_component *component = runtime->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
struct snd_soc_card *card = runtime->card;
const struct snd_soc_dapm_route *custom_map;
int num_routes;
diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c
index 67f06c95eec5..3b3df7c9008c 100644
--- a/sound/soc/intel/boards/bytcht_cx2072x.c
+++ b/sound/soc/intel/boards/bytcht_cx2072x.c
@@ -70,7 +70,7 @@ static const struct acpi_gpio_mapping byt_cht_cx2072x_acpi_gpios[] = {
static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
- struct snd_soc_component *codec = rtd->codec_dai->component;
+ struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
int ret;
if (devm_acpi_dev_add_driver_gpios(codec->dev,
@@ -80,7 +80,7 @@ static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd)
card->dapm.idle_bias_off = true;
/* set the default PLL rate, the clock is handled by the codec driver */
- ret = snd_soc_dai_set_sysclk(rtd->codec_dai, CX2072X_MCLK_EXTERNAL_PLL,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), CX2072X_MCLK_EXTERNAL_PLL,
19200000, SND_SOC_CLOCK_IN);
if (ret) {
dev_err(rtd->dev, "Could not set sysclk\n");
@@ -97,7 +97,7 @@ static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_component_set_jack(codec, &byt_cht_cx2072x_headset, NULL);
- snd_soc_dai_set_bclk_ratio(rtd->codec_dai, 50);
+ snd_soc_dai_set_bclk_ratio(asoc_rtd_to_codec(rtd, 0), 50);
return ret;
}
@@ -123,7 +123,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd,
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
@@ -132,7 +132,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c
index eda7a500cad6..5e96e7d02733 100644
--- a/sound/soc/intel/boards/bytcht_da7213.c
+++ b/sound/soc/intel/boards/bytcht_da7213.c
@@ -78,7 +78,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
@@ -87,7 +87,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
@@ -106,7 +106,7 @@ static int aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
@@ -127,7 +127,7 @@ static int aif1_hw_params(struct snd_pcm_substream *substream,
static int aif1_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_pll(codec_dai, 0,
@@ -231,7 +231,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev)
int ret_val = 0;
int i;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
card = &bytcht_da7213_card;
card->dev = &pdev->dev;
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 0adc5a5e134a..ddcd070100ef 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -157,7 +157,7 @@ static struct snd_soc_jack_pin byt_cht_es8316_jack_pins[] = {
static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime)
{
- struct snd_soc_component *codec = runtime->codec_dai->component;
+ struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component;
struct snd_soc_card *card = runtime->card;
struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card);
const struct snd_soc_dapm_route *custom_map;
@@ -212,7 +212,7 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime)
if (ret)
dev_err(card->dev, "unable to enable MCLK\n");
- ret = snd_soc_dai_set_sysclk(runtime->codec_dai, 0, 19200000,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0), 0, 19200000,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(card->dev, "can't set codec clock %d\n", ret);
@@ -262,7 +262,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd,
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS
@@ -272,7 +272,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c
index 479af808ef43..8c0dab1f4030 100644
--- a/sound/soc/intel/boards/bytcht_nocodec.c
+++ b/sound/soc/intel/boards/bytcht_nocodec.c
@@ -58,7 +58,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
@@ -68,7 +68,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 6bd9ae813be2..33fb8ea4e5cb 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -381,7 +381,7 @@ static int byt_rt5640_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
return byt_rt5640_prepare_and_enable_pll1(dai, params_rate(params));
}
@@ -805,7 +805,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card);
- struct snd_soc_component *component = runtime->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
const struct snd_soc_dapm_route *custom_map;
int num_routes;
int ret;
@@ -962,7 +962,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd,
* with explicit setting to I2S 2ch. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
@@ -971,7 +971,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 5074bb53f98e..214ef41e23e6 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -348,7 +348,7 @@ static int byt_rt5651_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
snd_pcm_format_t format = params_format(params);
int rate = params_rate(params);
int bclk_ratio;
@@ -540,7 +540,7 @@ static int byt_rt5651_add_codec_device_props(struct device *i2c_dev)
static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
- struct snd_soc_component *codec = runtime->codec_dai->component;
+ struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component;
struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
const struct snd_soc_dapm_route *custom_map;
int num_routes;
@@ -685,7 +685,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd,
* with explicit setting to I2S 2ch. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS
@@ -696,7 +696,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 70bb86f3342f..135701738a44 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -113,7 +113,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK,
@@ -257,7 +257,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
int ret = 0;
unsigned int fmt = 0;
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16);
if (ret < 0) {
dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret);
return ret;
@@ -266,7 +266,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS;
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt);
if (ret < 0) {
dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret);
return ret;
@@ -553,7 +553,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
/* override plaform name, if required */
snd_soc_card_cht.dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht,
diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c
index 501bad3976fb..f456150f89c2 100644
--- a/sound/soc/intel/boards/cht_bsw_nau8824.c
+++ b/sound/soc/intel/boards/cht_bsw_nau8824.c
@@ -73,7 +73,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, NAU8824_CLK_FLL_FS, 0,
@@ -96,7 +96,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
{
struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
struct snd_soc_jack *jack = &ctx->jack;
- struct snd_soc_dai *codec_dai = runtime->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0);
struct snd_soc_component *component = codec_dai->component;
int ret, jack_type;
@@ -259,7 +259,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
/* override plaform name, if required */
snd_soc_card_cht.dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index b5b016d493f1..e64eca56e426 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -208,7 +208,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
@@ -252,7 +252,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
- struct snd_soc_component *component = runtime->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
int jack_type;
int ret;
@@ -359,7 +359,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
* with explicit setting to I2S 2ch 16-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
- ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS
@@ -369,7 +369,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_fmt(rtd->codec_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS
@@ -379,7 +379,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
return ret;
}
- ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
@@ -393,7 +393,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
/*
* Default mode for SSP configuration is TDM 4 slot
*/
- ret = snd_soc_dai_set_fmt(rtd->codec_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBS_CFS);
@@ -403,7 +403,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
}
/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
return ret;
@@ -539,7 +539,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
if (!drv)
return -ENOMEM;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) {
if (acpi_dev_found(snd_soc_cards[i].codec_id) &&
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 9d657421730a..097023a3ec14 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -144,7 +144,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
@@ -176,7 +176,7 @@ static const struct acpi_gpio_mapping cht_rt5672_gpios[] = {
static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
- struct snd_soc_dai *codec_dai = runtime->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0);
struct snd_soc_component *component = codec_dai->component;
struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
@@ -255,7 +255,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
/*
* Default mode for SSP configuration is TDM 4 slot
*/
- ret = snd_soc_dai_set_fmt(rtd->codec_dai,
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBS_CFS);
@@ -265,7 +265,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
}
/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
return ret;
diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c
index dd80d0186a6c..8167b2977e1d 100644
--- a/sound/soc/intel/boards/cml_rt1011_rt5682.c
+++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c
@@ -85,7 +85,7 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = {
static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_jack *jack;
int ret;
@@ -125,7 +125,7 @@ static int cml_rt5682_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int clk_id, clk_freq, pll_out, ret;
clk_id = RT5682_PLL1_S_MCLK;
@@ -164,8 +164,7 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream,
srate = params_rate(params);
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
/* 100 Fs to drive 24 bit data */
ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK,
@@ -275,7 +274,7 @@ static int sof_card_late_probe(struct snd_soc_card *card)
static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -447,12 +446,12 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev)
const char *platform_name;
int ret;
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
if (!ctx)
return -ENOMEM;
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
snd_soc_card_cml.dev = &pdev->dev;
platform_name = mach->mach_params.platform;
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index 8e947bad143c..f13158e4a1fc 100644
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -136,8 +136,8 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_jack *jack;
int ret;
@@ -188,7 +188,7 @@ static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
/* Set valid bitmask & configuration for I2S in 24 bit */
@@ -208,7 +208,7 @@ static struct snd_soc_ops geminilake_rt5682_ops = {
static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct glk_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -225,7 +225,7 @@ static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
struct snd_soc_dapm_context *dapm;
int ret;
@@ -409,6 +409,7 @@ static struct snd_soc_dai_link geminilake_dais[] = {
.init = NULL,
.capture_only = 1,
.nonatomic = 1,
+ .dynamic = 1,
SND_SOC_DAILINK_REG(echoref, dummy, platform),
},
[GLK_DPCM_AUDIO_REF_CP] = {
@@ -604,7 +605,7 @@ static int geminilake_audio_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(card, ctx);
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret = snd_soc_fixup_dai_links_platform_name(card, platform_name);
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 3dadf9bff796..3ed53d7db4e6 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -56,7 +56,7 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000,
@@ -193,7 +193,7 @@ static int haswell_audio_probe(struct platform_device *pdev)
haswell_rt5640.dev = &pdev->dev;
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640,
mach->mach_params.platform);
if (ret)
diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c
index bc7f9a9ce9af..32cd90b8d4c4 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98357a.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c
@@ -159,8 +159,8 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_jack *jack;
int ret;
@@ -203,7 +203,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
{
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct kbl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -236,7 +236,7 @@ static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 7a13e9b35187..abd4e3839678 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -176,10 +176,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *runtime = substream->private_data;
+ struct snd_soc_dai *codec_dai;
int ret, j;
- for (j = 0; j < runtime->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+ for_each_rtd_codec_dais(runtime, j, codec_dai) {
if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) {
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
@@ -221,10 +221,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
int j, ret;
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
const char *name = codec_dai->component->name;
struct snd_soc_component *component = codec_dai->component;
struct snd_soc_dapm_context *dapm =
@@ -331,7 +331,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_jack *jack;
struct snd_soc_card *card = rtd->card;
int ret;
@@ -381,7 +381,7 @@ static int kabylake_dmic_init(struct snd_soc_pcm_runtime *rtd)
static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
{
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct kbl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -414,7 +414,7 @@ static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c
index e23dea9ab79a..6460e3f0c974 100644
--- a/sound/soc/intel/boards/kbl_rt5660.c
+++ b/sound/soc/intel/boards/kbl_rt5660.c
@@ -157,7 +157,7 @@ static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios);
@@ -210,7 +210,7 @@ static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd)
static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
{
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct kbl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -244,7 +244,7 @@ static int kabylake_rt5660_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai,
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index d8f2ff7139a9..658a9da3a40f 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -242,7 +242,7 @@ static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
@@ -258,7 +258,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_jack *jack;
/*
@@ -305,7 +305,7 @@ static int kabylake_rt5663_max98927_codec_init(struct snd_soc_pcm_runtime *rtd)
static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
{
struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct kbl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -431,7 +431,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
/* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */
@@ -472,7 +472,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai;
int ret = 0, j;
- for_each_rtd_codec_dai(rtd, j, codec_dai) {
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
/*
* Use channel 4 and 5 for the first amp
@@ -962,7 +962,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
kabylake_audio_card->dev = &pdev->dev;
snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 96c814f36458..1b1f8d7a4ea3 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -206,7 +206,7 @@ static struct snd_soc_codec_conf max98927_codec_conf[] = {
static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
int ret;
dapm = snd_soc_component_get_dapm(component);
@@ -221,7 +221,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_jack *jack;
/*
@@ -255,7 +255,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd)
static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
{
struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct kbl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -372,7 +372,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
/* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */
@@ -399,7 +399,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai;
int ret = 0, j;
- for_each_rtd_codec_dai(rtd, j, codec_dai) {
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) {
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16);
if (ret < 0) {
@@ -772,7 +772,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
kabylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&kabylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h
index d6150670ca05..e8545d13062f 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_common.h
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.h
@@ -49,6 +49,10 @@ static inline int skl_hda_hdmi_build_controls(struct snd_soc_card *card)
struct snd_soc_component *component;
struct skl_hda_hdmi_pcm *pcm;
+ /* HDMI disabled, do not create controls */
+ if (list_empty(&ctx->hdmi_pcm_list))
+ return 0;
+
pcm = list_first_entry(&ctx->hdmi_pcm_list, struct skl_hda_hdmi_pcm,
head);
component = pcm->codec_dai->component;
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 11eaee9ae41f..3be764299ab0 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -61,6 +61,9 @@ static const struct snd_soc_dapm_route skl_hda_map[] = {
{ "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" },
};
+SND_SOC_DAILINK_DEF(dummy_codec,
+ DAILINK_COMP_ARRAY(COMP_CODEC("snd-soc-dummy", "snd-soc-dummy-dai")));
+
static int skl_hda_card_late_probe(struct snd_soc_card *card)
{
return skl_hda_hdmi_jack_init(card);
@@ -114,13 +117,19 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
{
struct snd_soc_card *card = &hda_soc_card;
struct snd_soc_dai_link *dai_link;
- u32 codec_count, codec_mask;
+ u32 codec_count, codec_mask, idisp_mask;
int i, num_links, num_route;
codec_mask = mach_params->codec_mask;
codec_count = hweight_long(codec_mask);
+ idisp_mask = codec_mask & IDISP_CODEC_MASK;
+
+ if (!codec_count || codec_count > 2 ||
+ (codec_count == 2 && !idisp_mask))
+ return -EINVAL;
- if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) {
+ if (codec_mask == idisp_mask) {
+ /* topology with iDisp as the only HDA codec */
num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT;
num_route = IDISP_ROUTE_COUNT;
@@ -135,13 +144,19 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
skl_hda_be_dai_links[IDISP_DAI_COUNT +
HDAC_DAI_COUNT + i];
}
- } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) {
+ } else {
+ /* topology with external and iDisp HDA codecs */
num_links = ARRAY_SIZE(skl_hda_be_dai_links);
num_route = ARRAY_SIZE(skl_hda_map);
card->dapm_widgets = skl_hda_widgets;
card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
- } else {
- return -EINVAL;
+ if (!idisp_mask) {
+ for (i = 0; i < IDISP_DAI_COUNT; i++) {
+ skl_hda_be_dai_links[i].codecs = dummy_codec;
+ skl_hda_be_dai_links[i].num_codecs =
+ ARRAY_SIZE(dummy_codec);
+ }
+ }
}
card->num_links = num_links;
@@ -167,7 +182,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (!mach)
return -EINVAL;
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index e6de3b28d840..d7b8154c43a4 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
/*
* Headset buttons map to the google Reference headset.
@@ -182,7 +182,7 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
{
struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct skl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -200,7 +200,7 @@ static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
{
struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct skl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -218,7 +218,7 @@ static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
{
struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct skl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -236,7 +236,7 @@ static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
@@ -296,7 +296,7 @@ static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai,
@@ -660,7 +660,7 @@ static int skylake_audio_probe(struct platform_device *pdev)
skylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index c99c8b23e509..4b317bcf6ea0 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -161,12 +161,12 @@ static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd)
int ret;
/* Slot 1 for left */
- ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0x01, 0x01, 2, 48);
if (ret < 0)
return ret;
/* Slot 2 for right */
- ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48);
+ ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 1), 0x02, 0x02, 2, 48);
if (ret < 0)
return ret;
@@ -176,7 +176,7 @@ static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
{
int ret;
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
/*
* 4 buttons here map to the google Reference headset
@@ -201,7 +201,7 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
{
struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct skl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -219,7 +219,7 @@ static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
{
struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct skl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -238,7 +238,7 @@ static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
{
struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct skl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -256,7 +256,7 @@ static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
@@ -348,7 +348,7 @@ static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai,
@@ -686,6 +686,7 @@ static struct snd_soc_card skylake_audio_card = {
.codec_conf = ssm4567_codec_conf,
.num_configs = ARRAY_SIZE(ssm4567_codec_conf),
.fully_routed = true,
+ .disable_route_checks = true,
.late_probe = skylake_card_late_probe,
};
@@ -703,7 +704,7 @@ static int skylake_audio_probe(struct platform_device *pdev)
skylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index a9aec66a2351..903ae1b28ec9 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -112,7 +112,7 @@ static const struct snd_soc_dapm_route skylake_rt286_map[] = {
static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm;
- struct snd_soc_component *component = rtd->cpu_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
dapm = snd_soc_component_get_dapm(component);
snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
@@ -122,7 +122,7 @@ static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
int ret;
ret = snd_soc_card_jack_new(rtd->card, "Headset",
@@ -143,7 +143,7 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct skl_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -229,7 +229,7 @@ static int skylake_rt286_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c
index 8f44f13d2848..b707dd3b5625 100644
--- a/sound/soc/intel/boards/sof_da7219_max98373.c
+++ b/sound/soc/intel/boards/sof_da7219_max98373.c
@@ -2,7 +2,7 @@
// Copyright(c) 2019 Intel Corporation.
/*
- * Intel SOF Machine driver for DA7219 + MAX98373 codec
+ * Intel SOF Machine driver for DA7219 + MAX98373/MAX98360A codec
*/
#include <linux/input.h>
@@ -69,11 +69,20 @@ static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Right Spk"),
};
+static const struct snd_kcontrol_new m98360a_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Spk"),
+};
+
+/* For MAX98373 amp */
static const struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
SND_SOC_DAPM_SPK("Left Spk", NULL),
SND_SOC_DAPM_SPK("Right Spk", NULL),
+
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_POST_PMD |
SND_SOC_DAPM_PRE_PMU),
@@ -83,21 +92,45 @@ static const struct snd_soc_dapm_route audio_map[] = {
{ "Headphone Jack", NULL, "HPL" },
{ "Headphone Jack", NULL, "HPR" },
+ { "MIC", NULL, "Headset Mic" },
+
+ { "Headphone Jack", NULL, "Platform Clock" },
+ { "Headset Mic", NULL, "Platform Clock" },
+
{ "Left Spk", NULL, "Left BE_OUT" },
{ "Right Spk", NULL, "Right BE_OUT" },
+};
+
+/* For MAX98360A amp */
+static const struct snd_soc_dapm_widget max98360a_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
+ SND_SOC_DAPM_SPK("Spk", NULL),
+
+ SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+ platform_clock_control, SND_SOC_DAPM_POST_PMD |
+ SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_route max98360a_map[] = {
+ { "Headphone Jack", NULL, "HPL" },
+ { "Headphone Jack", NULL, "HPR" },
{ "MIC", NULL, "Headset Mic" },
{ "Headphone Jack", NULL, "Platform Clock" },
{ "Headset Mic", NULL, "Platform Clock" },
+
+ {"Spk", NULL, "Speaker"},
};
static struct snd_soc_jack headset;
static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_component *component = codec_dai->component;
struct snd_soc_jack *jack;
int ret;
@@ -140,7 +173,7 @@ static int ssp1_hw_params(struct snd_pcm_substream *substream,
int ret, j;
for (j = 0; j < runtime->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, j);
if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
/* vmon_slot_no = 0 imon_slot_no = 1 for TX slots */
@@ -181,7 +214,7 @@ static struct snd_soc_codec_conf max98373_codec_conf[] = {
static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -224,6 +257,9 @@ SND_SOC_DAILINK_DEF(ssp1_amps,
/* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI),
/* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI)));
+SND_SOC_DAILINK_DEF(ssp1_m98360a,
+ DAILINK_COMP_ARRAY(COMP_CODEC("MX98360A:00", "HiFi")));
+
SND_SOC_DAILINK_DEF(dmic_pin,
DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
SND_SOC_DAILINK_DEF(dmic_codec,
@@ -320,6 +356,21 @@ static struct snd_soc_card card_da7219_m98373 = {
.late_probe = card_late_probe,
};
+static struct snd_soc_card card_da7219_m98360a = {
+ .name = "da7219max98360a",
+ .owner = THIS_MODULE,
+ .dai_link = dais,
+ .num_links = ARRAY_SIZE(dais),
+ .controls = m98360a_controls,
+ .num_controls = ARRAY_SIZE(m98360a_controls),
+ .dapm_widgets = max98360a_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(max98360a_widgets),
+ .dapm_routes = max98360a_map,
+ .num_dapm_routes = ARRAY_SIZE(max98360a_map),
+ .fully_routed = true,
+ .late_probe = card_late_probe,
+};
+
static int audio_probe(struct platform_device *pdev)
{
static struct snd_soc_card *card;
@@ -327,15 +378,26 @@ static int audio_probe(struct platform_device *pdev)
struct card_private *ctx;
int ret;
- ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
if (!ctx)
return -ENOMEM;
+ /* By default dais[0] is configured for max98373 */
+ if (!strcmp(pdev->name, "sof_da7219_max98360a")) {
+ dais[0] = (struct snd_soc_dai_link) {
+ .name = "SSP1-Codec",
+ .id = 0,
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .ignore_pmdown_time = 1,
+ SND_SOC_DAILINK_REG(ssp1_pin, ssp1_m98360a, platform) };
+ }
+
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
card = (struct snd_soc_card *)pdev->id_entry->driver_data;
card->dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(card,
mach->mach_params.platform);
if (ret)
@@ -351,13 +413,17 @@ static const struct platform_device_id board_ids[] = {
.name = "sof_da7219_max98373",
.driver_data = (kernel_ulong_t)&card_da7219_m98373,
},
+ {
+ .name = "sof_da7219_max98360a",
+ .driver_data = (kernel_ulong_t)&card_da7219_m98360a,
+ },
{ }
};
static struct platform_driver audio = {
.probe = audio_probe,
.driver = {
- .name = "sof_da7219_max98373",
+ .name = "sof_da7219_max98_360a_373",
.pm = &snd_soc_pm_ops,
},
.id_table = board_ids,
@@ -368,4 +434,5 @@ module_platform_driver(audio)
MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver");
MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>");
MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_da7219_max98360a");
MODULE_ALIAS("platform:sof_da7219_max98373");
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c
new file mode 100644
index 000000000000..463b39a7ccfd
--- /dev/null
+++ b/sound/soc/intel/boards/sof_maxim_common.c
@@ -0,0 +1,80 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright(c) 2020 Intel Corporation. All rights reserved.
+#include <linux/string.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+#include <uapi/sound/asound.h>
+#include "sof_maxim_common.h"
+
+static const struct snd_soc_dapm_route max_98373_dapm_routes[] = {
+ /* speaker */
+ { "Left Spk", NULL, "Left BE_OUT" },
+ { "Right Spk", NULL, "Right BE_OUT" },
+};
+
+static struct snd_soc_codec_conf max_98373_codec_conf[] = {
+ {
+ .dlc = COMP_CODEC_CONF(MAX_98373_DEV0_NAME),
+ .name_prefix = "Right",
+ },
+ {
+ .dlc = COMP_CODEC_CONF(MAX_98373_DEV1_NAME),
+ .name_prefix = "Left",
+ },
+};
+
+struct snd_soc_dai_link_component max_98373_components[] = {
+ { /* For Left */
+ .name = MAX_98373_DEV0_NAME,
+ .dai_name = MAX_98373_CODEC_DAI,
+ },
+ { /* For Right */
+ .name = MAX_98373_DEV1_NAME,
+ .dai_name = MAX_98373_CODEC_DAI,
+ },
+};
+
+static int max98373_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
+ int j;
+
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
+ if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) {
+ /* DEV0 tdm slot configuration */
+ snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+ }
+ if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) {
+ /* DEV1 tdm slot configuration */
+ snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+ }
+ }
+ return 0;
+}
+
+struct snd_soc_ops max_98373_ops = {
+ .hw_params = max98373_hw_params,
+};
+
+int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes,
+ ARRAY_SIZE(max_98373_dapm_routes));
+ if (ret)
+ dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret);
+ return ret;
+}
+
+void sof_max98373_codec_conf(struct snd_soc_card *card)
+{
+ card->codec_conf = max_98373_codec_conf;
+ card->num_configs = ARRAY_SIZE(max_98373_codec_conf);
+}
diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h
new file mode 100644
index 000000000000..406bf0e81155
--- /dev/null
+++ b/sound/soc/intel/boards/sof_maxim_common.h
@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2020 Intel Corporation.
+ */
+
+/*
+ * This file defines data structures used in Machine Driver for Intel
+ * platforms with Maxim Codecs.
+ */
+#ifndef __SOF_MAXIM_COMMON_H
+#define __SOF_MAXIM_COMMON_H
+
+#include <sound/soc.h>
+
+#define MAX_98373_CODEC_DAI "max98373-aif1"
+#define MAX_98373_DEV0_NAME "i2c-MX98373:00"
+#define MAX_98373_DEV1_NAME "i2c-MX98373:01"
+
+extern struct snd_soc_dai_link_component max_98373_components[2];
+extern struct snd_soc_ops max_98373_ops;
+
+int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd);
+void sof_max98373_codec_conf(struct snd_soc_card *card);
+#endif /* __SOF_MAXIM_COMMON_H */
diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c
new file mode 100644
index 000000000000..fb7811899999
--- /dev/null
+++ b/sound/soc/intel/boards/sof_pcm512x.c
@@ -0,0 +1,448 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018-2020 Intel Corporation.
+
+/*
+ * Intel SOF Machine Driver for Intel platforms with TI PCM512x codec,
+ * e.g. Up or Up2 with Hifiberry DAC+ HAT
+ */
+#include <linux/clk.h>
+#include <linux/dmi.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/types.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "../../codecs/pcm512x.h"
+#include "../common/soc-intel-quirks.h"
+#include "hda_dsp_common.h"
+
+#define NAME_SIZE 32
+
+#define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0))
+#define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0))
+
+#define IDISP_CODEC_MASK 0x4
+
+/* Default: SSP5 */
+static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5);
+
+static bool is_legacy_cpu;
+
+struct sof_hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+struct sof_card_private {
+ struct list_head hdmi_pcm_list;
+ bool idisp_codec;
+};
+
+static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id)
+{
+ sof_pcm512x_quirk = (unsigned long)id->driver_data;
+ return 1;
+}
+
+static const struct dmi_system_id sof_pcm512x_quirk_table[] = {
+ {
+ .callback = sof_pcm512x_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "AAEON"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "UP-CHT01"),
+ },
+ .driver_data = (void *)(SOF_PCM512X_SSP_CODEC(2)),
+ },
+ {}
+};
+
+static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
+ struct sof_hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ /* dai_link id is 1:1 mapped to the PCM device */
+ pcm->device = rtd->dai_link->id;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
+}
+
+static int sof_pcm512x_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
+
+ snd_soc_component_update_bits(codec, PCM512x_GPIO_EN, 0x08, 0x08);
+ snd_soc_component_update_bits(codec, PCM512x_GPIO_OUTPUT_4, 0x0f, 0x02);
+ snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1,
+ 0x08, 0x08);
+
+ return 0;
+}
+
+static int aif1_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
+
+ snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1,
+ 0x08, 0x08);
+
+ return 0;
+}
+
+static void aif1_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
+
+ snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1,
+ 0x08, 0x00);
+}
+
+static const struct snd_soc_ops sof_pcm512x_ops = {
+ .startup = aif1_startup,
+ .shutdown = aif1_shutdown,
+};
+
+static struct snd_soc_dai_link_component platform_component[] = {
+ {
+ /* name might be overridden during probe */
+ .name = "0000:00:1f.3"
+ }
+};
+
+#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI)
+static int sof_card_late_probe(struct snd_soc_card *card)
+{
+ struct sof_card_private *ctx = snd_soc_card_get_drvdata(card);
+ struct sof_hdmi_pcm *pcm;
+
+ /* HDMI is not supported by SOF on Baytrail/CherryTrail */
+ if (is_legacy_cpu)
+ return 0;
+
+ if (list_empty(&ctx->hdmi_pcm_list))
+ return -EINVAL;
+
+ if (!ctx->idisp_codec)
+ return 0;
+
+ pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head);
+
+ return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component);
+}
+#else
+static int sof_card_late_probe(struct snd_soc_card *card)
+{
+ return 0;
+}
+#endif
+
+static const struct snd_kcontrol_new sof_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget sof_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_route sof_map[] = {
+ /* Speaker */
+ {"Ext Spk", NULL, "OUTR"},
+ {"Ext Spk", NULL, "OUTL"},
+};
+
+static const struct snd_soc_dapm_route dmic_map[] = {
+ /* digital mics */
+ {"DMic", NULL, "SoC DMIC"},
+};
+
+static int dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+ ARRAY_SIZE(dmic_widgets));
+ if (ret) {
+ dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+ /* Don't need to add routes if widget addition failed */
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+ ARRAY_SIZE(dmic_map));
+
+ if (ret)
+ dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+ return ret;
+}
+
+/* sof audio machine driver for pcm512x codec */
+static struct snd_soc_card sof_audio_card_pcm512x = {
+ .name = "pcm512x",
+ .owner = THIS_MODULE,
+ .controls = sof_controls,
+ .num_controls = ARRAY_SIZE(sof_controls),
+ .dapm_widgets = sof_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sof_widgets),
+ .dapm_routes = sof_map,
+ .num_dapm_routes = ARRAY_SIZE(sof_map),
+ .fully_routed = true,
+ .late_probe = sof_card_late_probe,
+};
+
+SND_SOC_DAILINK_DEF(pcm512x_component,
+ DAILINK_COMP_ARRAY(COMP_CODEC("i2c-104C5122:00", "pcm512x-hifi")));
+SND_SOC_DAILINK_DEF(dmic_component,
+ DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi")));
+
+static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
+ int ssp_codec,
+ int dmic_be_num,
+ int hdmi_num,
+ bool idisp_codec)
+{
+ struct snd_soc_dai_link_component *idisp_components;
+ struct snd_soc_dai_link_component *cpus;
+ struct snd_soc_dai_link *links;
+ int i, id = 0;
+
+ links = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links,
+ sizeof(struct snd_soc_dai_link), GFP_KERNEL);
+ cpus = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links,
+ sizeof(struct snd_soc_dai_link_component), GFP_KERNEL);
+ if (!links || !cpus)
+ goto devm_err;
+
+ /* codec SSP */
+ links[id].name = devm_kasprintf(dev, GFP_KERNEL,
+ "SSP%d-Codec", ssp_codec);
+ if (!links[id].name)
+ goto devm_err;
+
+ links[id].id = id;
+ links[id].codecs = pcm512x_component;
+ links[id].num_codecs = ARRAY_SIZE(pcm512x_component);
+ links[id].platforms = platform_component;
+ links[id].num_platforms = ARRAY_SIZE(platform_component);
+ links[id].init = sof_pcm512x_codec_init;
+ links[id].ops = &sof_pcm512x_ops;
+ links[id].nonatomic = true;
+ links[id].dpcm_playback = 1;
+ /*
+ * capture only supported with specific versions of the Hifiberry DAC+
+ * links[id].dpcm_capture = 1;
+ */
+ links[id].no_pcm = 1;
+ links[id].cpus = &cpus[id];
+ links[id].num_cpus = 1;
+ if (is_legacy_cpu) {
+ links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
+ "ssp%d-port",
+ ssp_codec);
+ if (!links[id].cpus->dai_name)
+ goto devm_err;
+ } else {
+ links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
+ "SSP%d Pin",
+ ssp_codec);
+ if (!links[id].cpus->dai_name)
+ goto devm_err;
+ }
+ id++;
+
+ /* dmic */
+ if (dmic_be_num > 0) {
+ /* at least we have dmic01 */
+ links[id].name = "dmic01";
+ links[id].cpus = &cpus[id];
+ links[id].cpus->dai_name = "DMIC01 Pin";
+ links[id].init = dmic_init;
+ if (dmic_be_num > 1) {
+ /* set up 2 BE links at most */
+ links[id + 1].name = "dmic16k";
+ links[id + 1].cpus = &cpus[id + 1];
+ links[id + 1].cpus->dai_name = "DMIC16k Pin";
+ dmic_be_num = 2;
+ }
+ }
+
+ for (i = 0; i < dmic_be_num; i++) {
+ links[id].id = id;
+ links[id].num_cpus = 1;
+ links[id].codecs = dmic_component;
+ links[id].num_codecs = ARRAY_SIZE(dmic_component);
+ links[id].platforms = platform_component;
+ links[id].num_platforms = ARRAY_SIZE(platform_component);
+ links[id].ignore_suspend = 1;
+ links[id].dpcm_capture = 1;
+ links[id].no_pcm = 1;
+ id++;
+ }
+
+ /* HDMI */
+ if (hdmi_num > 0) {
+ idisp_components = devm_kcalloc(dev, hdmi_num,
+ sizeof(struct snd_soc_dai_link_component),
+ GFP_KERNEL);
+ if (!idisp_components)
+ goto devm_err;
+ }
+ for (i = 1; i <= hdmi_num; i++) {
+ links[id].name = devm_kasprintf(dev, GFP_KERNEL,
+ "iDisp%d", i);
+ if (!links[id].name)
+ goto devm_err;
+
+ links[id].id = id;
+ links[id].cpus = &cpus[id];
+ links[id].num_cpus = 1;
+ links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
+ "iDisp%d Pin", i);
+ if (!links[id].cpus->dai_name)
+ goto devm_err;
+
+ /*
+ * topology cannot be loaded if codec is missing, so
+ * use the dummy codec if needed
+ */
+ if (idisp_codec) {
+ idisp_components[i - 1].name = "ehdaudio0D2";
+ idisp_components[i - 1].dai_name =
+ devm_kasprintf(dev, GFP_KERNEL,
+ "intel-hdmi-hifi%d", i);
+ } else {
+ idisp_components[i - 1].name = "snd-soc-dummy";
+ idisp_components[i - 1].dai_name = "snd-soc-dummy-dai";
+ }
+ if (!idisp_components[i - 1].dai_name)
+ goto devm_err;
+
+ links[id].codecs = &idisp_components[i - 1];
+ links[id].num_codecs = 1;
+ links[id].platforms = platform_component;
+ links[id].num_platforms = ARRAY_SIZE(platform_component);
+ links[id].init = sof_hdmi_init;
+ links[id].dpcm_playback = 1;
+ links[id].no_pcm = 1;
+ id++;
+ }
+
+ return links;
+devm_err:
+ return NULL;
+}
+
+static int sof_audio_probe(struct platform_device *pdev)
+{
+ struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
+ struct snd_soc_dai_link *dai_links;
+ struct sof_card_private *ctx;
+ int dmic_be_num, hdmi_num;
+ int ret, ssp_codec;
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ hdmi_num = 0;
+ if (soc_intel_is_byt() || soc_intel_is_cht()) {
+ is_legacy_cpu = true;
+ dmic_be_num = 0;
+ /* default quirk for legacy cpu */
+ sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(2);
+ } else {
+ dmic_be_num = 2;
+#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI)
+ if (mach->mach_params.common_hdmi_codec_drv &&
+ (mach->mach_params.codec_mask & IDISP_CODEC_MASK))
+ ctx->idisp_codec = true;
+
+ /* links are always present in topology */
+ hdmi_num = 3;
+#endif
+ }
+
+ dmi_check_system(sof_pcm512x_quirk_table);
+
+ dev_dbg(&pdev->dev, "sof_pcm512x_quirk = %lx\n", sof_pcm512x_quirk);
+
+ ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK;
+
+ /* compute number of dai links */
+ sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num;
+
+ dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec,
+ dmic_be_num, hdmi_num,
+ ctx->idisp_codec);
+ if (!dai_links)
+ return -ENOMEM;
+
+ sof_audio_card_pcm512x.dai_link = dai_links;
+
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+ sof_audio_card_pcm512x.dev = &pdev->dev;
+
+ /* set platform name for each dailink */
+ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x,
+ mach->mach_params.platform);
+ if (ret)
+ return ret;
+
+ snd_soc_card_set_drvdata(&sof_audio_card_pcm512x, ctx);
+
+ return devm_snd_soc_register_card(&pdev->dev,
+ &sof_audio_card_pcm512x);
+}
+
+static int sof_pcm512x_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_component *component = NULL;
+
+ for_each_card_components(card, component) {
+ if (!strcmp(component->name, pcm512x_component[0].name)) {
+ snd_soc_component_set_jack(component, NULL, NULL);
+ break;
+ }
+ }
+
+ return 0;
+}
+
+static struct platform_driver sof_audio = {
+ .probe = sof_audio_probe,
+ .remove = sof_pcm512x_remove,
+ .driver = {
+ .name = "sof_pcm512x",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+module_platform_driver(sof_audio)
+
+MODULE_DESCRIPTION("ASoC Intel(R) SOF + PCM512x Machine driver");
+MODULE_AUTHOR("Pierre-Louis Bossart");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_pcm512x");
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 5d878873a8e0..8c29431b5847 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -1,9 +1,9 @@
// SPDX-License-Identifier: GPL-2.0
-// Copyright(c) 2019 Intel Corporation.
+// Copyright(c) 2019-2020 Intel Corporation.
/*
* Intel SOF Machine Driver with Realtek rt5682 Codec
- * and speaker codec MAX98357A
+ * and speaker codec MAX98357A or RT1015.
*/
#include <linux/i2c.h>
#include <linux/input.h>
@@ -18,10 +18,12 @@
#include <sound/soc.h>
#include <sound/rt5682.h>
#include <sound/soc-acpi.h>
+#include "../../codecs/rt1015.h"
#include "../../codecs/rt5682.h"
#include "../../codecs/hdac_hdmi.h"
#include "../common/soc-intel-quirks.h"
#include "hda_dsp_common.h"
+#include "sof_maxim_common.h"
#define NAME_SIZE 32
@@ -39,6 +41,8 @@
#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10))
#define SOF_RT5682_NUM_HDMIDEV(quirk) \
((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK)
+#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13)
+#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(14)
/* Default: MCLK on, MCLK 19.2M, SSP0 */
static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
@@ -120,7 +124,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = {
static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct sof_hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
@@ -139,7 +143,7 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd)
static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_jack *jack;
int ret;
@@ -207,7 +211,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int clk_id, clk_freq, pll_out, ret;
if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) {
@@ -260,6 +264,42 @@ static struct snd_soc_ops sof_rt5682_ops = {
.hw_params = sof_rt5682_hw_params,
};
+static int sof_rt1015_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *codec_dai;
+ int i, ret;
+
+ if (!snd_soc_card_get_codec_dai(card, "rt1015-aif"))
+ return 0;
+
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK,
+ params_rate(params) * 50,
+ params_rate(params) * 256);
+ if (ret < 0) {
+ dev_err(card->dev, "failed to set pll\n");
+ return ret;
+ }
+ /* Configure sysclk for codec */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL,
+ params_rate(params) * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(card->dev, "failed to set sysclk\n");
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops sof_rt1015_ops = {
+ .hw_params = sof_rt1015_hw_params,
+};
+
static struct snd_soc_dai_link_component platform_component[] = {
{
/* name might be overridden during probe */
@@ -316,12 +356,17 @@ static const struct snd_kcontrol_new sof_controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Spk"),
+ SOC_DAPM_PIN_SWITCH("Left Spk"),
+ SOC_DAPM_PIN_SWITCH("Right Spk"),
+
};
static const struct snd_soc_dapm_widget sof_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_SPK("Spk", NULL),
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
};
static const struct snd_soc_dapm_widget dmic_widgets[] = {
@@ -342,11 +387,22 @@ static const struct snd_soc_dapm_route speaker_map[] = {
{ "Spk", NULL, "Speaker" },
};
+static const struct snd_soc_dapm_route speaker_map_lr[] = {
+ { "Left Spk", NULL, "Left SPO" },
+ { "Right Spk", NULL, "Right SPO" },
+};
+
static const struct snd_soc_dapm_route dmic_map[] = {
/* digital mics */
{"DMic", NULL, "SoC DMIC"},
};
+static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd)
+{
+ return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr,
+ ARRAY_SIZE(speaker_map_lr));
+}
+
static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
@@ -382,6 +438,17 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
+static struct snd_soc_codec_conf rt1015_amp_conf[] = {
+ {
+ .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"),
+ .name_prefix = "Left",
+ },
+ {
+ .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"),
+ .name_prefix = "Right",
+ },
+};
+
/* sof audio machine driver for rt5682 codec */
static struct snd_soc_card sof_audio_card_rt5682 = {
.name = "rt5682", /* the sof- prefix is added by the core */
@@ -417,6 +484,17 @@ static struct snd_soc_dai_link_component max98357a_component[] = {
}
};
+static struct snd_soc_dai_link_component rt1015_components[] = {
+ {
+ .name = "i2c-10EC1015:00",
+ .dai_name = "rt1015-aif",
+ },
+ {
+ .name = "i2c-10EC1015:01",
+ .dai_name = "rt1015-aif",
+ },
+};
+
static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
int ssp_codec,
int ssp_amp,
@@ -556,11 +634,24 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
goto devm_err;
links[id].id = id;
- links[id].codecs = max98357a_component;
- links[id].num_codecs = ARRAY_SIZE(max98357a_component);
+ if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) {
+ links[id].codecs = rt1015_components;
+ links[id].num_codecs = ARRAY_SIZE(rt1015_components);
+ links[id].init = speaker_codec_init_lr;
+ links[id].ops = &sof_rt1015_ops;
+ } else if (sof_rt5682_quirk &
+ SOF_MAX98373_SPEAKER_AMP_PRESENT) {
+ links[id].codecs = max_98373_components;
+ links[id].num_codecs = ARRAY_SIZE(max_98373_components);
+ links[id].init = max98373_spk_codec_init;
+ links[id].ops = &max_98373_ops;
+ } else {
+ links[id].codecs = max98357a_component;
+ links[id].num_codecs = ARRAY_SIZE(max98357a_component);
+ links[id].init = speaker_codec_init;
+ }
links[id].platforms = platform_component;
links[id].num_platforms = ARRAY_SIZE(platform_component);
- links[id].init = speaker_codec_init,
links[id].nonatomic = true;
links[id].dpcm_playback = 1;
links[id].no_pcm = 1;
@@ -604,7 +695,7 @@ static int sof_audio_probe(struct platform_device *pdev)
dmi_check_system(sof_rt5682_quirk_table);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
/* A speaker amp might not be present when the quirk claims one is.
* Detect this via whether the machine driver match includes quirk_data.
@@ -662,6 +753,9 @@ static int sof_audio_probe(struct platform_device *pdev)
if (sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT)
sof_audio_card_rt5682.num_links++;
+ if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT)
+ sof_max98373_codec_conf(&sof_audio_card_rt5682);
+
dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp,
dmic_be_num, hdmi_num);
if (!dai_links)
@@ -669,6 +763,11 @@ static int sof_audio_probe(struct platform_device *pdev)
sof_audio_card_rt5682.dai_link = dai_links;
+ if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) {
+ sof_audio_card_rt5682.codec_conf = rt1015_amp_conf;
+ sof_audio_card_rt5682.num_configs = ARRAY_SIZE(rt1015_amp_conf);
+ }
+
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
sof_audio_card_rt5682.dev = &pdev->dev;
@@ -714,6 +813,24 @@ static const struct platform_device_id board_ids[] = {
SOF_RT5682_SSP_AMP(1) |
SOF_RT5682_NUM_HDMIDEV(4)),
},
+ {
+ .name = "jsl_rt5682_rt1015",
+ .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+ SOF_RT5682_MCLK_24MHZ |
+ SOF_RT5682_SSP_CODEC(0) |
+ SOF_SPEAKER_AMP_PRESENT |
+ SOF_RT1015_SPEAKER_AMP_PRESENT |
+ SOF_RT5682_SSP_AMP(1)),
+ },
+ {
+ .name = "tgl_max98373_rt5682",
+ .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+ SOF_RT5682_SSP_CODEC(0) |
+ SOF_SPEAKER_AMP_PRESENT |
+ SOF_MAX98373_SPEAKER_AMP_PRESENT |
+ SOF_RT5682_SSP_AMP(1) |
+ SOF_RT5682_NUM_HDMIDEV(4)),
+ },
{ }
};
@@ -735,3 +852,5 @@ MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:sof_rt5682");
MODULE_ALIAS("platform:tgl_max98357a_rt5682");
+MODULE_ALIAS("platform:jsl_rt5682_rt1015");
+MODULE_ALIAS("platform:tgl_max98373_rt5682");
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
new file mode 100644
index 000000000000..a64dc563b47e
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -0,0 +1,962 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw - ASOC Machine driver for Intel SoundWire platforms
+ */
+
+#include <linux/device.h>
+#include <linux/dmi.h>
+#include <linux/module.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+
+unsigned long sof_sdw_quirk = SOF_RT711_JD_SRC_JD1;
+
+#define INC_ID(BE, CPU, LINK) do { (BE)++; (CPU)++; (LINK)++; } while (0)
+
+static int sof_sdw_quirk_cb(const struct dmi_system_id *id)
+{
+ sof_sdw_quirk = (unsigned long)id->driver_data;
+ return 1;
+}
+
+static const struct dmi_system_id sof_sdw_quirk_table[] = {
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "09C6")
+ },
+ .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+ SOF_RT715_DAI_ID_FIX),
+ },
+ {
+ /* early version of SKU 09C6 */
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0983")
+ },
+ .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+ SOF_RT715_DAI_ID_FIX),
+ },
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"),
+ },
+ .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+ SOF_RT715_DAI_ID_FIX |
+ SOF_SDW_FOUR_SPK),
+ },
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"),
+ },
+ .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+ SOF_RT715_DAI_ID_FIX |
+ SOF_SDW_FOUR_SPK),
+ },
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
+ DMI_MATCH(DMI_PRODUCT_NAME,
+ "Tiger Lake Client Platform"),
+ },
+ .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 |
+ SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC |
+ SOF_SSP_PORT(SOF_I2S_SSP2)),
+ },
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"),
+ },
+ .driver_data = (void *)SOF_SDW_PCH_DMIC,
+ },
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "CometLake Client"),
+ },
+ .driver_data = (void *)SOF_SDW_PCH_DMIC,
+ },
+ {
+ .callback = sof_sdw_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Google"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Volteer"),
+ },
+ .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC),
+ },
+
+ {}
+};
+
+static struct snd_soc_codec_conf codec_conf[] = {
+ {
+ .dlc = COMP_CODEC_CONF("sdw:0:25d:711:0"),
+ .name_prefix = "rt711",
+ },
+ /* rt1308 w/ I2S connection */
+ {
+ .dlc = COMP_CODEC_CONF("i2c-10EC1308:00"),
+ .name_prefix = "rt1308-1",
+ },
+ /* rt1308 left on link 1 */
+ {
+ .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0"),
+ .name_prefix = "rt1308-1",
+ },
+ /* two 1308s on link1 with different unique id */
+ {
+ .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:0"),
+ .name_prefix = "rt1308-1",
+ },
+ {
+ .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:2"),
+ .name_prefix = "rt1308-2",
+ },
+ /* rt1308 right on link 2 */
+ {
+ .dlc = COMP_CODEC_CONF("sdw:2:25d:1308:0"),
+ .name_prefix = "rt1308-2",
+ },
+ {
+ .dlc = COMP_CODEC_CONF("sdw:3:25d:715:0"),
+ .name_prefix = "rt715",
+ },
+ {
+ .dlc = COMP_CODEC_CONF("sdw:0:25d:5682:0"),
+ .name_prefix = "rt5682",
+ },
+};
+
+static struct snd_soc_dai_link_component dmic_component[] = {
+ {
+ .name = "dmic-codec",
+ .dai_name = "dmic-hifi",
+ }
+};
+
+static struct snd_soc_dai_link_component platform_component[] = {
+ {
+ /* name might be overridden during probe */
+ .name = "0000:00:1f.3"
+ }
+};
+
+/* these wrappers are only needed to avoid typecast compilation errors */
+static int sdw_startup(struct snd_pcm_substream *substream)
+{
+ return sdw_startup_stream(substream);
+}
+
+static void sdw_shutdown(struct snd_pcm_substream *substream)
+{
+ sdw_shutdown_stream(substream);
+}
+
+static const struct snd_soc_ops sdw_ops = {
+ .startup = sdw_startup,
+ .shutdown = sdw_shutdown,
+};
+
+static struct sof_sdw_codec_info codec_info_list[] = {
+ {
+ .id = 0x700,
+ .direction = {true, true},
+ .dai_name = "rt700-aif1",
+ .init = sof_sdw_rt700_init,
+ },
+ {
+ .id = 0x711,
+ .direction = {true, true},
+ .dai_name = "rt711-aif1",
+ .init = sof_sdw_rt711_init,
+ },
+ {
+ .id = 0x1308,
+ .acpi_id = "10EC1308",
+ .direction = {true, false},
+ .dai_name = "rt1308-aif",
+ .ops = &sof_sdw_rt1308_i2s_ops,
+ .init = sof_sdw_rt1308_init,
+ },
+ {
+ .id = 0x715,
+ .direction = {false, true},
+ .dai_name = "rt715-aif2",
+ .init = sof_sdw_rt715_init,
+ },
+ {
+ .id = 0x5682,
+ .direction = {true, true},
+ .dai_name = "rt5682-sdw",
+ .init = sof_sdw_rt5682_init,
+ },
+};
+
+static inline int find_codec_info_part(unsigned int part_id)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
+ if (part_id == codec_info_list[i].id)
+ break;
+
+ if (i == ARRAY_SIZE(codec_info_list))
+ return -EINVAL;
+
+ return i;
+}
+
+static inline int find_codec_info_acpi(const u8 *acpi_id)
+{
+ int i;
+
+ if (!acpi_id[0])
+ return -EINVAL;
+
+ for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
+ if (!memcmp(codec_info_list[i].acpi_id, acpi_id,
+ ACPI_ID_LEN))
+ break;
+
+ if (i == ARRAY_SIZE(codec_info_list))
+ return -EINVAL;
+
+ return i;
+}
+
+/*
+ * get BE dailink number and CPU DAI number based on sdw link adr.
+ * Since some sdw slaves may be aggregated, the CPU DAI number
+ * may be larger than the number of BE dailinks.
+ */
+static int get_sdw_dailink_info(const struct snd_soc_acpi_link_adr *links,
+ int *sdw_be_num, int *sdw_cpu_dai_num)
+{
+ const struct snd_soc_acpi_link_adr *link;
+ bool group_visited[SDW_MAX_GROUPS];
+ bool no_aggregation;
+ int i;
+
+ no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION;
+ *sdw_cpu_dai_num = 0;
+ *sdw_be_num = 0;
+
+ if (!links)
+ return -EINVAL;
+
+ for (i = 0; i < SDW_MAX_GROUPS; i++)
+ group_visited[i] = false;
+
+ for (link = links; link->num_adr; link++) {
+ const struct snd_soc_acpi_endpoint *endpoint;
+ int part_id, codec_index;
+ int stream;
+ u64 adr;
+
+ adr = link->adr_d->adr;
+ part_id = SDW_PART_ID(adr);
+ codec_index = find_codec_info_part(part_id);
+ if (codec_index < 0)
+ return codec_index;
+
+ endpoint = link->adr_d->endpoints;
+
+ /* count DAI number for playback and capture */
+ for_each_pcm_streams(stream) {
+ if (!codec_info_list[codec_index].direction[stream])
+ continue;
+
+ (*sdw_cpu_dai_num)++;
+
+ /* count BE for each non-aggregated slave or group */
+ if (!endpoint->aggregated || no_aggregation ||
+ !group_visited[endpoint->group_id])
+ (*sdw_be_num)++;
+ }
+
+ if (endpoint->aggregated)
+ group_visited[endpoint->group_id] = true;
+ }
+
+ return 0;
+}
+
+static void init_dai_link(struct snd_soc_dai_link *dai_links, int be_id,
+ char *name, int playback, int capture,
+ struct snd_soc_dai_link_component *cpus,
+ int cpus_num,
+ struct snd_soc_dai_link_component *codecs,
+ int codecs_num,
+ int (*init)(struct snd_soc_pcm_runtime *rtd),
+ const struct snd_soc_ops *ops)
+{
+ dai_links->id = be_id;
+ dai_links->name = name;
+ dai_links->platforms = platform_component;
+ dai_links->num_platforms = ARRAY_SIZE(platform_component);
+ dai_links->nonatomic = true;
+ dai_links->no_pcm = 1;
+ dai_links->cpus = cpus;
+ dai_links->num_cpus = cpus_num;
+ dai_links->codecs = codecs;
+ dai_links->num_codecs = codecs_num;
+ dai_links->dpcm_playback = playback;
+ dai_links->dpcm_capture = capture;
+ dai_links->init = init;
+ dai_links->ops = ops;
+}
+
+static bool is_unique_device(const struct snd_soc_acpi_link_adr *link,
+ unsigned int sdw_version,
+ unsigned int mfg_id,
+ unsigned int part_id,
+ unsigned int class_id,
+ int index_in_link
+ )
+{
+ int i;
+
+ for (i = 0; i < link->num_adr; i++) {
+ unsigned int sdw1_version, mfg1_id, part1_id, class1_id;
+ u64 adr;
+
+ /* skip itself */
+ if (i == index_in_link)
+ continue;
+
+ adr = link->adr_d[i].adr;
+
+ sdw1_version = SDW_VERSION(adr);
+ mfg1_id = SDW_MFG_ID(adr);
+ part1_id = SDW_PART_ID(adr);
+ class1_id = SDW_CLASS_ID(adr);
+
+ if (sdw_version == sdw1_version &&
+ mfg_id == mfg1_id &&
+ part_id == part1_id &&
+ class_id == class1_id)
+ return false;
+ }
+
+ return true;
+}
+
+static int create_codec_dai_name(struct device *dev,
+ const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link_component *codec,
+ int offset)
+{
+ int i;
+
+ for (i = 0; i < link->num_adr; i++) {
+ unsigned int sdw_version, unique_id, mfg_id;
+ unsigned int link_id, part_id, class_id;
+ int codec_index, comp_index;
+ char *codec_str;
+ u64 adr;
+
+ adr = link->adr_d[i].adr;
+
+ sdw_version = SDW_VERSION(adr);
+ link_id = SDW_DISCO_LINK_ID(adr);
+ unique_id = SDW_UNIQUE_ID(adr);
+ mfg_id = SDW_MFG_ID(adr);
+ part_id = SDW_PART_ID(adr);
+ class_id = SDW_CLASS_ID(adr);
+
+ comp_index = i + offset;
+ if (is_unique_device(link, sdw_version, mfg_id, part_id,
+ class_id, i)) {
+ codec_str = "sdw:%x:%x:%x:%x";
+ codec[comp_index].name =
+ devm_kasprintf(dev, GFP_KERNEL, codec_str,
+ link_id, mfg_id, part_id,
+ class_id);
+ } else {
+ codec_str = "sdw:%x:%x:%x:%x:%x";
+ codec[comp_index].name =
+ devm_kasprintf(dev, GFP_KERNEL, codec_str,
+ link_id, mfg_id, part_id,
+ class_id, unique_id);
+ }
+
+ if (!codec[comp_index].name)
+ return -ENOMEM;
+
+ codec_index = find_codec_info_part(part_id);
+ if (codec_index < 0)
+ return codec_index;
+
+ codec[comp_index].dai_name =
+ codec_info_list[codec_index].dai_name;
+ }
+
+ return 0;
+}
+
+static int set_codec_init_func(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ bool playback)
+{
+ int i;
+
+ for (i = 0; i < link->num_adr; i++) {
+ unsigned int part_id;
+ int codec_index;
+
+ part_id = SDW_PART_ID(link->adr_d[i].adr);
+ codec_index = find_codec_info_part(part_id);
+
+ if (codec_index < 0)
+ return codec_index;
+
+ if (codec_info_list[codec_index].init)
+ codec_info_list[codec_index].init(link, dai_links,
+ &codec_info_list[codec_index],
+ playback);
+ }
+
+ return 0;
+}
+
+/*
+ * check endpoint status in slaves and gather link ID for all slaves in
+ * the same group to generate different CPU DAI. Now only support
+ * one sdw link with all slaves set with only single group id.
+ *
+ * one slave on one sdw link with aggregated = 0
+ * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI
+ *
+ * two or more slaves on one sdw link with aggregated = 0
+ * one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs
+ *
+ * multiple links with multiple slaves with aggregated = 1
+ * one sdw BE DAI <---> 1 .. N CPU DAIs <----> 1 .. N codec DAIs
+ */
+static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link,
+ struct device *dev, int *cpu_dai_id, int *cpu_dai_num,
+ int *codec_num, int *group_id,
+ bool *group_generated)
+{
+ const struct snd_soc_acpi_adr_device *adr_d;
+ const struct snd_soc_acpi_link_adr *adr_next;
+ bool no_aggregation;
+ int index = 0;
+
+ no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION;
+ *codec_num = adr_link->num_adr;
+ adr_d = adr_link->adr_d;
+
+ /* make sure the link mask has a single bit set */
+ if (!is_power_of_2(adr_link->mask))
+ return -EINVAL;
+
+ cpu_dai_id[index++] = ffs(adr_link->mask) - 1;
+ if (!adr_d->endpoints->aggregated || no_aggregation) {
+ *cpu_dai_num = 1;
+ *group_id = 0;
+ return 0;
+ }
+
+ *group_id = adr_d->endpoints->group_id;
+
+ /* gather other link ID of slaves in the same group */
+ for (adr_next = adr_link + 1; adr_next && adr_next->num_adr;
+ adr_next++) {
+ const struct snd_soc_acpi_endpoint *endpoint;
+
+ endpoint = adr_next->adr_d->endpoints;
+ if (!endpoint->aggregated ||
+ endpoint->group_id != *group_id)
+ continue;
+
+ /* make sure the link mask has a single bit set */
+ if (!is_power_of_2(adr_next->mask))
+ return -EINVAL;
+
+ if (index >= SDW_MAX_CPU_DAIS) {
+ dev_err(dev, " cpu_dai_id array overflows");
+ return -EINVAL;
+ }
+
+ cpu_dai_id[index++] = ffs(adr_next->mask) - 1;
+ *codec_num += adr_next->num_adr;
+ }
+
+ /*
+ * indicate CPU DAIs for this group have been generated
+ * to avoid generating CPU DAIs for this group again.
+ */
+ group_generated[*group_id] = true;
+ *cpu_dai_num = index;
+
+ return 0;
+}
+
+static int create_sdw_dailink(struct device *dev, int *be_index,
+ struct snd_soc_dai_link *dai_links,
+ int sdw_be_num, int sdw_cpu_dai_num,
+ struct snd_soc_dai_link_component *cpus,
+ const struct snd_soc_acpi_link_adr *link,
+ int *cpu_id, bool *group_generated)
+{
+ const struct snd_soc_acpi_link_adr *link_next;
+ struct snd_soc_dai_link_component *codecs;
+ int cpu_dai_id[SDW_MAX_CPU_DAIS];
+ int cpu_dai_num, cpu_dai_index;
+ unsigned int part_id, group_id;
+ int codec_idx = 0;
+ int i = 0, j = 0;
+ int codec_index;
+ int codec_num;
+ int stream;
+ int ret;
+ int k;
+
+ ret = get_slave_info(link, dev, cpu_dai_id, &cpu_dai_num, &codec_num,
+ &group_id, group_generated);
+ if (ret)
+ return ret;
+
+ codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL);
+ if (!codecs)
+ return -ENOMEM;
+
+ /* generate codec name on different links in the same group */
+ for (link_next = link; link_next && link_next->num_adr &&
+ i < cpu_dai_num; link_next++) {
+ const struct snd_soc_acpi_endpoint *endpoints;
+
+ endpoints = link_next->adr_d->endpoints;
+ if (group_id && (!endpoints->aggregated ||
+ endpoints->group_id != group_id))
+ continue;
+
+ /* skip the link excluded by this processed group */
+ if (cpu_dai_id[i] != ffs(link_next->mask) - 1)
+ continue;
+
+ ret = create_codec_dai_name(dev, link_next, codecs, codec_idx);
+ if (ret < 0)
+ return ret;
+
+ /* check next link to create codec dai in the processed group */
+ i++;
+ codec_idx += link_next->num_adr;
+ }
+
+ /* find codec info to create BE DAI */
+ part_id = SDW_PART_ID(link->adr_d[0].adr);
+ codec_index = find_codec_info_part(part_id);
+ if (codec_index < 0)
+ return codec_index;
+
+ cpu_dai_index = *cpu_id;
+ for_each_pcm_streams(stream) {
+ char *name, *cpu_name;
+ int playback, capture;
+ static const char * const sdw_stream_name[] = {
+ "SDW%d-Playback",
+ "SDW%d-Capture",
+ };
+
+ if (!codec_info_list[codec_index].direction[stream])
+ continue;
+
+ /* create stream name according to first link id */
+ name = devm_kasprintf(dev, GFP_KERNEL,
+ sdw_stream_name[stream], cpu_dai_id[0]);
+ if (!name)
+ return -ENOMEM;
+
+ /*
+ * generate CPU DAI name base on the sdw link ID and
+ * PIN ID with offset of 2 according to sdw dai driver.
+ */
+ for (k = 0; k < cpu_dai_num; k++) {
+ cpu_name = devm_kasprintf(dev, GFP_KERNEL,
+ "SDW%d Pin%d", cpu_dai_id[k],
+ j + SDW_INTEL_BIDIR_PDI_BASE);
+ if (!cpu_name)
+ return -ENOMEM;
+
+ if (cpu_dai_index >= sdw_cpu_dai_num) {
+ dev_err(dev, "invalid cpu dai index %d",
+ cpu_dai_index);
+ return -EINVAL;
+ }
+
+ cpus[cpu_dai_index++].dai_name = cpu_name;
+ }
+
+ if (*be_index >= sdw_be_num) {
+ dev_err(dev, " invalid be dai index %d", *be_index);
+ return -EINVAL;
+ }
+
+ if (*cpu_id >= sdw_cpu_dai_num) {
+ dev_err(dev, " invalid cpu dai index %d", *cpu_id);
+ return -EINVAL;
+ }
+
+ playback = (stream == SNDRV_PCM_STREAM_PLAYBACK);
+ capture = (stream == SNDRV_PCM_STREAM_CAPTURE);
+ init_dai_link(dai_links + *be_index, *be_index, name,
+ playback, capture,
+ cpus + *cpu_id, cpu_dai_num,
+ codecs, codec_num,
+ NULL, &sdw_ops);
+
+ ret = set_codec_init_func(link, dai_links + (*be_index)++,
+ playback);
+ if (ret < 0) {
+ dev_err(dev, "failed to init codec %d", codec_index);
+ return ret;
+ }
+
+ *cpu_id += cpu_dai_num;
+ j++;
+ }
+
+ return 0;
+}
+
+/*
+ * DAI link ID of SSP & DMIC & HDMI are based on last
+ * link ID used by sdw link. Since be_id may be changed
+ * in init func of sdw codec, it is not equal to be_id
+ */
+static inline int get_next_be_id(struct snd_soc_dai_link *links,
+ int be_id)
+{
+ return links[be_id - 1].id + 1;
+}
+
+static int sof_card_dai_links_create(struct device *dev,
+ struct snd_soc_acpi_mach *mach,
+ struct snd_soc_card *card)
+{
+ int ssp_num, sdw_be_num = 0, hdmi_num = 0, dmic_num;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+ struct snd_soc_dai_link_component *idisp_components;
+#endif
+ struct snd_soc_dai_link_component *ssp_components;
+ struct snd_soc_acpi_mach_params *mach_params;
+ const struct snd_soc_acpi_link_adr *adr_link;
+ struct snd_soc_dai_link_component *cpus;
+ bool group_generated[SDW_MAX_GROUPS];
+ int ssp_codec_index, ssp_mask;
+ struct snd_soc_dai_link *links;
+ int num_links, link_id = 0;
+ char *name, *cpu_name;
+ int total_cpu_dai_num;
+ int sdw_cpu_dai_num;
+ int i, j, be_id = 0;
+ int cpu_id = 0;
+ int comp_num;
+ int ret;
+
+ /* reset amp_num to ensure amp_num++ starts from 0 in each probe */
+ for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
+ codec_info_list[i].amp_num = 0;
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+ hdmi_num = sof_sdw_quirk & SOF_SDW_TGL_HDMI ?
+ SOF_TGL_HDMI_COUNT : SOF_PRE_TGL_HDMI_COUNT;
+#endif
+
+ ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk);
+ /*
+ * on generic tgl platform, I2S or sdw mode is supported
+ * based on board rework. A ACPI device is registered in
+ * system only when I2S mode is supported, not sdw mode.
+ * Here check ACPI ID to confirm I2S is supported.
+ */
+ ssp_codec_index = find_codec_info_acpi(mach->id);
+ ssp_num = ssp_codec_index >= 0 ? hweight_long(ssp_mask) : 0;
+ comp_num = hdmi_num + ssp_num;
+
+ mach_params = &mach->mach_params;
+ ret = get_sdw_dailink_info(mach_params->links,
+ &sdw_be_num, &sdw_cpu_dai_num);
+ if (ret < 0) {
+ dev_err(dev, "failed to get sdw link info %d", ret);
+ return ret;
+ }
+
+ /* enable dmic01 & dmic16k */
+ dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC) ? 2 : 0;
+ comp_num += dmic_num;
+
+ dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d", sdw_be_num, ssp_num,
+ dmic_num, hdmi_num);
+
+ /* allocate BE dailinks */
+ num_links = comp_num + sdw_be_num;
+ links = devm_kcalloc(dev, num_links, sizeof(*links), GFP_KERNEL);
+
+ /* allocated CPU DAIs */
+ total_cpu_dai_num = comp_num + sdw_cpu_dai_num;
+ cpus = devm_kcalloc(dev, total_cpu_dai_num, sizeof(*cpus),
+ GFP_KERNEL);
+
+ if (!links || !cpus)
+ return -ENOMEM;
+
+ /* SDW */
+ if (!sdw_be_num)
+ goto SSP;
+
+ adr_link = mach_params->links;
+ if (!adr_link)
+ return -EINVAL;
+
+ /*
+ * SoundWire Slaves aggregated in the same group may be
+ * located on different hardware links. Clear array to indicate
+ * CPU DAIs for this group have not been generated.
+ */
+ for (i = 0; i < SDW_MAX_GROUPS; i++)
+ group_generated[i] = false;
+
+ /* generate DAI links by each sdw link */
+ for (; adr_link->num_adr; adr_link++) {
+ const struct snd_soc_acpi_endpoint *endpoint;
+
+ endpoint = adr_link->adr_d->endpoints;
+ if (endpoint->aggregated && !endpoint->group_id) {
+ dev_err(dev, "invalid group id on link %x",
+ adr_link->mask);
+ continue;
+ }
+
+ /* this group has been generated */
+ if (endpoint->aggregated &&
+ group_generated[endpoint->group_id])
+ continue;
+
+ ret = create_sdw_dailink(dev, &be_id, links, sdw_be_num,
+ sdw_cpu_dai_num, cpus, adr_link,
+ &cpu_id, group_generated);
+ if (ret < 0) {
+ dev_err(dev, "failed to create dai link %d", be_id);
+ return -ENOMEM;
+ }
+ }
+
+ /* non-sdw DAI follows sdw DAI */
+ link_id = be_id;
+
+ /* get BE ID for non-sdw DAI */
+ be_id = get_next_be_id(links, be_id);
+
+SSP:
+ /* SSP */
+ if (!ssp_num)
+ goto DMIC;
+
+ for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) {
+ struct sof_sdw_codec_info *info;
+ int playback, capture;
+ char *codec_name;
+
+ if (!(ssp_mask & 0x1))
+ continue;
+
+ name = devm_kasprintf(dev, GFP_KERNEL,
+ "SSP%d-Codec", i);
+ if (!name)
+ return -ENOMEM;
+
+ cpu_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i);
+ if (!cpu_name)
+ return -ENOMEM;
+
+ ssp_components = devm_kzalloc(dev, sizeof(*ssp_components),
+ GFP_KERNEL);
+ if (!ssp_components)
+ return -ENOMEM;
+
+ info = &codec_info_list[ssp_codec_index];
+ codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d",
+ info->acpi_id, j++);
+ if (!codec_name)
+ return -ENOMEM;
+
+ ssp_components->name = codec_name;
+ ssp_components->dai_name = info->dai_name;
+ cpus[cpu_id].dai_name = cpu_name;
+
+ playback = info->direction[SNDRV_PCM_STREAM_PLAYBACK];
+ capture = info->direction[SNDRV_PCM_STREAM_CAPTURE];
+ init_dai_link(links + link_id, be_id, name,
+ playback, capture,
+ cpus + cpu_id, 1,
+ ssp_components, 1,
+ NULL, info->ops);
+
+ ret = info->init(NULL, links + link_id, info, 0);
+ if (ret < 0)
+ return ret;
+
+ INC_ID(be_id, cpu_id, link_id);
+ }
+
+DMIC:
+ /* dmic */
+ if (dmic_num > 0) {
+ cpus[cpu_id].dai_name = "DMIC01 Pin";
+ init_dai_link(links + link_id, be_id, "dmic01",
+ 0, 1, // DMIC only supports capture
+ cpus + cpu_id, 1,
+ dmic_component, 1,
+ sof_sdw_dmic_init, NULL);
+ INC_ID(be_id, cpu_id, link_id);
+
+ cpus[cpu_id].dai_name = "DMIC16k Pin";
+ init_dai_link(links + link_id, be_id, "dmic16k",
+ 0, 1, // DMIC only supports capture
+ cpus + cpu_id, 1,
+ dmic_component, 1,
+ /* don't call sof_sdw_dmic_init() twice */
+ NULL, NULL);
+ INC_ID(be_id, cpu_id, link_id);
+ }
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+ /* HDMI */
+ if (hdmi_num > 0) {
+ idisp_components = devm_kcalloc(dev, hdmi_num,
+ sizeof(*idisp_components),
+ GFP_KERNEL);
+ if (!idisp_components)
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < hdmi_num; i++) {
+ name = devm_kasprintf(dev, GFP_KERNEL,
+ "iDisp%d", i + 1);
+ if (!name)
+ return -ENOMEM;
+
+ idisp_components[i].name = "ehdaudio0D2";
+ idisp_components[i].dai_name = devm_kasprintf(dev,
+ GFP_KERNEL,
+ "intel-hdmi-hifi%d",
+ i + 1);
+ if (!idisp_components[i].dai_name)
+ return -ENOMEM;
+
+ cpu_name = devm_kasprintf(dev, GFP_KERNEL,
+ "iDisp%d Pin", i + 1);
+ if (!cpu_name)
+ return -ENOMEM;
+
+ cpus[cpu_id].dai_name = cpu_name;
+ init_dai_link(links + link_id, be_id, name,
+ 1, 0, // HDMI only supports playback
+ cpus + cpu_id, 1,
+ idisp_components + i, 1,
+ sof_sdw_hdmi_init, NULL);
+ INC_ID(be_id, cpu_id, link_id);
+ }
+#endif
+
+ card->dai_link = links;
+ card->num_links = num_links;
+
+ return 0;
+}
+
+/* SoC card */
+static const char sdw_card_long_name[] = "Intel Soundwire SOF";
+
+static struct snd_soc_card card_sof_sdw = {
+ .name = "soundwire",
+ .late_probe = sof_sdw_hdmi_card_late_probe,
+ .codec_conf = codec_conf,
+ .num_configs = ARRAY_SIZE(codec_conf),
+};
+
+static int mc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &card_sof_sdw;
+ struct snd_soc_acpi_mach *mach;
+ struct mc_private *ctx;
+ int ret;
+
+ dev_dbg(&pdev->dev, "Entry %s\n", __func__);
+
+ ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
+ if (!ctx)
+ return -ENOMEM;
+
+ dmi_check_system(sof_sdw_quirk_table);
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+ INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+#endif
+
+ card->dev = &pdev->dev;
+
+ mach = pdev->dev.platform_data;
+ ret = sof_card_dai_links_create(&pdev->dev, mach,
+ card);
+ if (ret < 0)
+ return ret;
+
+ ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
+
+ snd_soc_card_set_drvdata(card, ctx);
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "cfg-spk:%d",
+ (sof_sdw_quirk & SOF_SDW_FOUR_SPK) ? 4 : 2);
+ if (!card->components)
+ return -ENOMEM;
+
+ card->long_name = sdw_card_long_name;
+
+ /* Register the card */
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
+ if (ret) {
+ dev_err(card->dev, "snd_soc_register_card failed %d\n", ret);
+ return ret;
+ }
+
+ platform_set_drvdata(pdev, card);
+
+ return ret;
+}
+
+static struct platform_driver sof_sdw_driver = {
+ .driver = {
+ .name = "sof_sdw",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = mc_probe,
+};
+
+module_platform_driver(sof_sdw_driver);
+
+MODULE_DESCRIPTION("ASoC SoundWire Generic Machine driver");
+MODULE_AUTHOR("Bard Liao <yung-chuan.liao@linux.intel.com>");
+MODULE_AUTHOR("Rander Wang <rander.wang@linux.intel.com>");
+MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_sdw");
diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h
new file mode 100644
index 000000000000..dd593ff3575b
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_common.h
@@ -0,0 +1,114 @@
+/* SPDX-License-Identifier: GPL-2.0
+ * Copyright (c) 2020 Intel Corporation
+ */
+
+/*
+ * sof_sdw_common.h - prototypes for common helpers
+ */
+
+#ifndef SND_SOC_SOF_SDW_COMMON_H
+#define SND_SOC_SOF_SDW_COMMON_H
+
+#include <linux/bits.h>
+#include <linux/types.h>
+
+#define MAX_NO_PROPS 2
+#define MAX_HDMI_NUM 4
+#define SDW_DMIC_DAI_ID 4
+#define SDW_MAX_CPU_DAIS 16
+#define SDW_INTEL_BIDIR_PDI_BASE 2
+
+/* 8 combinations with 4 links + unused group 0 */
+#define SDW_MAX_GROUPS 9
+
+enum {
+ SOF_RT711_JD_SRC_JD1 = 1,
+ SOF_RT711_JD_SRC_JD2 = 2,
+};
+
+enum {
+ SOF_PRE_TGL_HDMI_COUNT = 3,
+ SOF_TGL_HDMI_COUNT = 4,
+};
+
+enum {
+ SOF_I2S_SSP0 = BIT(0),
+ SOF_I2S_SSP1 = BIT(1),
+ SOF_I2S_SSP2 = BIT(2),
+ SOF_I2S_SSP3 = BIT(3),
+ SOF_I2S_SSP4 = BIT(4),
+ SOF_I2S_SSP5 = BIT(5),
+};
+
+#define SOF_RT711_JDSRC(quirk) ((quirk) & GENMASK(1, 0))
+#define SOF_SDW_FOUR_SPK BIT(2)
+#define SOF_SDW_TGL_HDMI BIT(3)
+#define SOF_SDW_PCH_DMIC BIT(4)
+#define SOF_SSP_PORT(x) (((x) & GENMASK(5, 0)) << 5)
+#define SOF_SSP_GET_PORT(quirk) (((quirk) >> 5) & GENMASK(5, 0))
+#define SOF_RT715_DAI_ID_FIX BIT(11)
+#define SOF_SDW_NO_AGGREGATION BIT(12)
+
+struct sof_sdw_codec_info {
+ const int id;
+ int amp_num;
+ const u8 acpi_id[ACPI_ID_LEN];
+ const bool direction[2]; // playback & capture support
+ const char *dai_name;
+ const struct snd_soc_ops *ops;
+
+ int (*init)(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+};
+
+struct mc_private {
+ struct list_head hdmi_pcm_list;
+ bool common_hdmi_codec_drv;
+ struct snd_soc_jack sdw_headset;
+};
+
+extern unsigned long sof_sdw_quirk;
+
+/* generic HDMI support */
+int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd);
+
+int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card);
+
+/* DMIC support */
+int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd);
+
+/* RT711 support */
+int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+
+/* RT700 support */
+int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+
+/* RT1308 support */
+extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops;
+
+int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+
+/* RT715 support */
+int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+
+/* RT5682 support */
+int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback);
+
+#endif
diff --git a/sound/soc/intel/boards/sof_sdw_dmic.c b/sound/soc/intel/boards/sof_sdw_dmic.c
new file mode 100644
index 000000000000..e92176bf0ad4
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_dmic.c
@@ -0,0 +1,42 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw_dmic - Helpers to handle dmic from generic machine driver
+ */
+
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
+ SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_route dmic_map[] = {
+ /* digital mics */
+ {"DMic", NULL, "SoC DMIC"},
+};
+
+int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+ ARRAY_SIZE(dmic_widgets));
+ if (ret) {
+ dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+ /* Don't need to add routes if widget addition failed */
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+ ARRAY_SIZE(dmic_map));
+
+ if (ret)
+ dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+ return ret;
+}
+
diff --git a/sound/soc/intel/boards/sof_sdw_hdmi.c b/sound/soc/intel/boards/sof_sdw_hdmi.c
new file mode 100644
index 000000000000..c7b5612a39e6
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_hdmi.c
@@ -0,0 +1,97 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw_hdmi - Helpers to handle HDMI from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/kernel.h>
+#include <linux/list.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+#include "../../codecs/hdac_hdmi.h"
+#include "hda_dsp_common.h"
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+static struct snd_soc_jack hdmi[MAX_HDMI_NUM];
+
+struct hdmi_pcm {
+ struct list_head head;
+ struct snd_soc_dai *codec_dai;
+ int device;
+};
+
+int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *dai = rtd->codec_dai;
+ struct hdmi_pcm *pcm;
+
+ pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+ if (!pcm)
+ return -ENOMEM;
+
+ /* dai_link id is 1:1 mapped to the PCM device */
+ pcm->device = rtd->dai_link->id;
+ pcm->codec_dai = dai;
+
+ list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+ return 0;
+}
+
+#define NAME_SIZE 32
+int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card)
+{
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+ struct hdmi_pcm *pcm;
+ struct snd_soc_component *component = NULL;
+ int err, i = 0;
+ char jack_name[NAME_SIZE];
+
+ pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm,
+ head);
+ component = pcm->codec_dai->component;
+
+ if (ctx->common_hdmi_codec_drv)
+ return hda_dsp_hdmi_build_controls(card, component);
+
+ list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+ component = pcm->codec_dai->component;
+ snprintf(jack_name, sizeof(jack_name),
+ "HDMI/DP, pcm=%d Jack", pcm->device);
+ err = snd_soc_card_jack_new(card, jack_name,
+ SND_JACK_AVOUT, &hdmi[i],
+ NULL, 0);
+
+ if (err)
+ return err;
+
+ err = snd_jack_add_new_kctl(hdmi[i].jack,
+ jack_name, SND_JACK_AVOUT);
+ if (err)
+ dev_warn(component->dev, "failed creating Jack kctl\n");
+
+ err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+ &hdmi[i]);
+ if (err < 0)
+ return err;
+
+ i++;
+ }
+
+ if (!component)
+ return -EINVAL;
+
+ return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
+#else
+int hdmi_card_late_probe(struct snd_soc_card *card)
+{
+ return 0;
+}
+#endif
diff --git a/sound/soc/intel/boards/sof_sdw_rt1308.c b/sound/soc/intel/boards/sof_sdw_rt1308.c
new file mode 100644
index 000000000000..321768e54d08
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_rt1308.c
@@ -0,0 +1,151 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw_rt1308 - Helpers to handle RT1308 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+#include "../../codecs/rt1308.h"
+
+static const struct snd_soc_dapm_widget rt1308_widgets[] = {
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/*
+ * dapm routes for rt1308 will be registered dynamically according
+ * to the number of rt1308 used. The first two entries will be registered
+ * for one codec case, and the last two entries are also registered
+ * if two 1308s are used.
+ */
+static const struct snd_soc_dapm_route rt1308_map[] = {
+ { "Speaker", NULL, "rt1308-1 SPOL" },
+ { "Speaker", NULL, "rt1308-1 SPOR" },
+ { "Speaker", NULL, "rt1308-2 SPOL" },
+ { "Speaker", NULL, "rt1308-2 SPOR" },
+};
+
+static const struct snd_kcontrol_new rt1308_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static int first_spk_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s spk:rt1308",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_add_card_controls(card, rt1308_controls,
+ ARRAY_SIZE(rt1308_controls));
+ if (ret) {
+ dev_err(card->dev, "rt1308 controls addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, rt1308_widgets,
+ ARRAY_SIZE(rt1308_widgets));
+ if (ret) {
+ dev_err(card->dev, "rt1308 widgets addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map, 2);
+ if (ret)
+ dev_err(rtd->dev, "failed to add first SPK map: %d\n", ret);
+
+ return ret;
+}
+
+static int second_spk_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ int ret;
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map + 2, 2);
+ if (ret)
+ dev_err(rtd->dev, "failed to add second SPK map: %d\n", ret);
+
+ return ret;
+}
+
+static int all_spk_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+
+ ret = first_spk_init(rtd);
+ if (ret)
+ return ret;
+
+ return second_spk_init(rtd);
+}
+
+static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int clk_id, clk_freq, pll_out;
+ int err;
+
+ clk_id = RT1308_PLL_S_MCLK;
+ clk_freq = 38400000;
+
+ pll_out = params_rate(params) * 512;
+
+ /* Set rt1308 pll */
+ err = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out);
+ if (err < 0) {
+ dev_err(card->dev, "Failed to set RT1308 PLL: %d\n", err);
+ return err;
+ }
+
+ /* Set rt1308 sysclk */
+ err = snd_soc_dai_set_sysclk(codec_dai, RT1308_FS_SYS_S_PLL, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (err < 0) {
+ dev_err(card->dev, "Failed to set RT1308 SYSCLK: %d\n", err);
+ return err;
+ }
+
+ return 0;
+}
+
+/* machine stream operations */
+struct snd_soc_ops sof_sdw_rt1308_i2s_ops = {
+ .hw_params = rt1308_i2s_hw_params,
+};
+
+int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback)
+{
+ info->amp_num++;
+ if (info->amp_num == 1)
+ dai_links->init = first_spk_init;
+
+ if (info->amp_num == 2) {
+ /*
+ * if two 1308s are in one dai link, the init function
+ * in this dai link will be first set for the first speaker,
+ * and it should be reset to initialize all speakers when
+ * the second speaker is found.
+ */
+ if (dai_links->init)
+ dai_links->init = all_spk_init;
+ else
+ dai_links->init = second_spk_init;
+ }
+
+ return 0;
+}
diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c
new file mode 100644
index 000000000000..5aa6211a1ed9
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_rt5682.c
@@ -0,0 +1,126 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw_rt5682 - Helpers to handle RT5682 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/input.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+
+static const struct snd_soc_dapm_widget rt5682_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route rt5682_map[] = {
+ /*Headphones*/
+ { "Headphone", NULL, "rt5682 HPOL" },
+ { "Headphone", NULL, "rt5682 HPOR" },
+ { "rt5682 IN1P", NULL, "Headset Mic" },
+};
+
+static const struct snd_kcontrol_new rt5682_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static struct snd_soc_jack_pin rt5682_jack_pins[] = {
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+ struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_jack *jack;
+ int ret;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s hs:rt5682",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_add_card_controls(card, rt5682_controls,
+ ARRAY_SIZE(rt5682_controls));
+ if (ret) {
+ dev_err(card->dev, "rt5682 control addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, rt5682_widgets,
+ ARRAY_SIZE(rt5682_widgets));
+ if (ret) {
+ dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, rt5682_map,
+ ARRAY_SIZE(rt5682_map));
+
+ if (ret) {
+ dev_err(card->dev, "rt5682 map addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+ SND_JACK_BTN_3,
+ &ctx->sdw_headset,
+ rt5682_jack_pins,
+ ARRAY_SIZE(rt5682_jack_pins));
+ if (ret) {
+ dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n",
+ ret);
+ return ret;
+ }
+
+ jack = &ctx->sdw_headset;
+
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+
+ ret = snd_soc_component_set_jack(component, jack, NULL);
+
+ if (ret)
+ dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n",
+ ret);
+
+ return ret;
+}
+
+int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback)
+{
+ /*
+ * headset should be initialized once.
+ * Do it with dai link for playback.
+ */
+ if (!playback)
+ return 0;
+
+ dai_links->init = rt5682_rtd_init;
+
+ return 0;
+}
diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c
new file mode 100644
index 000000000000..2ee4e6910d7f
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_rt700.c
@@ -0,0 +1,125 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw_rt700 - Helpers to handle RT700 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/input.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+
+static const struct snd_soc_dapm_widget rt700_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("AMIC", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route rt700_map[] = {
+ /* Headphones */
+ { "Headphones", NULL, "HP" },
+ { "Speaker", NULL, "SPK" },
+ { "MIC2", NULL, "AMIC" },
+};
+
+static const struct snd_kcontrol_new rt700_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphones"),
+ SOC_DAPM_PIN_SWITCH("AMIC"),
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static struct snd_soc_jack_pin rt700_jack_pins[] = {
+ {
+ .pin = "Headphones",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "AMIC",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+ struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_jack *jack;
+ int ret;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s hs:rt700",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_add_card_controls(card, rt700_controls,
+ ARRAY_SIZE(rt700_controls));
+ if (ret) {
+ dev_err(card->dev, "rt700 controls addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, rt700_widgets,
+ ARRAY_SIZE(rt700_widgets));
+ if (ret) {
+ dev_err(card->dev, "rt700 widgets addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, rt700_map,
+ ARRAY_SIZE(rt700_map));
+
+ if (ret) {
+ dev_err(card->dev, "rt700 map addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+ SND_JACK_BTN_3,
+ &ctx->sdw_headset,
+ rt700_jack_pins,
+ ARRAY_SIZE(rt700_jack_pins));
+ if (ret) {
+ dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n",
+ ret);
+ return ret;
+ }
+
+ jack = &ctx->sdw_headset;
+
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+ ret = snd_soc_component_set_jack(component, jack, NULL);
+ if (ret)
+ dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n",
+ ret);
+
+ return ret;
+}
+
+int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback)
+{
+ /*
+ * headset should be initialized once.
+ * Do it with dai link for playback.
+ */
+ if (!playback)
+ return 0;
+
+ dai_links->init = rt700_rtd_init;
+
+ return 0;
+}
diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c
new file mode 100644
index 000000000000..2a4917e3d561
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_rt711.c
@@ -0,0 +1,156 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw_rt711 - Helpers to handle RT711 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/input.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+
+/*
+ * Note this MUST be called before snd_soc_register_card(), so that the props
+ * are in place before the codec component driver's probe function parses them.
+ */
+static int rt711_add_codec_device_props(const char *sdw_dev_name)
+{
+ struct property_entry props[MAX_NO_PROPS] = {};
+ struct device *sdw_dev;
+ int ret;
+
+ sdw_dev = bus_find_device_by_name(&sdw_bus_type, NULL, sdw_dev_name);
+ if (!sdw_dev)
+ return -EPROBE_DEFER;
+
+ if (SOF_RT711_JDSRC(sof_sdw_quirk)) {
+ props[0] = PROPERTY_ENTRY_U32("realtek,jd-src",
+ SOF_RT711_JDSRC(sof_sdw_quirk));
+ }
+
+ ret = device_add_properties(sdw_dev, props);
+ put_device(sdw_dev);
+
+ return ret;
+}
+
+static const struct snd_soc_dapm_widget rt711_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route rt711_map[] = {
+ /* Headphones */
+ { "Headphone", NULL, "rt711 HP" },
+ { "rt711 MIC2", NULL, "Headset Mic" },
+};
+
+static const struct snd_kcontrol_new rt711_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static struct snd_soc_jack_pin rt711_jack_pins[] = {
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+ struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_jack *jack;
+ int ret;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s hs:rt711",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_add_card_controls(card, rt711_controls,
+ ARRAY_SIZE(rt711_controls));
+ if (ret) {
+ dev_err(card->dev, "rt711 controls addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_new_controls(&card->dapm, rt711_widgets,
+ ARRAY_SIZE(rt711_widgets));
+ if (ret) {
+ dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, rt711_map,
+ ARRAY_SIZE(rt711_map));
+
+ if (ret) {
+ dev_err(card->dev, "rt711 map addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+ SND_JACK_BTN_3,
+ &ctx->sdw_headset,
+ rt711_jack_pins,
+ ARRAY_SIZE(rt711_jack_pins));
+ if (ret) {
+ dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n",
+ ret);
+ return ret;
+ }
+
+ jack = &ctx->sdw_headset;
+
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+ ret = snd_soc_component_set_jack(component, jack, NULL);
+
+ if (ret)
+ dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n",
+ ret);
+
+ return ret;
+}
+
+int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback)
+{
+ int ret;
+
+ /*
+ * headset should be initialized once.
+ * Do it with dai link for playback.
+ */
+ if (!playback)
+ return 0;
+
+ ret = rt711_add_codec_device_props("sdw:0:25d:711:0");
+ if (ret < 0)
+ return ret;
+
+ dai_links->init = rt711_rtd_init;
+
+ return 0;
+}
diff --git a/sound/soc/intel/boards/sof_sdw_rt715.c b/sound/soc/intel/boards/sof_sdw_rt715.c
new file mode 100644
index 000000000000..321e1cbc03ed
--- /dev/null
+++ b/sound/soc/intel/boards/sof_sdw_rt715.c
@@ -0,0 +1,42 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ * sof_sdw_rt715 - Helpers to handle RT715 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+
+static int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_card *card = rtd->card;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s mic:rt715",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ return 0;
+}
+
+int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link,
+ struct snd_soc_dai_link *dai_links,
+ struct sof_sdw_codec_info *info,
+ bool playback)
+{
+ /*
+ * DAI ID is fixed at SDW_DMIC_DAI_ID for 715 to
+ * keep sdw DMIC and HDMI setting static in UCM
+ */
+ if (sof_sdw_quirk & SOF_RT715_DAI_ID_FIX)
+ dai_links->id = SDW_DMIC_DAI_ID;
+
+ dai_links->init = rt715_rtd_init;
+
+ return 0;
+}
diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
index 4a5adae1d785..f5092bc48364 100644
--- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
@@ -65,7 +65,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = {
},
{
.id = "104C5122",
- .drv_name = "bxt-pcm512x",
+ .drv_name = "sof_pcm512x",
.sof_fw_filename = "sof-apl.ri",
.sof_tplg_filename = "sof-apl-pcm512x.tplg",
},
diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
index d0fb43c2b9f6..2752dc955733 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
@@ -174,6 +174,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = {
.sof_fw_filename = "sof-cht.ri",
.sof_tplg_filename = "sof-cht-cx2072x.tplg",
},
+ {
+ .id = "104C5122",
+ .drv_name = "sof_pcm512x",
+ .sof_fw_filename = "sof-cht.ri",
+ .sof_tplg_filename = "sof-cht-src-50khz-pcm512x.tplg",
+ },
+
#if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
/*
* This is always last in the table so that it is selected only when
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index f55634c4c2e8..bcedec6c6117 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -59,42 +59,112 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = {
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines);
-static const u64 rt711_0_adr[] = {
- 0x000010025D071100
+static const struct snd_soc_acpi_endpoint single_endpoint = {
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
};
-static const u64 rt1308_1_adr[] = {
- 0x000110025D130800
+static const struct snd_soc_acpi_endpoint spk_l_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 0,
+ .group_id = 1,
};
-static const u64 rt1308_2_adr[] = {
- 0x000210025D130800
+static const struct snd_soc_acpi_endpoint spk_r_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 1,
+ .group_id = 1,
};
-static const u64 rt715_3_adr[] = {
- 0x000310025D071500
+static const struct snd_soc_acpi_adr_device rt700_1_adr[] = {
+ {
+ .adr = 0x000110025D070000,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_link_adr cml_rvp[] = {
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(rt700_1_adr),
+ .adr_d = rt700_1_adr,
+ },
+ {}
+};
+
+static const struct snd_soc_acpi_adr_device rt711_0_adr[] = {
+ {
+ .adr = 0x000010025D071100,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
+ {
+ .adr = 0x000110025D130800,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
+ {
+ .adr = 0x000210025D130800,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
+ {
+ .adr = 0x000110025D130800,
+ .num_endpoints = 1,
+ .endpoints = &spk_l_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = {
+ {
+ .adr = 0x000210025D130800,
+ .num_endpoints = 1,
+ .endpoints = &spk_r_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt715_3_adr[] = {
+ {
+ .adr = 0x000310025D071500,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
};
static const struct snd_soc_acpi_link_adr cml_3_in_1_default[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt711_0_adr),
- .adr = rt711_0_adr,
+ .adr_d = rt711_0_adr,
},
{
.mask = BIT(1),
- .num_adr = ARRAY_SIZE(rt1308_1_adr),
- .adr = rt1308_1_adr,
+ .num_adr = ARRAY_SIZE(rt1308_1_group1_adr),
+ .adr_d = rt1308_1_group1_adr,
},
{
.mask = BIT(2),
- .num_adr = ARRAY_SIZE(rt1308_2_adr),
- .adr = rt1308_2_adr,
+ .num_adr = ARRAY_SIZE(rt1308_2_group1_adr),
+ .adr_d = rt1308_2_group1_adr,
},
{
.mask = BIT(3),
.num_adr = ARRAY_SIZE(rt715_3_adr),
- .adr = rt715_3_adr,
+ .adr_d = rt715_3_adr,
},
{}
};
@@ -103,17 +173,17 @@ static const struct snd_soc_acpi_link_adr cml_3_in_1_mono_amp[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt711_0_adr),
- .adr = rt711_0_adr,
+ .adr_d = rt711_0_adr,
},
{
.mask = BIT(1),
.num_adr = ARRAY_SIZE(rt1308_1_adr),
- .adr = rt1308_1_adr,
+ .adr_d = rt1308_1_adr,
},
{
.mask = BIT(3),
.num_adr = ARRAY_SIZE(rt715_3_adr),
- .adr = rt715_3_adr,
+ .adr_d = rt715_3_adr,
},
{}
};
@@ -122,7 +192,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = {
{
.link_mask = 0xF, /* 4 active links required */
.links = cml_3_in_1_default,
- .drv_name = "sdw_rt711_rt1308_rt715",
+ .drv_name = "sof_sdw",
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-rt711-rt1308-rt715.tplg",
},
@@ -134,13 +204,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = {
*/
.link_mask = 0xF,
.links = cml_3_in_1_mono_amp,
- .drv_name = "sdw_rt711_rt1308_rt715",
+ .drv_name = "sof_sdw",
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-rt711-rt1308-mono-rt715.tplg",
},
{
.link_mask = 0x2, /* RT700 connected on Link1 */
- .drv_name = "sdw_rt700",
+ .links = cml_rvp,
+ .drv_name = "sof_sdw",
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-rt700.tplg",
},
diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
index 752733013d54..ef8500349f2f 100644
--- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
@@ -33,55 +33,112 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = {
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines);
-static const u64 rt700_0_adr[] = {
- 0x000010025D070000
+static const struct snd_soc_acpi_endpoint single_endpoint = {
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+};
+
+static const struct snd_soc_acpi_endpoint spk_l_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 0,
+ .group_id = 1,
+};
+
+static const struct snd_soc_acpi_endpoint spk_r_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 1,
+ .group_id = 1,
+};
+
+static const struct snd_soc_acpi_adr_device rt700_0_adr[] = {
+ {
+ .adr = 0x000010025D070000,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
};
static const struct snd_soc_acpi_link_adr icl_rvp[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt700_0_adr),
- .adr = rt700_0_adr,
+ .adr_d = rt700_0_adr,
},
{}
};
-static const u64 rt711_0_adr[] = {
- 0x000010025D071100
+static const struct snd_soc_acpi_adr_device rt711_0_adr[] = {
+ {
+ .adr = 0x000010025D071100,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
+ {
+ .adr = 0x000110025D130800,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
};
-static const u64 rt1308_1_adr[] = {
- 0x000110025D130800
+static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
+ {
+ .adr = 0x000210025D130800,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
};
-static const u64 rt1308_2_adr[] = {
- 0x000210025D130800
+static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
+ {
+ .adr = 0x000110025D130800,
+ .num_endpoints = 1,
+ .endpoints = &spk_l_endpoint,
+ }
};
-static const u64 rt715_3_adr[] = {
- 0x000310025D071500
+static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = {
+ {
+ .adr = 0x000210025D130800,
+ .num_endpoints = 1,
+ .endpoints = &spk_r_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt715_3_adr[] = {
+ {
+ .adr = 0x000310025D071500,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
};
static const struct snd_soc_acpi_link_adr icl_3_in_1_default[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt711_0_adr),
- .adr = rt711_0_adr,
+ .adr_d = rt711_0_adr,
},
{
.mask = BIT(1),
- .num_adr = ARRAY_SIZE(rt1308_1_adr),
- .adr = rt1308_1_adr,
+ .num_adr = ARRAY_SIZE(rt1308_1_group1_adr),
+ .adr_d = rt1308_1_group1_adr,
},
{
.mask = BIT(2),
- .num_adr = ARRAY_SIZE(rt1308_2_adr),
- .adr = rt1308_2_adr,
+ .num_adr = ARRAY_SIZE(rt1308_2_group1_adr),
+ .adr_d = rt1308_2_group1_adr,
},
{
.mask = BIT(3),
.num_adr = ARRAY_SIZE(rt715_3_adr),
- .adr = rt715_3_adr,
+ .adr_d = rt715_3_adr,
},
{}
};
@@ -90,17 +147,17 @@ static const struct snd_soc_acpi_link_adr icl_3_in_1_mono_amp[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt711_0_adr),
- .adr = rt711_0_adr,
+ .adr_d = rt711_0_adr,
},
{
.mask = BIT(1),
.num_adr = ARRAY_SIZE(rt1308_1_adr),
- .adr = rt1308_1_adr,
+ .adr_d = rt1308_1_adr,
},
{
.mask = BIT(3),
.num_adr = ARRAY_SIZE(rt715_3_adr),
- .adr = rt715_3_adr,
+ .adr_d = rt715_3_adr,
},
{}
};
@@ -109,21 +166,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[] = {
{
.link_mask = 0xF, /* 4 active links required */
.links = icl_3_in_1_default,
- .drv_name = "sdw_rt711_rt1308_rt715",
+ .drv_name = "sof_sdw",
.sof_fw_filename = "sof-icl.ri",
.sof_tplg_filename = "sof-icl-rt711-rt1308-rt715.tplg",
},
{
.link_mask = 0xB, /* 3 active links required */
.links = icl_3_in_1_mono_amp,
- .drv_name = "sdw_rt711_rt1308_rt715",
+ .drv_name = "sof_sdw",
.sof_fw_filename = "sof-icl.ri",
.sof_tplg_filename = "sof-icl-rt711-rt1308-rt715-mono.tplg",
},
{
.link_mask = 0x1, /* rt700 connected on link0 */
.links = icl_rvp,
- .drv_name = "sdw_rt700",
+ .drv_name = "sof_sdw",
.sof_fw_filename = "sof-icl.ri",
.sof_tplg_filename = "sof-icl-rt700.tplg",
},
diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
index ed2b125f6a11..4388a32718d8 100644
--- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
@@ -2,20 +2,50 @@
/*
* soc-apci-intel-jsl-match.c - tables and support for JSL ACPI enumeration.
*
- * Copyright (c) 2019, Intel Corporation.
+ * Copyright (c) 2019-2020, Intel Corporation.
*
*/
#include <sound/soc-acpi.h>
#include <sound/soc-acpi-intel-match.h>
+static struct snd_soc_acpi_codecs jsl_7219_98373_codecs = {
+ .num_codecs = 1,
+ .codecs = {"MX98373"}
+};
+
+static struct snd_soc_acpi_codecs rt1015_spk = {
+ .num_codecs = 1,
+ .codecs = {"10EC1015"}
+};
+
+/*
+ * When adding new entry to the snd_soc_acpi_intel_jsl_machines array,
+ * use .quirk_data member to distinguish different machine driver,
+ * and keep ACPI .id field unchanged for the common codec.
+ */
struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = {
{
.id = "DLGS7219",
.drv_name = "sof_da7219_max98373",
- .machine_quirk = snd_soc_acpi_codec_list,
.sof_fw_filename = "sof-jsl.ri",
.sof_tplg_filename = "sof-jsl-da7219.tplg",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &jsl_7219_98373_codecs,
+ },
+ {
+ .id = "DLGS7219",
+ .drv_name = "sof_da7219_max98360a",
+ .sof_fw_filename = "sof-jsl.ri",
+ .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg",
+ },
+ {
+ .id = "10EC5682",
+ .drv_name = "jsl_rt5682_rt1015",
+ .sof_fw_filename = "sof-jsl.ri",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &rt1015_spk,
+ .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg",
},
{},
};
diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
index 5984dd151f3e..449d9d2286ae 100644
--- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c
@@ -14,20 +14,61 @@ static struct snd_soc_acpi_codecs tgl_codecs = {
.codecs = {"MX98357A"}
};
-static const u64 rt711_0_adr[] = {
- 0x000010025D071100
+static const struct snd_soc_acpi_endpoint single_endpoint = {
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
};
-static const u64 rt1308_1_adr[] = {
- 0x000120025D130800,
- 0x000122025D130800
+static const struct snd_soc_acpi_endpoint spk_l_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 0,
+ .group_id = 1,
+};
+
+static const struct snd_soc_acpi_endpoint spk_r_endpoint = {
+ .num = 0,
+ .aggregated = 1,
+ .group_position = 1,
+ .group_id = 1,
+};
+
+static const struct snd_soc_acpi_adr_device rt711_0_adr[] = {
+ {
+ .adr = 0x000010025D071100,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
+ {
+ .adr = 0x000120025D130800,
+ .num_endpoints = 1,
+ .endpoints = &spk_l_endpoint,
+ },
+ {
+ .adr = 0x000122025D130800,
+ .num_endpoints = 1,
+ .endpoints = &spk_r_endpoint,
+ }
+};
+
+static const struct snd_soc_acpi_adr_device rt5682_0_adr[] = {
+ {
+ .adr = 0x000021025D568200,
+ .num_endpoints = 1,
+ .endpoints = &single_endpoint,
+ }
};
static const struct snd_soc_acpi_link_adr tgl_i2s_rt1308[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt711_0_adr),
- .adr = rt711_0_adr,
+ .adr_d = rt711_0_adr,
},
{}
};
@@ -36,24 +77,38 @@ static const struct snd_soc_acpi_link_adr tgl_rvp[] = {
{
.mask = BIT(0),
.num_adr = ARRAY_SIZE(rt711_0_adr),
- .adr = rt711_0_adr,
+ .adr_d = rt711_0_adr,
},
{
.mask = BIT(1),
.num_adr = ARRAY_SIZE(rt1308_1_adr),
- .adr = rt1308_1_adr,
+ .adr_d = rt1308_1_adr,
},
{}
};
+static const struct snd_soc_acpi_link_adr tgl_chromebook_base[] = {
+ {
+ .mask = BIT(0),
+ .num_adr = ARRAY_SIZE(rt5682_0_adr),
+ .adr_d = rt5682_0_adr,
+ },
+ {}
+};
+
+static struct snd_soc_acpi_codecs tgl_max98373_amp = {
+ .num_codecs = 1,
+ .codecs = {"MX98373"}
+};
+
struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = {
{
.id = "10EC1308",
- .drv_name = "rt711_rt1308",
+ .drv_name = "sof_sdw",
.link_mask = 0x1, /* RT711 on SoundWire link0 */
.links = tgl_i2s_rt1308,
.sof_fw_filename = "sof-tgl.ri",
- .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg",
+ .sof_tplg_filename = "sof-tgl-rt711-i2s-rt1308.tplg",
},
{
.id = "10EC5682",
@@ -63,6 +118,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = {
.sof_fw_filename = "sof-tgl.ri",
.sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg",
},
+ {
+ .id = "10EC5682",
+ .drv_name = "tgl_max98373_rt5682",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &tgl_max98373_amp,
+ .sof_fw_filename = "sof-tgl.ri",
+ .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg",
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines);
@@ -72,10 +135,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = {
{
.link_mask = 0x3, /* rt711 on link 0 and 2 rt1308s on link 1 */
.links = tgl_rvp,
- .drv_name = "sdw_rt711_rt1308_rt715",
+ .drv_name = "sof_sdw",
.sof_fw_filename = "sof-tgl.ri",
.sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg",
},
+ {
+ .link_mask = 0x1, /* this will only enable rt5682 for now */
+ .links = tgl_chromebook_base,
+ .drv_name = "sof_sdw",
+ .sof_fw_filename = "sof-tgl.ri",
+ .sof_tplg_filename = "sof-tgl-rt5682.tplg",
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_sdw_machines);
diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c
index 033d7c05d7fb..c183f8e94ee4 100644
--- a/sound/soc/intel/haswell/sst-haswell-pcm.c
+++ b/sound/soc/intel/haswell/sst-haswell-pcm.c
@@ -476,7 +476,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component,
u8 channels;
int ret, dai;
- dai = mod_map[rtd->cpu_dai->id].dai_id;
+ dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
pcm_data = &pdata->pcm[dai][substream->stream];
/* check if we are being called a subsequent time */
@@ -494,7 +494,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component,
}
pcm_data->allocated = false;
- pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id,
+ pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id,
hsw_notify_pointer, pcm_data);
if (pcm_data->stream == NULL) {
dev_err(rtd->dev, "error: failed to create stream\n");
@@ -509,7 +509,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component,
path_id = SST_HSW_STREAM_PATH_SSP0_IN;
/* DSP stream type depends on DAI ID */
- switch (rtd->cpu_dai->id) {
+ switch (asoc_rtd_to_cpu(rtd, 0)->id) {
case 0:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
stream_type = SST_HSW_STREAM_TYPE_SYSTEM;
@@ -533,7 +533,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component,
break;
default:
dev_err(rtd->dev, "error: invalid DAI ID %d\n",
- rtd->cpu_dai->id);
+ asoc_rtd_to_cpu(rtd, 0)->id);
return -EINVAL;
}
@@ -595,7 +595,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component,
dmab = snd_pcm_get_dma_buf(substream);
ret = create_adsp_page_table(substream, pdata, rtd, runtime->dma_area,
- runtime->dma_bytes, rtd->cpu_dai->id);
+ runtime->dma_bytes, asoc_rtd_to_cpu(rtd, 0)->id);
if (ret < 0)
return ret;
@@ -608,7 +608,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component,
pages = runtime->dma_bytes / PAGE_SIZE;
ret = sst_hsw_stream_buffer(hsw, pcm_data->stream,
- pdata->dmab[rtd->cpu_dai->id][substream->stream].addr,
+ pdata->dmab[asoc_rtd_to_cpu(rtd, 0)->id][substream->stream].addr,
pages, runtime->dma_bytes, 0,
snd_sgbuf_get_addr(dmab, 0) >> PAGE_SHIFT);
if (ret < 0) {
@@ -661,7 +661,7 @@ static int hsw_pcm_trigger(struct snd_soc_component *component,
snd_pcm_uframes_t pos;
int dai;
- dai = mod_map[rtd->cpu_dai->id].dai_id;
+ dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
pcm_data = &pdata->pcm[dai][substream->stream];
sst_stream = pcm_data->stream;
@@ -770,7 +770,7 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_soc_component *component,
u32 position;
int dai;
- dai = mod_map[rtd->cpu_dai->id].dai_id;
+ dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
pcm_data = &pdata->pcm[dai][substream->stream];
position = sst_hsw_get_dsp_position(hsw, pcm_data->stream);
@@ -791,7 +791,7 @@ static int hsw_pcm_open(struct snd_soc_component *component,
struct sst_hsw *hsw = pdata->hsw;
int dai;
- dai = mod_map[rtd->cpu_dai->id].dai_id;
+ dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
pcm_data = &pdata->pcm[dai][substream->stream];
mutex_lock(&pcm_data->mutex);
@@ -801,7 +801,7 @@ static int hsw_pcm_open(struct snd_soc_component *component,
snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware);
- pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id,
+ pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id,
hsw_notify_pointer, pcm_data);
if (pcm_data->stream == NULL) {
dev_err(rtd->dev, "error: failed to create stream\n");
@@ -824,7 +824,7 @@ static int hsw_pcm_close(struct snd_soc_component *component,
struct sst_hsw *hsw = pdata->hsw;
int ret, dai;
- dai = mod_map[rtd->cpu_dai->id].dai_id;
+ dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
pcm_data = &pdata->pcm[dai][substream->stream];
mutex_lock(&pcm_data->mutex);
@@ -923,9 +923,9 @@ static int hsw_pcm_new(struct snd_soc_component *component,
hsw_pcm_hardware.buffer_bytes_max);
}
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream)
- priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm;
+ priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm;
if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream)
- priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm;
+ priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm;
return 0;
}
diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index 92a82e6b5fe6..38b9d7494083 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -17,7 +17,6 @@
#include "skl.h"
#define BXT_BASEFW_TIMEOUT 3000
-#define BXT_INIT_TIMEOUT 300
#define BXT_ROM_INIT_TIMEOUT 70
#define BXT_IPC_PURGE_FW 0x01004000
@@ -38,8 +37,6 @@
/* Delay before scheduling D0i3 entry */
#define BXT_D0I3_DELAY 5000
-#define BXT_FW_ROM_INIT_RETRY 3
-
static unsigned int bxt_get_errorcode(struct sst_dsp *ctx)
{
return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE);
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
index 4f64f097e9ae..c6abcd5aa67b 100644
--- a/sound/soc/intel/skylake/cnl-sst.c
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -57,18 +57,34 @@ static int cnl_prepare_fw(struct sst_dsp *ctx, const void *fwdata, u32 fwsize)
ctx->dsp_ops.stream_tag = stream_tag;
memcpy(ctx->dmab.area, fwdata, fwsize);
+ ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK);
+ if (ret < 0) {
+ dev_err(ctx->dev, "dsp core0 power up failed\n");
+ ret = -EIO;
+ goto base_fw_load_failed;
+ }
+
/* purge FW request */
sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR,
CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE |
((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID)));
- ret = cnl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK);
+ ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK);
if (ret < 0) {
- dev_err(ctx->dev, "dsp boot core failed ret: %d\n", ret);
+ dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret);
ret = -EIO;
goto base_fw_load_failed;
}
+ ret = sst_dsp_register_poll(ctx, CNL_ADSP_REG_HIPCIDA,
+ CNL_ADSP_REG_HIPCIDA_DONE,
+ CNL_ADSP_REG_HIPCIDA_DONE,
+ BXT_INIT_TIMEOUT, "HIPCIDA Done");
+ if (ret < 0) {
+ dev_err(ctx->dev, "timeout for purge request: %d\n", ret);
+ goto base_fw_load_failed;
+ }
+
/* enable interrupt */
cnl_ipc_int_enable(ctx);
cnl_ipc_op_int_enable(ctx);
@@ -109,7 +125,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx)
{
struct firmware stripped_fw;
struct skl_dev *cnl = ctx->thread_context;
- int ret;
+ int ret, i;
if (!ctx->fw) {
ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev);
@@ -131,12 +147,16 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx)
stripped_fw.size = ctx->fw->size;
skl_dsp_strip_extended_manifest(&stripped_fw);
- ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
- if (ret < 0) {
- dev_err(ctx->dev, "prepare firmware failed: %d\n", ret);
- goto cnl_load_base_firmware_failed;
+ for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) {
+ ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
+ if (!ret)
+ break;
+ dev_dbg(ctx->dev, "prepare firmware failed: %d\n", ret);
}
+ if (ret < 0)
+ goto cnl_load_base_firmware_failed;
+
ret = sst_transfer_fw_host_dma(ctx);
if (ret < 0) {
dev_err(ctx->dev, "transfer firmware failed: %d\n", ret);
@@ -158,6 +178,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx)
return 0;
cnl_load_base_firmware_failed:
+ dev_err(ctx->dev, "firmware load failed: %d\n", ret);
release_firmware(ctx->fw);
ctx->fw = NULL;
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 19f328d71f24..d9c8f5cb389e 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -182,7 +182,8 @@ void skl_nhlt_remove_sysfs(struct skl_dev *skl)
{
struct device *dev = &skl->pci->dev;
- sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr);
+ if (skl->nhlt)
+ sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr);
}
/*
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index b99509675d29..89dcccdfb1cd 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -112,10 +112,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream,
struct snd_soc_dapm_widget *w;
struct skl_dev *skl = bus_to_skl(bus);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- w = dai->playback_widget;
- else
- w = dai->capture_widget;
+ w = snd_soc_dai_get_widget(dai, substream->stream);
if (w->ignore_suspend && enable)
skl->supend_active++;
@@ -475,10 +472,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
if (!mconfig)
return -EIO;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- w = dai->playback_widget;
- else
- w = dai->capture_widget;
+ w = snd_soc_dai_get_widget(dai, substream->stream);
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
@@ -551,7 +545,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
struct hdac_bus *bus = dev_get_drvdata(dai->dev);
struct hdac_ext_stream *link_dev;
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct skl_pipe_params p_params = {0};
struct hdac_ext_link *link;
int stream_tag;
@@ -650,7 +644,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream,
link_dev->link_prepared = 0;
- link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name);
+ link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name);
if (!link)
return -EINVAL;
@@ -1080,7 +1074,7 @@ static int skl_platform_soc_open(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai_link *dai_link = rtd->dai_link;
- dev_dbg(rtd->cpu_dai->dev, "In %s:%s\n", __func__,
+ dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "In %s:%s\n", __func__,
dai_link->cpus->dai_name);
snd_soc_set_runtime_hwparams(substream, &azx_pcm_hw);
@@ -1232,7 +1226,7 @@ static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream,
u64 nsec)
{
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
u64 codec_frames, codec_nsecs;
if (!codec_dai->driver->ops->delay)
@@ -1287,7 +1281,7 @@ static int skl_platform_soc_get_time_info(
static int skl_platform_soc_new(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
struct hdac_bus *bus = dev_get_drvdata(dai->dev);
struct snd_pcm *pcm = rtd->pcm;
unsigned int size;
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index cdfec0fca577..1df9ef422f61 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -67,6 +67,8 @@ struct skl_dev;
#define SKL_FW_INIT 0x1
#define SKL_FW_RFW_START 0xf
+#define BXT_FW_ROM_INIT_RETRY 3
+#define BXT_INIT_TIMEOUT 300
#define SKL_ADSPIC_IPC 1
#define SKL_ADSPIS_IPC 1
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index f755ca2484cf..63182bfd7941 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -130,6 +130,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset)
struct hdac_ext_link *hlink;
int ret;
+ snd_hdac_set_codec_wakeup(bus, true);
skl_enable_miscbdcge(bus->dev, false);
ret = snd_hdac_bus_init_chip(bus, full_reset);
@@ -138,6 +139,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset)
writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
skl_enable_miscbdcge(bus->dev, true);
+ snd_hdac_set_codec_wakeup(bus, false);
return ret;
}
@@ -359,7 +361,7 @@ static int skl_resume(struct device *dev)
struct pci_dev *pci = to_pci_dev(dev);
struct hdac_bus *bus = pci_get_drvdata(pci);
struct skl_dev *skl = bus_to_skl(bus);
- struct hdac_ext_link *hlink = NULL;
+ struct hdac_ext_link *hlink;
int ret;
/*
@@ -481,13 +483,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = {
static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl,
struct snd_soc_acpi_mach *machines)
{
- struct hdac_bus *bus = skl_to_bus(skl);
struct snd_soc_acpi_mach *mach;
- /* check if we have any codecs detected on bus */
- if (bus->codec_mask == 0)
- return NULL;
-
/* point to common table */
mach = snd_soc_acpi_intel_hda_machines;
@@ -636,6 +633,9 @@ static int skl_clock_device_register(struct skl_dev *skl)
struct platform_device_info pdevinfo = {NULL};
struct skl_clk_pdata *clk_pdata;
+ if (!skl->nhlt)
+ return 0;
+
clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata),
GFP_KERNEL);
if (!clk_pdata)
@@ -794,7 +794,7 @@ static void skl_probe_work(struct work_struct *work)
{
struct skl_dev *skl = container_of(work, struct skl_dev, probe_work);
struct hdac_bus *bus = skl_to_bus(skl);
- struct hdac_ext_link *hlink = NULL;
+ struct hdac_ext_link *hlink;
int err;
if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
@@ -803,6 +803,9 @@ static void skl_probe_work(struct work_struct *work)
return;
}
+ skl_init_pci(skl);
+ skl_dum_set(bus);
+
err = skl_init_chip(bus, true);
if (err < 0) {
dev_err(bus->dev, "Init chip failed with err: %d\n", err);
@@ -918,8 +921,6 @@ static int skl_first_init(struct hdac_bus *bus)
return -ENXIO;
}
- snd_hdac_bus_reset_link(bus, true);
-
snd_hdac_bus_parse_capabilities(bus);
/* check if PPCAP exists */
@@ -967,11 +968,7 @@ static int skl_first_init(struct hdac_bus *bus)
if (err < 0)
return err;
- /* initialize chip */
- skl_init_pci(skl);
- skl_dum_set(bus);
-
- return skl_init_chip(bus, true);
+ return 0;
}
static int skl_probe(struct pci_dev *pci,
@@ -1064,8 +1061,6 @@ static int skl_probe(struct pci_dev *pci,
if (bus->mlcap)
snd_hdac_ext_bus_get_ml_capabilities(bus);
- snd_hdac_bus_stop_chip(bus);
-
/* create device for soc dmic */
err = skl_dmic_device_register(skl);
if (err < 0) {
@@ -1082,7 +1077,8 @@ out_dsp_free:
out_clk_free:
skl_clock_device_unregister(skl);
out_nhlt_free:
- intel_nhlt_free(skl->nhlt);
+ if (skl->nhlt)
+ intel_nhlt_free(skl->nhlt);
out_free:
skl_free(bus);
@@ -1131,7 +1127,8 @@ static void skl_remove(struct pci_dev *pci)
skl_dmic_device_unregister(skl);
skl_clock_device_unregister(skl);
skl_nhlt_remove_sysfs(skl);
- intel_nhlt_free(skl->nhlt);
+ if (skl->nhlt)
+ intel_nhlt_free(skl->nhlt);
skl_free(bus);
dev_set_drvdata(&pci->dev, NULL);
}
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 9d5405881209..6f6f8dad0356 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -49,12 +49,8 @@
#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12
#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8
-#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24
-#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16
-#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_MASK \
- (0xf << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET)
-#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_MASK \
- (0x1f << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET)
+#define JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24
+#define JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16
#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19)
#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16)
@@ -83,16 +79,23 @@
#define JZ_AIC_I2S_STATUS_BUSY BIT(2)
#define JZ_AIC_CLK_DIV_MASK 0xf
-#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_SHIFT 0
#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
#define I2SDIV_IDV_SHIFT 8
#define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT)
enum jz47xx_i2s_version {
JZ_I2S_JZ4740,
+ JZ_I2S_JZ4760,
+ JZ_I2S_JZ4770,
JZ_I2S_JZ4780,
};
+struct i2s_soc_info {
+ enum jz47xx_i2s_version version;
+ struct snd_soc_dai_driver *dai;
+};
+
struct jz4740_i2s {
struct resource *mem;
void __iomem *base;
@@ -104,7 +107,7 @@ struct jz4740_i2s {
struct snd_dmaengine_dai_dma_data playback_dma_data;
struct snd_dmaengine_dai_dma_data capture_dma_data;
- enum jz47xx_i2s_version version;
+ const struct i2s_soc_info *soc_info;
};
static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s,
@@ -284,7 +287,7 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK;
ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
- if (i2s->version >= JZ_I2S_JZ4780) {
+ if (i2s->soc_info->version >= JZ_I2S_JZ4770) {
div_reg &= ~I2SDIV_IDV_MASK;
div_reg |= (div - 1) << I2SDIV_IDV_SHIFT;
} else {
@@ -398,9 +401,9 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai)
snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data,
&i2s->capture_dma_data);
- if (i2s->version >= JZ_I2S_JZ4780) {
- conf = (7 << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) |
- (8 << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) |
+ if (i2s->soc_info->version >= JZ_I2S_JZ4760) {
+ conf = (7 << JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) |
+ (8 << JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) |
JZ_AIC_CONF_OVERFLOW_PLAY_LAST |
JZ_AIC_CONF_I2S |
JZ_AIC_CONF_INTERNAL_CODEC;
@@ -457,7 +460,17 @@ static struct snd_soc_dai_driver jz4740_i2s_dai = {
.ops = &jz4740_i2s_dai_ops,
};
-static struct snd_soc_dai_driver jz4780_i2s_dai = {
+static const struct i2s_soc_info jz4740_i2s_soc_info = {
+ .version = JZ_I2S_JZ4740,
+ .dai = &jz4740_i2s_dai,
+};
+
+static const struct i2s_soc_info jz4760_i2s_soc_info = {
+ .version = JZ_I2S_JZ4760,
+ .dai = &jz4740_i2s_dai,
+};
+
+static struct snd_soc_dai_driver jz4770_i2s_dai = {
.probe = jz4740_i2s_dai_probe,
.remove = jz4740_i2s_dai_remove,
.playback = {
@@ -475,6 +488,16 @@ static struct snd_soc_dai_driver jz4780_i2s_dai = {
.ops = &jz4740_i2s_dai_ops,
};
+static const struct i2s_soc_info jz4770_i2s_soc_info = {
+ .version = JZ_I2S_JZ4770,
+ .dai = &jz4770_i2s_dai,
+};
+
+static const struct i2s_soc_info jz4780_i2s_soc_info = {
+ .version = JZ_I2S_JZ4780,
+ .dai = &jz4770_i2s_dai,
+};
+
static const struct snd_soc_component_driver jz4740_i2s_component = {
.name = "jz4740-i2s",
.suspend = jz4740_i2s_suspend,
@@ -483,8 +506,10 @@ static const struct snd_soc_component_driver jz4740_i2s_component = {
#ifdef CONFIG_OF
static const struct of_device_id jz4740_of_matches[] = {
- { .compatible = "ingenic,jz4740-i2s", .data = (void *)JZ_I2S_JZ4740 },
- { .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 },
+ { .compatible = "ingenic,jz4740-i2s", .data = &jz4740_i2s_soc_info },
+ { .compatible = "ingenic,jz4760-i2s", .data = &jz4760_i2s_soc_info },
+ { .compatible = "ingenic,jz4770-i2s", .data = &jz4770_i2s_soc_info },
+ { .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info },
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, jz4740_of_matches);
@@ -492,45 +517,40 @@ MODULE_DEVICE_TABLE(of, jz4740_of_matches);
static int jz4740_i2s_dev_probe(struct platform_device *pdev)
{
+ struct device *dev = &pdev->dev;
struct jz4740_i2s *i2s;
struct resource *mem;
int ret;
- i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
+ i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL);
if (!i2s)
return -ENOMEM;
- i2s->version =
- (enum jz47xx_i2s_version)of_device_get_match_data(&pdev->dev);
+ i2s->soc_info = device_get_match_data(dev);
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- i2s->base = devm_ioremap_resource(&pdev->dev, mem);
+ i2s->base = devm_ioremap_resource(dev, mem);
if (IS_ERR(i2s->base))
return PTR_ERR(i2s->base);
i2s->phys_base = mem->start;
- i2s->clk_aic = devm_clk_get(&pdev->dev, "aic");
+ i2s->clk_aic = devm_clk_get(dev, "aic");
if (IS_ERR(i2s->clk_aic))
return PTR_ERR(i2s->clk_aic);
- i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s");
+ i2s->clk_i2s = devm_clk_get(dev, "i2s");
if (IS_ERR(i2s->clk_i2s))
return PTR_ERR(i2s->clk_i2s);
platform_set_drvdata(pdev, i2s);
- if (i2s->version == JZ_I2S_JZ4780)
- ret = devm_snd_soc_register_component(&pdev->dev,
- &jz4740_i2s_component, &jz4780_i2s_dai, 1);
- else
- ret = devm_snd_soc_register_component(&pdev->dev,
- &jz4740_i2s_component, &jz4740_i2s_dai, 1);
-
+ ret = devm_snd_soc_register_component(dev, &jz4740_i2s_component,
+ i2s->soc_info->dai, 1);
if (ret)
return ret;
- return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
+ return devm_snd_dmaengine_pcm_register(dev, NULL,
SND_DMAENGINE_PCM_FLAG_COMPAT);
}
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
index 8c3c808bda9a..4f66b011f1b4 100644
--- a/sound/soc/kirkwood/armada-370-db.c
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -19,7 +19,7 @@ static int a370db_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
unsigned int freq;
switch (params_rate(params)) {
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index f882b4003edf..e037826b2451 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -20,7 +20,7 @@
static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
{
struct snd_soc_pcm_runtime *soc_runtime = subs->private_data;
- return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai);
+ return snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(soc_runtime, 0));
}
static const struct snd_pcm_hardware kirkwood_dma_snd_hw = {
diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
index 4254f3a954dd..375e3b492922 100644
--- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c
+++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
@@ -40,7 +40,7 @@ int mtk_afe_fe_startup(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
struct snd_pcm_runtime *runtime = substream->runtime;
- int memif_num = rtd->cpu_dai->id;
+ int memif_num = asoc_rtd_to_cpu(rtd, 0)->id;
struct mtk_base_afe_memif *memif = &afe->memif[memif_num];
const struct snd_pcm_hardware *mtk_afe_hardware = afe->mtk_afe_hardware;
int ret;
@@ -100,7 +100,7 @@ void mtk_afe_fe_shutdown(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
- struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id];
+ struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id];
int irq_id;
irq_id = memif->irq_usage;
@@ -122,7 +122,7 @@ int mtk_afe_fe_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
- int id = rtd->cpu_dai->id;
+ int id = asoc_rtd_to_cpu(rtd, 0)->id;
struct mtk_base_afe_memif *memif = &afe->memif[id];
int ret;
unsigned int channels = params_channels(params);
@@ -199,7 +199,7 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_pcm_runtime * const runtime = substream->runtime;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
- int id = rtd->cpu_dai->id;
+ int id = asoc_rtd_to_cpu(rtd, 0)->id;
struct mtk_base_afe_memif *memif = &afe->memif[id];
struct mtk_base_afe_irq *irqs = &afe->irqs[memif->irq_usage];
const struct mtk_base_irq_data *irq_data = irqs->irq_data;
@@ -265,7 +265,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
- int id = rtd->cpu_dai->id;
+ int id = asoc_rtd_to_cpu(rtd, 0)->id;
int pbuf_size;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
index 44dfef713905..0a1a65c86f0e 100644
--- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c
+++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c
@@ -82,7 +82,7 @@ snd_pcm_uframes_t mtk_afe_pcm_pointer(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
- struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id];
+ struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id];
const struct mtk_base_memif_data *memif_data = memif->data;
struct regmap *regmap = afe->regmap;
struct device *dev = afe->dev;
diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
index 488603a0c4b1..f0250b0dd734 100644
--- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
+++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c
@@ -497,7 +497,7 @@ static int mt2701_memif_fs(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int fs;
- if (rtd->cpu_dai->id != MT2701_MEMIF_ULBT)
+ if (asoc_rtd_to_cpu(rtd, 0)->id != MT2701_MEMIF_ULBT)
fs = mt2701_afe_i2s_fs(rate);
else
fs = (rate == 16000 ? 1 : 0);
diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
index b6941796efca..c47af9b6949b 100644
--- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c
+++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
@@ -128,8 +128,8 @@ static int mt2701_cs42448_be_ops_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
unsigned int mclk_rate;
unsigned int rate = params_rate(params);
unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4;
diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
index 8c4c89e4c616..0122e7df067f 100644
--- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c
+++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
@@ -25,8 +25,8 @@ static int mt2701_wm8960_be_ops_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int mclk_rate;
unsigned int rate = params_rate(params);
unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4;
diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
index 378bfc16ef52..7f930556d961 100644
--- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
+++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c
@@ -143,7 +143,7 @@ static int mt6797_memif_fs(struct snd_pcm_substream *substream,
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
- int id = rtd->cpu_dai->id;
+ int id = asoc_rtd_to_cpu(rtd, 0)->id;
return mt6797_rate_transform(afe->dev, rate, id);
}
diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
index 461e4de8c918..1e3f2d786066 100644
--- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
+++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c
@@ -485,7 +485,7 @@ static int mt8173_memif_fs(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
- struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id];
+ struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id];
int fs;
if (memif->data->id == MT8173_AFE_MEMIF_DAI ||
diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c
index 22c00600c999..37693d354e66 100644
--- a/sound/soc/mediatek/mt8173/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c
@@ -53,7 +53,7 @@ static int mt8173_max98090_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
return snd_soc_dai_set_sysclk(codec_dai, 0, params_rate(params) * 256,
SND_SOC_CLOCK_IN);
@@ -67,7 +67,7 @@ static int mt8173_max98090_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
struct snd_soc_card *card = runtime->card;
- struct snd_soc_component *component = runtime->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
/* enable jack detection */
ret = snd_soc_card_jack_new(card, "Headphone", SND_JACK_HEADPHONE,
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
index 2e1e61d8f127..51009a172777 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
@@ -47,7 +47,7 @@ static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai;
int i, ret;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
/* pll from mclk 12.288M */
ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
params_rate(params) * 512);
@@ -73,7 +73,7 @@ static struct snd_soc_jack mt8173_rt5650_rt5514_jack;
static int mt8173_rt5650_rt5514_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
- struct snd_soc_component *component = runtime->codec_dais[0]->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
int ret;
rt5645_sel_asrc_clk_src(component,
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index ebcc0b86286b..247ac7690805 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -51,7 +51,7 @@ static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai;
int i, ret;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
/* pll from mclk 12.288M */
ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
params_rate(params) * 512);
@@ -77,8 +77,8 @@ static struct snd_soc_jack mt8173_rt5650_rt5676_jack;
static int mt8173_rt5650_rt5676_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
- struct snd_soc_component *component = runtime->codec_dais[0]->component;
- struct snd_soc_component *component_sub = runtime->codec_dais[1]->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
+ struct snd_soc_component *component_sub = asoc_rtd_to_codec(runtime, 1)->component;
int ret;
rt5645_sel_asrc_clk_src(component,
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index ef6f23675286..2065c94dbf99 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -11,6 +11,7 @@
#include <linux/of_gpio.h>
#include <sound/soc.h>
#include <sound/jack.h>
+#include <sound/hdmi-codec.h>
#include "../../codecs/rt5645.h"
#define MCLK_FOR_CODECS 12288000
@@ -77,7 +78,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
break;
}
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
/* pll from mclk */
ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock,
params_rate(params) * 512);
@@ -98,13 +99,13 @@ static const struct snd_soc_ops mt8173_rt5650_ops = {
.hw_params = mt8173_rt5650_hw_params,
};
-static struct snd_soc_jack mt8173_rt5650_jack;
+static struct snd_soc_jack mt8173_rt5650_jack, mt8173_rt5650_hdmi_jack;
static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
- struct snd_soc_component *component = runtime->codec_dais[0]->component;
- const char *codec_capture_dai = runtime->codec_dais[1]->name;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
+ const char *codec_capture_dai = asoc_rtd_to_codec(runtime, 1)->name;
int ret;
rt5645_sel_asrc_clk_src(component,
@@ -144,6 +145,19 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
&mt8173_rt5650_jack);
}
+static int mt8173_rt5650_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+
+ ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT,
+ &mt8173_rt5650_hdmi_jack, NULL, 0);
+ if (ret)
+ return ret;
+
+ return hdmi_codec_set_jack_detect(asoc_rtd_to_codec(rtd, 0)->component,
+ &mt8173_rt5650_hdmi_jack);
+}
+
enum {
DAI_LINK_PLAYBACK,
DAI_LINK_CAPTURE,
@@ -222,6 +236,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = {
.name = "HDMI BE",
.no_pcm = 1,
.dpcm_playback = 1,
+ .init = mt8173_rt5650_hdmi_init,
SND_SOC_DAILINK_REG(hdmi_be),
},
};
diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c
index 6e2270bbb10e..c8ded53bde1d 100644
--- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c
+++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c
@@ -146,7 +146,7 @@ static int mt8183_memif_fs(struct snd_pcm_substream *substream,
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
- int id = rtd->cpu_dai->id;
+ int id = asoc_rtd_to_cpu(rtd, 0)->id;
return mt8183_rate_transform(afe->dev, rate, id);
}
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index c65493721e90..5b3dfa79b4ae 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -16,7 +16,9 @@
#include "../../codecs/da7219-aad.h"
#include "../../codecs/da7219.h"
-static struct snd_soc_jack headset_jack;
+struct mt8183_da7219_max98357_priv {
+ struct snd_soc_jack headset_jack;
+};
static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
@@ -26,7 +28,7 @@ static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
unsigned int mclk_fs_ratio = 128;
unsigned int mclk_fs = rate * mclk_fs_ratio;
- return snd_soc_dai_set_sysclk(rtd->cpu_dai,
+ return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0),
0, mclk_fs, SND_SOC_CLOCK_OUT);
}
@@ -38,19 +40,19 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
unsigned int rate = params_rate(params);
unsigned int mclk_fs_ratio = 256;
unsigned int mclk_fs = rate * mclk_fs_ratio;
unsigned int freq;
int ret = 0, j;
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0,
mclk_fs, SND_SOC_CLOCK_OUT);
if (ret < 0)
dev_err(rtd->dev, "failed to set cpu dai sysclk\n");
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, "da7219.5-001a")) {
ret = snd_soc_dai_set_sysclk(codec_dai,
@@ -80,10 +82,10 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream,
static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
int ret = 0, j;
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, "da7219.5-001a")) {
ret = snd_soc_dai_set_pll(codec_dai,
@@ -116,6 +118,46 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
+static int
+mt8183_da7219_max98357_bt_sco_startup(
+ struct snd_pcm_substream *substream)
+{
+ static const unsigned int rates[] = {
+ 8000, 16000
+ };
+ static const struct snd_pcm_hw_constraint_list constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+ static const unsigned int channels[] = {
+ 1,
+ };
+ static const struct snd_pcm_hw_constraint_list constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+ };
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+ runtime->hw.channels_max = 1;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+ return 0;
+}
+
+static const struct snd_soc_ops mt8183_da7219_max98357_bt_sco_ops = {
+ .startup = mt8183_da7219_max98357_bt_sco_startup,
+};
+
/* FE */
SND_SOC_DAILINK_DEFS(playback1,
DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
@@ -222,6 +264,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
SND_SOC_DPCM_TRIGGER_PRE},
.dynamic = 1,
.dpcm_playback = 1,
+ .ops = &mt8183_da7219_max98357_bt_sco_ops,
SND_SOC_DAILINK_REG(playback2),
},
{
@@ -240,6 +283,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
SND_SOC_DPCM_TRIGGER_PRE},
.dynamic = 1,
.dpcm_capture = 1,
+ .ops = &mt8183_da7219_max98357_bt_sco_ops,
SND_SOC_DAILINK_REG(capture1),
},
{
@@ -351,8 +395,12 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
{
.name = "TDM",
.no_pcm = 1,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_IB_IF |
+ SND_SOC_DAIFMT_CBM_CFM,
.dpcm_playback = 1,
.ignore_suspend = 1,
+ .be_hw_params_fixup = mt8183_i2s_hw_params_fixup,
SND_SOC_DAILINK_REG(tdm),
},
};
@@ -372,9 +420,31 @@ static struct snd_soc_codec_conf mt6358_codec_conf[] = {
},
};
+static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speakers"),
+};
+
+static const
+struct snd_soc_dapm_widget mt8183_da7219_max98357_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Speakers", NULL),
+ SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL",
+ "aud_tdm_out_on", "aud_tdm_out_off"),
+};
+
+static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = {
+ {"Speakers", NULL, "Speaker"},
+ {"I2S Playback", NULL, "TDM_OUT_PINCTRL"},
+};
+
static struct snd_soc_card mt8183_da7219_max98357_card = {
.name = "mt8183_da7219_max98357",
.owner = THIS_MODULE,
+ .controls = mt8183_da7219_max98357_snd_controls,
+ .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls),
+ .dapm_widgets = mt8183_da7219_max98357_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets),
+ .dapm_routes = mt8183_da7219_max98357_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes),
.dai_link = mt8183_da7219_max98357_dai_links,
.num_links = ARRAY_SIZE(mt8183_da7219_max98357_dai_links),
.aux_dev = &mt8183_da7219_max98357_headset_dev,
@@ -387,6 +457,8 @@ static int
mt8183_da7219_max98357_headset_init(struct snd_soc_component *component)
{
int ret;
+ struct mt8183_da7219_max98357_priv *priv =
+ snd_soc_card_get_drvdata(component->card);
/* Enable Headset and 4 Buttons Jack detection */
ret = snd_soc_card_jack_new(&mt8183_da7219_max98357_card,
@@ -394,12 +466,12 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component)
SND_JACK_HEADSET |
SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3,
- &headset_jack,
+ &priv->headset_jack,
NULL, 0);
if (ret)
return ret;
- da7219_aad_jack_det(component, &headset_jack);
+ da7219_aad_jack_det(component, &priv->headset_jack);
return ret;
}
@@ -409,7 +481,8 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev)
struct snd_soc_card *card = &mt8183_da7219_max98357_card;
struct device_node *platform_node;
struct snd_soc_dai_link *dai_link;
- struct pinctrl *default_pins;
+ struct mt8183_da7219_max98357_priv *priv;
+ struct pinctrl *pinctrl;
int ret, i;
card->dev = &pdev->dev;
@@ -436,22 +509,21 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev)
return -EINVAL;
}
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret) {
- dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ snd_soc_card_set_drvdata(card, priv);
+
+ pinctrl = devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT);
+ if (IS_ERR(pinctrl)) {
+ ret = PTR_ERR(pinctrl);
+ dev_err(&pdev->dev, "%s failed to select default state %d\n",
__func__, ret);
return ret;
}
- default_pins =
- devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT);
- if (IS_ERR(default_pins)) {
- dev_err(&pdev->dev, "%s set pins failed\n",
- __func__);
- return PTR_ERR(default_pins);
- }
-
- return ret;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
#ifdef CONFIG_OF
@@ -478,4 +550,3 @@ MODULE_DESCRIPTION("MT8183-DA7219-MAX98357 ALSA SoC machine driver");
MODULE_AUTHOR("Shunli Wang <shunli.wang@mediatek.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("mt8183_da7219_max98357 soc card");
-
diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
index 0555f7d73d05..1fca8df109b4 100644
--- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c
@@ -41,7 +41,7 @@ static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
unsigned int mclk_fs_ratio = 128;
unsigned int mclk_fs = rate * mclk_fs_ratio;
- return snd_soc_dai_set_sysclk(rtd->cpu_dai,
+ return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0),
0, mclk_fs, SND_SOC_CLOCK_OUT);
}
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 2e3676147cea..8b6295283989 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -2,6 +2,16 @@
menu "ASoC support for Amlogic platforms"
depends on ARCH_MESON || COMPILE_TEST
+config SND_MESON_AIU
+ tristate "Amlogic AIU"
+ select SND_MESON_CODEC_GLUE
+ select SND_PCM_IEC958
+ imply SND_SOC_MESON_T9015
+ imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI
+ help
+ Select Y or M to add support for the Audio output subsystem found
+ in the Amlogic Meson8, Meson8b and GX SoC families
+
config SND_MESON_AXG_FIFO
tristate
select REGMAP_MMIO
@@ -50,6 +60,7 @@ config SND_MESON_AXG_TDMOUT
config SND_MESON_AXG_SOUND_CARD
tristate "Amlogic AXG Sound Card Support"
select SND_MESON_AXG_TDM_INTERFACE
+ select SND_MESON_CARD_UTILS
imply SND_MESON_AXG_FRDDR
imply SND_MESON_AXG_TODDR
imply SND_MESON_AXG_TDMIN
@@ -85,11 +96,41 @@ config SND_MESON_AXG_PDM
Select Y or M to add support for PDM input embedded
in the Amlogic AXG SoC family
+config SND_MESON_CARD_UTILS
+ tristate
+
+config SND_MESON_CODEC_GLUE
+ tristate
+
+config SND_MESON_GX_SOUND_CARD
+ tristate "Amlogic GX Sound Card Support"
+ select SND_MESON_CARD_UTILS
+ imply SND_MESON_AIU
+ help
+ Select Y or M to add support for the GXBB/GXL SoC sound card
+
+config SND_MESON_G12A_TOACODEC
+ tristate "Amlogic G12A To Internal DAC Control Support"
+ select SND_MESON_CODEC_GLUE
+ select REGMAP_MMIO
+ imply SND_SOC_MESON_T9015
+ help
+ Select Y or M to add support for the internal audio DAC on the
+ g12a SoC family
+
config SND_MESON_G12A_TOHDMITX
tristate "Amlogic G12A To HDMI TX Control Support"
select REGMAP_MMIO
+ select SND_MESON_CODEC_GLUE
imply SND_SOC_HDMI_CODEC
help
Select Y or M to add support for HDMI audio on the g12a SoC
family
+
+config SND_SOC_MESON_T9015
+ tristate "Amlogic T9015 DAC"
+ select REGMAP_MMIO
+ help
+ Say Y or M if you want to add support for the internal DAC found
+ on GXL, G12 and SM1 SoC family.
endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
index 1a8b1470ed84..e446bc980481 100644
--- a/sound/soc/meson/Makefile
+++ b/sound/soc/meson/Makefile
@@ -1,5 +1,13 @@
# SPDX-License-Identifier: (GPL-2.0 OR MIT)
+snd-soc-meson-aiu-objs := aiu.o
+snd-soc-meson-aiu-objs += aiu-acodec-ctrl.o
+snd-soc-meson-aiu-objs += aiu-codec-ctrl.o
+snd-soc-meson-aiu-objs += aiu-encoder-i2s.o
+snd-soc-meson-aiu-objs += aiu-encoder-spdif.o
+snd-soc-meson-aiu-objs += aiu-fifo.o
+snd-soc-meson-aiu-objs += aiu-fifo-i2s.o
+snd-soc-meson-aiu-objs += aiu-fifo-spdif.o
snd-soc-meson-axg-fifo-objs := axg-fifo.o
snd-soc-meson-axg-frddr-objs := axg-frddr.o
snd-soc-meson-axg-toddr-objs := axg-toddr.o
@@ -11,8 +19,14 @@ snd-soc-meson-axg-sound-card-objs := axg-card.o
snd-soc-meson-axg-spdifin-objs := axg-spdifin.o
snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
snd-soc-meson-axg-pdm-objs := axg-pdm.o
+snd-soc-meson-card-utils-objs := meson-card-utils.o
+snd-soc-meson-codec-glue-objs := meson-codec-glue.o
+snd-soc-meson-gx-sound-card-objs := gx-card.o
+snd-soc-meson-g12a-toacodec-objs := g12a-toacodec.o
snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o
+snd-soc-meson-t9015-objs := t9015.o
+obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o
obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o
@@ -24,4 +38,9 @@ obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o
obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o
obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o
+obj-$(CONFIG_SND_MESON_CARD_UTILS) += snd-soc-meson-card-utils.o
+obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o
+obj-$(CONFIG_SND_MESON_GX_SOUND_CARD) += snd-soc-meson-gx-sound-card.o
+obj-$(CONFIG_SND_MESON_G12A_TOACODEC) += snd-soc-meson-g12a-toacodec.o
obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o
+obj-$(CONFIG_SND_SOC_MESON_T9015) += snd-soc-meson-t9015.o
diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c
new file mode 100644
index 000000000000..7078197e0cc5
--- /dev/null
+++ b/sound/soc/meson/aiu-acodec-ctrl.c
@@ -0,0 +1,203 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-aiu.h>
+#include "aiu.h"
+#include "meson-codec-glue.h"
+
+#define CTRL_DIN_EN 15
+#define CTRL_CLK_INV BIT(14)
+#define CTRL_LRCLK_INV BIT(13)
+#define CTRL_I2S_IN_BCLK_SRC BIT(11)
+#define CTRL_DIN_LRCLK_SRC_SHIFT 6
+#define CTRL_DIN_LRCLK_SRC (0x3 << CTRL_DIN_LRCLK_SRC_SHIFT)
+#define CTRL_BCLK_MCLK_SRC GENMASK(5, 4)
+#define CTRL_DIN_SKEW GENMASK(3, 2)
+#define CTRL_I2S_OUT_LANE_SRC 0
+
+#define AIU_ACODEC_OUT_CHMAX 2
+
+static const char * const aiu_acodec_ctrl_mux_texts[] = {
+ "DISABLED", "I2S", "PCM",
+};
+
+static int aiu_acodec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_kcontrol_component(kcontrol);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int mux, changed;
+
+ mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+ changed = snd_soc_component_test_bits(component, e->reg,
+ CTRL_DIN_LRCLK_SRC,
+ FIELD_PREP(CTRL_DIN_LRCLK_SRC,
+ mux));
+
+ if (!changed)
+ return 0;
+
+ /* Force disconnect of the mux while updating */
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+ snd_soc_component_update_bits(component, e->reg,
+ CTRL_DIN_LRCLK_SRC |
+ CTRL_BCLK_MCLK_SRC,
+ FIELD_PREP(CTRL_DIN_LRCLK_SRC, mux) |
+ FIELD_PREP(CTRL_BCLK_MCLK_SRC, mux));
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+ return 0;
+}
+
+static SOC_ENUM_SINGLE_DECL(aiu_acodec_ctrl_mux_enum, AIU_ACODEC_CTRL,
+ CTRL_DIN_LRCLK_SRC_SHIFT,
+ aiu_acodec_ctrl_mux_texts);
+
+static const struct snd_kcontrol_new aiu_acodec_ctrl_mux =
+ SOC_DAPM_ENUM_EXT("ACodec Source", aiu_acodec_ctrl_mux_enum,
+ snd_soc_dapm_get_enum_double,
+ aiu_acodec_ctrl_mux_put_enum);
+
+static const struct snd_kcontrol_new aiu_acodec_ctrl_out_enable =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", AIU_ACODEC_CTRL,
+ CTRL_DIN_EN, 1, 0);
+
+static const struct snd_soc_dapm_widget aiu_acodec_ctrl_widgets[] = {
+ SND_SOC_DAPM_MUX("ACODEC SRC", SND_SOC_NOPM, 0, 0,
+ &aiu_acodec_ctrl_mux),
+ SND_SOC_DAPM_SWITCH("ACODEC OUT EN", SND_SOC_NOPM, 0, 0,
+ &aiu_acodec_ctrl_out_enable),
+};
+
+static int aiu_acodec_ctrl_input_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct meson_codec_glue_input *data;
+ int ret;
+
+ ret = meson_codec_glue_input_hw_params(substream, params, dai);
+ if (ret)
+ return ret;
+
+ /* The glue will provide 1 lane out of the 4 to the output */
+ data = meson_codec_glue_input_get_data(dai);
+ data->params.channels_min = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX,
+ data->params.channels_min);
+ data->params.channels_max = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX,
+ data->params.channels_max);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops aiu_acodec_ctrl_input_ops = {
+ .hw_params = aiu_acodec_ctrl_input_hw_params,
+ .set_fmt = meson_codec_glue_input_set_fmt,
+};
+
+static const struct snd_soc_dai_ops aiu_acodec_ctrl_output_ops = {
+ .startup = meson_codec_glue_output_startup,
+};
+
+#define AIU_ACODEC_CTRL_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+#define AIU_ACODEC_STREAM(xname, xsuffix, xchmax) \
+{ \
+ .stream_name = xname " " xsuffix, \
+ .channels_min = 1, \
+ .channels_max = (xchmax), \
+ .rate_min = 5512, \
+ .rate_max = 192000, \
+ .formats = AIU_ACODEC_CTRL_FORMATS, \
+}
+
+#define AIU_ACODEC_INPUT(xname) { \
+ .name = "ACODEC CTRL " xname, \
+ .playback = AIU_ACODEC_STREAM(xname, "Playback", 8), \
+ .ops = &aiu_acodec_ctrl_input_ops, \
+ .probe = meson_codec_glue_input_dai_probe, \
+ .remove = meson_codec_glue_input_dai_remove, \
+}
+
+#define AIU_ACODEC_OUTPUT(xname) { \
+ .name = "ACODEC CTRL " xname, \
+ .capture = AIU_ACODEC_STREAM(xname, "Capture", AIU_ACODEC_OUT_CHMAX), \
+ .ops = &aiu_acodec_ctrl_output_ops, \
+}
+
+static struct snd_soc_dai_driver aiu_acodec_ctrl_dai_drv[] = {
+ [CTRL_I2S] = AIU_ACODEC_INPUT("ACODEC I2S IN"),
+ [CTRL_PCM] = AIU_ACODEC_INPUT("ACODEC PCM IN"),
+ [CTRL_OUT] = AIU_ACODEC_OUTPUT("ACODEC OUT"),
+};
+
+static const struct snd_soc_dapm_route aiu_acodec_ctrl_routes[] = {
+ { "ACODEC SRC", "I2S", "ACODEC I2S IN Playback" },
+ { "ACODEC SRC", "PCM", "ACODEC PCM IN Playback" },
+ { "ACODEC OUT EN", "Switch", "ACODEC SRC" },
+ { "ACODEC OUT Capture", NULL, "ACODEC OUT EN" },
+};
+
+static const struct snd_kcontrol_new aiu_acodec_ctrl_controls[] = {
+ SOC_SINGLE("ACODEC I2S Lane Select", AIU_ACODEC_CTRL,
+ CTRL_I2S_OUT_LANE_SRC, 3, 0),
+};
+
+static int aiu_acodec_of_xlate_dai_name(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name)
+{
+ return aiu_of_xlate_dai_name(component, args, dai_name, AIU_ACODEC);
+}
+
+static int aiu_acodec_ctrl_component_probe(struct snd_soc_component *component)
+{
+ /*
+ * NOTE: Din Skew setting
+ * According to the documentation, the following update adds one delay
+ * to the din line. Without this, the output saturates. This happens
+ * regardless of the link format (i2s or left_j) so it is not clear what
+ * it actually does but it seems to be required
+ */
+ snd_soc_component_update_bits(component, AIU_ACODEC_CTRL,
+ CTRL_DIN_SKEW,
+ FIELD_PREP(CTRL_DIN_SKEW, 2));
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver aiu_acodec_ctrl_component = {
+ .name = "AIU Internal DAC Codec Control",
+ .probe = aiu_acodec_ctrl_component_probe,
+ .controls = aiu_acodec_ctrl_controls,
+ .num_controls = ARRAY_SIZE(aiu_acodec_ctrl_controls),
+ .dapm_widgets = aiu_acodec_ctrl_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aiu_acodec_ctrl_widgets),
+ .dapm_routes = aiu_acodec_ctrl_routes,
+ .num_dapm_routes = ARRAY_SIZE(aiu_acodec_ctrl_routes),
+ .of_xlate_dai_name = aiu_acodec_of_xlate_dai_name,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+int aiu_acodec_ctrl_register_component(struct device *dev)
+{
+ return snd_soc_register_component(dev, &aiu_acodec_ctrl_component,
+ aiu_acodec_ctrl_dai_drv,
+ ARRAY_SIZE(aiu_acodec_ctrl_dai_drv));
+}
diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c
new file mode 100644
index 000000000000..4b773d3e8b07
--- /dev/null
+++ b/sound/soc/meson/aiu-codec-ctrl.c
@@ -0,0 +1,151 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-aiu.h>
+#include "aiu.h"
+#include "meson-codec-glue.h"
+
+#define CTRL_CLK_SEL GENMASK(1, 0)
+#define CTRL_DATA_SEL_SHIFT 4
+#define CTRL_DATA_SEL (0x3 << CTRL_DATA_SEL_SHIFT)
+
+static const char * const aiu_codec_ctrl_mux_texts[] = {
+ "DISABLED", "PCM", "I2S",
+};
+
+static int aiu_codec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_kcontrol_component(kcontrol);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int mux, changed;
+
+ mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+ changed = snd_soc_component_test_bits(component, e->reg,
+ CTRL_DATA_SEL,
+ FIELD_PREP(CTRL_DATA_SEL, mux));
+
+ if (!changed)
+ return 0;
+
+ /* Force disconnect of the mux while updating */
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+ /* Reset the source first */
+ snd_soc_component_update_bits(component, e->reg,
+ CTRL_CLK_SEL |
+ CTRL_DATA_SEL,
+ FIELD_PREP(CTRL_CLK_SEL, 0) |
+ FIELD_PREP(CTRL_DATA_SEL, 0));
+
+ /* Set the appropriate source */
+ snd_soc_component_update_bits(component, e->reg,
+ CTRL_CLK_SEL |
+ CTRL_DATA_SEL,
+ FIELD_PREP(CTRL_CLK_SEL, mux) |
+ FIELD_PREP(CTRL_DATA_SEL, mux));
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+ return 0;
+}
+
+static SOC_ENUM_SINGLE_DECL(aiu_hdmi_ctrl_mux_enum, AIU_HDMI_CLK_DATA_CTRL,
+ CTRL_DATA_SEL_SHIFT,
+ aiu_codec_ctrl_mux_texts);
+
+static const struct snd_kcontrol_new aiu_hdmi_ctrl_mux =
+ SOC_DAPM_ENUM_EXT("HDMI Source", aiu_hdmi_ctrl_mux_enum,
+ snd_soc_dapm_get_enum_double,
+ aiu_codec_ctrl_mux_put_enum);
+
+static const struct snd_soc_dapm_widget aiu_hdmi_ctrl_widgets[] = {
+ SND_SOC_DAPM_MUX("HDMI CTRL SRC", SND_SOC_NOPM, 0, 0,
+ &aiu_hdmi_ctrl_mux),
+};
+
+static const struct snd_soc_dai_ops aiu_codec_ctrl_input_ops = {
+ .hw_params = meson_codec_glue_input_hw_params,
+ .set_fmt = meson_codec_glue_input_set_fmt,
+};
+
+static const struct snd_soc_dai_ops aiu_codec_ctrl_output_ops = {
+ .startup = meson_codec_glue_output_startup,
+};
+
+#define AIU_CODEC_CTRL_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+#define AIU_CODEC_CTRL_STREAM(xname, xsuffix) \
+{ \
+ .stream_name = xname " " xsuffix, \
+ .channels_min = 1, \
+ .channels_max = 8, \
+ .rate_min = 5512, \
+ .rate_max = 192000, \
+ .formats = AIU_CODEC_CTRL_FORMATS, \
+}
+
+#define AIU_CODEC_CTRL_INPUT(xname) { \
+ .name = "CODEC CTRL " xname, \
+ .playback = AIU_CODEC_CTRL_STREAM(xname, "Playback"), \
+ .ops = &aiu_codec_ctrl_input_ops, \
+ .probe = meson_codec_glue_input_dai_probe, \
+ .remove = meson_codec_glue_input_dai_remove, \
+}
+
+#define AIU_CODEC_CTRL_OUTPUT(xname) { \
+ .name = "CODEC CTRL " xname, \
+ .capture = AIU_CODEC_CTRL_STREAM(xname, "Capture"), \
+ .ops = &aiu_codec_ctrl_output_ops, \
+}
+
+static struct snd_soc_dai_driver aiu_hdmi_ctrl_dai_drv[] = {
+ [CTRL_I2S] = AIU_CODEC_CTRL_INPUT("HDMI I2S IN"),
+ [CTRL_PCM] = AIU_CODEC_CTRL_INPUT("HDMI PCM IN"),
+ [CTRL_OUT] = AIU_CODEC_CTRL_OUTPUT("HDMI OUT"),
+};
+
+static const struct snd_soc_dapm_route aiu_hdmi_ctrl_routes[] = {
+ { "HDMI CTRL SRC", "I2S", "HDMI I2S IN Playback" },
+ { "HDMI CTRL SRC", "PCM", "HDMI PCM IN Playback" },
+ { "HDMI OUT Capture", NULL, "HDMI CTRL SRC" },
+};
+
+static int aiu_hdmi_of_xlate_dai_name(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name)
+{
+ return aiu_of_xlate_dai_name(component, args, dai_name, AIU_HDMI);
+}
+
+static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = {
+ .name = "AIU HDMI Codec Control",
+ .dapm_widgets = aiu_hdmi_ctrl_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aiu_hdmi_ctrl_widgets),
+ .dapm_routes = aiu_hdmi_ctrl_routes,
+ .num_dapm_routes = ARRAY_SIZE(aiu_hdmi_ctrl_routes),
+ .of_xlate_dai_name = aiu_hdmi_of_xlate_dai_name,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+int aiu_hdmi_ctrl_register_component(struct device *dev)
+{
+ return snd_soc_register_component(dev, &aiu_hdmi_ctrl_component,
+ aiu_hdmi_ctrl_dai_drv,
+ ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv));
+}
+
diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c
new file mode 100644
index 000000000000..832e22d275fe
--- /dev/null
+++ b/sound/soc/meson/aiu-encoder-i2s.c
@@ -0,0 +1,365 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+
+#define AIU_I2S_SOURCE_DESC_MODE_8CH BIT(0)
+#define AIU_I2S_SOURCE_DESC_MODE_24BIT BIT(5)
+#define AIU_I2S_SOURCE_DESC_MODE_32BIT BIT(9)
+#define AIU_I2S_SOURCE_DESC_MODE_SPLIT BIT(11)
+#define AIU_RST_SOFT_I2S_FAST BIT(0)
+
+#define AIU_I2S_DAC_CFG_MSB_FIRST BIT(2)
+#define AIU_I2S_MISC_HOLD_EN BIT(2)
+#define AIU_CLK_CTRL_I2S_DIV_EN BIT(0)
+#define AIU_CLK_CTRL_I2S_DIV GENMASK(3, 2)
+#define AIU_CLK_CTRL_AOCLK_INVERT BIT(6)
+#define AIU_CLK_CTRL_LRCLK_INVERT BIT(7)
+#define AIU_CLK_CTRL_LRCLK_SKEW GENMASK(9, 8)
+#define AIU_CLK_CTRL_MORE_HDMI_AMCLK BIT(6)
+#define AIU_CLK_CTRL_MORE_I2S_DIV GENMASK(5, 0)
+#define AIU_CODEC_DAC_LRCLK_CTRL_DIV GENMASK(11, 0)
+
+static void aiu_encoder_i2s_divider_enable(struct snd_soc_component *component,
+ bool enable)
+{
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+ AIU_CLK_CTRL_I2S_DIV_EN,
+ enable ? AIU_CLK_CTRL_I2S_DIV_EN : 0);
+}
+
+static void aiu_encoder_i2s_hold(struct snd_soc_component *component,
+ bool enable)
+{
+ snd_soc_component_update_bits(component, AIU_I2S_MISC,
+ AIU_I2S_MISC_HOLD_EN,
+ enable ? AIU_I2S_MISC_HOLD_EN : 0);
+}
+
+static int aiu_encoder_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ aiu_encoder_i2s_hold(component, false);
+ return 0;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ aiu_encoder_i2s_hold(component, true);
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component,
+ struct snd_pcm_hw_params *params)
+{
+ /* Always operate in split (classic interleaved) mode */
+ unsigned int desc = AIU_I2S_SOURCE_DESC_MODE_SPLIT;
+ unsigned int val;
+
+ /* Reset required to update the pipeline */
+ snd_soc_component_write(component, AIU_RST_SOFT, AIU_RST_SOFT_I2S_FAST);
+ snd_soc_component_read(component, AIU_I2S_SYNC, &val);
+
+ switch (params_physical_width(params)) {
+ case 16: /* Nothing to do */
+ break;
+
+ case 32:
+ desc |= (AIU_I2S_SOURCE_DESC_MODE_24BIT |
+ AIU_I2S_SOURCE_DESC_MODE_32BIT);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ switch (params_channels(params)) {
+ case 2: /* Nothing to do */
+ break;
+ case 8:
+ desc |= AIU_I2S_SOURCE_DESC_MODE_8CH;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, AIU_I2S_SOURCE_DESC,
+ AIU_I2S_SOURCE_DESC_MODE_8CH |
+ AIU_I2S_SOURCE_DESC_MODE_24BIT |
+ AIU_I2S_SOURCE_DESC_MODE_32BIT |
+ AIU_I2S_SOURCE_DESC_MODE_SPLIT,
+ desc);
+
+ return 0;
+}
+
+static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component,
+ struct snd_pcm_hw_params *params,
+ unsigned int bs)
+{
+ switch (bs) {
+ case 1:
+ case 2:
+ case 4:
+ case 8:
+ /* These are the only valid legacy dividers */
+ break;
+
+ default:
+ dev_err(component->dev, "Unsupported i2s divider: %u\n", bs);
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+ AIU_CLK_CTRL_I2S_DIV,
+ FIELD_PREP(AIU_CLK_CTRL_I2S_DIV,
+ __ffs(bs)));
+
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE,
+ AIU_CLK_CTRL_MORE_I2S_DIV,
+ FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV,
+ 0));
+
+ return 0;
+}
+
+static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component,
+ struct snd_pcm_hw_params *params,
+ unsigned int bs)
+{
+ /*
+ * NOTE: this HW is odd.
+ * In most configuration, the i2s divider is 'mclk / blck'.
+ * However, in 16 bits - 8ch mode, this factor needs to be
+ * increased by 50% to get the correct output rate.
+ * No idea why !
+ */
+ if (params_width(params) == 16 && params_channels(params) == 8) {
+ if (bs % 2) {
+ dev_err(component->dev,
+ "Cannot increase i2s divider by 50%%\n");
+ return -EINVAL;
+ }
+ bs += bs / 2;
+ }
+
+ /* Use CLK_MORE for mclk to bclk divider */
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+ AIU_CLK_CTRL_I2S_DIV,
+ FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0));
+
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE,
+ AIU_CLK_CTRL_MORE_I2S_DIV,
+ FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV,
+ bs - 1));
+
+ return 0;
+}
+
+static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component,
+ struct snd_pcm_hw_params *params)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(component);
+ unsigned int srate = params_rate(params);
+ unsigned int fs, bs;
+ int ret;
+
+ /* Get the oversampling factor */
+ fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate);
+
+ if (fs % 64)
+ return -EINVAL;
+
+ /* Send data MSB first */
+ snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG,
+ AIU_I2S_DAC_CFG_MSB_FIRST,
+ AIU_I2S_DAC_CFG_MSB_FIRST);
+
+ /* Set bclk to lrlck ratio */
+ snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL,
+ AIU_CODEC_DAC_LRCLK_CTRL_DIV,
+ FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV,
+ 64 - 1));
+
+ bs = fs / 64;
+
+ if (aiu->platform->has_clk_ctrl_more_i2s_div)
+ ret = aiu_encoder_i2s_set_more_div(component, params, bs);
+ else
+ ret = aiu_encoder_i2s_set_legacy_div(component, params, bs);
+
+ if (ret)
+ return ret;
+
+ /* Make sure amclk is used for HDMI i2s as well */
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE,
+ AIU_CLK_CTRL_MORE_HDMI_AMCLK,
+ AIU_CLK_CTRL_MORE_HDMI_AMCLK);
+
+ return 0;
+}
+
+static int aiu_encoder_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ int ret;
+
+ /* Disable the clock while changing the settings */
+ aiu_encoder_i2s_divider_enable(component, false);
+
+ ret = aiu_encoder_i2s_setup_desc(component, params);
+ if (ret) {
+ dev_err(dai->dev, "setting i2s desc failed\n");
+ return ret;
+ }
+
+ ret = aiu_encoder_i2s_set_clocks(component, params);
+ if (ret) {
+ dev_err(dai->dev, "setting i2s clocks failed\n");
+ return ret;
+ }
+
+ aiu_encoder_i2s_divider_enable(component, true);
+
+ return 0;
+}
+
+static int aiu_encoder_i2s_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+
+ aiu_encoder_i2s_divider_enable(component, false);
+
+ return 0;
+}
+
+static int aiu_encoder_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_component *component = dai->component;
+ unsigned int inv = fmt & SND_SOC_DAIFMT_INV_MASK;
+ unsigned int val = 0;
+ unsigned int skew;
+
+ /* Only CPU Master / Codec Slave supported ATM */
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+ return -EINVAL;
+
+ if (inv == SND_SOC_DAIFMT_NB_IF ||
+ inv == SND_SOC_DAIFMT_IB_IF)
+ val |= AIU_CLK_CTRL_LRCLK_INVERT;
+
+ if (inv == SND_SOC_DAIFMT_IB_NF ||
+ inv == SND_SOC_DAIFMT_IB_IF)
+ val |= AIU_CLK_CTRL_AOCLK_INVERT;
+
+ /* Signal skew */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* Invert sample clock for i2s */
+ val ^= AIU_CLK_CTRL_LRCLK_INVERT;
+ skew = 1;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ skew = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ val |= FIELD_PREP(AIU_CLK_CTRL_LRCLK_SKEW, skew);
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+ AIU_CLK_CTRL_LRCLK_INVERT |
+ AIU_CLK_CTRL_AOCLK_INVERT |
+ AIU_CLK_CTRL_LRCLK_SKEW,
+ val);
+
+ return 0;
+}
+
+static int aiu_encoder_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+ int ret;
+
+ if (WARN_ON(clk_id != 0))
+ return -EINVAL;
+
+ if (dir == SND_SOC_CLOCK_IN)
+ return 0;
+
+ ret = clk_set_rate(aiu->i2s.clks[MCLK].clk, freq);
+ if (ret)
+ dev_err(dai->dev, "Failed to set sysclk to %uHz", freq);
+
+ return ret;
+}
+
+static const unsigned int hw_channels[] = {2, 8};
+static const struct snd_pcm_hw_constraint_list hw_channel_constraints = {
+ .list = hw_channels,
+ .count = ARRAY_SIZE(hw_channels),
+ .mask = 0,
+};
+
+static int aiu_encoder_i2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+ int ret;
+
+ /* Make sure the encoder gets either 2 or 8 channels */
+ ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &hw_channel_constraints);
+ if (ret) {
+ dev_err(dai->dev, "adding channels constraints failed\n");
+ return ret;
+ }
+
+ ret = clk_bulk_prepare_enable(aiu->i2s.clk_num, aiu->i2s.clks);
+ if (ret)
+ dev_err(dai->dev, "failed to enable i2s clocks\n");
+
+ return ret;
+}
+
+static void aiu_encoder_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+
+ clk_bulk_disable_unprepare(aiu->i2s.clk_num, aiu->i2s.clks);
+}
+
+const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops = {
+ .trigger = aiu_encoder_i2s_trigger,
+ .hw_params = aiu_encoder_i2s_hw_params,
+ .hw_free = aiu_encoder_i2s_hw_free,
+ .set_fmt = aiu_encoder_i2s_set_fmt,
+ .set_sysclk = aiu_encoder_i2s_set_sysclk,
+ .startup = aiu_encoder_i2s_startup,
+ .shutdown = aiu_encoder_i2s_shutdown,
+};
+
diff --git a/sound/soc/meson/aiu-encoder-spdif.c b/sound/soc/meson/aiu-encoder-spdif.c
new file mode 100644
index 000000000000..de850913975f
--- /dev/null
+++ b/sound/soc/meson/aiu-encoder-spdif.c
@@ -0,0 +1,209 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm_iec958.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+
+#define AIU_958_MISC_NON_PCM BIT(0)
+#define AIU_958_MISC_MODE_16BITS BIT(1)
+#define AIU_958_MISC_16BITS_ALIGN GENMASK(6, 5)
+#define AIU_958_MISC_MODE_32BITS BIT(7)
+#define AIU_958_MISC_U_FROM_STREAM BIT(12)
+#define AIU_958_MISC_FORCE_LR BIT(13)
+#define AIU_958_CTRL_HOLD_EN BIT(0)
+#define AIU_CLK_CTRL_958_DIV_EN BIT(1)
+#define AIU_CLK_CTRL_958_DIV GENMASK(5, 4)
+#define AIU_CLK_CTRL_958_DIV_MORE BIT(12)
+
+#define AIU_CS_WORD_LEN 4
+#define AIU_958_INTERNAL_DIV 2
+
+static void
+aiu_encoder_spdif_divider_enable(struct snd_soc_component *component,
+ bool enable)
+{
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+ AIU_CLK_CTRL_958_DIV_EN,
+ enable ? AIU_CLK_CTRL_958_DIV_EN : 0);
+}
+
+static void aiu_encoder_spdif_hold(struct snd_soc_component *component,
+ bool enable)
+{
+ snd_soc_component_update_bits(component, AIU_958_CTRL,
+ AIU_958_CTRL_HOLD_EN,
+ enable ? AIU_958_CTRL_HOLD_EN : 0);
+}
+
+static int
+aiu_encoder_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ aiu_encoder_spdif_hold(component, false);
+ return 0;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ aiu_encoder_spdif_hold(component, true);
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int aiu_encoder_spdif_setup_cs_word(struct snd_soc_component *component,
+ struct snd_pcm_hw_params *params)
+{
+ u8 cs[AIU_CS_WORD_LEN];
+ unsigned int val;
+ int ret;
+
+ ret = snd_pcm_create_iec958_consumer_hw_params(params, cs,
+ AIU_CS_WORD_LEN);
+ if (ret < 0)
+ return ret;
+
+ /* Write the 1st half word */
+ val = cs[1] | cs[0] << 8;
+ snd_soc_component_write(component, AIU_958_CHSTAT_L0, val);
+ snd_soc_component_write(component, AIU_958_CHSTAT_R0, val);
+
+ /* Write the 2nd half word */
+ val = cs[3] | cs[2] << 8;
+ snd_soc_component_write(component, AIU_958_CHSTAT_L1, val);
+ snd_soc_component_write(component, AIU_958_CHSTAT_R1, val);
+
+ return 0;
+}
+
+static int aiu_encoder_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct aiu *aiu = snd_soc_component_get_drvdata(component);
+ unsigned int val = 0, mrate;
+ int ret;
+
+ /* Disable the clock while changing the settings */
+ aiu_encoder_spdif_divider_enable(component, false);
+
+ switch (params_physical_width(params)) {
+ case 16:
+ val |= AIU_958_MISC_MODE_16BITS;
+ val |= FIELD_PREP(AIU_958_MISC_16BITS_ALIGN, 2);
+ break;
+ case 32:
+ val |= AIU_958_MISC_MODE_32BITS;
+ break;
+ default:
+ dev_err(dai->dev, "Unsupport physical width\n");
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, AIU_958_MISC,
+ AIU_958_MISC_NON_PCM |
+ AIU_958_MISC_MODE_16BITS |
+ AIU_958_MISC_16BITS_ALIGN |
+ AIU_958_MISC_MODE_32BITS |
+ AIU_958_MISC_FORCE_LR |
+ AIU_958_MISC_U_FROM_STREAM,
+ val);
+
+ /* Set the stream channel status word */
+ ret = aiu_encoder_spdif_setup_cs_word(component, params);
+ if (ret) {
+ dev_err(dai->dev, "failed to set channel status word\n");
+ return ret;
+ }
+
+ snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+ AIU_CLK_CTRL_958_DIV |
+ AIU_CLK_CTRL_958_DIV_MORE,
+ FIELD_PREP(AIU_CLK_CTRL_958_DIV,
+ __ffs(AIU_958_INTERNAL_DIV)));
+
+ /* 2 * 32bits per subframe * 2 channels = 128 */
+ mrate = params_rate(params) * 128 * AIU_958_INTERNAL_DIV;
+ ret = clk_set_rate(aiu->spdif.clks[MCLK].clk, mrate);
+ if (ret) {
+ dev_err(dai->dev, "failed to set mclk rate\n");
+ return ret;
+ }
+
+ aiu_encoder_spdif_divider_enable(component, true);
+
+ return 0;
+}
+
+static int aiu_encoder_spdif_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+
+ aiu_encoder_spdif_divider_enable(component, false);
+
+ return 0;
+}
+
+static int aiu_encoder_spdif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+ int ret;
+
+ /*
+ * NOTE: Make sure the spdif block is on its own divider.
+ *
+ * The spdif can be clocked by the i2s master clock or its own
+ * clock. We should (in theory) change the source depending on the
+ * origin of the data.
+ *
+ * However, considering the clocking scheme used on these platforms,
+ * the master clocks will pick the same PLL source when they are
+ * playing from the same FIFO. The clock should be in sync so, it
+ * should not be necessary to reparent the spdif master clock.
+ */
+ ret = clk_set_parent(aiu->spdif.clks[MCLK].clk,
+ aiu->spdif_mclk);
+ if (ret)
+ return ret;
+
+ ret = clk_bulk_prepare_enable(aiu->spdif.clk_num, aiu->spdif.clks);
+ if (ret)
+ dev_err(dai->dev, "failed to enable spdif clocks\n");
+
+ return ret;
+}
+
+static void aiu_encoder_spdif_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+
+ clk_bulk_disable_unprepare(aiu->spdif.clk_num, aiu->spdif.clks);
+}
+
+const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops = {
+ .trigger = aiu_encoder_spdif_trigger,
+ .hw_params = aiu_encoder_spdif_hw_params,
+ .hw_free = aiu_encoder_spdif_hw_free,
+ .startup = aiu_encoder_spdif_startup,
+ .shutdown = aiu_encoder_spdif_shutdown,
+};
diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c
new file mode 100644
index 000000000000..9a5271ce80fe
--- /dev/null
+++ b/sound/soc/meson/aiu-fifo-i2s.c
@@ -0,0 +1,153 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+#include "aiu-fifo.h"
+
+#define AIU_I2S_SOURCE_DESC_MODE_8CH BIT(0)
+#define AIU_I2S_SOURCE_DESC_MODE_24BIT BIT(5)
+#define AIU_I2S_SOURCE_DESC_MODE_32BIT BIT(9)
+#define AIU_I2S_SOURCE_DESC_MODE_SPLIT BIT(11)
+#define AIU_MEM_I2S_MASKS_IRQ_BLOCK GENMASK(31, 16)
+#define AIU_MEM_I2S_CONTROL_MODE_16BIT BIT(6)
+#define AIU_MEM_I2S_BUF_CNTL_INIT BIT(0)
+#define AIU_RST_SOFT_I2S_FAST BIT(0)
+
+#define AIU_FIFO_I2S_BLOCK 256
+
+static struct snd_pcm_hardware fifo_i2s_pcm = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = AIU_FORMATS,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 8,
+ .period_bytes_min = AIU_FIFO_I2S_BLOCK,
+ .period_bytes_max = AIU_FIFO_I2S_BLOCK * USHRT_MAX,
+ .periods_min = 2,
+ .periods_max = UINT_MAX,
+
+ /* No real justification for this */
+ .buffer_bytes_max = 1 * 1024 * 1024,
+};
+
+static int aiu_fifo_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ unsigned int val;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ snd_soc_component_write(component, AIU_RST_SOFT,
+ AIU_RST_SOFT_I2S_FAST);
+ snd_soc_component_read(component, AIU_I2S_SYNC, &val);
+ break;
+ }
+
+ return aiu_fifo_trigger(substream, cmd, dai);
+}
+
+static int aiu_fifo_i2s_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ int ret;
+
+ ret = aiu_fifo_prepare(substream, dai);
+ if (ret)
+ return ret;
+
+ snd_soc_component_update_bits(component,
+ AIU_MEM_I2S_BUF_CNTL,
+ AIU_MEM_I2S_BUF_CNTL_INIT,
+ AIU_MEM_I2S_BUF_CNTL_INIT);
+ snd_soc_component_update_bits(component,
+ AIU_MEM_I2S_BUF_CNTL,
+ AIU_MEM_I2S_BUF_CNTL_INIT, 0);
+
+ return 0;
+}
+
+static int aiu_fifo_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+ unsigned int val;
+ int ret;
+
+ ret = aiu_fifo_hw_params(substream, params, dai);
+ if (ret)
+ return ret;
+
+ switch (params_physical_width(params)) {
+ case 16:
+ val = AIU_MEM_I2S_CONTROL_MODE_16BIT;
+ break;
+ case 32:
+ val = 0;
+ break;
+ default:
+ dev_err(dai->dev, "Unsupported physical width %u\n",
+ params_physical_width(params));
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, AIU_MEM_I2S_CONTROL,
+ AIU_MEM_I2S_CONTROL_MODE_16BIT,
+ val);
+
+ /* Setup the irq periodicity */
+ val = params_period_bytes(params) / fifo->fifo_block;
+ val = FIELD_PREP(AIU_MEM_I2S_MASKS_IRQ_BLOCK, val);
+ snd_soc_component_update_bits(component, AIU_MEM_I2S_MASKS,
+ AIU_MEM_I2S_MASKS_IRQ_BLOCK, val);
+
+ return 0;
+}
+
+const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops = {
+ .trigger = aiu_fifo_i2s_trigger,
+ .prepare = aiu_fifo_i2s_prepare,
+ .hw_params = aiu_fifo_i2s_hw_params,
+ .hw_free = aiu_fifo_hw_free,
+ .startup = aiu_fifo_startup,
+ .shutdown = aiu_fifo_shutdown,
+};
+
+int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct aiu *aiu = snd_soc_component_get_drvdata(component);
+ struct aiu_fifo *fifo;
+ int ret;
+
+ ret = aiu_fifo_dai_probe(dai);
+ if (ret)
+ return ret;
+
+ fifo = dai->playback_dma_data;
+
+ fifo->pcm = &fifo_i2s_pcm;
+ fifo->mem_offset = AIU_MEM_I2S_START;
+ fifo->fifo_block = AIU_FIFO_I2S_BLOCK;
+ fifo->pclk = aiu->i2s.clks[PCLK].clk;
+ fifo->irq = aiu->i2s.irq;
+
+ return 0;
+}
diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c
new file mode 100644
index 000000000000..44eb6faacf44
--- /dev/null
+++ b/sound/soc/meson/aiu-fifo-spdif.c
@@ -0,0 +1,186 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+#include "aiu-fifo.h"
+
+#define AIU_IEC958_DCU_FF_CTRL_EN BIT(0)
+#define AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE BIT(1)
+#define AIU_IEC958_DCU_FF_CTRL_IRQ_MODE GENMASK(3, 2)
+#define AIU_IEC958_DCU_FF_CTRL_IRQ_OUT_THD BIT(2)
+#define AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ BIT(3)
+#define AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN BIT(4)
+#define AIU_IEC958_DCU_FF_CTRL_BYTE_SEEK BIT(5)
+#define AIU_IEC958_DCU_FF_CTRL_CONTINUE BIT(6)
+#define AIU_MEM_IEC958_CONTROL_ENDIAN GENMASK(5, 3)
+#define AIU_MEM_IEC958_CONTROL_RD_DDR BIT(6)
+#define AIU_MEM_IEC958_CONTROL_MODE_16BIT BIT(7)
+#define AIU_MEM_IEC958_CONTROL_MODE_LINEAR BIT(8)
+#define AIU_MEM_IEC958_BUF_CNTL_INIT BIT(0)
+
+#define AIU_FIFO_SPDIF_BLOCK 8
+
+static struct snd_pcm_hardware fifo_spdif_pcm = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = AIU_FORMATS,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .period_bytes_min = AIU_FIFO_SPDIF_BLOCK,
+ .period_bytes_max = AIU_FIFO_SPDIF_BLOCK * USHRT_MAX,
+ .periods_min = 2,
+ .periods_max = UINT_MAX,
+
+ /* No real justification for this */
+ .buffer_bytes_max = 1 * 1024 * 1024,
+};
+
+static void fifo_spdif_dcu_enable(struct snd_soc_component *component,
+ bool enable)
+{
+ snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL,
+ AIU_IEC958_DCU_FF_CTRL_EN,
+ enable ? AIU_IEC958_DCU_FF_CTRL_EN : 0);
+}
+
+static int fifo_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ int ret;
+
+ ret = aiu_fifo_trigger(substream, cmd, dai);
+ if (ret)
+ return ret;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ fifo_spdif_dcu_enable(component, true);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_STOP:
+ fifo_spdif_dcu_enable(component, false);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int fifo_spdif_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ int ret;
+
+ ret = aiu_fifo_prepare(substream, dai);
+ if (ret)
+ return ret;
+
+ snd_soc_component_update_bits(component,
+ AIU_MEM_IEC958_BUF_CNTL,
+ AIU_MEM_IEC958_BUF_CNTL_INIT,
+ AIU_MEM_IEC958_BUF_CNTL_INIT);
+ snd_soc_component_update_bits(component,
+ AIU_MEM_IEC958_BUF_CNTL,
+ AIU_MEM_IEC958_BUF_CNTL_INIT, 0);
+
+ return 0;
+}
+
+static int fifo_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ unsigned int val;
+ int ret;
+
+ ret = aiu_fifo_hw_params(substream, params, dai);
+ if (ret)
+ return ret;
+
+ val = AIU_MEM_IEC958_CONTROL_RD_DDR |
+ AIU_MEM_IEC958_CONTROL_MODE_LINEAR;
+
+ switch (params_physical_width(params)) {
+ case 16:
+ val |= AIU_MEM_IEC958_CONTROL_MODE_16BIT;
+ break;
+ case 32:
+ break;
+ default:
+ dev_err(dai->dev, "Unsupported physical width %u\n",
+ params_physical_width(params));
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, AIU_MEM_IEC958_CONTROL,
+ AIU_MEM_IEC958_CONTROL_ENDIAN |
+ AIU_MEM_IEC958_CONTROL_RD_DDR |
+ AIU_MEM_IEC958_CONTROL_MODE_LINEAR |
+ AIU_MEM_IEC958_CONTROL_MODE_16BIT,
+ val);
+
+ /* Number bytes read by the FIFO between each IRQ */
+ snd_soc_component_write(component, AIU_IEC958_BPF,
+ params_period_bytes(params));
+
+ /*
+ * AUTO_DISABLE and SYNC_HEAD are enabled by default but
+ * this should be disabled in PCM (uncompressed) mode
+ */
+ snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL,
+ AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE |
+ AIU_IEC958_DCU_FF_CTRL_IRQ_MODE |
+ AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN,
+ AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ);
+
+ return 0;
+}
+
+const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops = {
+ .trigger = fifo_spdif_trigger,
+ .prepare = fifo_spdif_prepare,
+ .hw_params = fifo_spdif_hw_params,
+ .hw_free = aiu_fifo_hw_free,
+ .startup = aiu_fifo_startup,
+ .shutdown = aiu_fifo_shutdown,
+};
+
+int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct aiu *aiu = snd_soc_component_get_drvdata(component);
+ struct aiu_fifo *fifo;
+ int ret;
+
+ ret = aiu_fifo_dai_probe(dai);
+ if (ret)
+ return ret;
+
+ fifo = dai->playback_dma_data;
+
+ fifo->pcm = &fifo_spdif_pcm;
+ fifo->mem_offset = AIU_MEM_IEC958_START;
+ fifo->fifo_block = 1;
+ fifo->pclk = aiu->spdif.clks[PCLK].clk;
+ fifo->irq = aiu->spdif.irq;
+
+ return 0;
+}
diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c
new file mode 100644
index 000000000000..d9cede4c33ff
--- /dev/null
+++ b/sound/soc/meson/aiu-fifo.c
@@ -0,0 +1,223 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu-fifo.h"
+
+#define AIU_MEM_START 0x00
+#define AIU_MEM_RD 0x04
+#define AIU_MEM_END 0x08
+#define AIU_MEM_MASKS 0x0c
+#define AIU_MEM_MASK_CH_RD GENMASK(7, 0)
+#define AIU_MEM_MASK_CH_MEM GENMASK(15, 8)
+#define AIU_MEM_CONTROL 0x10
+#define AIU_MEM_CONTROL_INIT BIT(0)
+#define AIU_MEM_CONTROL_FILL_EN BIT(1)
+#define AIU_MEM_CONTROL_EMPTY_EN BIT(2)
+
+static struct snd_soc_dai *aiu_fifo_dai(struct snd_pcm_substream *ss)
+{
+ struct snd_soc_pcm_runtime *rtd = ss->private_data;
+
+ return asoc_rtd_to_cpu(rtd, 0);
+}
+
+snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_soc_dai *dai = aiu_fifo_dai(substream);
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int addr;
+
+ snd_soc_component_read(component, fifo->mem_offset + AIU_MEM_RD,
+ &addr);
+
+ return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr);
+}
+
+static void aiu_fifo_enable(struct snd_soc_dai *dai, bool enable)
+{
+ struct snd_soc_component *component = dai->component;
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+ unsigned int en_mask = (AIU_MEM_CONTROL_FILL_EN |
+ AIU_MEM_CONTROL_EMPTY_EN);
+
+ snd_soc_component_update_bits(component,
+ fifo->mem_offset + AIU_MEM_CONTROL,
+ en_mask, enable ? en_mask : 0);
+}
+
+int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ aiu_fifo_enable(dai, true);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_STOP:
+ aiu_fifo_enable(dai, false);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+int aiu_fifo_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+
+ snd_soc_component_update_bits(component,
+ fifo->mem_offset + AIU_MEM_CONTROL,
+ AIU_MEM_CONTROL_INIT,
+ AIU_MEM_CONTROL_INIT);
+ snd_soc_component_update_bits(component,
+ fifo->mem_offset + AIU_MEM_CONTROL,
+ AIU_MEM_CONTROL_INIT, 0);
+ return 0;
+}
+
+int aiu_fifo_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_component *component = dai->component;
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+ dma_addr_t end;
+ int ret;
+
+ ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (ret < 0)
+ return ret;
+
+ /* Setup the fifo boundaries */
+ end = runtime->dma_addr + runtime->dma_bytes - fifo->fifo_block;
+ snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_START,
+ runtime->dma_addr);
+ snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_RD,
+ runtime->dma_addr);
+ snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_END,
+ end);
+
+ /* Setup the fifo to read all the memory - no skip */
+ snd_soc_component_update_bits(component,
+ fifo->mem_offset + AIU_MEM_MASKS,
+ AIU_MEM_MASK_CH_RD | AIU_MEM_MASK_CH_MEM,
+ FIELD_PREP(AIU_MEM_MASK_CH_RD, 0xff) |
+ FIELD_PREP(AIU_MEM_MASK_CH_MEM, 0xff));
+
+ return 0;
+}
+
+int aiu_fifo_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static irqreturn_t aiu_fifo_isr(int irq, void *dev_id)
+{
+ struct snd_pcm_substream *playback = dev_id;
+
+ snd_pcm_period_elapsed(playback);
+
+ return IRQ_HANDLED;
+}
+
+int aiu_fifo_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+ int ret;
+
+ snd_soc_set_runtime_hwparams(substream, fifo->pcm);
+
+ /*
+ * Make sure the buffer and period size are multiple of the fifo burst
+ * size
+ */
+ ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ fifo->fifo_block);
+ if (ret)
+ return ret;
+
+ ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ fifo->fifo_block);
+ if (ret)
+ return ret;
+
+ ret = clk_prepare_enable(fifo->pclk);
+ if (ret)
+ return ret;
+
+ ret = request_irq(fifo->irq, aiu_fifo_isr, 0, dev_name(dai->dev),
+ substream);
+ if (ret)
+ clk_disable_unprepare(fifo->pclk);
+
+ return ret;
+}
+
+void aiu_fifo_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+
+ free_irq(fifo->irq, substream);
+ clk_disable_unprepare(fifo->pclk);
+}
+
+int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_substream *substream =
+ rtd->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ struct snd_card *card = rtd->card->snd_card;
+ struct aiu_fifo *fifo = dai->playback_dma_data;
+ size_t size = fifo->pcm->buffer_bytes_max;
+
+ snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ card->dev, size, size);
+
+ return 0;
+}
+
+int aiu_fifo_dai_probe(struct snd_soc_dai *dai)
+{
+ struct aiu_fifo *fifo;
+
+ fifo = kzalloc(sizeof(*fifo), GFP_KERNEL);
+ if (!fifo)
+ return -ENOMEM;
+
+ dai->playback_dma_data = fifo;
+
+ return 0;
+}
+
+int aiu_fifo_dai_remove(struct snd_soc_dai *dai)
+{
+ kfree(dai->playback_dma_data);
+
+ return 0;
+}
+
diff --git a/sound/soc/meson/aiu-fifo.h b/sound/soc/meson/aiu-fifo.h
new file mode 100644
index 000000000000..42ce266677cc
--- /dev/null
+++ b/sound/soc/meson/aiu-fifo.h
@@ -0,0 +1,50 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */
+/*
+ * Copyright (c) 2020 BayLibre, SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AIU_FIFO_H
+#define _MESON_AIU_FIFO_H
+
+struct snd_pcm_hardware;
+struct snd_soc_component_driver;
+struct snd_soc_dai_driver;
+struct clk;
+struct snd_pcm_ops;
+struct snd_pcm_substream;
+struct snd_soc_dai;
+struct snd_pcm_hw_params;
+struct platform_device;
+
+struct aiu_fifo {
+ struct snd_pcm_hardware *pcm;
+ unsigned int mem_offset;
+ unsigned int fifo_block;
+ struct clk *pclk;
+ int irq;
+};
+
+int aiu_fifo_dai_probe(struct snd_soc_dai *dai);
+int aiu_fifo_dai_remove(struct snd_soc_dai *dai);
+
+snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream);
+
+int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai);
+int aiu_fifo_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+int aiu_fifo_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai);
+int aiu_fifo_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+int aiu_fifo_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+void aiu_fifo_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dai *dai);
+
+#endif /* _MESON_AIU_FIFO_H */
diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c
new file mode 100644
index 000000000000..dc35ca79021c
--- /dev/null
+++ b/sound/soc/meson/aiu.c
@@ -0,0 +1,388 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <linux/reset.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-aiu.h>
+#include "aiu.h"
+#include "aiu-fifo.h"
+
+#define AIU_I2S_MISC_958_SRC_SHIFT 3
+
+static const char * const aiu_spdif_encode_sel_texts[] = {
+ "SPDIF", "I2S",
+};
+
+static SOC_ENUM_SINGLE_DECL(aiu_spdif_encode_sel_enum, AIU_I2S_MISC,
+ AIU_I2S_MISC_958_SRC_SHIFT,
+ aiu_spdif_encode_sel_texts);
+
+static const struct snd_kcontrol_new aiu_spdif_encode_mux =
+ SOC_DAPM_ENUM("SPDIF Buffer Src", aiu_spdif_encode_sel_enum);
+
+static const struct snd_soc_dapm_widget aiu_cpu_dapm_widgets[] = {
+ SND_SOC_DAPM_MUX("SPDIF SRC SEL", SND_SOC_NOPM, 0, 0,
+ &aiu_spdif_encode_mux),
+};
+
+static const struct snd_soc_dapm_route aiu_cpu_dapm_routes[] = {
+ { "I2S Encoder Playback", NULL, "I2S FIFO Playback" },
+ { "SPDIF SRC SEL", "SPDIF", "SPDIF FIFO Playback" },
+ { "SPDIF SRC SEL", "I2S", "I2S FIFO Playback" },
+ { "SPDIF Encoder Playback", NULL, "SPDIF SRC SEL" },
+};
+
+int aiu_of_xlate_dai_name(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name,
+ unsigned int component_id)
+{
+ struct snd_soc_dai *dai;
+ int id;
+
+ if (args->args_count != 2)
+ return -EINVAL;
+
+ if (args->args[0] != component_id)
+ return -EINVAL;
+
+ id = args->args[1];
+
+ if (id < 0 || id >= component->num_dai)
+ return -EINVAL;
+
+ for_each_component_dais(component, dai) {
+ if (id == 0)
+ break;
+ id--;
+ }
+
+ *dai_name = dai->driver->name;
+
+ return 0;
+}
+
+static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name)
+{
+ return aiu_of_xlate_dai_name(component, args, dai_name, AIU_CPU);
+}
+
+static int aiu_cpu_component_probe(struct snd_soc_component *component)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(component);
+
+ /* Required for the SPDIF Source control operation */
+ return clk_prepare_enable(aiu->i2s.clks[PCLK].clk);
+}
+
+static void aiu_cpu_component_remove(struct snd_soc_component *component)
+{
+ struct aiu *aiu = snd_soc_component_get_drvdata(component);
+
+ clk_disable_unprepare(aiu->i2s.clks[PCLK].clk);
+}
+
+static const struct snd_soc_component_driver aiu_cpu_component = {
+ .name = "AIU CPU",
+ .dapm_widgets = aiu_cpu_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aiu_cpu_dapm_widgets),
+ .dapm_routes = aiu_cpu_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(aiu_cpu_dapm_routes),
+ .of_xlate_dai_name = aiu_cpu_of_xlate_dai_name,
+ .pointer = aiu_fifo_pointer,
+ .probe = aiu_cpu_component_probe,
+ .remove = aiu_cpu_component_remove,
+};
+
+static struct snd_soc_dai_driver aiu_cpu_dai_drv[] = {
+ [CPU_I2S_FIFO] = {
+ .name = "I2S FIFO",
+ .playback = {
+ .stream_name = "I2S FIFO Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .formats = AIU_FORMATS,
+ },
+ .ops = &aiu_fifo_i2s_dai_ops,
+ .pcm_new = aiu_fifo_pcm_new,
+ .probe = aiu_fifo_i2s_dai_probe,
+ .remove = aiu_fifo_dai_remove,
+ },
+ [CPU_SPDIF_FIFO] = {
+ .name = "SPDIF FIFO",
+ .playback = {
+ .stream_name = "SPDIF FIFO Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .formats = AIU_FORMATS,
+ },
+ .ops = &aiu_fifo_spdif_dai_ops,
+ .pcm_new = aiu_fifo_pcm_new,
+ .probe = aiu_fifo_spdif_dai_probe,
+ .remove = aiu_fifo_dai_remove,
+ },
+ [CPU_I2S_ENCODER] = {
+ .name = "I2S Encoder",
+ .playback = {
+ .stream_name = "I2S Encoder Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = AIU_FORMATS,
+ },
+ .ops = &aiu_encoder_i2s_dai_ops,
+ },
+ [CPU_SPDIF_ENCODER] = {
+ .name = "SPDIF Encoder",
+ .playback = {
+ .stream_name = "SPDIF Encoder Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000),
+ .formats = AIU_FORMATS,
+ },
+ .ops = &aiu_encoder_spdif_dai_ops,
+ }
+};
+
+static const struct regmap_config aiu_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+ .max_register = 0x2ac,
+};
+
+static int aiu_clk_bulk_get(struct device *dev,
+ const char * const *ids,
+ unsigned int num,
+ struct aiu_interface *interface)
+{
+ struct clk_bulk_data *clks;
+ int i, ret;
+
+ clks = devm_kcalloc(dev, num, sizeof(*clks), GFP_KERNEL);
+ if (!clks)
+ return -ENOMEM;
+
+ for (i = 0; i < num; i++)
+ clks[i].id = ids[i];
+
+ ret = devm_clk_bulk_get(dev, num, clks);
+ if (ret < 0)
+ return ret;
+
+ interface->clks = clks;
+ interface->clk_num = num;
+ return 0;
+}
+
+static const char * const aiu_i2s_ids[] = {
+ [PCLK] = "i2s_pclk",
+ [AOCLK] = "i2s_aoclk",
+ [MCLK] = "i2s_mclk",
+ [MIXER] = "i2s_mixer",
+};
+
+static const char * const aiu_spdif_ids[] = {
+ [PCLK] = "spdif_pclk",
+ [AOCLK] = "spdif_aoclk",
+ [MCLK] = "spdif_mclk_sel"
+};
+
+static int aiu_clk_get(struct device *dev)
+{
+ struct aiu *aiu = dev_get_drvdata(dev);
+ int ret;
+
+ aiu->pclk = devm_clk_get(dev, "pclk");
+ if (IS_ERR(aiu->pclk)) {
+ if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER)
+ dev_err(dev, "Can't get the aiu pclk\n");
+ return PTR_ERR(aiu->pclk);
+ }
+
+ aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk");
+ if (IS_ERR(aiu->spdif_mclk)) {
+ if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER)
+ dev_err(dev, "Can't get the aiu spdif master clock\n");
+ return PTR_ERR(aiu->spdif_mclk);
+ }
+
+ ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids),
+ &aiu->i2s);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "Can't get the i2s clocks\n");
+ return ret;
+ }
+
+ ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids),
+ &aiu->spdif);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "Can't get the spdif clocks\n");
+ return ret;
+ }
+
+ ret = clk_prepare_enable(aiu->pclk);
+ if (ret) {
+ dev_err(dev, "peripheral clock enable failed\n");
+ return ret;
+ }
+
+ ret = devm_add_action_or_reset(dev,
+ (void(*)(void *))clk_disable_unprepare,
+ aiu->pclk);
+ if (ret)
+ dev_err(dev, "failed to add reset action on pclk");
+
+ return ret;
+}
+
+static int aiu_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ void __iomem *regs;
+ struct regmap *map;
+ struct aiu *aiu;
+ int ret;
+
+ aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL);
+ if (!aiu)
+ return -ENOMEM;
+
+ aiu->platform = device_get_match_data(dev);
+ if (!aiu->platform)
+ return -ENODEV;
+
+ platform_set_drvdata(pdev, aiu);
+
+ ret = device_reset(dev);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "Failed to reset device\n");
+ return ret;
+ }
+
+ regs = devm_platform_ioremap_resource(pdev, 0);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ map = devm_regmap_init_mmio(dev, regs, &aiu_regmap_cfg);
+ if (IS_ERR(map)) {
+ dev_err(dev, "failed to init regmap: %ld\n",
+ PTR_ERR(map));
+ return PTR_ERR(map);
+ }
+
+ aiu->i2s.irq = platform_get_irq_byname(pdev, "i2s");
+ if (aiu->i2s.irq < 0)
+ return aiu->i2s.irq;
+
+ aiu->spdif.irq = platform_get_irq_byname(pdev, "spdif");
+ if (aiu->spdif.irq < 0)
+ return aiu->spdif.irq;
+
+ ret = aiu_clk_get(dev);
+ if (ret)
+ return ret;
+
+ /* Register the cpu component of the aiu */
+ ret = snd_soc_register_component(dev, &aiu_cpu_component,
+ aiu_cpu_dai_drv,
+ ARRAY_SIZE(aiu_cpu_dai_drv));
+ if (ret) {
+ dev_err(dev, "Failed to register cpu component\n");
+ return ret;
+ }
+
+ /* Register the hdmi codec control component */
+ ret = aiu_hdmi_ctrl_register_component(dev);
+ if (ret) {
+ dev_err(dev, "Failed to register hdmi control component\n");
+ goto err;
+ }
+
+ /* Register the internal dac control component on gxl */
+ if (aiu->platform->has_acodec) {
+ ret = aiu_acodec_ctrl_register_component(dev);
+ if (ret) {
+ dev_err(dev,
+ "Failed to register acodec control component\n");
+ goto err;
+ }
+ }
+
+ return 0;
+err:
+ snd_soc_unregister_component(dev);
+ return ret;
+}
+
+static int aiu_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_component(&pdev->dev);
+
+ return 0;
+}
+
+static const struct aiu_platform_data aiu_gxbb_pdata = {
+ .has_acodec = false,
+ .has_clk_ctrl_more_i2s_div = true,
+};
+
+static const struct aiu_platform_data aiu_gxl_pdata = {
+ .has_acodec = true,
+ .has_clk_ctrl_more_i2s_div = true,
+};
+
+static const struct aiu_platform_data aiu_meson8_pdata = {
+ .has_acodec = false,
+ .has_clk_ctrl_more_i2s_div = false,
+};
+
+static const struct of_device_id aiu_of_match[] = {
+ { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata },
+ { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata },
+ { .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata },
+ { .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata },
+ {}
+};
+MODULE_DEVICE_TABLE(of, aiu_of_match);
+
+static struct platform_driver aiu_pdrv = {
+ .probe = aiu_probe,
+ .remove = aiu_remove,
+ .driver = {
+ .name = "meson-aiu",
+ .of_match_table = aiu_of_match,
+ },
+};
+module_platform_driver(aiu_pdrv);
+
+MODULE_DESCRIPTION("Meson AIU Driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h
new file mode 100644
index 000000000000..87aa19ac4af3
--- /dev/null
+++ b/sound/soc/meson/aiu.h
@@ -0,0 +1,89 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */
+/*
+ * Copyright (c) 2018 BayLibre, SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_AIU_H
+#define _MESON_AIU_H
+
+struct clk;
+struct clk_bulk_data;
+struct device;
+struct of_phandle_args;
+struct snd_soc_dai;
+struct snd_soc_dai_ops;
+
+enum aiu_clk_ids {
+ PCLK = 0,
+ AOCLK,
+ MCLK,
+ MIXER
+};
+
+struct aiu_interface {
+ struct clk_bulk_data *clks;
+ unsigned int clk_num;
+ int irq;
+};
+
+struct aiu_platform_data {
+ bool has_acodec;
+ bool has_clk_ctrl_more_i2s_div;
+};
+
+struct aiu {
+ struct clk *pclk;
+ struct clk *spdif_mclk;
+ struct aiu_interface i2s;
+ struct aiu_interface spdif;
+ const struct aiu_platform_data *platform;
+};
+
+#define AIU_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+int aiu_of_xlate_dai_name(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name,
+ unsigned int component_id);
+
+int aiu_hdmi_ctrl_register_component(struct device *dev);
+int aiu_acodec_ctrl_register_component(struct device *dev);
+
+int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai);
+int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai);
+
+extern const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops;
+extern const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops;
+extern const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops;
+extern const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops;
+
+#define AIU_IEC958_BPF 0x000
+#define AIU_958_MISC 0x010
+#define AIU_IEC958_DCU_FF_CTRL 0x01c
+#define AIU_958_CHSTAT_L0 0x020
+#define AIU_958_CHSTAT_L1 0x024
+#define AIU_958_CTRL 0x028
+#define AIU_I2S_SOURCE_DESC 0x034
+#define AIU_I2S_DAC_CFG 0x040
+#define AIU_I2S_SYNC 0x044
+#define AIU_I2S_MISC 0x048
+#define AIU_RST_SOFT 0x054
+#define AIU_CLK_CTRL 0x058
+#define AIU_CLK_CTRL_MORE 0x064
+#define AIU_CODEC_DAC_LRCLK_CTRL 0x0a0
+#define AIU_HDMI_CLK_DATA_CTRL 0x0a8
+#define AIU_ACODEC_CTRL 0x0b0
+#define AIU_958_CHSTAT_R0 0x0c0
+#define AIU_958_CHSTAT_R1 0x0c4
+#define AIU_MEM_I2S_START 0x180
+#define AIU_MEM_I2S_MASKS 0x18c
+#define AIU_MEM_I2S_CONTROL 0x190
+#define AIU_MEM_IEC958_START 0x194
+#define AIU_MEM_IEC958_CONTROL 0x1a4
+#define AIU_MEM_I2S_BUF_CNTL 0x1d8
+#define AIU_MEM_IEC958_BUF_CNTL 0x1fc
+
+#endif /* _MESON_AIU_H */
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 1f698adde506..af46845f4ef2 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -9,11 +9,7 @@
#include <sound/soc-dai.h>
#include "axg-tdm.h"
-
-struct axg_card {
- struct snd_soc_card card;
- void **link_data;
-};
+#include "meson-card.h"
struct axg_dai_link_tdm_mask {
u32 tx;
@@ -41,161 +37,15 @@ static const struct snd_soc_pcm_stream codec_params = {
.channels_max = 8,
};
-#define PREFIX "amlogic,"
-
-static int axg_card_reallocate_links(struct axg_card *priv,
- unsigned int num_links)
-{
- struct snd_soc_dai_link *links;
- void **ldata;
-
- links = krealloc(priv->card.dai_link,
- num_links * sizeof(*priv->card.dai_link),
- GFP_KERNEL | __GFP_ZERO);
- ldata = krealloc(priv->link_data,
- num_links * sizeof(*priv->link_data),
- GFP_KERNEL | __GFP_ZERO);
-
- if (!links || !ldata) {
- dev_err(priv->card.dev, "failed to allocate links\n");
- return -ENOMEM;
- }
-
- priv->card.dai_link = links;
- priv->link_data = ldata;
- priv->card.num_links = num_links;
- return 0;
-}
-
-static int axg_card_parse_dai(struct snd_soc_card *card,
- struct device_node *node,
- struct device_node **dai_of_node,
- const char **dai_name)
-{
- struct of_phandle_args args;
- int ret;
-
- if (!dai_name || !dai_of_node || !node)
- return -EINVAL;
-
- ret = of_parse_phandle_with_args(node, "sound-dai",
- "#sound-dai-cells", 0, &args);
- if (ret) {
- if (ret != -EPROBE_DEFER)
- dev_err(card->dev, "can't parse dai %d\n", ret);
- return ret;
- }
- *dai_of_node = args.np;
-
- return snd_soc_get_dai_name(&args, dai_name);
-}
-
-static int axg_card_set_link_name(struct snd_soc_card *card,
- struct snd_soc_dai_link *link,
- struct device_node *node,
- const char *prefix)
-{
- char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
- prefix, node->full_name);
- if (!name)
- return -ENOMEM;
-
- link->name = name;
- link->stream_name = name;
-
- return 0;
-}
-
-static void axg_card_clean_references(struct axg_card *priv)
-{
- struct snd_soc_card *card = &priv->card;
- struct snd_soc_dai_link *link;
- struct snd_soc_dai_link_component *codec;
- struct snd_soc_aux_dev *aux;
- int i, j;
-
- if (card->dai_link) {
- for_each_card_prelinks(card, i, link) {
- if (link->cpus)
- of_node_put(link->cpus->of_node);
- for_each_link_codecs(link, j, codec)
- of_node_put(codec->of_node);
- }
- }
-
- if (card->aux_dev) {
- for_each_card_pre_auxs(card, i, aux)
- of_node_put(aux->dlc.of_node);
- }
-
- kfree(card->dai_link);
- kfree(priv->link_data);
-}
-
-static int axg_card_add_aux_devices(struct snd_soc_card *card)
-{
- struct device_node *node = card->dev->of_node;
- struct snd_soc_aux_dev *aux;
- int num, i;
-
- num = of_count_phandle_with_args(node, "audio-aux-devs", NULL);
- if (num == -ENOENT) {
- /*
- * It is ok to have no auxiliary devices but for this card it
- * is a strange situtation. Let's warn the about it.
- */
- dev_warn(card->dev, "card has no auxiliary devices\n");
- return 0;
- } else if (num < 0) {
- dev_err(card->dev, "error getting auxiliary devices: %d\n",
- num);
- return num;
- }
-
- aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL);
- if (!aux)
- return -ENOMEM;
- card->aux_dev = aux;
- card->num_aux_devs = num;
-
- for_each_card_pre_auxs(card, i, aux) {
- aux->dlc.of_node =
- of_parse_phandle(node, "audio-aux-devs", i);
- if (!aux->dlc.of_node)
- return -EINVAL;
- }
-
- return 0;
-}
-
static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
struct axg_dai_link_tdm_data *be =
(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
- struct snd_soc_dai *codec_dai;
- unsigned int mclk;
- int ret, i;
-
- if (be->mclk_fs) {
- mclk = params_rate(params) * be->mclk_fs;
-
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
- SND_SOC_CLOCK_IN);
- if (ret && ret != -ENOTSUPP)
- return ret;
- }
-
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk,
- SND_SOC_CLOCK_OUT);
- if (ret && ret != -ENOTSUPP)
- return ret;
- }
- return 0;
+ return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs);
}
static const struct snd_soc_ops axg_card_tdm_be_ops = {
@@ -204,13 +54,13 @@ static const struct snd_soc_ops axg_card_tdm_be_ops = {
static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
struct axg_dai_link_tdm_data *be =
(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
struct snd_soc_dai *codec_dai;
int ret, i;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
ret = snd_soc_dai_set_tdm_slot(codec_dai,
be->codec_masks[i].tx,
be->codec_masks[i].rx,
@@ -222,10 +72,10 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
}
}
- ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask,
+ ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), be->tx_mask, be->rx_mask,
be->slots, be->slot_width);
if (ret) {
- dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+ dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n");
return ret;
}
@@ -234,16 +84,16 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd)
{
- struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
struct axg_dai_link_tdm_data *be =
(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
int ret;
/* The loopback rx_mask is the pad tx_mask */
- ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask,
+ ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), NULL, be->tx_mask,
be->slots, be->slot_width);
if (ret) {
- dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+ dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n");
return ret;
}
@@ -253,14 +103,14 @@ static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd)
static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
int *index)
{
- struct axg_card *priv = snd_soc_card_get_drvdata(card);
+ struct meson_card *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_dai_link *pad = &card->dai_link[*index];
struct snd_soc_dai_link *lb;
struct snd_soc_dai_link_component *dlc;
int ret;
/* extend links */
- ret = axg_card_reallocate_links(priv, card->num_links + 1);
+ ret = meson_card_reallocate_links(card, card->num_links + 1);
if (ret)
return ret;
@@ -304,32 +154,6 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
return 0;
}
-static unsigned int axg_card_parse_daifmt(struct device_node *node,
- struct device_node *cpu_node)
-{
- struct device_node *bitclkmaster = NULL;
- struct device_node *framemaster = NULL;
- unsigned int daifmt;
-
- daifmt = snd_soc_of_parse_daifmt(node, PREFIX,
- &bitclkmaster, &framemaster);
- daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
-
- /* If no master is provided, default to cpu master */
- if (!bitclkmaster || bitclkmaster == cpu_node) {
- daifmt |= (!framemaster || framemaster == cpu_node) ?
- SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM;
- } else {
- daifmt |= (!framemaster || framemaster == cpu_node) ?
- SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM;
- }
-
- of_node_put(bitclkmaster);
- of_node_put(framemaster);
-
- return daifmt;
-}
-
static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card,
struct snd_soc_dai_link *link,
struct device_node *node,
@@ -424,7 +248,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card,
struct device_node *node,
int *index)
{
- struct axg_card *priv = snd_soc_card_get_drvdata(card);
+ struct meson_card *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_dai_link *link = &card->dai_link[*index];
struct axg_dai_link_tdm_data *be;
int ret;
@@ -438,7 +262,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card,
/* Setup tdm link */
link->ops = &axg_card_tdm_be_ops;
link->init = axg_card_tdm_dai_init;
- link->dai_fmt = axg_card_parse_daifmt(node, link->cpus->of_node);
+ link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node);
of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
@@ -462,97 +286,25 @@ static int axg_card_parse_tdm(struct snd_soc_card *card,
return 0;
}
-static int axg_card_set_be_link(struct snd_soc_card *card,
- struct snd_soc_dai_link *link,
- struct device_node *node)
-{
- struct snd_soc_dai_link_component *codec;
- struct device_node *np;
- int ret, num_codecs;
-
- link->no_pcm = 1;
- link->dpcm_playback = 1;
- link->dpcm_capture = 1;
-
- num_codecs = of_get_child_count(node);
- if (!num_codecs) {
- dev_err(card->dev, "be link %s has no codec\n",
- node->full_name);
- return -EINVAL;
- }
-
- codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL);
- if (!codec)
- return -ENOMEM;
-
- link->codecs = codec;
- link->num_codecs = num_codecs;
-
- for_each_child_of_node(node, np) {
- ret = axg_card_parse_dai(card, np, &codec->of_node,
- &codec->dai_name);
- if (ret) {
- of_node_put(np);
- return ret;
- }
-
- codec++;
- }
-
- ret = axg_card_set_link_name(card, link, node, "be");
- if (ret)
- dev_err(card->dev, "error setting %pOFn link name\n", np);
-
- return ret;
-}
-
-static int axg_card_set_fe_link(struct snd_soc_card *card,
- struct snd_soc_dai_link *link,
- struct device_node *node,
- bool is_playback)
-{
- struct snd_soc_dai_link_component *codec;
-
- codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL);
- if (!codec)
- return -ENOMEM;
-
- link->codecs = codec;
- link->num_codecs = 1;
-
- link->dynamic = 1;
- link->dpcm_merged_format = 1;
- link->dpcm_merged_chan = 1;
- link->dpcm_merged_rate = 1;
- link->codecs->dai_name = "snd-soc-dummy-dai";
- link->codecs->name = "snd-soc-dummy";
-
- if (is_playback)
- link->dpcm_playback = 1;
- else
- link->dpcm_capture = 1;
-
- return axg_card_set_link_name(card, link, node, "fe");
-}
-
static int axg_card_cpu_is_capture_fe(struct device_node *np)
{
- return of_device_is_compatible(np, PREFIX "axg-toddr");
+ return of_device_is_compatible(np, DT_PREFIX "axg-toddr");
}
static int axg_card_cpu_is_playback_fe(struct device_node *np)
{
- return of_device_is_compatible(np, PREFIX "axg-frddr");
+ return of_device_is_compatible(np, DT_PREFIX "axg-frddr");
}
static int axg_card_cpu_is_tdm_iface(struct device_node *np)
{
- return of_device_is_compatible(np, PREFIX "axg-tdm-iface");
+ return of_device_is_compatible(np, DT_PREFIX "axg-tdm-iface");
}
static int axg_card_cpu_is_codec(struct device_node *np)
{
- return of_device_is_compatible(np, PREFIX "g12a-tohdmitx");
+ return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx") ||
+ of_device_is_compatible(np, DT_PREFIX "g12a-toacodec");
}
static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
@@ -569,17 +321,17 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
dai_link->cpus = cpu;
dai_link->num_cpus = 1;
- ret = axg_card_parse_dai(card, np, &dai_link->cpus->of_node,
- &dai_link->cpus->dai_name);
+ ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node,
+ &dai_link->cpus->dai_name);
if (ret)
return ret;
if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node))
- ret = axg_card_set_fe_link(card, dai_link, np, true);
+ ret = meson_card_set_fe_link(card, dai_link, np, true);
else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node))
- ret = axg_card_set_fe_link(card, dai_link, np, false);
+ ret = meson_card_set_fe_link(card, dai_link, np, false);
else
- ret = axg_card_set_be_link(card, dai_link, np);
+ ret = meson_card_set_be_link(card, dai_link, np);
if (ret)
return ret;
@@ -592,121 +344,21 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
return ret;
}
-static int axg_card_add_links(struct snd_soc_card *card)
-{
- struct axg_card *priv = snd_soc_card_get_drvdata(card);
- struct device_node *node = card->dev->of_node;
- struct device_node *np;
- int num, i, ret;
-
- num = of_get_child_count(node);
- if (!num) {
- dev_err(card->dev, "card has no links\n");
- return -EINVAL;
- }
-
- ret = axg_card_reallocate_links(priv, num);
- if (ret)
- return ret;
-
- i = 0;
- for_each_child_of_node(node, np) {
- ret = axg_card_add_link(card, np, &i);
- if (ret) {
- of_node_put(np);
- return ret;
- }
-
- i++;
- }
-
- return 0;
-}
-
-static int axg_card_parse_of_optional(struct snd_soc_card *card,
- const char *propname,
- int (*func)(struct snd_soc_card *c,
- const char *p))
-{
- /* If property is not provided, don't fail ... */
- if (!of_property_read_bool(card->dev->of_node, propname))
- return 0;
-
- /* ... but do fail if it is provided and the parsing fails */
- return func(card, propname);
-}
+static const struct meson_card_match_data axg_card_match_data = {
+ .add_link = axg_card_add_link,
+};
static const struct of_device_id axg_card_of_match[] = {
- { .compatible = "amlogic,axg-sound-card", },
- {}
+ {
+ .compatible = "amlogic,axg-sound-card",
+ .data = &axg_card_match_data,
+ }, {}
};
MODULE_DEVICE_TABLE(of, axg_card_of_match);
-static int axg_card_probe(struct platform_device *pdev)
-{
- struct device *dev = &pdev->dev;
- struct axg_card *priv;
- int ret;
-
- priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
- if (!priv)
- return -ENOMEM;
-
- platform_set_drvdata(pdev, priv);
- snd_soc_card_set_drvdata(&priv->card, priv);
-
- priv->card.owner = THIS_MODULE;
- priv->card.dev = dev;
-
- ret = snd_soc_of_parse_card_name(&priv->card, "model");
- if (ret < 0)
- return ret;
-
- ret = axg_card_parse_of_optional(&priv->card, "audio-routing",
- snd_soc_of_parse_audio_routing);
- if (ret) {
- dev_err(dev, "error while parsing routing\n");
- return ret;
- }
-
- ret = axg_card_parse_of_optional(&priv->card, "audio-widgets",
- snd_soc_of_parse_audio_simple_widgets);
- if (ret) {
- dev_err(dev, "error while parsing widgets\n");
- return ret;
- }
-
- ret = axg_card_add_links(&priv->card);
- if (ret)
- goto out_err;
-
- ret = axg_card_add_aux_devices(&priv->card);
- if (ret)
- goto out_err;
-
- ret = devm_snd_soc_register_card(dev, &priv->card);
- if (ret)
- goto out_err;
-
- return 0;
-
-out_err:
- axg_card_clean_references(priv);
- return ret;
-}
-
-static int axg_card_remove(struct platform_device *pdev)
-{
- struct axg_card *priv = platform_get_drvdata(pdev);
-
- axg_card_clean_references(priv);
-
- return 0;
-}
-
static struct platform_driver axg_card_pdrv = {
- .probe = axg_card_probe,
- .remove = axg_card_remove,
+ .probe = meson_card_probe,
+ .remove = meson_card_remove,
.driver = {
.name = "axg-sound-card",
.of_match_table = axg_card_of_match,
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index c12b0d5e8ebf..2e9b56b29d31 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -47,7 +47,7 @@ static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss)
{
struct snd_soc_pcm_runtime *rtd = ss->private_data;
- return rtd->cpu_dai;
+ return asoc_rtd_to_cpu(rtd, 0);
}
static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss)
diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c
new file mode 100644
index 000000000000..9339fabccb79
--- /dev/null
+++ b/sound/soc/meson/g12a-toacodec.c
@@ -0,0 +1,252 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <linux/reset.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-g12a-toacodec.h>
+#include "axg-tdm.h"
+#include "meson-codec-glue.h"
+
+#define G12A_TOACODEC_DRV_NAME "g12a-toacodec"
+
+#define TOACODEC_CTRL0 0x0
+#define CTRL0_ENABLE_SHIFT 31
+#define CTRL0_DAT_SEL_SHIFT 14
+#define CTRL0_DAT_SEL (0x3 << CTRL0_DAT_SEL_SHIFT)
+#define CTRL0_LANE_SEL 12
+#define CTRL0_LRCLK_SEL GENMASK(9, 8)
+#define CTRL0_BLK_CAP_INV BIT(7)
+#define CTRL0_BCLK_O_INV BIT(6)
+#define CTRL0_BCLK_SEL GENMASK(5, 4)
+#define CTRL0_MCLK_SEL GENMASK(2, 0)
+
+#define TOACODEC_OUT_CHMAX 2
+
+static const char * const g12a_toacodec_mux_texts[] = {
+ "I2S A", "I2S B", "I2S C",
+};
+
+static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_kcontrol_component(kcontrol);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int mux, changed;
+
+ mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+ changed = snd_soc_component_test_bits(component, e->reg,
+ CTRL0_DAT_SEL,
+ FIELD_PREP(CTRL0_DAT_SEL, mux));
+
+ if (!changed)
+ return 0;
+
+ /* Force disconnect of the mux while updating */
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+ snd_soc_component_update_bits(component, e->reg,
+ CTRL0_DAT_SEL |
+ CTRL0_LRCLK_SEL |
+ CTRL0_BCLK_SEL,
+ FIELD_PREP(CTRL0_DAT_SEL, mux) |
+ FIELD_PREP(CTRL0_LRCLK_SEL, mux) |
+ FIELD_PREP(CTRL0_BCLK_SEL, mux));
+
+ /*
+ * FIXME:
+ * On this soc, the glue gets the MCLK directly from the clock
+ * controller instead of going the through the TDM interface.
+ *
+ * Here we assume interface A uses clock A, etc ... While it is
+ * true for now, it could be different. Instead the glue should
+ * find out the clock used by the interface and select the same
+ * source. For that, we will need regmap backed clock mux which
+ * is a work in progress
+ */
+ snd_soc_component_update_bits(component, e->reg,
+ CTRL0_MCLK_SEL,
+ FIELD_PREP(CTRL0_MCLK_SEL, mux));
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+ return 0;
+}
+
+static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0,
+ CTRL0_DAT_SEL_SHIFT,
+ g12a_toacodec_mux_texts);
+
+static const struct snd_kcontrol_new g12a_toacodec_mux =
+ SOC_DAPM_ENUM_EXT("Source", g12a_toacodec_mux_enum,
+ snd_soc_dapm_get_enum_double,
+ g12a_toacodec_mux_put_enum);
+
+static const struct snd_kcontrol_new g12a_toacodec_out_enable =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOACODEC_CTRL0,
+ CTRL0_ENABLE_SHIFT, 1, 0);
+
+static const struct snd_soc_dapm_widget g12a_toacodec_widgets[] = {
+ SND_SOC_DAPM_MUX("SRC", SND_SOC_NOPM, 0, 0,
+ &g12a_toacodec_mux),
+ SND_SOC_DAPM_SWITCH("OUT EN", SND_SOC_NOPM, 0, 0,
+ &g12a_toacodec_out_enable),
+};
+
+static int g12a_toacodec_input_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct meson_codec_glue_input *data;
+ int ret;
+
+ ret = meson_codec_glue_input_hw_params(substream, params, dai);
+ if (ret)
+ return ret;
+
+ /* The glue will provide 1 lane out of the 4 to the output */
+ data = meson_codec_glue_input_get_data(dai);
+ data->params.channels_min = min_t(unsigned int, TOACODEC_OUT_CHMAX,
+ data->params.channels_min);
+ data->params.channels_max = min_t(unsigned int, TOACODEC_OUT_CHMAX,
+ data->params.channels_max);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops g12a_toacodec_input_ops = {
+ .hw_params = g12a_toacodec_input_hw_params,
+ .set_fmt = meson_codec_glue_input_set_fmt,
+};
+
+static const struct snd_soc_dai_ops g12a_toacodec_output_ops = {
+ .startup = meson_codec_glue_output_startup,
+};
+
+#define TOACODEC_STREAM(xname, xsuffix, xchmax) \
+{ \
+ .stream_name = xname " " xsuffix, \
+ .channels_min = 1, \
+ .channels_max = (xchmax), \
+ .rate_min = 5512, \
+ .rate_max = 192000, \
+ .formats = AXG_TDM_FORMATS, \
+}
+
+#define TOACODEC_INPUT(xname, xid) { \
+ .name = xname, \
+ .id = (xid), \
+ .playback = TOACODEC_STREAM(xname, "Playback", 8), \
+ .ops = &g12a_toacodec_input_ops, \
+ .probe = meson_codec_glue_input_dai_probe, \
+ .remove = meson_codec_glue_input_dai_remove, \
+}
+
+#define TOACODEC_OUTPUT(xname, xid) { \
+ .name = xname, \
+ .id = (xid), \
+ .capture = TOACODEC_STREAM(xname, "Capture", TOACODEC_OUT_CHMAX), \
+ .ops = &g12a_toacodec_output_ops, \
+}
+
+static struct snd_soc_dai_driver g12a_toacodec_dai_drv[] = {
+ TOACODEC_INPUT("IN A", TOACODEC_IN_A),
+ TOACODEC_INPUT("IN B", TOACODEC_IN_B),
+ TOACODEC_INPUT("IN C", TOACODEC_IN_C),
+ TOACODEC_OUTPUT("OUT", TOACODEC_OUT),
+};
+
+static int g12a_toacodec_component_probe(struct snd_soc_component *c)
+{
+ /* Initialize the static clock parameters */
+ return snd_soc_component_write(c, TOACODEC_CTRL0,
+ CTRL0_BLK_CAP_INV);
+}
+
+static const struct snd_soc_dapm_route g12a_toacodec_routes[] = {
+ { "SRC", "I2S A", "IN A Playback" },
+ { "SRC", "I2S B", "IN B Playback" },
+ { "SRC", "I2S C", "IN C Playback" },
+ { "OUT EN", "Switch", "SRC" },
+ { "OUT Capture", NULL, "OUT EN" },
+};
+
+static const struct snd_kcontrol_new g12a_toacodec_controls[] = {
+ SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0),
+};
+
+static const struct snd_soc_component_driver g12a_toacodec_component_drv = {
+ .probe = g12a_toacodec_component_probe,
+ .controls = g12a_toacodec_controls,
+ .num_controls = ARRAY_SIZE(g12a_toacodec_controls),
+ .dapm_widgets = g12a_toacodec_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(g12a_toacodec_widgets),
+ .dapm_routes = g12a_toacodec_routes,
+ .num_dapm_routes = ARRAY_SIZE(g12a_toacodec_routes),
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+static const struct regmap_config g12a_toacodec_regmap_cfg = {
+ .reg_bits = 32,
+ .val_bits = 32,
+ .reg_stride = 4,
+};
+
+static const struct of_device_id g12a_toacodec_of_match[] = {
+ { .compatible = "amlogic,g12a-toacodec", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, g12a_toacodec_of_match);
+
+static int g12a_toacodec_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ void __iomem *regs;
+ struct regmap *map;
+ int ret;
+
+ ret = device_reset(dev);
+ if (ret)
+ return ret;
+
+ regs = devm_platform_ioremap_resource(pdev, 0);
+ if (IS_ERR(regs))
+ return PTR_ERR(regs);
+
+ map = devm_regmap_init_mmio(dev, regs, &g12a_toacodec_regmap_cfg);
+ if (IS_ERR(map)) {
+ dev_err(dev, "failed to init regmap: %ld\n",
+ PTR_ERR(map));
+ return PTR_ERR(map);
+ }
+
+ return devm_snd_soc_register_component(dev,
+ &g12a_toacodec_component_drv, g12a_toacodec_dai_drv,
+ ARRAY_SIZE(g12a_toacodec_dai_drv));
+}
+
+static struct platform_driver g12a_toacodec_pdrv = {
+ .driver = {
+ .name = G12A_TOACODEC_DRV_NAME,
+ .of_match_table = g12a_toacodec_of_match,
+ },
+ .probe = g12a_toacodec_probe,
+};
+module_platform_driver(g12a_toacodec_pdrv);
+
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_DESCRIPTION("Amlogic G12a To Internal DAC Codec Driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c
index 8a0db28a6a40..9b2b59536ced 100644
--- a/sound/soc/meson/g12a-tohdmitx.c
+++ b/sound/soc/meson/g12a-tohdmitx.c
@@ -13,112 +13,51 @@
#include <sound/soc-dai.h>
#include <dt-bindings/sound/meson-g12a-tohdmitx.h>
+#include "meson-codec-glue.h"
#define G12A_TOHDMITX_DRV_NAME "g12a-tohdmitx"
#define TOHDMITX_CTRL0 0x0
#define CTRL0_ENABLE_SHIFT 31
-#define CTRL0_I2S_DAT_SEL GENMASK(13, 12)
+#define CTRL0_I2S_DAT_SEL_SHIFT 12
+#define CTRL0_I2S_DAT_SEL (0x3 << CTRL0_I2S_DAT_SEL_SHIFT)
#define CTRL0_I2S_LRCLK_SEL GENMASK(9, 8)
#define CTRL0_I2S_BLK_CAP_INV BIT(7)
#define CTRL0_I2S_BCLK_O_INV BIT(6)
#define CTRL0_I2S_BCLK_SEL GENMASK(5, 4)
#define CTRL0_SPDIF_CLK_CAP_INV BIT(3)
#define CTRL0_SPDIF_CLK_O_INV BIT(2)
-#define CTRL0_SPDIF_SEL BIT(1)
+#define CTRL0_SPDIF_SEL_SHIFT 1
+#define CTRL0_SPDIF_SEL (0x1 << CTRL0_SPDIF_SEL_SHIFT)
#define CTRL0_SPDIF_CLK_SEL BIT(0)
-struct g12a_tohdmitx_input {
- struct snd_soc_pcm_stream params;
- unsigned int fmt;
-};
-
-static struct snd_soc_dapm_widget *
-g12a_tohdmitx_get_input(struct snd_soc_dapm_widget *w)
-{
- struct snd_soc_dapm_path *p = NULL;
- struct snd_soc_dapm_widget *in;
-
- snd_soc_dapm_widget_for_each_source_path(w, p) {
- if (!p->connect)
- continue;
-
- /* Check that we still are in the same component */
- if (snd_soc_dapm_to_component(w->dapm) !=
- snd_soc_dapm_to_component(p->source->dapm))
- continue;
-
- if (p->source->id == snd_soc_dapm_dai_in)
- return p->source;
-
- in = g12a_tohdmitx_get_input(p->source);
- if (in)
- return in;
- }
-
- return NULL;
-}
-
-static struct g12a_tohdmitx_input *
-g12a_tohdmitx_get_input_data(struct snd_soc_dapm_widget *w)
-{
- struct snd_soc_dapm_widget *in =
- g12a_tohdmitx_get_input(w);
- struct snd_soc_dai *dai;
-
- if (WARN_ON(!in))
- return NULL;
-
- dai = in->priv;
-
- return dai->playback_dma_data;
-}
-
static const char * const g12a_tohdmitx_i2s_mux_texts[] = {
"I2S A", "I2S B", "I2S C",
};
-static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_i2s_mux_enum,
- g12a_tohdmitx_i2s_mux_texts);
-
-static int g12a_tohdmitx_get_input_val(struct snd_soc_component *component,
- unsigned int mask)
-{
- unsigned int val;
-
- snd_soc_component_read(component, TOHDMITX_CTRL0, &val);
- return (val & mask) >> __ffs(mask);
-}
-
-static int g12a_tohdmitx_i2s_mux_get_enum(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_component *component =
- snd_soc_dapm_kcontrol_component(kcontrol);
-
- ucontrol->value.enumerated.item[0] =
- g12a_tohdmitx_get_input_val(component, CTRL0_I2S_DAT_SEL);
-
- return 0;
-}
-
static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component =
snd_soc_dapm_kcontrol_component(kcontrol);
struct snd_soc_dapm_context *dapm =
snd_soc_dapm_kcontrol_dapm(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int mux = ucontrol->value.enumerated.item[0];
- unsigned int val = g12a_tohdmitx_get_input_val(component,
- CTRL0_I2S_DAT_SEL);
+ unsigned int mux, changed;
+
+ mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+ changed = snd_soc_component_test_bits(component, e->reg,
+ CTRL0_I2S_DAT_SEL,
+ FIELD_PREP(CTRL0_I2S_DAT_SEL,
+ mux));
+
+ if (!changed)
+ return 0;
/* Force disconnect of the mux while updating */
- if (val != mux)
- snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
- snd_soc_component_update_bits(component, TOHDMITX_CTRL0,
+ snd_soc_component_update_bits(component, e->reg,
CTRL0_I2S_DAT_SEL |
CTRL0_I2S_LRCLK_SEL |
CTRL0_I2S_BCLK_SEL,
@@ -131,30 +70,19 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol,
return 0;
}
+static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_i2s_mux_enum, TOHDMITX_CTRL0,
+ CTRL0_I2S_DAT_SEL_SHIFT,
+ g12a_tohdmitx_i2s_mux_texts);
+
static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux =
SOC_DAPM_ENUM_EXT("I2S Source", g12a_tohdmitx_i2s_mux_enum,
- g12a_tohdmitx_i2s_mux_get_enum,
+ snd_soc_dapm_get_enum_double,
g12a_tohdmitx_i2s_mux_put_enum);
static const char * const g12a_tohdmitx_spdif_mux_texts[] = {
"SPDIF A", "SPDIF B",
};
-static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_spdif_mux_enum,
- g12a_tohdmitx_spdif_mux_texts);
-
-static int g12a_tohdmitx_spdif_mux_get_enum(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_component *component =
- snd_soc_dapm_kcontrol_component(kcontrol);
-
- ucontrol->value.enumerated.item[0] =
- g12a_tohdmitx_get_input_val(component, CTRL0_SPDIF_SEL);
-
- return 0;
-}
-
static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -163,13 +91,18 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol,
struct snd_soc_dapm_context *dapm =
snd_soc_dapm_kcontrol_dapm(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int mux = ucontrol->value.enumerated.item[0];
- unsigned int val = g12a_tohdmitx_get_input_val(component,
- CTRL0_SPDIF_SEL);
+ unsigned int mux, changed;
+
+ mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+ changed = snd_soc_component_test_bits(component, TOHDMITX_CTRL0,
+ CTRL0_SPDIF_SEL,
+ FIELD_PREP(CTRL0_SPDIF_SEL, mux));
+
+ if (!changed)
+ return 0;
/* Force disconnect of the mux while updating */
- if (val != mux)
- snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
snd_soc_component_update_bits(component, TOHDMITX_CTRL0,
CTRL0_SPDIF_SEL |
@@ -182,9 +115,13 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol,
return 0;
}
+static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_spdif_mux_enum, TOHDMITX_CTRL0,
+ CTRL0_SPDIF_SEL_SHIFT,
+ g12a_tohdmitx_spdif_mux_texts);
+
static const struct snd_kcontrol_new g12a_tohdmitx_spdif_mux =
SOC_DAPM_ENUM_EXT("SPDIF Source", g12a_tohdmitx_spdif_mux_enum,
- g12a_tohdmitx_spdif_mux_get_enum,
+ snd_soc_dapm_get_enum_double,
g12a_tohdmitx_spdif_mux_put_enum);
static const struct snd_kcontrol_new g12a_tohdmitx_out_enable =
@@ -202,83 +139,13 @@ static const struct snd_soc_dapm_widget g12a_tohdmitx_widgets[] = {
&g12a_tohdmitx_out_enable),
};
-static int g12a_tohdmitx_input_probe(struct snd_soc_dai *dai)
-{
- struct g12a_tohdmitx_input *data;
-
- data = kzalloc(sizeof(*data), GFP_KERNEL);
- if (!data)
- return -ENOMEM;
-
- dai->playback_dma_data = data;
- return 0;
-}
-
-static int g12a_tohdmitx_input_remove(struct snd_soc_dai *dai)
-{
- kfree(dai->playback_dma_data);
- return 0;
-}
-
-static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
-{
- struct g12a_tohdmitx_input *data = dai->playback_dma_data;
-
- data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params));
- data->params.rate_min = params_rate(params);
- data->params.rate_max = params_rate(params);
- data->params.formats = 1 << params_format(params);
- data->params.channels_min = params_channels(params);
- data->params.channels_max = params_channels(params);
- data->params.sig_bits = dai->driver->playback.sig_bits;
-
- return 0;
-}
-
-
-static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai,
- unsigned int fmt)
-{
- struct g12a_tohdmitx_input *data = dai->playback_dma_data;
-
- /* Save the source stream format for the downstream link */
- data->fmt = fmt;
- return 0;
-}
-
-static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct g12a_tohdmitx_input *in_data =
- g12a_tohdmitx_get_input_data(dai->capture_widget);
-
- if (!in_data)
- return -ENODEV;
-
- if (WARN_ON(!rtd->dai_link->params)) {
- dev_warn(dai->dev, "codec2codec link expected\n");
- return -EINVAL;
- }
-
- /* Replace link params with the input params */
- rtd->dai_link->params = &in_data->params;
-
- if (!in_data->fmt)
- return 0;
-
- return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt);
-}
-
static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = {
- .hw_params = g12a_tohdmitx_input_hw_params,
- .set_fmt = g12a_tohdmitx_input_set_fmt,
+ .hw_params = meson_codec_glue_input_hw_params,
+ .set_fmt = meson_codec_glue_input_set_fmt,
};
static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = {
- .startup = g12a_tohdmitx_output_startup,
+ .startup = meson_codec_glue_output_startup,
};
#define TOHDMITX_SPDIF_FORMATS \
@@ -305,8 +172,8 @@ static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = {
.id = (xid), \
.playback = TOHDMITX_STREAM(xname, "Playback", xfmt, xchmax), \
.ops = &g12a_tohdmitx_input_ops, \
- .probe = g12a_tohdmitx_input_probe, \
- .remove = g12a_tohdmitx_input_remove, \
+ .probe = meson_codec_glue_input_dai_probe, \
+ .remove = meson_codec_glue_input_dai_remove, \
}
#define TOHDMITX_OUT(xname, xid, xfmt, xchmax) { \
diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c
new file mode 100644
index 000000000000..7b01dcb73e5e
--- /dev/null
+++ b/sound/soc/meson/gx-card.c
@@ -0,0 +1,141 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "meson-card.h"
+
+struct gx_dai_link_i2s_data {
+ unsigned int mclk_fs;
+};
+
+/*
+ * Base params for the codec to codec links
+ * Those will be over-written by the CPU side of the link
+ */
+static const struct snd_soc_pcm_stream codec_params = {
+ .formats = SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 5525,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 8,
+};
+
+static int gx_card_i2s_be_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct gx_dai_link_i2s_data *be =
+ (struct gx_dai_link_i2s_data *)priv->link_data[rtd->num];
+
+ return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs);
+}
+
+static const struct snd_soc_ops gx_card_i2s_be_ops = {
+ .hw_params = gx_card_i2s_be_hw_params,
+};
+
+static int gx_card_parse_i2s(struct snd_soc_card *card,
+ struct device_node *node,
+ int *index)
+{
+ struct meson_card *priv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai_link *link = &card->dai_link[*index];
+ struct gx_dai_link_i2s_data *be;
+
+ /* Allocate i2s link parameters */
+ be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL);
+ if (!be)
+ return -ENOMEM;
+ priv->link_data[*index] = be;
+
+ /* Setup i2s link */
+ link->ops = &gx_card_i2s_be_ops;
+ link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node);
+
+ of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
+
+ return 0;
+}
+
+static int gx_card_cpu_identify(struct snd_soc_dai_link_component *c,
+ char *match)
+{
+ if (of_device_is_compatible(c->of_node, DT_PREFIX "aiu")) {
+ if (strstr(c->dai_name, match))
+ return 1;
+ }
+
+ /* dai not matched */
+ return 0;
+}
+
+static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np,
+ int *index)
+{
+ struct snd_soc_dai_link *dai_link = &card->dai_link[*index];
+ struct snd_soc_dai_link_component *cpu;
+ int ret;
+
+ cpu = devm_kzalloc(card->dev, sizeof(*cpu), GFP_KERNEL);
+ if (!cpu)
+ return -ENOMEM;
+
+ dai_link->cpus = cpu;
+ dai_link->num_cpus = 1;
+
+ ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node,
+ &dai_link->cpus->dai_name);
+ if (ret)
+ return ret;
+
+ if (gx_card_cpu_identify(dai_link->cpus, "FIFO"))
+ ret = meson_card_set_fe_link(card, dai_link, np, true);
+ else
+ ret = meson_card_set_be_link(card, dai_link, np);
+
+ if (ret)
+ return ret;
+
+ /* Check if the cpu is the i2s encoder and parse i2s data */
+ if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
+ ret = gx_card_parse_i2s(card, np, index);
+
+ /* Or apply codec to codec params if necessary */
+ else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL"))
+ dai_link->params = &codec_params;
+
+ return ret;
+}
+
+static const struct meson_card_match_data gx_card_match_data = {
+ .add_link = gx_card_add_link,
+};
+
+static const struct of_device_id gx_card_of_match[] = {
+ {
+ .compatible = "amlogic,gx-sound-card",
+ .data = &gx_card_match_data,
+ }, {}
+};
+MODULE_DEVICE_TABLE(of, gx_card_of_match);
+
+static struct platform_driver gx_card_pdrv = {
+ .probe = meson_card_probe,
+ .remove = meson_card_remove,
+ .driver = {
+ .name = "gx-sound-card",
+ .of_match_table = gx_card_of_match,
+ },
+};
+module_platform_driver(gx_card_pdrv);
+
+MODULE_DESCRIPTION("Amlogic GX ALSA machine driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
new file mode 100644
index 000000000000..2ca8c98e204f
--- /dev/null
+++ b/sound/soc/meson/meson-card-utils.c
@@ -0,0 +1,385 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+#include "meson-card.h"
+
+int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ unsigned int mclk_fs)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
+ unsigned int mclk;
+ int ret, i;
+
+ if (!mclk_fs)
+ return 0;
+
+ mclk = params_rate(params) * mclk_fs;
+
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+ SND_SOC_CLOCK_IN);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk,
+ SND_SOC_CLOCK_OUT);
+ if (ret && ret != -ENOTSUPP)
+ return ret;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(meson_card_i2s_set_sysclk);
+
+int meson_card_reallocate_links(struct snd_soc_card *card,
+ unsigned int num_links)
+{
+ struct meson_card *priv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_dai_link *links;
+ void **ldata;
+
+ links = krealloc(priv->card.dai_link,
+ num_links * sizeof(*priv->card.dai_link),
+ GFP_KERNEL | __GFP_ZERO);
+ ldata = krealloc(priv->link_data,
+ num_links * sizeof(*priv->link_data),
+ GFP_KERNEL | __GFP_ZERO);
+
+ if (!links || !ldata) {
+ dev_err(priv->card.dev, "failed to allocate links\n");
+ return -ENOMEM;
+ }
+
+ priv->card.dai_link = links;
+ priv->link_data = ldata;
+ priv->card.num_links = num_links;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(meson_card_reallocate_links);
+
+int meson_card_parse_dai(struct snd_soc_card *card,
+ struct device_node *node,
+ struct device_node **dai_of_node,
+ const char **dai_name)
+{
+ struct of_phandle_args args;
+ int ret;
+
+ if (!dai_name || !dai_of_node || !node)
+ return -EINVAL;
+
+ ret = of_parse_phandle_with_args(node, "sound-dai",
+ "#sound-dai-cells", 0, &args);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(card->dev, "can't parse dai %d\n", ret);
+ return ret;
+ }
+ *dai_of_node = args.np;
+
+ return snd_soc_get_dai_name(&args, dai_name);
+}
+EXPORT_SYMBOL_GPL(meson_card_parse_dai);
+
+static int meson_card_set_link_name(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node,
+ const char *prefix)
+{
+ char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
+ prefix, node->full_name);
+ if (!name)
+ return -ENOMEM;
+
+ link->name = name;
+ link->stream_name = name;
+
+ return 0;
+}
+
+unsigned int meson_card_parse_daifmt(struct device_node *node,
+ struct device_node *cpu_node)
+{
+ struct device_node *bitclkmaster = NULL;
+ struct device_node *framemaster = NULL;
+ unsigned int daifmt;
+
+ daifmt = snd_soc_of_parse_daifmt(node, DT_PREFIX,
+ &bitclkmaster, &framemaster);
+ daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+
+ /* If no master is provided, default to cpu master */
+ if (!bitclkmaster || bitclkmaster == cpu_node) {
+ daifmt |= (!framemaster || framemaster == cpu_node) ?
+ SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM;
+ } else {
+ daifmt |= (!framemaster || framemaster == cpu_node) ?
+ SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM;
+ }
+
+ of_node_put(bitclkmaster);
+ of_node_put(framemaster);
+
+ return daifmt;
+}
+EXPORT_SYMBOL_GPL(meson_card_parse_daifmt);
+
+int meson_card_set_be_link(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node)
+{
+ struct snd_soc_dai_link_component *codec;
+ struct device_node *np;
+ int ret, num_codecs;
+
+ link->no_pcm = 1;
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
+
+ num_codecs = of_get_child_count(node);
+ if (!num_codecs) {
+ dev_err(card->dev, "be link %s has no codec\n",
+ node->full_name);
+ return -EINVAL;
+ }
+
+ codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL);
+ if (!codec)
+ return -ENOMEM;
+
+ link->codecs = codec;
+ link->num_codecs = num_codecs;
+
+ for_each_child_of_node(node, np) {
+ ret = meson_card_parse_dai(card, np, &codec->of_node,
+ &codec->dai_name);
+ if (ret) {
+ of_node_put(np);
+ return ret;
+ }
+
+ codec++;
+ }
+
+ ret = meson_card_set_link_name(card, link, node, "be");
+ if (ret)
+ dev_err(card->dev, "error setting %pOFn link name\n", np);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(meson_card_set_be_link);
+
+int meson_card_set_fe_link(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node,
+ bool is_playback)
+{
+ struct snd_soc_dai_link_component *codec;
+
+ codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL);
+ if (!codec)
+ return -ENOMEM;
+
+ link->codecs = codec;
+ link->num_codecs = 1;
+
+ link->dynamic = 1;
+ link->dpcm_merged_format = 1;
+ link->dpcm_merged_chan = 1;
+ link->dpcm_merged_rate = 1;
+ link->codecs->dai_name = "snd-soc-dummy-dai";
+ link->codecs->name = "snd-soc-dummy";
+
+ if (is_playback)
+ link->dpcm_playback = 1;
+ else
+ link->dpcm_capture = 1;
+
+ return meson_card_set_link_name(card, link, node, "fe");
+}
+EXPORT_SYMBOL_GPL(meson_card_set_fe_link);
+
+static int meson_card_add_links(struct snd_soc_card *card)
+{
+ struct meson_card *priv = snd_soc_card_get_drvdata(card);
+ struct device_node *node = card->dev->of_node;
+ struct device_node *np;
+ int num, i, ret;
+
+ num = of_get_child_count(node);
+ if (!num) {
+ dev_err(card->dev, "card has no links\n");
+ return -EINVAL;
+ }
+
+ ret = meson_card_reallocate_links(card, num);
+ if (ret)
+ return ret;
+
+ i = 0;
+ for_each_child_of_node(node, np) {
+ ret = priv->match_data->add_link(card, np, &i);
+ if (ret) {
+ of_node_put(np);
+ return ret;
+ }
+
+ i++;
+ }
+
+ return 0;
+}
+
+static int meson_card_parse_of_optional(struct snd_soc_card *card,
+ const char *propname,
+ int (*func)(struct snd_soc_card *c,
+ const char *p))
+{
+ /* If property is not provided, don't fail ... */
+ if (!of_property_read_bool(card->dev->of_node, propname))
+ return 0;
+
+ /* ... but do fail if it is provided and the parsing fails */
+ return func(card, propname);
+}
+
+static int meson_card_add_aux_devices(struct snd_soc_card *card)
+{
+ struct device_node *node = card->dev->of_node;
+ struct snd_soc_aux_dev *aux;
+ int num, i;
+
+ num = of_count_phandle_with_args(node, "audio-aux-devs", NULL);
+ if (num == -ENOENT) {
+ return 0;
+ } else if (num < 0) {
+ dev_err(card->dev, "error getting auxiliary devices: %d\n",
+ num);
+ return num;
+ }
+
+ aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL);
+ if (!aux)
+ return -ENOMEM;
+ card->aux_dev = aux;
+ card->num_aux_devs = num;
+
+ for_each_card_pre_auxs(card, i, aux) {
+ aux->dlc.of_node =
+ of_parse_phandle(node, "audio-aux-devs", i);
+ if (!aux->dlc.of_node)
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static void meson_card_clean_references(struct meson_card *priv)
+{
+ struct snd_soc_card *card = &priv->card;
+ struct snd_soc_dai_link *link;
+ struct snd_soc_dai_link_component *codec;
+ struct snd_soc_aux_dev *aux;
+ int i, j;
+
+ if (card->dai_link) {
+ for_each_card_prelinks(card, i, link) {
+ if (link->cpus)
+ of_node_put(link->cpus->of_node);
+ for_each_link_codecs(link, j, codec)
+ of_node_put(codec->of_node);
+ }
+ }
+
+ if (card->aux_dev) {
+ for_each_card_pre_auxs(card, i, aux)
+ of_node_put(aux->dlc.of_node);
+ }
+
+ kfree(card->dai_link);
+ kfree(priv->link_data);
+}
+
+int meson_card_probe(struct platform_device *pdev)
+{
+ const struct meson_card_match_data *data;
+ struct device *dev = &pdev->dev;
+ struct meson_card *priv;
+ int ret;
+
+ data = of_device_get_match_data(dev);
+ if (!data) {
+ dev_err(dev, "failed to match device\n");
+ return -ENODEV;
+ }
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, priv);
+ snd_soc_card_set_drvdata(&priv->card, priv);
+
+ priv->card.owner = THIS_MODULE;
+ priv->card.dev = dev;
+ priv->match_data = data;
+
+ ret = snd_soc_of_parse_card_name(&priv->card, "model");
+ if (ret < 0)
+ return ret;
+
+ ret = meson_card_parse_of_optional(&priv->card, "audio-routing",
+ snd_soc_of_parse_audio_routing);
+ if (ret) {
+ dev_err(dev, "error while parsing routing\n");
+ return ret;
+ }
+
+ ret = meson_card_parse_of_optional(&priv->card, "audio-widgets",
+ snd_soc_of_parse_audio_simple_widgets);
+ if (ret) {
+ dev_err(dev, "error while parsing widgets\n");
+ return ret;
+ }
+
+ ret = meson_card_add_links(&priv->card);
+ if (ret)
+ goto out_err;
+
+ ret = meson_card_add_aux_devices(&priv->card);
+ if (ret)
+ goto out_err;
+
+ ret = devm_snd_soc_register_card(dev, &priv->card);
+ if (ret)
+ goto out_err;
+
+ return 0;
+
+out_err:
+ meson_card_clean_references(priv);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(meson_card_probe);
+
+int meson_card_remove(struct platform_device *pdev)
+{
+ struct meson_card *priv = platform_get_drvdata(pdev);
+
+ meson_card_clean_references(priv);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(meson_card_remove);
+
+MODULE_DESCRIPTION("Amlogic Sound Card Utils");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/meson-card.h b/sound/soc/meson/meson-card.h
new file mode 100644
index 000000000000..74314071c80d
--- /dev/null
+++ b/sound/soc/meson/meson-card.h
@@ -0,0 +1,55 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright (c) 2020 BayLibre, SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_SND_CARD_H
+#define _MESON_SND_CARD_H
+
+struct device_node;
+struct platform_device;
+
+struct snd_soc_card;
+struct snd_pcm_substream;
+struct snd_pcm_hw_params;
+
+#define DT_PREFIX "amlogic,"
+
+struct meson_card_match_data {
+ int (*add_link)(struct snd_soc_card *card,
+ struct device_node *node,
+ int *index);
+};
+
+struct meson_card {
+ const struct meson_card_match_data *match_data;
+ struct snd_soc_card card;
+ void **link_data;
+};
+
+unsigned int meson_card_parse_daifmt(struct device_node *node,
+ struct device_node *cpu_node);
+
+int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ unsigned int mclk_fs);
+
+int meson_card_reallocate_links(struct snd_soc_card *card,
+ unsigned int num_links);
+int meson_card_parse_dai(struct snd_soc_card *card,
+ struct device_node *node,
+ struct device_node **dai_of_node,
+ const char **dai_name);
+int meson_card_set_be_link(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node);
+int meson_card_set_fe_link(struct snd_soc_card *card,
+ struct snd_soc_dai_link *link,
+ struct device_node *node,
+ bool is_playback);
+
+int meson_card_probe(struct platform_device *pdev);
+int meson_card_remove(struct platform_device *pdev);
+
+#endif /* _MESON_SND_CARD_H */
diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c
new file mode 100644
index 000000000000..524a33472337
--- /dev/null
+++ b/sound/soc/meson/meson-codec-glue.c
@@ -0,0 +1,149 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2019 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "meson-codec-glue.h"
+
+static struct snd_soc_dapm_widget *
+meson_codec_glue_get_input(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_path *p = NULL;
+ struct snd_soc_dapm_widget *in;
+
+ snd_soc_dapm_widget_for_each_source_path(w, p) {
+ if (!p->connect)
+ continue;
+
+ /* Check that we still are in the same component */
+ if (snd_soc_dapm_to_component(w->dapm) !=
+ snd_soc_dapm_to_component(p->source->dapm))
+ continue;
+
+ if (p->source->id == snd_soc_dapm_dai_in)
+ return p->source;
+
+ in = meson_codec_glue_get_input(p->source);
+ if (in)
+ return in;
+ }
+
+ return NULL;
+}
+
+static void meson_codec_glue_input_set_data(struct snd_soc_dai *dai,
+ struct meson_codec_glue_input *data)
+{
+ dai->playback_dma_data = data;
+}
+
+struct meson_codec_glue_input *
+meson_codec_glue_input_get_data(struct snd_soc_dai *dai)
+{
+ return dai->playback_dma_data;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_get_data);
+
+static struct meson_codec_glue_input *
+meson_codec_glue_output_get_input_data(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_widget *in =
+ meson_codec_glue_get_input(w);
+ struct snd_soc_dai *dai;
+
+ if (WARN_ON(!in))
+ return NULL;
+
+ dai = in->priv;
+
+ return meson_codec_glue_input_get_data(dai);
+}
+
+int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct meson_codec_glue_input *data =
+ meson_codec_glue_input_get_data(dai);
+
+ data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params));
+ data->params.rate_min = params_rate(params);
+ data->params.rate_max = params_rate(params);
+ data->params.formats = 1ULL << (__force int) params_format(params);
+ data->params.channels_min = params_channels(params);
+ data->params.channels_max = params_channels(params);
+ data->params.sig_bits = dai->driver->playback.sig_bits;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_hw_params);
+
+int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt)
+{
+ struct meson_codec_glue_input *data =
+ meson_codec_glue_input_get_data(dai);
+
+ /* Save the source stream format for the downstream link */
+ data->fmt = fmt;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_set_fmt);
+
+int meson_codec_glue_output_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct meson_codec_glue_input *in_data =
+ meson_codec_glue_output_get_input_data(dai->capture_widget);
+
+ if (!in_data)
+ return -ENODEV;
+
+ if (WARN_ON(!rtd->dai_link->params)) {
+ dev_warn(dai->dev, "codec2codec link expected\n");
+ return -EINVAL;
+ }
+
+ /* Replace link params with the input params */
+ rtd->dai_link->params = &in_data->params;
+
+ if (!in_data->fmt)
+ return 0;
+
+ return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt);
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_output_startup);
+
+int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai)
+{
+ struct meson_codec_glue_input *data;
+
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ meson_codec_glue_input_set_data(dai, data);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_probe);
+
+int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai)
+{
+ struct meson_codec_glue_input *data =
+ meson_codec_glue_input_get_data(dai);
+
+ kfree(data);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_remove);
+
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_DESCRIPTION("Amlogic Codec Glue Helpers");
+MODULE_LICENSE("GPL v2");
+
diff --git a/sound/soc/meson/meson-codec-glue.h b/sound/soc/meson/meson-codec-glue.h
new file mode 100644
index 000000000000..07f99446c0c6
--- /dev/null
+++ b/sound/soc/meson/meson-codec-glue.h
@@ -0,0 +1,32 @@
+/* SPDX-License-Identifier: GPL-2.0
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <jbrunet@baylibre.com>
+ */
+
+#ifndef _MESON_CODEC_GLUE_H
+#define _MESON_CODEC_GLUE_H
+
+#include <sound/soc.h>
+
+struct meson_codec_glue_input {
+ struct snd_soc_pcm_stream params;
+ unsigned int fmt;
+};
+
+/* Input helpers */
+struct meson_codec_glue_input *
+meson_codec_glue_input_get_data(struct snd_soc_dai *dai);
+int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai);
+int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt);
+int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai);
+int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai);
+
+/* Output helpers */
+int meson_codec_glue_output_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai);
+
+#endif /* _MESON_CODEC_GLUE_H */
diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c
new file mode 100644
index 000000000000..56d2592c16d5
--- /dev/null
+++ b/sound/soc/meson/t9015.c
@@ -0,0 +1,333 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <linux/reset.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#define BLOCK_EN 0x00
+#define LORN_EN 0
+#define LORP_EN 1
+#define LOLN_EN 2
+#define LOLP_EN 3
+#define DACR_EN 4
+#define DACL_EN 5
+#define DACR_INV 20
+#define DACL_INV 21
+#define DACR_SRC 22
+#define DACL_SRC 23
+#define REFP_BUF_EN BIT(12)
+#define BIAS_CURRENT_EN BIT(13)
+#define VMID_GEN_FAST BIT(14)
+#define VMID_GEN_EN BIT(15)
+#define I2S_MODE BIT(30)
+#define VOL_CTRL0 0x04
+#define GAIN_H 31
+#define GAIN_L 23
+#define VOL_CTRL1 0x08
+#define DAC_MONO 8
+#define RAMP_RATE 10
+#define VC_RAMP_MODE 12
+#define MUTE_MODE 13
+#define UNMUTE_MODE 14
+#define DAC_SOFT_MUTE 15
+#define DACR_VC 16
+#define DACL_VC 24
+#define LINEOUT_CFG 0x0c
+#define LORN_POL 0
+#define LORP_POL 4
+#define LOLN_POL 8
+#define LOLP_POL 12
+#define POWER_CFG 0x10
+
+struct t9015 {
+ struct clk *pclk;
+ struct regulator *avdd;
+};
+
+static int t9015_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_component *component = dai->component;
+ unsigned int val;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val = I2S_MODE;
+ break;
+
+ case SND_SOC_DAIFMT_CBS_CFS:
+ val = 0;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_component_update_bits(component, BLOCK_EN, I2S_MODE, val);
+
+ if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) &&
+ ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_LEFT_J))
+ return -EINVAL;
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops t9015_dai_ops = {
+ .set_fmt = t9015_dai_set_fmt,
+};
+
+static struct snd_soc_dai_driver t9015_dai = {
+ .name = "t9015-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = (SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ },
+ .ops = &t9015_dai_ops,
+};
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -9525, 0);
+
+static const char * const ramp_rate_txt[] = { "Fast", "Slow" };
+static SOC_ENUM_SINGLE_DECL(ramp_rate_enum, VOL_CTRL1, RAMP_RATE,
+ ramp_rate_txt);
+
+static const char * const dacr_in_txt[] = { "Right", "Left" };
+static SOC_ENUM_SINGLE_DECL(dacr_in_enum, BLOCK_EN, DACR_SRC, dacr_in_txt);
+
+static const char * const dacl_in_txt[] = { "Left", "Right" };
+static SOC_ENUM_SINGLE_DECL(dacl_in_enum, BLOCK_EN, DACL_SRC, dacl_in_txt);
+
+static const char * const mono_txt[] = { "Stereo", "Mono"};
+static SOC_ENUM_SINGLE_DECL(mono_enum, VOL_CTRL1, DAC_MONO, mono_txt);
+
+static const struct snd_kcontrol_new t9015_snd_controls[] = {
+ /* Volume Controls */
+ SOC_ENUM("Playback Channel Mode", mono_enum),
+ SOC_SINGLE("Playback Switch", VOL_CTRL1, DAC_SOFT_MUTE, 1, 1),
+ SOC_DOUBLE_TLV("Playback Volume", VOL_CTRL1, DACL_VC, DACR_VC,
+ 0xff, 0, dac_vol_tlv),
+
+ /* Ramp Controls */
+ SOC_ENUM("Ramp Rate", ramp_rate_enum),
+ SOC_SINGLE("Volume Ramp Switch", VOL_CTRL1, VC_RAMP_MODE, 1, 0),
+ SOC_SINGLE("Mute Ramp Switch", VOL_CTRL1, MUTE_MODE, 1, 0),
+ SOC_SINGLE("Unmute Ramp Switch", VOL_CTRL1, UNMUTE_MODE, 1, 0),
+};
+
+static const struct snd_kcontrol_new t9015_right_dac_mux =
+ SOC_DAPM_ENUM("Right DAC Source", dacr_in_enum);
+static const struct snd_kcontrol_new t9015_left_dac_mux =
+ SOC_DAPM_ENUM("Left DAC Source", dacl_in_enum);
+
+static const struct snd_soc_dapm_widget t9015_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("Right IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("Left IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_MUX("Right DAC Sel", SND_SOC_NOPM, 0, 0,
+ &t9015_right_dac_mux),
+ SND_SOC_DAPM_MUX("Left DAC Sel", SND_SOC_NOPM, 0, 0,
+ &t9015_left_dac_mux),
+ SND_SOC_DAPM_DAC("Right DAC", NULL, BLOCK_EN, DACR_EN, 0),
+ SND_SOC_DAPM_DAC("Left DAC", NULL, BLOCK_EN, DACL_EN, 0),
+ SND_SOC_DAPM_OUT_DRV("Right- Driver", BLOCK_EN, LORN_EN, 0,
+ NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Right+ Driver", BLOCK_EN, LORP_EN, 0,
+ NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Left- Driver", BLOCK_EN, LOLN_EN, 0,
+ NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Left+ Driver", BLOCK_EN, LOLP_EN, 0,
+ NULL, 0),
+ SND_SOC_DAPM_OUTPUT("LORN"),
+ SND_SOC_DAPM_OUTPUT("LORP"),
+ SND_SOC_DAPM_OUTPUT("LOLN"),
+ SND_SOC_DAPM_OUTPUT("LOLP"),
+};
+
+static const struct snd_soc_dapm_route t9015_dapm_routes[] = {
+ { "Right IN", NULL, "Playback" },
+ { "Left IN", NULL, "Playback" },
+ { "Right DAC Sel", "Right", "Right IN" },
+ { "Right DAC Sel", "Left", "Left IN" },
+ { "Left DAC Sel", "Right", "Right IN" },
+ { "Left DAC Sel", "Left", "Left IN" },
+ { "Right DAC", NULL, "Right DAC Sel" },
+ { "Left DAC", NULL, "Left DAC Sel" },
+ { "Right- Driver", NULL, "Right DAC" },
+ { "Right+ Driver", NULL, "Right DAC" },
+ { "Left- Driver", NULL, "Left DAC" },
+ { "Left+ Driver", NULL, "Left DAC" },
+ { "LORN", NULL, "Right- Driver", },
+ { "LORP", NULL, "Right+ Driver", },
+ { "LOLN", NULL, "Left- Driver", },
+ { "LOLP", NULL, "Left+ Driver", },
+};
+
+static int t9015_set_bias_level(struct snd_soc_component *component,
+ enum snd_soc_bias_level level)
+{
+ struct t9015 *priv = snd_soc_component_get_drvdata(component);
+ enum snd_soc_bias_level now =
+ snd_soc_component_get_bias_level(component);
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ snd_soc_component_update_bits(component, BLOCK_EN,
+ BIAS_CURRENT_EN,
+ BIAS_CURRENT_EN);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ snd_soc_component_update_bits(component, BLOCK_EN,
+ BIAS_CURRENT_EN,
+ 0);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ ret = regulator_enable(priv->avdd);
+ if (ret) {
+ dev_err(component->dev, "AVDD enable failed\n");
+ return ret;
+ }
+
+ if (now == SND_SOC_BIAS_OFF) {
+ snd_soc_component_update_bits(component, BLOCK_EN,
+ VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN,
+ VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN);
+
+ mdelay(200);
+ snd_soc_component_update_bits(component, BLOCK_EN,
+ VMID_GEN_FAST,
+ 0);
+ }
+
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_component_update_bits(component, BLOCK_EN,
+ VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN,
+ 0);
+
+ regulator_disable(priv->avdd);
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver t9015_codec_driver = {
+ .set_bias_level = t9015_set_bias_level,
+ .controls = t9015_snd_controls,
+ .num_controls = ARRAY_SIZE(t9015_snd_controls),
+ .dapm_widgets = t9015_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(t9015_dapm_widgets),
+ .dapm_routes = t9015_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(t9015_dapm_routes),
+ .suspend_bias_off = 1,
+ .endianness = 1,
+ .non_legacy_dai_naming = 1,
+};
+
+static const struct regmap_config t9015_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = POWER_CFG,
+};
+
+static int t9015_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct t9015 *priv;
+ void __iomem *regs;
+ struct regmap *regmap;
+ int ret;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, priv);
+
+ priv->pclk = devm_clk_get(dev, "pclk");
+ if (IS_ERR(priv->pclk)) {
+ if (PTR_ERR(priv->pclk) != -EPROBE_DEFER)
+ dev_err(dev, "failed to get core clock\n");
+ return PTR_ERR(priv->pclk);
+ }
+
+ priv->avdd = devm_regulator_get(dev, "AVDD");
+ if (IS_ERR(priv->avdd)) {
+ if (PTR_ERR(priv->avdd) != -EPROBE_DEFER)
+ dev_err(dev, "failed to AVDD\n");
+ return PTR_ERR(priv->avdd);
+ }
+
+ ret = clk_prepare_enable(priv->pclk);
+ if (ret) {
+ dev_err(dev, "core clock enable failed\n");
+ return ret;
+ }
+
+ ret = devm_add_action_or_reset(dev,
+ (void(*)(void *))clk_disable_unprepare,
+ priv->pclk);
+ if (ret)
+ return ret;
+
+ ret = device_reset(dev);
+ if (ret) {
+ dev_err(dev, "reset failed\n");
+ return ret;
+ }
+
+ regs = devm_platform_ioremap_resource(pdev, 0);
+ if (IS_ERR(regs)) {
+ dev_err(dev, "register map failed\n");
+ return PTR_ERR(regs);
+ }
+
+ regmap = devm_regmap_init_mmio(dev, regs, &t9015_regmap_config);
+ if (IS_ERR(regmap)) {
+ dev_err(dev, "regmap init failed\n");
+ return PTR_ERR(regmap);
+ }
+
+ /*
+ * Initialize output polarity:
+ * ATM the output polarity is fixed but in the future it might useful
+ * to add DT property to set this depending on the platform needs
+ */
+ regmap_write(regmap, LINEOUT_CFG, 0x1111);
+
+ return devm_snd_soc_register_component(dev, &t9015_codec_driver,
+ &t9015_dai, 1);
+}
+
+static const struct of_device_id t9015_ids[] = {
+ { .compatible = "amlogic,t9015", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, t9015_ids);
+
+static struct platform_driver t9015_driver = {
+ .driver = {
+ .name = "t9015-codec",
+ .of_match_table = of_match_ptr(t9015_ids),
+ },
+ .probe = t9015_probe,
+};
+
+module_platform_driver(t9015_driver);
+
+MODULE_DESCRIPTION("ASoC Amlogic T9015 codec driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 9841e1da9782..f46d7aca8cf6 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -20,8 +20,8 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int rate = params_rate(params);
u32 mclk;
int ret;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 295cfffa4646..d4c0f580a565 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -81,6 +81,9 @@ config SND_PXA2XX_SOC_TOSA
depends on SND_PXA2XX_SOC && MACH_TOSA
depends on MFD_TC6393XB
depends on AC97_BUS=n
+ select REGMAP
+ select AC97_BUS_NEW
+ select AC97_BUS_COMPAT
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
@@ -91,6 +94,9 @@ config SND_PXA2XX_SOC_E740
tristate "SoC AC97 Audio support for e740"
depends on SND_PXA2XX_SOC && MACH_E740
depends on AC97_BUS=n
+ select REGMAP
+ select AC97_BUS_NEW
+ select AC97_BUS_COMPAT
select SND_SOC_WM9705
select SND_PXA2XX_SOC_AC97
help
@@ -101,6 +107,7 @@ config SND_PXA2XX_SOC_E750
tristate "SoC AC97 Audio support for e750"
depends on SND_PXA2XX_SOC && MACH_E750
depends on AC97_BUS=n
+ select REGMAP
select SND_SOC_WM9705
select SND_PXA2XX_SOC_AC97
help
@@ -111,7 +118,10 @@ config SND_PXA2XX_SOC_E800
tristate "SoC AC97 Audio support for e800"
depends on SND_PXA2XX_SOC && MACH_E800
depends on AC97_BUS=n
+ select REGMAP
select SND_SOC_WM9712
+ select AC97_BUS_NEW
+ select AC97_BUS_COMPAT
select SND_PXA2XX_SOC_AC97
help
Say Y if you want to add support for SoC audio on the
@@ -122,6 +132,9 @@ config SND_PXA2XX_SOC_EM_X270
depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
MACH_CM_X300)
depends on AC97_BUS=n
+ select REGMAP
+ select AC97_BUS_NEW
+ select AC97_BUS_COMPAT
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
@@ -133,6 +146,9 @@ config SND_PXA2XX_SOC_PALM27X
depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
MACH_PALMT5 || MACH_PALMTE2)
depends on AC97_BUS=n
+ select REGMAP
+ select AC97_BUS_NEW
+ select AC97_BUS_COMPAT
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
@@ -163,7 +179,10 @@ config SND_SOC_ZYLONITE
tristate "SoC Audio support for Marvell Zylonite"
depends on SND_PXA2XX_SOC && MACH_ZYLONITE
depends on AC97_BUS=n
+ select AC97_BUS_NEW
+ select AC97_BUS_COMPAT
select SND_PXA2XX_SOC_AC97
+ select REGMAP
select SND_PXA_SOC_SSP
select SND_SOC_WM9713
help
@@ -193,6 +212,9 @@ config SND_PXA2XX_SOC_MIOA701
tristate "SoC Audio support for MIO A701"
depends on SND_PXA2XX_SOC && MACH_MIOA701
depends on AC97_BUS=n
+ select REGMAP
+ select AC97_BUS_NEW
+ select AC97_BUS_COMPAT
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9713
help
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 53b1435ced3f..016a91199485 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -44,8 +44,8 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int freq_out, sspa_mclk, sysclk;
if (params_rate(params) > 11025) {
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index d81082323fb4..6fbef9a0afa7 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -116,8 +116,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int clk = 0;
int ret = 0;
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 0139343dbcce..b4da9a9a6521 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -54,8 +54,8 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret = 0;
/* set the I2S system clock as output */
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index 514e17724fc3..3014e8244ab4 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -12,8 +12,8 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int clk = 0;
int ret;
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 6483cff5b73d..e4c818f4cd62 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -83,8 +83,8 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int width;
int ret = 0;
@@ -121,8 +121,8 @@ static int magician_capture_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret = 0;
/* set codec DAI configuration */
@@ -358,10 +358,10 @@ static int __init magician_init(void)
adapter = i2c_get_adapter(0);
if (!adapter)
return -ENODEV;
- client = i2c_new_device(adapter, i2c_board_info);
+ client = i2c_new_client_device(adapter, i2c_board_info);
i2c_put_adapter(adapter);
- if (!client)
- return -ENODEV;
+ if (IS_ERR(client))
+ return PTR_ERR(client);
ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
if (ret)
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 76e054d514a8..bf27b277c01f 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -73,7 +73,7 @@ static int rear_amp_event(struct snd_soc_dapm_widget *widget,
struct snd_soc_component *component;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- component = rtd->codec_dai->component;
+ component = asoc_rtd_to_codec(rtd, 0)->component;
return rear_amp_power(component, SND_SOC_DAPM_EVENT_ON(event));
}
@@ -117,7 +117,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
/* Prepare GPIO8 for rear speaker amplifier */
snd_soc_component_update_bits(component, AC97_GPIO_CFG, 0x100, 0x100);
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 287b5da739e5..3fe6c4c5a3ab 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -112,7 +112,7 @@ static int mmp_pcm_open(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct platform_device *pdev = to_platform_device(component->dev);
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct mmp_dma_data dma_data;
struct resource *r;
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
index e701637a9ae9..3548a2634a63 100644
--- a/sound/soc/pxa/mmp-sspa.c
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -251,7 +251,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
struct ssp_device *sspa = sspa_priv->sspa;
struct snd_dmaengine_dai_dma_data *dma_params;
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 59ef04d0467a..287984a564c8 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -90,8 +90,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int clk = 0;
int ret = 0;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 5f1c477b5833..9a32bf72127a 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -96,7 +96,7 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
if (IS_ERR(clk_i2s))
return PTR_ERR(clk_i2s);
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index f7babffb7228..6d8174f62935 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -117,8 +117,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int clk = 0;
int ret = 0;
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index d8f79e2266b1..d5f2961b1a3e 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -61,7 +61,7 @@ static const struct snd_soc_dapm_route ttc_audio_map[] = {
static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
/* Headset jack detection */
snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE |
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index f9a33cb36f5b..6eee1aefc89a 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -34,8 +34,8 @@ static int z2_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int clk = 0;
int ret = 0;
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 567dc133ea92..447b59b8bd33 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -66,7 +66,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
if (clk_pout)
- snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
+ snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0,
clk_get_rate(pout), 0);
return 0;
@@ -76,8 +76,8 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int wm9713_div = 0;
int ret = 0;
int rate = params_rate(params);
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 6530d2462a9e..f51b28d1b94d 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -99,7 +99,7 @@ config SND_SOC_MSM8996
config SND_SOC_SDM845
tristate "SoC Machine driver for SDM845 boards"
- depends on QCOM_APR && CROS_EC && I2C
+ depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE
select SND_SOC_QDSP6
select SND_SOC_QCOM_COMMON
select SND_SOC_RT5663
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index ac75838bbfab..2ef090f4af9e 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -33,9 +33,9 @@ struct apq8016_sbc_data {
static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
struct snd_soc_component *component;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_card *card = rtd->card;
struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card);
int i, rval;
@@ -90,10 +90,9 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
pdata->jack_setup = true;
}
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
- component = dai->component;
+ component = codec_dai->component;
/* Set default mclk for internal codec */
rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE,
SND_SOC_CLOCK_IN);
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 94363fd6846a..d55e3ad96716 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -31,8 +31,8 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
int ret = 0;
@@ -66,7 +66,7 @@ static struct snd_soc_ops apq8096_ops = {
static int apq8096_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/*
* Codec SLIMBUS configuration
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index b05091c283b7..34f7fd1bab1c 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -55,7 +55,7 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0);
struct lpass_data *drvdata = snd_soc_component_get_drvdata(component);
struct lpass_variant *v = drvdata->variant;
int ret, dma_ch, dir = substream->stream;
@@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component,
struct snd_pcm_substream *substream;
int i;
- for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ for_each_pcm_streams(i) {
substream = pcm->streams[i].substream;
if (substream) {
snd_dma_free_pages(&substream->dma_buffer);
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index c0d422d0ab94..f6c7cddf08e8 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -41,6 +41,9 @@
#define Q6ASM_DAI_TX 1
#define Q6ASM_DAI_RX 2
+#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1)
+#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
+
enum stream_state {
Q6ASM_STREAM_IDLE = 0,
Q6ASM_STREAM_STOPPED,
@@ -69,6 +72,8 @@ struct q6asm_dai_rtd {
};
struct q6asm_dai_data {
+ struct snd_soc_dai_driver *dais;
+ int num_dais;
long long int sid;
};
@@ -250,7 +255,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
- prtd->bits_per_sample);
+ 0, prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
prtd->bits_per_sample);
@@ -328,7 +333,7 @@ static int q6asm_dai_open(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
struct q6asm_dai_rtd *prtd;
struct q6asm_dai_data *pdata;
struct device *dev = component->dev;
@@ -540,7 +545,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
struct snd_soc_pcm_runtime *rtd = stream->private_data;
struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct snd_compr_runtime *runtime = stream->runtime;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct q6asm_dai_data *pdata;
struct device *dev = c->dev;
struct q6asm_dai_rtd *prtd;
@@ -627,10 +632,17 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
int dir = stream->direction;
struct q6asm_dai_data *pdata;
struct q6asm_flac_cfg flac_cfg;
+ struct q6asm_wma_cfg wma_cfg;
+ struct q6asm_alac_cfg alac_cfg;
+ struct q6asm_ape_cfg ape_cfg;
+ unsigned int wma_v9 = 0;
struct device *dev = c->dev;
int ret;
union snd_codec_options *codec_options;
struct snd_dec_flac *flac;
+ struct snd_dec_wma *wma;
+ struct snd_dec_alac *alac;
+ struct snd_dec_ape *ape;
codec_options = &(prtd->codec_param.codec.options);
@@ -652,7 +664,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
prtd->bits_per_sample = 16;
if (dir == SND_COMPRESS_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, params->codec.id,
- prtd->bits_per_sample);
+ params->codec.profile, prtd->bits_per_sample);
if (ret < 0) {
dev_err(dev, "q6asm_open_write failed\n");
@@ -692,6 +704,126 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
return -EIO;
}
break;
+
+ case SND_AUDIOCODEC_WMA:
+ wma = &codec_options->wma_d;
+
+ memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+ wma_cfg.sample_rate = params->codec.sample_rate;
+ wma_cfg.num_channels = params->codec.ch_in;
+ wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
+ wma_cfg.block_align = params->codec.align;
+ wma_cfg.bits_per_sample = prtd->bits_per_sample;
+ wma_cfg.enc_options = wma->encoder_option;
+ wma_cfg.adv_enc_options = wma->adv_encoder_option;
+ wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+ if (wma_cfg.num_channels == 1)
+ wma_cfg.channel_mask = 4; /* Mono Center */
+ else if (wma_cfg.num_channels == 2)
+ wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+ else
+ return -EINVAL;
+
+ /* check the codec profile */
+ switch (params->codec.profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ wma_cfg.fmtag = 0x161;
+ wma_v9 = 1;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10:
+ wma_cfg.fmtag = 0x166;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ wma_cfg.fmtag = 0x162;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ wma_cfg.fmtag = 0x163;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ wma_cfg.fmtag = 0x167;
+ break;
+
+ default:
+ dev_err(dev, "Unknown WMA profile:%x\n",
+ params->codec.profile);
+ return -EIO;
+ }
+
+ if (wma_v9)
+ ret = q6asm_stream_media_format_block_wma_v9(
+ prtd->audio_client, &wma_cfg);
+ else
+ ret = q6asm_stream_media_format_block_wma_v10(
+ prtd->audio_client, &wma_cfg);
+ if (ret < 0) {
+ dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_ALAC:
+ memset(&alac_cfg, 0x0, sizeof(alac_cfg));
+ alac = &codec_options->alac_d;
+
+ alac_cfg.sample_rate = params->codec.sample_rate;
+ alac_cfg.avg_bit_rate = params->codec.bit_rate;
+ alac_cfg.bit_depth = prtd->bits_per_sample;
+ alac_cfg.num_channels = params->codec.ch_in;
+
+ alac_cfg.frame_length = alac->frame_length;
+ alac_cfg.pb = alac->pb;
+ alac_cfg.mb = alac->mb;
+ alac_cfg.kb = alac->kb;
+ alac_cfg.max_run = alac->max_run;
+ alac_cfg.compatible_version = alac->compatible_version;
+ alac_cfg.max_frame_bytes = alac->max_frame_bytes;
+
+ switch (params->codec.ch_in) {
+ case 1:
+ alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
+ break;
+ case 2:
+ alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
+ break;
+ }
+ ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+ &alac_cfg);
+ if (ret < 0) {
+ dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_APE:
+ memset(&ape_cfg, 0x0, sizeof(ape_cfg));
+ ape = &codec_options->ape_d;
+
+ ape_cfg.sample_rate = params->codec.sample_rate;
+ ape_cfg.num_channels = params->codec.ch_in;
+ ape_cfg.bits_per_sample = prtd->bits_per_sample;
+
+ ape_cfg.compatible_version = ape->compatible_version;
+ ape_cfg.compression_level = ape->compression_level;
+ ape_cfg.format_flags = ape->format_flags;
+ ape_cfg.blocks_per_frame = ape->blocks_per_frame;
+ ape_cfg.final_frame_blocks = ape->final_frame_blocks;
+ ape_cfg.total_frames = ape->total_frames;
+ ape_cfg.seek_table_present = ape->seek_table_present;
+
+ ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+ &ape_cfg);
+ if (ret < 0) {
+ dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
default:
break;
}
@@ -791,9 +923,12 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream,
caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
- caps->num_codecs = 2;
+ caps->num_codecs = 5;
caps->codecs[0] = SND_AUDIOCODEC_MP3;
caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+ caps->codecs[2] = SND_AUDIOCODEC_WMA;
+ caps->codecs[3] = SND_AUDIOCODEC_ALAC;
+ caps->codecs[4] = SND_AUDIOCODEC_APE;
return 0;
}
@@ -889,7 +1024,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = {
.compr_ops = &q6asm_dai_compr_ops,
};
-static struct snd_soc_dai_driver q6asm_fe_dais[] = {
+static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
Q6ASM_FEDAI_DRIVER(1),
Q6ASM_FEDAI_DRIVER(2),
Q6ASM_FEDAI_DRIVER(3),
@@ -903,10 +1038,22 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = {
static int of_q6asm_parse_dai_data(struct device *dev,
struct q6asm_dai_data *pdata)
{
- static struct snd_soc_dai_driver *dai_drv;
+ struct snd_soc_dai_driver *dai_drv;
struct snd_soc_pcm_stream empty_stream;
struct device_node *node;
- int ret, id, dir;
+ int ret, id, dir, idx = 0;
+
+
+ pdata->num_dais = of_get_child_count(dev->of_node);
+ if (!pdata->num_dais) {
+ dev_err(dev, "No dais found in DT\n");
+ return -EINVAL;
+ }
+
+ pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
+ GFP_KERNEL);
+ if (!pdata->dais)
+ return -ENOMEM;
memset(&empty_stream, 0, sizeof(empty_stream));
@@ -917,7 +1064,8 @@ static int of_q6asm_parse_dai_data(struct device *dev,
continue;
}
- dai_drv = &q6asm_fe_dais[id];
+ dai_drv = &pdata->dais[idx++];
+ *dai_drv = q6asm_fe_dais_template[id];
ret = of_property_read_u32(node, "direction", &dir);
if (ret)
@@ -955,11 +1103,12 @@ static int q6asm_dai_probe(struct platform_device *pdev)
dev_set_drvdata(dev, pdata);
- of_q6asm_parse_dai_data(dev, pdata);
+ rc = of_q6asm_parse_dai_data(dev, pdata);
+ if (rc)
+ return rc;
return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
- q6asm_fe_dais,
- ARRAY_SIZE(q6asm_fe_dais));
+ pdata->dais, pdata->num_dais);
}
static const struct of_device_id q6asm_dai_device_id[] = {
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 36e0eab13a98..0e0e8f7a460a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -39,6 +39,8 @@
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_FLAC 0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
#define ASM_DATA_CMD_READ_V2 0x00010DAC
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
@@ -46,6 +48,8 @@
#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+#define ASM_MEDIA_FMT_ALAC 0x00012f31
+#define ASM_MEDIA_FMT_APE 0x00012f32
#define ASM_LEGACY_STREAM_SESSION 0
@@ -104,6 +108,63 @@ struct asm_flac_fmt_blk_v2 {
u16 reserved;
} __packed;
+struct asm_wmastdv9_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 advanced_enc_options1;
+ u32 advanced_enc_options2;
+} __packed;
+
+struct asm_alac_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+} __packed;
+
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
u32 param_size;
@@ -858,7 +919,7 @@ err:
* Return: Will be an negative value on error or zero on success
*/
int q6asm_open_write(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample)
+ u32 codec_profile, uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_write_v3 *open;
struct apr_pkt *pkt;
@@ -894,6 +955,30 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
case SND_AUDIOCODEC_FLAC:
open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
break;
+ case SND_AUDIOCODEC_WMA:
+ switch (codec_profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+ break;
+ case SND_AUDIOPROFILE_WMA10:
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+ break;
+ default:
+ dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+ codec_profile);
+ rc = -EINVAL;
+ goto err;
+ }
+ break;
+ case SND_AUDIOCODEC_ALAC:
+ open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+ break;
+ case SND_AUDIOCODEC_APE:
+ open->dec_fmt_id = ASM_MEDIA_FMT_APE;
+ break;
default:
dev_err(ac->dev, "Invalid format 0x%x\n", format);
rc = -EINVAL;
@@ -1075,6 +1160,162 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmastdv9_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->reserved = 0;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmaprov10_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->advanced_enc_options1 = cfg->adv_enc_options;
+ fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ struct q6asm_alac_cfg *cfg)
+{
+ struct asm_alac_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->frame_length = cfg->frame_length;
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->bit_depth = cfg->bit_depth;
+ fmt->num_channels = cfg->num_channels;
+ fmt->max_run = cfg->max_run;
+ fmt->max_frame_bytes = cfg->max_frame_bytes;
+ fmt->avg_bit_rate = cfg->avg_bit_rate;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->channel_layout_tag = cfg->channel_layout_tag;
+ fmt->pb = cfg->pb;
+ fmt->mb = cfg->mb;
+ fmt->kb = cfg->kb;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ struct q6asm_ape_cfg *cfg)
+{
+ struct asm_ape_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->compression_level = cfg->compression_level;
+ fmt->format_flags = cfg->format_flags;
+ fmt->blocks_per_frame = cfg->blocks_per_frame;
+ fmt->final_frame_blocks = cfg->final_frame_blocks;
+ fmt->total_frames = cfg->total_frames;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->seek_table_present = cfg->seek_table_present;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 6764f55f7078..38a207d6cd95 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -45,6 +45,47 @@ struct q6asm_flac_cfg {
u16 md5_sum;
};
+struct q6asm_wma_cfg {
+ u32 fmtag;
+ u32 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u32 block_align;
+ u32 bits_per_sample;
+ u32 channel_mask;
+ u32 enc_options;
+ u32 adv_enc_options;
+ u32 adv_enc_options2;
+};
+
+struct q6asm_alac_cfg {
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+};
+
+struct q6asm_ape_cfg {
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+};
+
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
struct audio_client;
@@ -55,7 +96,7 @@ void q6asm_audio_client_free(struct audio_client *ac);
int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags);
int q6asm_open_write(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample);
+ u32 codec_profile, uint16_t bits_per_sample);
int q6asm_open_read(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample);
@@ -69,6 +110,14 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint16_t bits_per_sample);
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
struct q6asm_flac_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ struct q6asm_alac_cfg *cfg);
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ struct q6asm_ape_cfg *cfg);
int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
uint32_t lsw_ts);
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 20724102e85a..46e50612b92c 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -918,25 +918,6 @@ static const struct snd_soc_dapm_route intercon[] = {
{"MM_UL6", NULL, "MultiMedia6 Mixer"},
{"MM_UL7", NULL, "MultiMedia7 Mixer"},
{"MM_UL8", NULL, "MultiMedia8 Mixer"},
-
- {"MM_DL1", NULL, "MultiMedia1 Playback" },
- {"MM_DL2", NULL, "MultiMedia2 Playback" },
- {"MM_DL3", NULL, "MultiMedia3 Playback" },
- {"MM_DL4", NULL, "MultiMedia4 Playback" },
- {"MM_DL5", NULL, "MultiMedia5 Playback" },
- {"MM_DL6", NULL, "MultiMedia6 Playback" },
- {"MM_DL7", NULL, "MultiMedia7 Playback" },
- {"MM_DL8", NULL, "MultiMedia8 Playback" },
-
- {"MultiMedia1 Capture", NULL, "MM_UL1"},
- {"MultiMedia2 Capture", NULL, "MM_UL2"},
- {"MultiMedia3 Capture", NULL, "MM_UL3"},
- {"MultiMedia4 Capture", NULL, "MM_UL4"},
- {"MultiMedia5 Capture", NULL, "MM_UL5"},
- {"MultiMedia6 Capture", NULL, "MM_UL6"},
- {"MultiMedia7 Capture", NULL, "MM_UL7"},
- {"MultiMedia8 Capture", NULL, "MM_UL8"},
-
};
static int routing_hw_params(struct snd_soc_component *component,
@@ -945,7 +926,7 @@ static int routing_hw_params(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct msm_routing_data *data = dev_get_drvdata(component->dev);
- unsigned int be_id = rtd->cpu_dai->id;
+ unsigned int be_id = asoc_rtd_to_cpu(rtd, 0)->id;
struct session_data *session;
int path_type;
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 3b5547a27aad..b2de65c7f95c 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -11,6 +11,7 @@
#include <sound/pcm_params.h>
#include <sound/jack.h>
#include <sound/soc.h>
+#include <linux/soundwire/sdw.h>
#include <uapi/linux/input-event-codes.h>
#include "common.h"
#include "qdsp6/q6afe.h"
@@ -31,10 +32,12 @@
struct sdm845_snd_data {
struct snd_soc_jack jack;
bool jack_setup;
+ bool stream_prepared[SLIM_MAX_RX_PORTS];
struct snd_soc_card *card;
uint32_t pri_mi2s_clk_count;
uint32_t sec_mi2s_clk_count;
uint32_t quat_tdm_clk_count;
+ struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS];
};
static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
@@ -43,14 +46,21 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
+ struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card);
u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ struct sdw_stream_runtime *sruntime;
u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
int ret = 0, i;
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i],
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ sruntime = snd_soc_dai_get_sdw_stream(codec_dai,
+ substream->stream);
+ if (sruntime != ERR_PTR(-ENOTSUPP))
+ pdata->sruntime[cpu_dai->id] = sruntime;
+
+ ret = snd_soc_dai_get_channel_map(codec_dai,
&tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch);
if (ret != 0 && ret != -ENOTSUPP) {
@@ -76,7 +86,8 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai;
int ret = 0, j;
int channels, slot_width;
@@ -125,8 +136,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
}
}
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name_prefix, "Left")) {
ret = snd_soc_dai_set_tdm_slot(
@@ -161,8 +171,8 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret = 0;
switch (cpu_dai->id) {
@@ -210,11 +220,10 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_component *component;
struct snd_soc_card *card = rtd->card;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
struct snd_jack *jack;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
/*
* Codec SLIMBUS configuration
* RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13
@@ -266,8 +275,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
}
break;
case SLIMBUS_0_RX...SLIMBUS_6_TX:
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i],
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ rval = snd_soc_dai_set_channel_map(codec_dai,
ARRAY_SIZE(tx_ch),
tx_ch,
ARRAY_SIZE(rx_ch),
@@ -275,7 +284,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
if (rval != 0 && rval != -ENOTSUPP)
return rval;
- snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0,
+ snd_soc_dai_set_sysclk(codec_dai, 0,
WCD934X_DEFAULT_MCLK_RATE,
SNDRV_PCM_STREAM_PLAYBACK);
}
@@ -295,8 +304,8 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int j;
int ret;
@@ -345,8 +354,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B;
- for (j = 0; j < rtd->num_codecs; j++) {
- codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dais(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name_prefix,
"Left")) {
@@ -386,7 +394,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_card *card = rtd->card;
struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
switch (cpu_dai->id) {
case PRIMARY_MI2S_RX:
@@ -427,8 +435,65 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
}
}
+static int sdm845_snd_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
+ int ret;
+
+ if (!sruntime)
+ return 0;
+
+ if (data->stream_prepared[cpu_dai->id]) {
+ sdw_disable_stream(sruntime);
+ sdw_deprepare_stream(sruntime);
+ data->stream_prepared[cpu_dai->id] = false;
+ }
+
+ ret = sdw_prepare_stream(sruntime);
+ if (ret)
+ return ret;
+
+ /**
+ * NOTE: there is a strict hw requirement about the ordering of port
+ * enables and actual WSA881x PA enable. PA enable should only happen
+ * after soundwire ports are enabled if not DC on the line is
+ * accumulated resulting in Click/Pop Noise
+ * PA enable/mute are handled as part of codec DAPM and digital mute.
+ */
+
+ ret = sdw_enable_stream(sruntime);
+ if (ret) {
+ sdw_deprepare_stream(sruntime);
+ return ret;
+ }
+ data->stream_prepared[cpu_dai->id] = true;
+
+ return ret;
+}
+
+static int sdm845_snd_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
+
+ if (sruntime && data->stream_prepared[cpu_dai->id]) {
+ sdw_disable_stream(sruntime);
+ sdw_deprepare_stream(sruntime);
+ data->stream_prepared[cpu_dai->id] = false;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_ops sdm845_be_ops = {
.hw_params = sdm845_snd_hw_params,
+ .hw_free = sdm845_snd_hw_free,
+ .prepare = sdm845_snd_prepare,
.startup = sdm845_snd_startup,
.shutdown = sdm845_snd_shutdown,
};
diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c
index e6666e597265..3a6e18709b9e 100644
--- a/sound/soc/qcom/storm.c
+++ b/sound/soc/qcom/storm.c
@@ -39,7 +39,7 @@ static int storm_ops_hw_params(struct snd_pcm_substream *substream,
*/
sysclk_freq = rate * bitwidth * 2 * STORM_SYSCLK_MULT;
- ret = snd_soc_dai_set_sysclk(soc_runtime->cpu_dai, 0, sysclk_freq, 0);
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(soc_runtime, 0), 0, sysclk_freq, 0);
if (ret) {
dev_err(card->dev, "error setting sysclk to %u: %d\n",
sysclk_freq, ret);
diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c
index 767700c34ee2..01078155a914 100644
--- a/sound/soc/rockchip/rk3288_hdmi_analog.c
+++ b/sound/soc/rockchip/rk3288_hdmi_analog.c
@@ -67,8 +67,8 @@ static int rk_hw_params(struct snd_pcm_substream *substream,
{
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int mclk;
switch (params_rate(params)) {
diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index d951100bf770..f45e5aaa4b30 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -57,7 +57,7 @@ static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substrea
mclk = params_rate(params) * SOUND_FS;
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0);
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0);
if (ret) {
dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n",
__func__, mclk, ret);
@@ -71,8 +71,8 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
unsigned int mclk;
int ret;
@@ -103,8 +103,8 @@ static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int mclk, ret;
/* in bypass mode, the mclk has to be one of the frequencies below */
@@ -153,8 +153,8 @@ static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream,
static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dais[0]->component;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
/* We need default MCLK and PLL settings for the accessory detection */
@@ -206,7 +206,7 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream,
mclk = params_rate(params) * SOUND_FS;
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0);
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0);
if (ret) {
dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n",
__func__, mclk, ret);
diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c
index 60930fa85aa4..1f527d3763ce 100644
--- a/sound/soc/rockchip/rockchip_max98090.c
+++ b/sound/soc/rockchip/rockchip_max98090.c
@@ -146,8 +146,8 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream,
{
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int mclk;
switch (params_rate(params)) {
@@ -227,7 +227,7 @@ static struct snd_soc_jack rk_hdmi_jack;
static int rk_hdmi_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
- struct snd_soc_component *component = runtime->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
int ret;
/* enable jack detection */
diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c
index 26b67b245484..0617ccf4e42c 100644
--- a/sound/soc/rockchip/rockchip_rt5645.c
+++ b/sound/soc/rockchip/rockchip_rt5645.c
@@ -56,8 +56,8 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream,
{
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int mclk;
switch (params_rate(params)) {
@@ -113,7 +113,7 @@ static int rk_init(struct snd_soc_pcm_runtime *runtime)
return ret;
}
- return rt5645_set_jack_detect(runtime->codec_dai->component,
+ return rt5645_set_jack_detect(asoc_rtd_to_codec(runtime, 0)->component,
&headset_jack,
&headset_jack,
&headset_jack);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 1a0b163ca47b..112911dc271b 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -151,7 +151,7 @@ config SND_SOC_TOBERMORY
config SND_SOC_BELLS
tristate "Audio support for Wolfson Bells"
- depends on MFD_ARIZONA && I2C && SPI_MASTER
+ depends on MFD_ARIZONA && MFD_WM5102 && MFD_WM5110 && I2C && SPI_MASTER
depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST
select SND_SAMSUNG_I2S
select SND_SOC_WM5102
@@ -204,7 +204,7 @@ config SND_SOC_ARNDALE
config SND_SOC_SAMSUNG_TM2_WM5110
tristate "SoC I2S Audio support for WM5110 on TM2 board"
- depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER
+ depends on SND_SOC_SAMSUNG && MFD_ARIZONA && MFD_WM5110 && I2C && SPI_MASTER
depends on GPIOLIB || COMPILE_TEST
select SND_SOC_MAX98504
select SND_SOC_WM5110
diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c
index d64602950cbd..c81ece78e036 100644
--- a/sound/soc/samsung/arndale.c
+++ b/sound/soc/samsung/arndale.c
@@ -21,8 +21,8 @@ static int arndale_rt5631_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int rfs, ret;
unsigned long rclk;
@@ -56,7 +56,7 @@ static int arndale_wm1811_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
unsigned int rfs, rclk;
/* Ensure AIF1CLK is >= 3 MHz for optimal performance */
@@ -174,7 +174,9 @@ static int arndale_audio_probe(struct platform_device *pdev)
ret = devm_snd_soc_register_card(card->dev, card);
if (ret) {
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret);
+ if (ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev,
+ "snd_soc_register_card() failed: %d\n", ret);
goto err_put_of_nodes;
}
return 0;
diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c
index 5de633497f83..8b83f39c3ac9 100644
--- a/sound/soc/samsung/bells.c
+++ b/sound/soc/samsung/bells.c
@@ -60,7 +60,7 @@ static int bells_set_bias_level(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
component = codec_dai->component;
if (dapm->dev != codec_dai->dev)
@@ -106,7 +106,7 @@ static int bells_set_bias_level_post(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
component = codec_dai->component;
if (dapm->dev != codec_dai->dev)
@@ -152,11 +152,11 @@ static int bells_late_probe(struct snd_soc_card *card)
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_AP_DSP]);
- wm0010 = rtd->codec_dai->component;
+ wm0010 = asoc_rtd_to_codec(rtd, 0)->component;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]);
- component = rtd->codec_dai->component;
- aif1_dai = rtd->codec_dai;
+ component = asoc_rtd_to_codec(rtd, 0)->component;
+ aif1_dai = asoc_rtd_to_codec(rtd, 0);
ret = snd_soc_component_set_sysclk(component, ARIZONA_CLK_SYSCLK,
ARIZONA_CLK_SRC_FLL1,
@@ -195,7 +195,7 @@ static int bells_late_probe(struct snd_soc_card *card)
}
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_CP]);
- aif2_dai = rtd->cpu_dai;
+ aif2_dai = asoc_rtd_to_cpu(rtd, 0);
ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
if (ret != 0) {
@@ -207,8 +207,8 @@ static int bells_late_probe(struct snd_soc_card *card)
return 0;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_SUB]);
- aif3_dai = rtd->cpu_dai;
- wm9081_dai = rtd->codec_dai;
+ aif3_dai = asoc_rtd_to_cpu(rtd, 0);
+ wm9081_dai = asoc_rtd_to_codec(rtd, 0);
ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0);
if (ret != 0) {
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index a95c34e53a2b..9139a1e7e200 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -68,7 +68,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int div;
int ret;
unsigned int rate = params_rate(params);
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index a57bb989a0ef..f86e3028b402 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -932,7 +932,7 @@ static int i2s_trigger(struct snd_pcm_substream *substream,
struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai);
int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct i2s_dai *i2s = to_info(rtd->cpu_dai);
+ struct i2s_dai *i2s = to_info(asoc_rtd_to_cpu(rtd, 0));
unsigned long flags;
switch (cmd) {
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 949d2e029962..30899016cf08 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -33,8 +33,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct s3c_i2sv2_rate_calc div;
unsigned int clk = 0;
int ret = 0;
diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index 59904f44118b..f4375c49f7f4 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -23,7 +23,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- aif1_dai = rtd->codec_dai;
+ aif1_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != aif1_dai->dev)
return 0;
@@ -70,7 +70,7 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- aif1_dai = rtd->codec_dai;
+ aif1_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != aif1_dai->dev)
return 0;
@@ -105,7 +105,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
sample_rate = params_rate(params);
@@ -181,7 +181,7 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
- aif2_dai = rtd->cpu_dai;
+ aif2_dai = asoc_rtd_to_cpu(rtd, 0);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -264,11 +264,11 @@ static int littlemill_late_probe(struct snd_soc_card *card)
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- component = rtd->codec_dai->component;
- aif1_dai = rtd->codec_dai;
+ component = asoc_rtd_to_codec(rtd, 0)->component;
+ aif1_dai = asoc_rtd_to_codec(rtd, 0);
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
- aif2_dai = rtd->cpu_dai;
+ aif2_dai = asoc_rtd_to_cpu(rtd, 0);
ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2,
32768, SND_SOC_CLOCK_IN);
@@ -325,7 +325,7 @@ static int littlemill_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
+ if (ret && ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c
index 098eefc764db..998d10cf8c94 100644
--- a/sound/soc/samsung/lowland.c
+++ b/sound/soc/samsung/lowland.c
@@ -32,7 +32,7 @@ static struct snd_soc_jack_pin lowland_headset_pins[] = {
static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
int ret;
ret = snd_soc_component_set_sysclk(component, WM5100_CLK_SYSCLK,
@@ -65,7 +65,7 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd)
static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
snd_soc_dapm_nc_pin(&rtd->card->dapm, "LINEOUT");
@@ -183,7 +183,7 @@ static int lowland_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
+ if (ret && ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 1339e41e9860..b7ce1da854ce 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -26,8 +26,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int pll_out = 0, bclk = 0;
int ret = 0;
unsigned long iis_clkrate;
@@ -100,7 +100,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
@@ -118,7 +118,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
unsigned int pcmdiv = 0;
int ret = 0;
unsigned long iis_clkrate;
@@ -155,7 +155,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index f0f5fa9c27d3..6eda5af989fe 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -98,7 +98,7 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream,
return ret;
if (rtd->num_codecs > 1) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[1];
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 1);
ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk_freq,
SND_SOC_CLOCK_IN);
@@ -311,7 +311,9 @@ static int odroid_audio_probe(struct platform_device *pdev)
ret = devm_snd_soc_register_card(dev, card);
if (ret < 0) {
- dev_err(dev, "snd_soc_register_card() failed: %d\n", ret);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "snd_soc_register_card() failed: %d\n",
+ ret);
goto err_put_clk_i2s;
}
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index f6e67d0e7882..a5b1a12b3496 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -212,7 +212,7 @@ static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
unsigned long flags;
dev_dbg(pcm->dev, "Entered %s\n", __func__);
@@ -256,7 +256,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *socdai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
void __iomem *regs = pcm->regs;
struct clk *clk;
int sclk_div, sync_div;
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 4b247e91ae5b..3afe63c0923e 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -149,7 +149,7 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int div;
int ret;
unsigned int rate = params_rate(params);
diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 593be1b668d6..358887848293 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -380,7 +380,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct s3c_i2sv2_info *i2s = to_info(rtd->cpu_dai);
+ struct s3c_i2sv2_info *i2s = to_info(asoc_rtd_to_cpu(rtd, 0));
int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
unsigned long irqs;
int ret = 0;
diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c
index 4543705b8d87..fd2a4da086f3 100644
--- a/sound/soc/samsung/s3c24xx_simtec.c
+++ b/sound/soc/samsung/s3c24xx_simtec.c
@@ -160,8 +160,8 @@ static int simtec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, 0,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index 55d2a802a6cb..abb5c4713c53 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -51,7 +51,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card);
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret = 0;
mutex_lock(&priv->clk_lock);
@@ -119,8 +119,8 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int clk = 0;
int ret = 0;
int clk_source, fs_mode;
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index fab3db9fdb98..36bef136d57f 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -25,8 +25,8 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int clk = 0;
int ret;
diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c
index 4baef84d29ee..776a270261bf 100644
--- a/sound/soc/samsung/smdk_spdif.c
+++ b/sound/soc/samsung/smdk_spdif.c
@@ -101,7 +101,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned long pll_out, rclk_rate;
int ret, ratio;
diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c
index d096ff912260..02074c34a2b2 100644
--- a/sound/soc/samsung/smdk_wm8580.c
+++ b/sound/soc/samsung/smdk_wm8580.c
@@ -23,7 +23,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
unsigned int pll_out;
int rfs, ret;
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 28f8be000aa1..a9f345f19a8a 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -45,7 +45,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
unsigned int pll_out;
int ret;
@@ -178,7 +178,7 @@ static int smdk_audio_probe(struct platform_device *pdev)
ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
+ if (ret && ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
return ret;
diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c
index 2e3dc7320c62..746930dde5d7 100644
--- a/sound/soc/samsung/smdk_wm8994pcm.c
+++ b/sound/soc/samsung/smdk_wm8994pcm.c
@@ -44,8 +44,8 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned long mclk_freq;
int rfs, ret;
@@ -118,7 +118,7 @@ static int snd_smdk_probe(struct platform_device *pdev)
smdk_pcm.dev = &pdev->dev;
ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm);
- if (ret)
+ if (ret && ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
return ret;
diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c
index f075aae9561a..40c5de8df0ff 100644
--- a/sound/soc/samsung/snow.c
+++ b/sound/soc/samsung/snow.c
@@ -110,9 +110,9 @@ static int snow_late_probe(struct snd_soc_card *card)
/* In the multi-codec case codec_dais 0 is MAX98095 and 1 is HDMI. */
if (rtd->num_codecs > 1)
- codec_dai = rtd->codec_dais[0];
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
else
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
/* Set the MCLK rate for the codec */
return snd_soc_dai_set_sysclk(codec_dai, 0,
@@ -216,7 +216,9 @@ static int snow_probe(struct platform_device *pdev)
ret = devm_snd_soc_register_card(dev, card);
if (ret) {
- dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ if (ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev,
+ "snd_soc_register_card failed (%d)\n", ret);
return ret;
}
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index 1a9f08a50394..759fc6644329 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -142,7 +142,7 @@ static int spdif_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai);
+ struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0));
unsigned long flags;
dev_dbg(spdif->dev, "Entered %s\n", __func__);
@@ -178,7 +178,7 @@ static int spdif_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *socdai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai);
+ struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0));
void __iomem *regs = spdif->regs;
struct snd_dmaengine_dai_dma_data *dma_data;
u32 con, clkcon, cstas;
@@ -194,7 +194,7 @@ static int spdif_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data);
+ snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_data);
spin_lock_irqsave(&spdif->lock, flags);
@@ -280,7 +280,7 @@ static void spdif_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai);
+ struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0));
void __iomem *regs = spdif->regs;
u32 con, clkcon;
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index ea0d1ec67f01..f5f6ba00d073 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -25,7 +25,7 @@ static int speyside_set_bias_level(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
@@ -61,7 +61,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
@@ -131,7 +131,7 @@ static void speyside_set_polarity(struct snd_soc_component *component,
static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
int ret;
ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0);
@@ -143,7 +143,7 @@ static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd)
static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *dai = rtd->codec_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_component *component = dai->component;
int ret;
@@ -330,7 +330,7 @@ static int speyside_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
+ if (ret && ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index 10ff14b856f2..6dfd540e2d74 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -93,7 +93,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card);
switch (params_rate(params)) {
@@ -134,7 +134,7 @@ static int tm2_aif2_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
unsigned int asyncclk_rate;
int ret;
@@ -188,7 +188,7 @@ static int tm2_aif2_hw_params(struct snd_pcm_substream *substream,
static int tm2_aif2_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
int ret;
/* disable FLL2 */
@@ -209,7 +209,7 @@ static int tm2_hdmi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
unsigned int bfs;
int bitwidth, ret;
@@ -284,7 +284,7 @@ static int tm2_set_bias_level(struct snd_soc_card *card,
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- if (dapm->dev != rtd->codec_dai->dev)
+ if (dapm->dev != asoc_rtd_to_codec(rtd, 0)->dev)
return 0;
switch (level) {
@@ -315,8 +315,8 @@ static int tm2_late_probe(struct snd_soc_card *card)
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF1]);
- aif1_dai = rtd->codec_dai;
- priv->component = rtd->codec_dai->component;
+ aif1_dai = asoc_rtd_to_codec(rtd, 0);
+ priv->component = asoc_rtd_to_codec(rtd, 0)->component;
ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0);
if (ret < 0) {
@@ -325,7 +325,7 @@ static int tm2_late_probe(struct snd_soc_card *card)
}
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF2]);
- aif2_dai = rtd->codec_dai;
+ aif2_dai = asoc_rtd_to_codec(rtd, 0);
ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
if (ret < 0) {
@@ -611,7 +611,8 @@ static int tm2_probe(struct platform_device *pdev)
ret = devm_snd_soc_register_card(dev, card);
if (ret < 0) {
- dev_err(dev, "Failed to register card: %d\n", ret);
+ if (ret != -EPROBE_DEFER)
+ dev_err(dev, "Failed to register card: %d\n", ret);
goto dai_node_put;
}
diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c
index fdce28cc26c4..c962d2c2a7f7 100644
--- a/sound/soc/samsung/tobermory.c
+++ b/sound/soc/samsung/tobermory.c
@@ -23,7 +23,7 @@ static int tobermory_set_bias_level(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
@@ -66,7 +66,7 @@ static int tobermory_set_bias_level_post(struct snd_soc_card *card,
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- codec_dai = rtd->codec_dai;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
@@ -181,8 +181,8 @@ static int tobermory_late_probe(struct snd_soc_card *card)
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- component = rtd->codec_dai->component;
- codec_dai = rtd->codec_dai;
+ component = asoc_rtd_to_codec(rtd, 0)->component;
+ codec_dai = asoc_rtd_to_codec(rtd, 0);
ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
32768, SND_SOC_CLOCK_IN);
@@ -229,7 +229,7 @@ static int tobermory_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
+ if (ret && ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
ret);
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index eee1a1e994cb..a35de78f14a9 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -119,7 +119,7 @@ static int camelot_pcm_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+ struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
int ret, dmairq;
@@ -132,7 +132,7 @@ static int camelot_pcm_open(struct snd_soc_component *component,
ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam);
if (unlikely(ret)) {
pr_debug("audio unit %d irqs already taken!\n",
- rtd->cpu_dai->id);
+ asoc_rtd_to_cpu(rtd, 0)->id);
return -EBUSY;
}
(void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam);
@@ -141,7 +141,7 @@ static int camelot_pcm_open(struct snd_soc_component *component,
ret = dmabrg_request_irq(dmairq, camelot_txdma, cam);
if (unlikely(ret)) {
pr_debug("audio unit %d irqs already taken!\n",
- rtd->cpu_dai->id);
+ asoc_rtd_to_cpu(rtd, 0)->id);
return -EBUSY;
}
(void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam);
@@ -153,7 +153,7 @@ static int camelot_pcm_close(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+ struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
int dmairq;
@@ -175,7 +175,7 @@ static int camelot_hw_params(struct snd_soc_component *component,
struct snd_pcm_hw_params *hw_params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+ struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
int ret;
@@ -194,7 +194,7 @@ static int camelot_prepare(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+ struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
pr_debug("PCM data: addr 0x%08lx len %d\n",
(u32)runtime->dma_addr, runtime->dma_bytes);
@@ -242,7 +242,7 @@ static int camelot_trigger(struct snd_soc_component *component,
struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+ struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
switch (cmd) {
@@ -270,7 +270,7 @@ static snd_pcm_uframes_t camelot_pos(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+ struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
unsigned long pos;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 4b35ef402604..1c3c4fdc9bef 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -408,7 +408,7 @@ static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- return rtd->cpu_dai;
+ return asoc_rtd_to_cpu(rtd, 0);
}
static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *dai)
@@ -1938,8 +1938,7 @@ static int fsi_probe(struct platform_device *pdev)
if (!master)
return -ENOMEM;
- master->base = devm_ioremap(&pdev->dev,
- res->start, resource_size(res));
+ master->base = devm_ioremap(&pdev->dev, res->start, resource_size(res));
if (!master->base) {
dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n");
return -ENXIO;
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index 991557e25eba..d5702fbf176b 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -46,7 +46,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int ret;
unsigned int rate = params_rate(params);
@@ -67,7 +67,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream,
clk_set_rate(&siumckb_clk, codec_freq);
dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq);
- ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, SIU_CLKB_EXT,
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), SIU_CLKB_EXT,
codec_freq / 2, SND_SOC_CLOCK_IN);
if (!ret)
@@ -79,7 +79,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream,
static int migor_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
if (use_count) {
use_count--;
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 0bfcb77e5f65..4349f2fb823f 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -696,7 +696,7 @@ struct snd_soc_dai *rsnd_substream_to_dai(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- return rtd->cpu_dai;
+ return asoc_rtd_to_cpu(rtd, 0);
}
static
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 392a1c5b15d3..50062eb79adb 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -810,9 +810,10 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
int playback = 0, capture = 0;
int i;
- if (rtd->num_codecs > 1) {
+ if (rtd->num_cpus > 1 ||
+ rtd->num_codecs > 1) {
dev_err(rtd->card->dev,
- "Compress ASoC: Multicodec not supported\n");
+ "Compress ASoC: Multi CPU/Codec not supported\n");
return -EINVAL;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 068d809c349a..843b8b1c89d4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -365,19 +365,20 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int playback = SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
dev_dbg(rtd->dev,
"ASoC: pop wq checking: %s status: %s waiting: %s\n",
codec_dai->driver->playback.stream_name,
- codec_dai->playback_active ? "active" : "inactive",
+ codec_dai->stream_active[playback] ? "active" : "inactive",
rtd->pop_wait ? "yes" : "no");
/* are we waiting on this codec DAI stream */
if (rtd->pop_wait == 1) {
rtd->pop_wait = 0;
- snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+ snd_soc_dapm_stream_event(rtd, playback,
SND_SOC_DAPM_STREAM_STOP);
}
@@ -431,6 +432,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
struct snd_soc_component *component;
struct device *dev;
int ret;
+ int stream;
/*
* for rtd->dev
@@ -465,23 +467,31 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
rtd->dev = dev;
INIT_LIST_HEAD(&rtd->list);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
+ for_each_pcm_streams(stream) {
+ INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients);
+ INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients);
+ }
dev_set_drvdata(dev, rtd);
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
/*
- * for rtd->codec_dais
+ * for rtd->dais
*/
- rtd->codec_dais = devm_kcalloc(dev, dai_link->num_codecs,
+ rtd->dais = devm_kcalloc(dev, dai_link->num_cpus + dai_link->num_codecs,
sizeof(struct snd_soc_dai *),
GFP_KERNEL);
- if (!rtd->codec_dais)
+ if (!rtd->dais)
goto free_rtd;
/*
+ * dais = [][][][][][][][][][][][][][][][][][]
+ * ^cpu_dais ^codec_dais
+ * |--- num_cpus ---|--- num_codecs --|
+ */
+ rtd->cpu_dais = &rtd->dais[0];
+ rtd->codec_dais = &rtd->dais[dai_link->num_cpus];
+
+ /*
* rtd remaining settings
*/
rtd->card = card;
@@ -514,6 +524,7 @@ int snd_soc_suspend(struct device *dev)
struct snd_soc_card *card = dev_get_drvdata(dev);
struct snd_soc_component *component;
struct snd_soc_pcm_runtime *rtd;
+ int playback = SNDRV_PCM_STREAM_PLAYBACK;
int i;
/* If the card is not initialized yet there is nothing to do */
@@ -536,10 +547,9 @@ int snd_soc_suspend(struct device *dev)
if (rtd->dai_link->ignore_suspend)
continue;
- for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->playback_active)
- snd_soc_dai_digital_mute(dai, 1,
- SNDRV_PCM_STREAM_PLAYBACK);
+ for_each_rtd_codec_dais(rtd, i, dai) {
+ if (dai->stream_active[playback])
+ snd_soc_dai_digital_mute(dai, 1, playback);
}
}
@@ -558,17 +568,14 @@ int snd_soc_suspend(struct device *dev)
snd_soc_flush_all_delayed_work(card);
for_each_card_rtds(card, rtd) {
+ int stream;
if (rtd->dai_link->ignore_suspend)
continue;
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_PLAYBACK,
- SND_SOC_DAPM_STREAM_SUSPEND);
-
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_CAPTURE,
- SND_SOC_DAPM_STREAM_SUSPEND);
+ for_each_pcm_streams(stream)
+ snd_soc_dapm_stream_event(rtd, stream,
+ SND_SOC_DAPM_STREAM_SUSPEND);
}
/* Recheck all endpoints too, their state is affected by suspend */
@@ -664,30 +671,27 @@ static void soc_resume_deferred(struct work_struct *work)
}
for_each_card_rtds(card, rtd) {
+ int stream;
if (rtd->dai_link->ignore_suspend)
continue;
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_PLAYBACK,
- SND_SOC_DAPM_STREAM_RESUME);
-
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_CAPTURE,
- SND_SOC_DAPM_STREAM_RESUME);
+ for_each_pcm_streams(stream)
+ snd_soc_dapm_stream_event(rtd, stream,
+ SND_SOC_DAPM_STREAM_RESUME);
}
/* unmute any active DACs */
for_each_card_rtds(card, rtd) {
struct snd_soc_dai *dai;
+ int playback = SNDRV_PCM_STREAM_PLAYBACK;
if (rtd->dai_link->ignore_suspend)
continue;
- for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->playback_active)
- snd_soc_dai_digital_mute(dai, 0,
- SNDRV_PCM_STREAM_PLAYBACK);
+ for_each_rtd_codec_dais(rtd, i, dai) {
+ if (dai->stream_active[playback])
+ snd_soc_dai_digital_mute(dai, 0, playback);
}
}
@@ -837,7 +841,7 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card,
struct snd_soc_dai_link *link)
{
int i;
- struct snd_soc_dai_link_component *codec, *platform;
+ struct snd_soc_dai_link_component *cpu, *codec, *platform;
for_each_link_codecs(link, i, codec) {
/*
@@ -886,44 +890,38 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card,
return -EPROBE_DEFER;
}
- /* FIXME */
- if (link->num_cpus > 1) {
- dev_err(card->dev,
- "ASoC: multi cpu is not yet supported %s\n",
- link->name);
- return -EINVAL;
- }
-
- /*
- * CPU device may be specified by either name or OF node, but
- * can be left unspecified, and will be matched based on DAI
- * name alone..
- */
- if (link->cpus->name && link->cpus->of_node) {
- dev_err(card->dev,
- "ASoC: Neither/both cpu name/of_node are set for %s\n",
- link->name);
- return -EINVAL;
- }
+ for_each_link_cpus(link, i, cpu) {
+ /*
+ * CPU device may be specified by either name or OF node, but
+ * can be left unspecified, and will be matched based on DAI
+ * name alone..
+ */
+ if (cpu->name && cpu->of_node) {
+ dev_err(card->dev,
+ "ASoC: Neither/both cpu name/of_node are set for %s\n",
+ link->name);
+ return -EINVAL;
+ }
- /*
- * Defer card registration if cpu dai component is not added to
- * component list.
- */
- if ((link->cpus->of_node || link->cpus->name) &&
- !soc_find_component(link->cpus))
- return -EPROBE_DEFER;
+ /*
+ * Defer card registration if cpu dai component is not added to
+ * component list.
+ */
+ if ((cpu->of_node || cpu->name) &&
+ !soc_find_component(cpu))
+ return -EPROBE_DEFER;
- /*
- * At least one of CPU DAI name or CPU device name/node must be
- * specified
- */
- if (!link->cpus->dai_name &&
- !(link->cpus->name || link->cpus->of_node)) {
- dev_err(card->dev,
- "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
- link->name);
- return -EINVAL;
+ /*
+ * At least one of CPU DAI name or CPU device name/node must be
+ * specified
+ */
+ if (!cpu->dai_name &&
+ !(cpu->name || cpu->of_node)) {
+ dev_err(card->dev,
+ "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
+ link->name);
+ return -EINVAL;
+ }
}
return 0;
@@ -966,7 +964,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
struct snd_soc_dai_link *dai_link)
{
struct snd_soc_pcm_runtime *rtd;
- struct snd_soc_dai_link_component *codec, *platform;
+ struct snd_soc_dai_link_component *codec, *platform, *cpu;
struct snd_soc_component *component;
int i, ret;
@@ -991,14 +989,19 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
if (!rtd)
return -ENOMEM;
- /* FIXME: we need multi CPU support in the future */
- rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus);
- if (!rtd->cpu_dai) {
- dev_info(card->dev, "ASoC: CPU DAI %s not registered\n",
- dai_link->cpus->dai_name);
- goto _err_defer;
+ rtd->num_cpus = dai_link->num_cpus;
+ for_each_link_cpus(dai_link, i, cpu) {
+ rtd->cpu_dais[i] = snd_soc_find_dai(cpu);
+ if (!rtd->cpu_dais[i]) {
+ dev_info(card->dev, "ASoC: CPU DAI %s not registered\n",
+ cpu->dai_name);
+ goto _err_defer;
+ }
+ snd_soc_rtd_add_component(rtd, rtd->cpu_dais[i]->component);
}
- snd_soc_rtd_add_component(rtd, rtd->cpu_dai->component);
+
+ /* Single cpu links expect cpu and cpu_dai in runtime data */
+ rtd->cpu_dai = rtd->cpu_dais[0];
/* Find CODEC from registered CODECs */
rtd->num_codecs = dai_link->num_codecs;
@@ -1034,20 +1037,20 @@ _err_defer:
}
EXPORT_SYMBOL_GPL(snd_soc_add_pcm_runtime);
-static int soc_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
- struct snd_soc_pcm_runtime *rtd)
+static int soc_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai;
int i, ret = 0;
- for (i = 0; i < num_dais; ++i) {
- struct snd_soc_dai_driver *drv = dais[i]->driver;
+ for_each_rtd_dais(rtd, i, dai) {
+ struct snd_soc_dai_driver *drv = dai->driver;
if (drv->pcm_new)
- ret = drv->pcm_new(rtd, dais[i]);
+ ret = drv->pcm_new(rtd, dai);
if (ret < 0) {
- dev_err(dais[i]->dev,
+ dev_err(dai->dev,
"ASoC: Failed to bind %s with pcm device\n",
- dais[i]->name);
+ dai->name);
return ret;
}
}
@@ -1118,12 +1121,8 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card,
dai_link->stream_name, ret);
return ret;
}
- ret = soc_dai_pcm_new(&cpu_dai, 1, rtd);
- if (ret < 0)
- return ret;
- ret = soc_dai_pcm_new(rtd->codec_dais,
- rtd->num_codecs, rtd);
- return ret;
+
+ return soc_dai_pcm_new(rtd);
}
static void soc_set_name_prefix(struct snd_soc_card *card,
@@ -1256,8 +1255,18 @@ static int soc_probe_component(struct snd_soc_card *card,
ret = snd_soc_dapm_add_routes(dapm,
component->driver->dapm_routes,
component->driver->num_dapm_routes);
- if (ret < 0)
- goto err_probe;
+ if (ret < 0) {
+ if (card->disable_route_checks) {
+ dev_info(card->dev,
+ "%s: disable_route_checks set, ignoring errors on add_routes\n",
+ __func__);
+ } else {
+ dev_err(card->dev,
+ "%s: snd_soc_dapm_add_routes failed: %d\n",
+ __func__, ret);
+ goto err_probe;
+ }
+ }
/* see for_each_card_components */
list_add(&component->card_list, &card->component_dev_list);
@@ -1309,24 +1318,22 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order)
static void soc_remove_link_dais(struct snd_soc_card *card)
{
int i;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
struct snd_soc_pcm_runtime *rtd;
int order;
for_each_comp_order(order) {
for_each_card_rtds(card, rtd) {
- /* remove the CODEC DAI */
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- soc_remove_dai(codec_dai, order);
-
- soc_remove_dai(rtd->cpu_dai, order);
+ /* remove DAIs */
+ for_each_rtd_dais(rtd, i, dai)
+ soc_remove_dai(dai, order);
}
}
}
static int soc_probe_link_dais(struct snd_soc_card *card)
{
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
struct snd_soc_pcm_runtime *rtd;
int i, order, ret;
@@ -1337,13 +1344,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card)
"ASoC: probe %s dai link %d late %d\n",
card->name, rtd->num, order);
- ret = soc_probe_dai(rtd->cpu_dai, order);
- if (ret)
- return ret;
-
- /* probe the CODEC DAI */
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- ret = soc_probe_dai(codec_dai, order);
+ /* probe the CPU DAI */
+ for_each_rtd_dais(rtd, i, dai) {
+ ret = soc_probe_dai(dai, order);
if (ret)
return ret;
}
@@ -1471,12 +1474,13 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
unsigned int dai_fmt)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai;
struct snd_soc_dai *codec_dai;
+ unsigned int inv_dai_fmt;
unsigned int i;
int ret;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
if (ret != 0 && ret != -ENOTSUPP) {
dev_warn(codec_dai->dev,
@@ -1489,33 +1493,33 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
* Flip the polarity for the "CPU" end of a CODEC<->CODEC link
* the component which has non_legacy_dai_naming is Codec
*/
- if (cpu_dai->component->driver->non_legacy_dai_naming) {
- unsigned int inv_dai_fmt;
-
- inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK;
- switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBM_CFM:
- inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
- break;
- case SND_SOC_DAIFMT_CBM_CFS:
- inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
- break;
- case SND_SOC_DAIFMT_CBS_CFM:
- inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
- break;
- case SND_SOC_DAIFMT_CBS_CFS:
- inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
- break;
- }
-
- dai_fmt = inv_dai_fmt;
+ inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK;
+ switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ break;
}
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ unsigned int fmt = dai_fmt;
- ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt);
- if (ret != 0 && ret != -ENOTSUPP) {
- dev_warn(cpu_dai->dev,
- "ASoC: Failed to set DAI format: %d\n", ret);
- return ret;
+ if (cpu_dai->component->driver->non_legacy_dai_naming)
+ fmt = inv_dai_fmt;
+
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+ if (ret != 0 && ret != -ENOTSUPP) {
+ dev_warn(cpu_dai->dev,
+ "ASoC: Failed to set DAI format: %d\n", ret);
+ return ret;
+ }
}
return 0;
@@ -1938,8 +1942,18 @@ static int snd_soc_bind_card(struct snd_soc_card *card)
ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
card->num_dapm_routes);
- if (ret < 0)
- goto probe_end;
+ if (ret < 0) {
+ if (card->disable_route_checks) {
+ dev_info(card->dev,
+ "%s: disable_route_checks set, ignoring errors on add_routes\n",
+ __func__);
+ } else {
+ dev_err(card->dev,
+ "%s: snd_soc_dapm_add_routes failed: %d\n",
+ __func__, ret);
+ goto probe_end;
+ }
+ }
ret = snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes,
card->num_of_dapm_routes);
@@ -3102,6 +3116,14 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
*dai_name = dai->driver->name;
if (!*dai_name)
*dai_name = pos->name;
+ } else if (ret) {
+ /*
+ * if another error than ENOTSUPP is returned go on and
+ * check if another component is provided with the same
+ * node. This may happen if a device provides several
+ * components
+ */
+ continue;
}
break;
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 51031e330179..19142f6e533c 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -295,17 +295,24 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
{
int ret = 0;
- if (dai->driver->ops->startup)
+ if (!dai->started &&
+ dai->driver->ops->startup)
ret = dai->driver->ops->startup(substream, dai);
+ if (ret == 0)
+ dai->started = 1;
+
return ret;
}
void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
struct snd_pcm_substream *substream)
{
- if (dai->driver->ops->shutdown)
+ if (dai->started &&
+ dai->driver->ops->shutdown)
dai->driver->ops->shutdown(substream, dai);
+
+ dai->started = 0;
}
int snd_soc_dai_prepare(struct snd_soc_dai *dai,
@@ -383,12 +390,7 @@ int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
*/
bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
{
- struct snd_soc_pcm_stream *stream;
-
- if (dir == SNDRV_PCM_STREAM_PLAYBACK)
- stream = &dai->driver->playback;
- else
- stream = &dai->driver->capture;
+ struct snd_soc_pcm_stream *stream = snd_soc_dai_get_pcm_stream(dai, dir);
/* If the codec specifies any channels at all, it supports the stream */
return stream->channels_min;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 9fb54e6fe254..04da7928c873 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -302,7 +302,7 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card)
mutex_lock(&card->dapm_mutex);
- list_for_each_entry(w, &card->widgets, list) {
+ for_each_card_widgets(card, w) {
if (w->is_ep) {
dapm_mark_dirty(w, "Rechecking endpoints");
if (w->is_ep & SND_SOC_DAPM_EP_SINK)
@@ -589,7 +589,7 @@ static void dapm_reset(struct snd_soc_card *card)
memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
- list_for_each_entry(w, &card->widgets, list) {
+ for_each_card_widgets(card, w) {
w->new_power = w->power;
w->power_checked = false;
}
@@ -833,7 +833,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
*kcontrol = NULL;
- list_for_each_entry(w, &dapm->card->widgets, list) {
+ for_each_card_widgets(dapm->card, w) {
if (w == kcontrolw || w->dapm != kcontrolw->dapm)
continue;
for (i = 0; i < w->num_kcontrols; i++) {
@@ -1105,6 +1105,11 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
}
}
+static void dapm_widget_list_free(struct snd_soc_dapm_widget_list **list)
+{
+ kfree(*list);
+}
+
static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list,
struct list_head *widgets)
{
@@ -1310,6 +1315,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
return paths;
}
+void snd_soc_dapm_dai_free_widgets(struct snd_soc_dapm_widget_list **list)
+{
+ dapm_widget_list_free(list);
+}
+
/*
* Handler for regulator supply widget.
*/
@@ -1706,9 +1716,8 @@ static void dapm_seq_run(struct snd_soc_card *card,
i, cur_subseq);
}
- list_for_each_entry(d, &card->dapm_list, list) {
+ for_each_card_dapms(card, d)
soc_dapm_async_complete(d);
- }
}
static void dapm_widget_update(struct snd_soc_card *card)
@@ -1724,9 +1733,7 @@ static void dapm_widget_update(struct snd_soc_card *card)
wlist = dapm_kcontrol_get_wlist(update->kcontrol);
- for (wi = 0; wi < wlist->num_widgets; wi++) {
- w = wlist->widgets[wi];
-
+ for_each_dapm_widgets(wlist, wi, w) {
if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) {
ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG);
if (ret != 0)
@@ -1753,9 +1760,7 @@ static void dapm_widget_update(struct snd_soc_card *card)
w->name, ret);
}
- for (wi = 0; wi < wlist->num_widgets; wi++) {
- w = wlist->widgets[wi];
-
+ for_each_dapm_widgets(wlist, wi, w) {
if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) {
ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG);
if (ret != 0)
@@ -1943,7 +1948,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
trace_snd_soc_dapm_start(card);
- list_for_each_entry(d, &card->dapm_list, list) {
+ for_each_card_dapms(card, d) {
if (dapm_idle_bias_off(d))
d->target_bias_level = SND_SOC_BIAS_OFF;
else
@@ -1962,7 +1967,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
dapm_power_one_widget(w, &up_list, &down_list);
}
- list_for_each_entry(w, &card->widgets, list) {
+ for_each_card_widgets(card, w) {
switch (w->id) {
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
@@ -2007,10 +2012,10 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
* they're not ground referenced.
*/
bias = SND_SOC_BIAS_OFF;
- list_for_each_entry(d, &card->dapm_list, list)
+ for_each_card_dapms(card, d)
if (d->target_bias_level > bias)
bias = d->target_bias_level;
- list_for_each_entry(d, &card->dapm_list, list)
+ for_each_card_dapms(card, d)
if (!dapm_idle_bias_off(d))
d->target_bias_level = bias;
@@ -2019,7 +2024,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
/* Run card bias changes at first */
dapm_pre_sequence_async(&card->dapm, 0);
/* Run other bias changes in parallel */
- list_for_each_entry(d, &card->dapm_list, list) {
+ for_each_card_dapms(card, d) {
if (d != &card->dapm && d->bias_level != d->target_bias_level)
async_schedule_domain(dapm_pre_sequence_async, d,
&async_domain);
@@ -2043,7 +2048,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
dapm_seq_run(card, &up_list, event, true);
/* Run all the bias changes in parallel */
- list_for_each_entry(d, &card->dapm_list, list) {
+ for_each_card_dapms(card, d) {
if (d != &card->dapm && d->bias_level != d->target_bias_level)
async_schedule_domain(dapm_post_sequence_async, d,
&async_domain);
@@ -2053,7 +2058,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
dapm_post_sequence_async(&card->dapm, 0);
/* do we need to notify any clients that DAPM event is complete */
- list_for_each_entry(d, &card->dapm_list, list) {
+ for_each_card_dapms(card, d) {
if (!d->component)
continue;
@@ -2286,7 +2291,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm,
card->update = NULL;
mutex_unlock(&card->dapm_mutex);
if (ret > 0)
- soc_dpcm_runtime_update(card);
+ snd_soc_dpcm_runtime_update(card);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power);
@@ -2351,7 +2356,7 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm,
card->update = NULL;
mutex_unlock(&card->dapm_mutex);
if (ret > 0)
- soc_dpcm_runtime_update(card);
+ snd_soc_dpcm_runtime_update(card);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power);
@@ -2371,7 +2376,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt,
if (!cmpnt->card)
return 0;
- list_for_each_entry(w, &cmpnt->card->widgets, list) {
+ for_each_card_widgets(cmpnt->card, w) {
if (w->dapm != dapm)
continue;
@@ -2431,7 +2436,7 @@ static ssize_t dapm_widget_show(struct device *dev,
mutex_lock(&rtd->card->dapm_mutex);
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
struct snd_soc_component *cmpnt = codec_dai->component;
count += dapm_widget_show_component(cmpnt, buf + count);
@@ -2491,7 +2496,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_widget *w, *next_w;
- list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) {
+ for_each_card_widgets_safe(dapm->card, w, next_w) {
if (w->dapm != dapm)
continue;
snd_soc_dapm_free_widget(w);
@@ -2506,7 +2511,7 @@ static struct snd_soc_dapm_widget *dapm_find_widget(
struct snd_soc_dapm_widget *w;
struct snd_soc_dapm_widget *fallback = NULL;
- list_for_each_entry(w, &dapm->card->widgets, list) {
+ for_each_card_widgets(dapm->card, w) {
if (!strcmp(w->name, pin)) {
if (w->dapm == dapm)
return w;
@@ -2624,10 +2629,7 @@ static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream,
struct snd_soc_dapm_widget *w;
int ret;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK)
- w = dai->playback_widget;
- else
- w = dai->capture_widget;
+ w = snd_soc_dai_get_widget(dai, dir);
if (!w)
return 0;
@@ -2908,7 +2910,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
* find src and dest widgets over all widgets but favor a widget from
* current DAPM context
*/
- list_for_each_entry(w, &dapm->card->widgets, list) {
+ for_each_card_widgets(dapm->card, w) {
if (!wsink && !(strcmp(w->name, sink))) {
wtsink = w;
if (w->dapm == dapm) {
@@ -3187,7 +3189,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
- list_for_each_entry(w, &card->widgets, list)
+ for_each_card_widgets(card, w)
{
if (w->new)
continue;
@@ -3394,7 +3396,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
mutex_unlock(&card->dapm_mutex);
if (ret > 0)
- soc_dpcm_runtime_update(card);
+ snd_soc_dpcm_runtime_update(card);
return change;
}
@@ -3499,7 +3501,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
mutex_unlock(&card->dapm_mutex);
if (ret > 0)
- soc_dpcm_runtime_update(card);
+ snd_soc_dpcm_runtime_update(card);
return change;
}
@@ -3604,6 +3606,9 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
ret = PTR_ERR(w->pinctrl);
goto request_failed;
}
+
+ /* set to sleep_state when initializing */
+ dapm_pinctrl_event(w, NULL, SND_SOC_DAPM_POST_PMD);
break;
case snd_soc_dapm_clock_supply:
w->clk = devm_clk_get(dapm->dev, w->name);
@@ -3698,6 +3703,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
w->dapm = dapm;
INIT_LIST_HEAD(&w->list);
INIT_LIST_HEAD(&w->dirty);
+ /* see for_each_card_widgets */
list_add_tail(&w->list, &dapm->card->widgets);
snd_soc_dapm_for_each_direction(dir) {
@@ -4222,7 +4228,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
struct snd_soc_dai *dai;
/* For each DAI widget... */
- list_for_each_entry(dai_w, &card->widgets, list) {
+ for_each_card_widgets(card, dai_w) {
switch (dai_w->id) {
case snd_soc_dapm_dai_in:
case snd_soc_dapm_dai_out:
@@ -4241,7 +4247,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
dai = dai_w->priv;
/* ...find all widgets with the same stream and link them */
- list_for_each_entry(w, &card->widgets, list) {
+ for_each_card_widgets(card, w) {
if (w->dapm != dai_w->dapm)
continue;
@@ -4271,16 +4277,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
return 0;
}
-static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
- struct snd_soc_pcm_runtime *rtd)
+static void dapm_add_valid_dai_widget(struct snd_soc_card *card,
+ struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dai *codec_dai,
+ struct snd_soc_dai *cpu_dai)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
struct snd_pcm_substream *substream;
struct snd_pcm_str *streams = rtd->pcm->streams;
- int i;
if (rtd->dai_link->params) {
playback_cpu = cpu_dai->capture_widget;
@@ -4292,77 +4297,92 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
capture_cpu = capture;
}
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- /* connect BE DAI playback if widgets are valid */
- codec = codec_dai->playback_widget;
-
- if (playback_cpu && codec) {
- if (!playback) {
- substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
- playback = snd_soc_dapm_new_dai(card, substream,
- "playback");
- if (IS_ERR(playback)) {
- dev_err(rtd->dev,
- "ASoC: Failed to create DAI %s: %ld\n",
- codec_dai->name,
- PTR_ERR(playback));
- continue;
- }
-
- snd_soc_dapm_add_path(&card->dapm, playback_cpu,
- playback, NULL, NULL);
+ /* connect BE DAI playback if widgets are valid */
+ codec = codec_dai->playback_widget;
+
+ if (playback_cpu && codec) {
+ if (!playback) {
+ substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ playback = snd_soc_dapm_new_dai(card, substream,
+ "playback");
+ if (IS_ERR(playback)) {
+ dev_err(rtd->dev,
+ "ASoC: Failed to create DAI %s: %ld\n",
+ codec_dai->name,
+ PTR_ERR(playback));
+ goto capture;
}
- dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- cpu_dai->component->name, playback_cpu->name,
- codec_dai->component->name, codec->name);
-
- snd_soc_dapm_add_path(&card->dapm, playback, codec,
- NULL, NULL);
+ snd_soc_dapm_add_path(&card->dapm, playback_cpu,
+ playback, NULL, NULL);
}
- }
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- /* connect BE DAI capture if widgets are valid */
- codec = codec_dai->capture_widget;
-
- if (codec && capture_cpu) {
- if (!capture) {
- substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
- capture = snd_soc_dapm_new_dai(card, substream,
- "capture");
- if (IS_ERR(capture)) {
- dev_err(rtd->dev,
- "ASoC: Failed to create DAI %s: %ld\n",
- codec_dai->name,
- PTR_ERR(capture));
- continue;
- }
-
- snd_soc_dapm_add_path(&card->dapm, capture,
- capture_cpu, NULL, NULL);
+ dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+ cpu_dai->component->name, playback_cpu->name,
+ codec_dai->component->name, codec->name);
+
+ snd_soc_dapm_add_path(&card->dapm, playback, codec,
+ NULL, NULL);
+ }
+
+capture:
+ /* connect BE DAI capture if widgets are valid */
+ codec = codec_dai->capture_widget;
+
+ if (codec && capture_cpu) {
+ if (!capture) {
+ substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ capture = snd_soc_dapm_new_dai(card, substream,
+ "capture");
+ if (IS_ERR(capture)) {
+ dev_err(rtd->dev,
+ "ASoC: Failed to create DAI %s: %ld\n",
+ codec_dai->name,
+ PTR_ERR(capture));
+ return;
}
- dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- codec_dai->component->name, codec->name,
- cpu_dai->component->name, capture_cpu->name);
-
- snd_soc_dapm_add_path(&card->dapm, codec, capture,
- NULL, NULL);
+ snd_soc_dapm_add_path(&card->dapm, capture,
+ capture_cpu, NULL, NULL);
}
+
+ dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+ codec_dai->component->name, codec->name,
+ cpu_dai->component->name, capture_cpu->name);
+
+ snd_soc_dapm_add_path(&card->dapm, codec, capture,
+ NULL, NULL);
}
}
+static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
+ struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *codec_dai;
+ int i;
+
+ if (rtd->num_cpus == 1) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai)
+ dapm_add_valid_dai_widget(card, rtd, codec_dai,
+ rtd->cpu_dais[0]);
+ } else if (rtd->num_codecs == rtd->num_cpus) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai)
+ dapm_add_valid_dai_widget(card, rtd, codec_dai,
+ rtd->cpu_dais[i]);
+ } else {
+ dev_err(card->dev,
+ "N cpus to M codecs link is not supported yet\n");
+ }
+
+}
+
static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
int event)
{
struct snd_soc_dapm_widget *w;
unsigned int ep;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- w = dai->playback_widget;
- else
- w = dai->capture_widget;
+ w = snd_soc_dai_get_widget(dai, stream);
if (w) {
dapm_mark_dirty(w, "stream event");
@@ -4413,12 +4433,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
int event)
{
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
int i;
- soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- soc_dapm_dai_stream_event(codec_dai, stream, event);
+ for_each_rtd_dais(rtd, i, dai)
+ soc_dapm_dai_stream_event(dai, stream, event);
dapm_power_widgets(rtd->card, event);
}
@@ -4754,6 +4773,7 @@ void snd_soc_dapm_init(struct snd_soc_dapm_context *dapm,
}
INIT_LIST_HEAD(&dapm->list);
+ /* see for_each_card_dapms */
list_add(&dapm->list, &card->dapm_list);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_init);
@@ -4767,7 +4787,7 @@ static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm)
mutex_lock(&card->dapm_mutex);
- list_for_each_entry(w, &dapm->card->widgets, list) {
+ for_each_card_widgets(dapm->card, w) {
if (w->dapm != dapm)
continue;
if (w->power) {
@@ -4800,7 +4820,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
{
struct snd_soc_dapm_context *dapm;
- list_for_each_entry(dapm, &card->dapm_list, list) {
+ for_each_card_dapms(card, dapm) {
if (dapm != &card->dapm) {
soc_dapm_shutdown_dapm(dapm);
if (dapm->bias_level == SND_SOC_BIAS_STANDBY)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 2cc25651661c..facf1922a714 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -62,6 +62,12 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
struct snd_dmaengine_dai_dma_data *dma_data;
int ret;
+ if (rtd->num_cpus > 1) {
+ dev_err(rtd->dev,
+ "%s doesn't support Multi CPU yet\n", __func__);
+ return -EINVAL;
+ }
+
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
@@ -118,6 +124,12 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component,
struct snd_dmaengine_dai_dma_data *dma_data;
struct snd_pcm_hardware hw;
+ if (rtd->num_cpus > 1) {
+ dev_err(rtd->dev,
+ "%s doesn't support Multi CPU yet\n", __func__);
+ return -EINVAL;
+ }
+
if (pcm->config && pcm->config->pcm_hardware)
return snd_soc_set_runtime_hwparams(substream,
pcm->config->pcm_hardware);
@@ -185,6 +197,12 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel(
struct snd_dmaengine_dai_dma_data *dma_data;
dma_filter_fn fn = NULL;
+ if (rtd->num_cpus > 1) {
+ dev_err(rtd->dev,
+ "%s doesn't support Multi CPU yet\n", __func__);
+ return NULL;
+ }
+
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0])
@@ -237,7 +255,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component,
max_buffer_size = SIZE_MAX;
}
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
+ for_each_pcm_streams(i) {
substream = rtd->pcm->streams[i].substream;
if (!substream)
continue;
@@ -371,8 +389,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
dev = config->dma_dev;
}
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
- i++) {
+ for_each_pcm_streams(i) {
if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
name = "rx-tx";
else
@@ -401,8 +418,7 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
{
unsigned int i;
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
- i++) {
+ for_each_pcm_streams(i) {
if (!pcm->chan[i])
continue;
dma_release_channel(pcm->chan[i]);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2c59b3688ca0..e256d438ee68 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -28,6 +28,180 @@
#define DPCM_MAX_BE_USERS 8
+#ifdef CONFIG_DEBUG_FS
+static const char *dpcm_state_string(enum snd_soc_dpcm_state state)
+{
+ switch (state) {
+ case SND_SOC_DPCM_STATE_NEW:
+ return "new";
+ case SND_SOC_DPCM_STATE_OPEN:
+ return "open";
+ case SND_SOC_DPCM_STATE_HW_PARAMS:
+ return "hw_params";
+ case SND_SOC_DPCM_STATE_PREPARE:
+ return "prepare";
+ case SND_SOC_DPCM_STATE_START:
+ return "start";
+ case SND_SOC_DPCM_STATE_STOP:
+ return "stop";
+ case SND_SOC_DPCM_STATE_SUSPEND:
+ return "suspend";
+ case SND_SOC_DPCM_STATE_PAUSED:
+ return "paused";
+ case SND_SOC_DPCM_STATE_HW_FREE:
+ return "hw_free";
+ case SND_SOC_DPCM_STATE_CLOSE:
+ return "close";
+ }
+
+ return "unknown";
+}
+
+static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
+ int stream, char *buf, size_t size)
+{
+ struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params;
+ struct snd_soc_dpcm *dpcm;
+ ssize_t offset = 0;
+ unsigned long flags;
+
+ /* FE state */
+ offset += scnprintf(buf + offset, size - offset,
+ "[%s - %s]\n", fe->dai_link->name,
+ stream ? "Capture" : "Playback");
+
+ offset += scnprintf(buf + offset, size - offset, "State: %s\n",
+ dpcm_state_string(fe->dpcm[stream].state));
+
+ if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+ offset += scnprintf(buf + offset, size - offset,
+ "Hardware Params: "
+ "Format = %s, Channels = %d, Rate = %d\n",
+ snd_pcm_format_name(params_format(params)),
+ params_channels(params),
+ params_rate(params));
+
+ /* BEs state */
+ offset += scnprintf(buf + offset, size - offset, "Backends:\n");
+
+ if (list_empty(&fe->dpcm[stream].be_clients)) {
+ offset += scnprintf(buf + offset, size - offset,
+ " No active DSP links\n");
+ goto out;
+ }
+
+ spin_lock_irqsave(&fe->card->dpcm_lock, flags);
+ for_each_dpcm_be(fe, stream, dpcm) {
+ struct snd_soc_pcm_runtime *be = dpcm->be;
+ params = &dpcm->hw_params;
+
+ offset += scnprintf(buf + offset, size - offset,
+ "- %s\n", be->dai_link->name);
+
+ offset += scnprintf(buf + offset, size - offset,
+ " State: %s\n",
+ dpcm_state_string(be->dpcm[stream].state));
+
+ if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+ offset += scnprintf(buf + offset, size - offset,
+ " Hardware Params: "
+ "Format = %s, Channels = %d, Rate = %d\n",
+ snd_pcm_format_name(params_format(params)),
+ params_channels(params),
+ params_rate(params));
+ }
+ spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
+out:
+ return offset;
+}
+
+static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ struct snd_soc_pcm_runtime *fe = file->private_data;
+ ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
+ int stream;
+ char *buf;
+
+ if (fe->num_cpus > 1) {
+ dev_err(fe->dev,
+ "%s doesn't support Multi CPU yet\n", __func__);
+ return -EINVAL;
+ }
+
+ buf = kmalloc(out_count, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ for_each_pcm_streams(stream)
+ if (snd_soc_dai_stream_valid(fe->cpu_dai, stream))
+ offset += dpcm_show_state(fe, stream,
+ buf + offset,
+ out_count - offset);
+
+ ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
+
+ kfree(buf);
+ return ret;
+}
+
+static const struct file_operations dpcm_state_fops = {
+ .open = simple_open,
+ .read = dpcm_state_read_file,
+ .llseek = default_llseek,
+};
+
+void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
+{
+ if (!rtd->dai_link)
+ return;
+
+ if (!rtd->dai_link->dynamic)
+ return;
+
+ if (!rtd->card->debugfs_card_root)
+ return;
+
+ rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name,
+ rtd->card->debugfs_card_root);
+
+ debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root,
+ rtd, &dpcm_state_fops);
+}
+
+static void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, int stream)
+{
+ char *name;
+
+ name = kasprintf(GFP_KERNEL, "%s:%s", dpcm->be->dai_link->name,
+ stream ? "capture" : "playback");
+ if (name) {
+ dpcm->debugfs_state = debugfs_create_dir(
+ name, dpcm->fe->debugfs_dpcm_root);
+ debugfs_create_u32("state", 0644, dpcm->debugfs_state,
+ &dpcm->state);
+ kfree(name);
+ }
+}
+
+static void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm)
+{
+ debugfs_remove_recursive(dpcm->debugfs_state);
+}
+
+#else
+static inline void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm,
+ int stream)
+{
+}
+
+static inline void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm)
+{
+}
+#endif
+
static int soc_rtd_startup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_substream *substream)
{
@@ -82,6 +256,21 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd,
return 0;
}
+static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd,
+ int stream, int action)
+{
+ struct snd_soc_dai *dai;
+ int i;
+
+ lockdep_assert_held(&rtd->card->pcm_mutex);
+
+ for_each_rtd_dais(rtd, i, dai) {
+ dai->stream_active[stream] += action;
+ dai->active += action;
+ dai->component->active += action;
+ }
+}
+
/**
* snd_soc_runtime_activate() - Increment active count for PCM runtime components
* @rtd: ASoC PCM runtime that is activated
@@ -94,29 +283,9 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd,
*/
void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
- int i;
-
- lockdep_assert_held(&rtd->card->pcm_mutex);
-
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active++;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- codec_dai->capture_active++;
- }
-
- cpu_dai->active++;
- cpu_dai->component->active++;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- codec_dai->active++;
- codec_dai->component->active++;
- }
+ snd_soc_runtime_action(rtd, stream, 1);
}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_activate);
/**
* snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components
@@ -130,29 +299,9 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
*/
void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
- int i;
-
- lockdep_assert_held(&rtd->card->pcm_mutex);
-
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active--;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- codec_dai->playback_active--;
- } else {
- cpu_dai->capture_active--;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- codec_dai->capture_active--;
- }
-
- cpu_dai->active--;
- cpu_dai->component->active--;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- codec_dai->component->active--;
- codec_dai->active--;
- }
+ snd_soc_runtime_action(rtd, stream, -1);
}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_deactivate);
/**
* snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay
@@ -287,8 +436,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
+ struct snd_soc_dai *cpu_dai;
unsigned int rate, channels, sample_bits, symmetry, i;
rate = params_rate(params);
@@ -296,40 +445,51 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
sample_bits = snd_pcm_format_physical_width(params_format(params));
/* reject unmatched parameters when applying symmetry */
- symmetry = cpu_dai->driver->symmetric_rates ||
- rtd->dai_link->symmetric_rates;
+ symmetry = rtd->dai_link->symmetric_rates;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- symmetry |= codec_dai->driver->symmetric_rates;
+ for_each_rtd_cpu_dais(rtd, i, dai)
+ symmetry |= dai->driver->symmetric_rates;
- if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
- dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
- cpu_dai->rate, rate);
- return -EINVAL;
+ if (symmetry) {
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ if (cpu_dai->rate && cpu_dai->rate != rate) {
+ dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
+ cpu_dai->rate, rate);
+ return -EINVAL;
+ }
+ }
}
- symmetry = cpu_dai->driver->symmetric_channels ||
- rtd->dai_link->symmetric_channels;
+ symmetry = rtd->dai_link->symmetric_channels;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- symmetry |= codec_dai->driver->symmetric_channels;
+ for_each_rtd_dais(rtd, i, dai)
+ symmetry |= dai->driver->symmetric_channels;
- if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
- dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
- cpu_dai->channels, channels);
- return -EINVAL;
+ if (symmetry) {
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ if (cpu_dai->channels &&
+ cpu_dai->channels != channels) {
+ dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
+ cpu_dai->channels, channels);
+ return -EINVAL;
+ }
+ }
}
- symmetry = cpu_dai->driver->symmetric_samplebits ||
- rtd->dai_link->symmetric_samplebits;
+ symmetry = rtd->dai_link->symmetric_samplebits;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- symmetry |= codec_dai->driver->symmetric_samplebits;
+ for_each_rtd_dais(rtd, i, dai)
+ symmetry |= dai->driver->symmetric_samplebits;
- if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
- dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
- cpu_dai->sample_bits, sample_bits);
- return -EINVAL;
+ if (symmetry) {
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ if (cpu_dai->sample_bits &&
+ cpu_dai->sample_bits != sample_bits) {
+ dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
+ cpu_dai->sample_bits, sample_bits);
+ return -EINVAL;
+ }
+ }
}
return 0;
@@ -338,20 +498,19 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver;
struct snd_soc_dai_link *link = rtd->dai_link;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
unsigned int symmetry, i;
- symmetry = cpu_driver->symmetric_rates || link->symmetric_rates ||
- cpu_driver->symmetric_channels || link->symmetric_channels ||
- cpu_driver->symmetric_samplebits || link->symmetric_samplebits;
+ symmetry = link->symmetric_rates ||
+ link->symmetric_channels ||
+ link->symmetric_samplebits;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
+ for_each_rtd_dais(rtd, i, dai)
symmetry = symmetry ||
- codec_dai->driver->symmetric_rates ||
- codec_dai->driver->symmetric_channels ||
- codec_dai->driver->symmetric_samplebits;
+ dai->driver->symmetric_rates ||
+ dai->driver->symmetric_channels ||
+ dai->driver->symmetric_samplebits;
return symmetry;
}
@@ -373,77 +532,98 @@ static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits)
static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai;
struct snd_soc_dai *codec_dai;
+ struct snd_soc_pcm_stream *pcm_codec, *pcm_cpu;
+ int stream = substream->stream;
int i;
- unsigned int bits = 0, cpu_bits;
+ unsigned int bits = 0, cpu_bits = 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if (codec_dai->driver->playback.sig_bits == 0) {
- bits = 0;
- break;
- }
- bits = max(codec_dai->driver->playback.sig_bits, bits);
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ pcm_codec = snd_soc_dai_get_pcm_stream(codec_dai, stream);
+
+ if (pcm_codec->sig_bits == 0) {
+ bits = 0;
+ break;
}
- cpu_bits = cpu_dai->driver->playback.sig_bits;
- } else {
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if (codec_dai->driver->capture.sig_bits == 0) {
- bits = 0;
- break;
- }
- bits = max(codec_dai->driver->capture.sig_bits, bits);
+ bits = max(pcm_codec->sig_bits, bits);
+ }
+
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ pcm_cpu = snd_soc_dai_get_pcm_stream(cpu_dai, stream);
+
+ if (pcm_cpu->sig_bits == 0) {
+ cpu_bits = 0;
+ break;
}
- cpu_bits = cpu_dai->driver->capture.sig_bits;
+ cpu_bits = max(pcm_cpu->sig_bits, cpu_bits);
}
soc_pcm_set_msb(substream, bits);
soc_pcm_set_msb(substream, cpu_bits);
}
-static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+/**
+ * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream
+ * @rtd: ASoC PCM runtime
+ * @hw: PCM hardware parameters (output)
+ * @stream: Direction of the PCM stream
+ *
+ * Calculates the subset of stream parameters supported by all DAIs
+ * associated with the PCM stream.
+ */
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hardware *hw, int stream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_pcm_hardware *hw = &runtime->hw;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai;
- struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver;
- struct snd_soc_dai_driver *codec_dai_drv;
+ struct snd_soc_dai *cpu_dai;
struct snd_soc_pcm_stream *codec_stream;
struct snd_soc_pcm_stream *cpu_stream;
unsigned int chan_min = 0, chan_max = UINT_MAX;
+ unsigned int cpu_chan_min = 0, cpu_chan_max = UINT_MAX;
unsigned int rate_min = 0, rate_max = UINT_MAX;
- unsigned int rates = UINT_MAX;
+ unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX;
+ unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX;
u64 formats = ULLONG_MAX;
int i;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_stream = &cpu_dai_drv->playback;
- else
- cpu_stream = &cpu_dai_drv->capture;
+ /* first calculate min/max only for CPUs in the DAI link */
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
- /* first calculate min/max only for CODECs in the DAI link */
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ /*
+ * Skip CPUs which don't support the current stream type.
+ * Otherwise, since the rate, channel, and format values will
+ * zero in that case, we would have no usable settings left,
+ * causing the resulting setup to fail.
+ */
+ if (!snd_soc_dai_stream_valid(cpu_dai, stream))
+ continue;
+
+ cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream);
+
+ cpu_chan_min = max(cpu_chan_min, cpu_stream->channels_min);
+ cpu_chan_max = min(cpu_chan_max, cpu_stream->channels_max);
+ cpu_rate_min = max(cpu_rate_min, cpu_stream->rate_min);
+ cpu_rate_max = min_not_zero(cpu_rate_max, cpu_stream->rate_max);
+ formats &= cpu_stream->formats;
+ cpu_rates = snd_pcm_rate_mask_intersect(cpu_stream->rates,
+ cpu_rates);
+ }
+
+ /* second calculate min/max only for CODECs in the DAI link */
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
/*
* Skip CODECs which don't support the current stream type.
* Otherwise, since the rate, channel, and format values will
* zero in that case, we would have no usable settings left,
* causing the resulting setup to fail.
- * At least one CODEC should match, otherwise we should have
- * bailed out on a higher level, since there would be no
- * CODEC to support the transfer direction in that case.
*/
- if (!snd_soc_dai_stream_valid(codec_dai,
- substream->stream))
+ if (!snd_soc_dai_stream_valid(codec_dai, stream))
continue;
- codec_dai_drv = codec_dai->driver;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- codec_stream = &codec_dai_drv->playback;
- else
- codec_stream = &codec_dai_drv->capture;
+ codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream);
+
chan_min = max(chan_min, codec_stream->channels_min);
chan_max = min(chan_max, codec_stream->channels_max);
rate_min = max(rate_min, codec_stream->rate_min);
@@ -452,74 +632,107 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates);
}
+ /* Verify both a valid CPU DAI and a valid CODEC DAI were found */
+ if (!chan_min || !cpu_chan_min)
+ return -EINVAL;
+
/*
* chan min/max cannot be enforced if there are multiple CODEC DAIs
- * connected to a single CPU DAI, use CPU DAI's directly and let
+ * connected to CPU DAI(s), use CPU DAI's directly and let
* channel allocation be fixed up later
*/
if (rtd->num_codecs > 1) {
- chan_min = cpu_stream->channels_min;
- chan_max = cpu_stream->channels_max;
+ chan_min = cpu_chan_min;
+ chan_max = cpu_chan_max;
}
- hw->channels_min = max(chan_min, cpu_stream->channels_min);
- hw->channels_max = min(chan_max, cpu_stream->channels_max);
- if (hw->formats)
- hw->formats &= formats & cpu_stream->formats;
- else
- hw->formats = formats & cpu_stream->formats;
- hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates);
+ /* finally find a intersection between CODECs and CPUs */
+ hw->channels_min = max(chan_min, cpu_chan_min);
+ hw->channels_max = min(chan_max, cpu_chan_max);
+ hw->formats = formats;
+ hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates);
- snd_pcm_limit_hw_rates(runtime);
+ snd_pcm_hw_limit_rates(hw);
- hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
+ hw->rate_min = max(hw->rate_min, cpu_rate_min);
hw->rate_min = max(hw->rate_min, rate_min);
- hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
+ hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max);
hw->rate_max = min_not_zero(hw->rate_max, rate_max);
+
+ return 0;
}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw);
-static int soc_pcm_components_open(struct snd_pcm_substream *substream,
- struct snd_soc_component **last)
+static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_hardware *hw = &substream->runtime->hw;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u64 formats = hw->formats;
+
+ /*
+ * At least one CPU and one CODEC should match. Otherwise, we should
+ * have bailed out on a higher level, since there would be no CPU or
+ * CODEC to support the transfer direction in that case.
+ */
+ snd_soc_runtime_calc_hw(rtd, hw, substream->stream);
+
+ if (formats)
+ hw->formats &= formats;
+}
+
+static int soc_pcm_components_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_component *last = NULL;
struct snd_soc_component *component;
int i, ret = 0;
for_each_rtd_components(rtd, i, component) {
- *last = component;
+ last = component;
ret = snd_soc_component_module_get_when_open(component);
if (ret < 0) {
dev_err(component->dev,
"ASoC: can't get module %s\n",
component->name);
- return ret;
+ break;
}
ret = snd_soc_component_open(component, substream);
if (ret < 0) {
+ snd_soc_component_module_put_when_close(component);
dev_err(component->dev,
"ASoC: can't open component %s: %d\n",
component->name, ret);
- return ret;
+ break;
}
}
- *last = NULL;
- return 0;
+
+ if (ret < 0) {
+ /* rollback on error */
+ for_each_rtd_components(rtd, i, component) {
+ if (component == last)
+ break;
+
+ snd_soc_component_close(component, substream);
+ snd_soc_component_module_put_when_close(component);
+ }
+ }
+
+ return ret;
}
-static int soc_pcm_components_close(struct snd_pcm_substream *substream,
- struct snd_soc_component *last)
+static int soc_pcm_components_close(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component;
- int i, ret = 0;
+ int i, r, ret = 0;
for_each_rtd_components(rtd, i, component) {
- if (component == last)
- break;
+ r = snd_soc_component_close(component, substream);
+ if (r < 0)
+ ret = r; /* use last ret */
- ret |= snd_soc_component_close(component, substream);
snd_soc_component_module_put_when_close(component);
}
@@ -527,6 +740,45 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream,
}
/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and components are also
+ * shutdown.
+ */
+static int soc_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_component *component;
+ struct snd_soc_dai *dai;
+ int i;
+
+ mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
+
+ snd_soc_runtime_deactivate(rtd, substream->stream);
+
+ for_each_rtd_dais(rtd, i, dai)
+ snd_soc_dai_shutdown(dai, substream);
+
+ soc_rtd_shutdown(rtd, substream);
+
+ soc_pcm_components_close(substream);
+
+ snd_soc_dapm_stream_stop(rtd, substream->stream);
+
+ mutex_unlock(&rtd->card->pcm_mutex);
+
+ for_each_rtd_components(rtd, i, component) {
+ pm_runtime_mark_last_busy(component->dev);
+ pm_runtime_put_autosuspend(component->dev);
+ }
+
+ for_each_rtd_components(rtd, i, component)
+ if (!component->active)
+ pinctrl_pm_select_sleep_state(component->dev);
+
+ return 0;
+}
+
+/*
* Called by ALSA when a PCM substream is opened, the runtime->hw record is
* then initialized and any private data can be allocated. This also calls
* startup for the cpu DAI, component, machine and codec DAI.
@@ -536,9 +788,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
const char *codec_dai_name = "multicodec";
+ const char *cpu_dai_name = "multicpu";
int i, ret = 0;
for_each_rtd_components(rtd, i, component)
@@ -549,38 +801,31 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
- /* startup the audio subsystem */
- ret = snd_soc_dai_startup(cpu_dai, substream);
- if (ret < 0) {
- dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n",
- cpu_dai->name, ret);
- goto out;
- }
-
- ret = soc_pcm_components_open(substream, &component);
+ ret = soc_pcm_components_open(substream);
if (ret < 0)
goto component_err;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- ret = snd_soc_dai_startup(codec_dai, substream);
+ ret = soc_rtd_startup(rtd, substream);
+ if (ret < 0) {
+ pr_err("ASoC: %s startup failed: %d\n",
+ rtd->dai_link->name, ret);
+ goto rtd_startup_err;
+ }
+
+ /* startup the audio subsystem */
+ for_each_rtd_dais(rtd, i, dai) {
+ ret = snd_soc_dai_startup(dai, substream);
if (ret < 0) {
- dev_err(codec_dai->dev,
- "ASoC: can't open codec %s: %d\n",
- codec_dai->name, ret);
- goto codec_dai_err;
+ dev_err(dai->dev,
+ "ASoC: can't open DAI %s: %d\n",
+ dai->name, ret);
+ goto config_err;
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- codec_dai->tx_mask = 0;
+ dai->tx_mask = 0;
else
- codec_dai->rx_mask = 0;
- }
-
- ret = soc_rtd_startup(rtd, substream);
- if (ret < 0) {
- pr_err("ASoC: %s startup failed: %d\n",
- rtd->dai_link->name, ret);
- goto machine_err;
+ dai->rx_mask = 0;
}
/* Dynamic PCM DAI links compat checks use dynamic capabilities */
@@ -593,46 +838,43 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
if (rtd->num_codecs == 1)
codec_dai_name = rtd->codec_dai->name;
+ if (rtd->num_cpus == 1)
+ cpu_dai_name = rtd->cpu_dai->name;
+
if (soc_pcm_has_symmetry(substream))
runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
ret = -EINVAL;
if (!runtime->hw.rates) {
printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
- codec_dai_name, cpu_dai->name);
+ codec_dai_name, cpu_dai_name);
goto config_err;
}
if (!runtime->hw.formats) {
printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n",
- codec_dai_name, cpu_dai->name);
+ codec_dai_name, cpu_dai_name);
goto config_err;
}
if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
runtime->hw.channels_min > runtime->hw.channels_max) {
printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n",
- codec_dai_name, cpu_dai->name);
+ codec_dai_name, cpu_dai_name);
goto config_err;
}
soc_pcm_apply_msb(substream);
/* Symmetry only applies if we've already got an active stream. */
- if (cpu_dai->active) {
- ret = soc_pcm_apply_symmetry(substream, cpu_dai);
- if (ret != 0)
- goto config_err;
- }
-
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if (codec_dai->active) {
- ret = soc_pcm_apply_symmetry(substream, codec_dai);
+ for_each_rtd_dais(rtd, i, dai) {
+ if (dai->active) {
+ ret = soc_pcm_apply_symmetry(substream, dai);
if (ret != 0)
goto config_err;
}
}
pr_debug("ASoC: %s <-> %s info:\n",
- codec_dai_name, cpu_dai->name);
+ codec_dai_name, cpu_dai_name);
pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates);
pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min,
runtime->hw.channels_max);
@@ -647,20 +889,13 @@ dynamic:
return 0;
config_err:
- soc_rtd_shutdown(rtd, substream);
-
-machine_err:
- i = rtd->num_codecs;
-
-codec_dai_err:
- for_each_rtd_codec_dai_rollback(rtd, i, codec_dai)
- snd_soc_dai_shutdown(codec_dai, substream);
+ for_each_rtd_dais(rtd, i, dai)
+ snd_soc_dai_shutdown(dai, substream);
+ soc_rtd_shutdown(rtd, substream);
+rtd_startup_err:
+ soc_pcm_components_close(substream);
component_err:
- soc_pcm_components_close(substream, component);
-
- snd_soc_dai_shutdown(cpu_dai, substream);
-out:
mutex_unlock(&rtd->card->pcm_mutex);
for_each_rtd_components(rtd, i, component) {
@@ -686,59 +921,6 @@ static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
}
/*
- * Called by ALSA when a PCM substream is closed. Private data can be
- * freed here. The cpu DAI, codec DAI, machine and components are also
- * shutdown.
- */
-static int soc_pcm_close(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
- int i;
-
- mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
-
- snd_soc_runtime_deactivate(rtd, substream->stream);
-
- /* clear the corresponding DAIs rate when inactive */
- if (!cpu_dai->active)
- cpu_dai->rate = 0;
-
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if (!codec_dai->active)
- codec_dai->rate = 0;
- }
-
- snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream);
-
- snd_soc_dai_shutdown(cpu_dai, substream);
-
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- snd_soc_dai_shutdown(codec_dai, substream);
-
- soc_rtd_shutdown(rtd, substream);
-
- soc_pcm_components_close(substream, NULL);
-
- snd_soc_dapm_stream_stop(rtd, substream->stream);
-
- mutex_unlock(&rtd->card->pcm_mutex);
-
- for_each_rtd_components(rtd, i, component) {
- pm_runtime_mark_last_busy(component->dev);
- pm_runtime_put_autosuspend(component->dev);
- }
-
- for_each_rtd_components(rtd, i, component)
- if (!component->active)
- pinctrl_pm_select_sleep_state(component->dev);
-
- return 0;
-}
-
-/*
* Called by ALSA when the PCM substream is prepared, can set format, sample
* rate, etc. This function is non atomic and can be called multiple times,
* it can refer to the runtime info.
@@ -747,8 +929,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
int i, ret = 0;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -769,23 +950,15 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
}
}
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- ret = snd_soc_dai_prepare(codec_dai, substream);
+ for_each_rtd_dais(rtd, i, dai) {
+ ret = snd_soc_dai_prepare(dai, substream);
if (ret < 0) {
- dev_err(codec_dai->dev,
- "ASoC: codec DAI prepare error: %d\n",
- ret);
+ dev_err(dai->dev,
+ "ASoC: DAI prepare error: %d\n", ret);
goto out;
}
}
- ret = snd_soc_dai_prepare(cpu_dai, substream);
- if (ret < 0) {
- dev_err(cpu_dai->dev,
- "ASoC: cpu DAI prepare error: %d\n", ret);
- goto out;
- }
-
/* cancel any delayed stream shutdown that is pending */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
rtd->pop_wait) {
@@ -796,10 +969,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
snd_soc_dapm_stream_event(rtd, substream->stream,
SND_SOC_DAPM_STREAM_START);
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- snd_soc_dai_digital_mute(codec_dai, 0,
- substream->stream);
- snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);
+ for_each_rtd_dais(rtd, i, dai)
+ snd_soc_dai_digital_mute(dai, 0, substream->stream);
out:
mutex_unlock(&rtd->card->pcm_mutex);
@@ -822,13 +993,15 @@ static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component;
- int i, ret = 0;
+ int i, r, ret = 0;
for_each_rtd_components(rtd, i, component) {
if (component == last)
break;
- ret |= snd_soc_component_hw_free(component, substream);
+ r = snd_soc_component_hw_free(component, substream);
+ if (r < 0)
+ ret = r; /* use last ret */
}
return ret;
@@ -844,7 +1017,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai;
struct snd_soc_dai *codec_dai;
int i, ret = 0;
@@ -861,7 +1034,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
goto out;
}
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
struct snd_pcm_hw_params codec_params;
/*
@@ -908,17 +1081,26 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
snd_soc_dapm_update_dai(substream, &codec_params, codec_dai);
}
- ret = snd_soc_dai_hw_params(cpu_dai, substream, params);
- if (ret < 0)
- goto interface_err;
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ /*
+ * Skip CPUs which don't support the current stream
+ * type. See soc_pcm_init_runtime_hw() for more details
+ */
+ if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream))
+ continue;
- /* store the parameters for each DAIs */
- cpu_dai->rate = params_rate(params);
- cpu_dai->channels = params_channels(params);
- cpu_dai->sample_bits =
- snd_pcm_format_physical_width(params_format(params));
+ ret = snd_soc_dai_hw_params(cpu_dai, substream, params);
+ if (ret < 0)
+ goto interface_err;
+
+ /* store the parameters for each DAI */
+ cpu_dai->rate = params_rate(params);
+ cpu_dai->channels = params_channels(params);
+ cpu_dai->sample_bits =
+ snd_pcm_format_physical_width(params_format(params));
- snd_soc_dapm_update_dai(substream, params, cpu_dai);
+ snd_soc_dapm_update_dai(substream, params, cpu_dai);
+ }
for_each_rtd_components(rtd, i, component) {
ret = snd_soc_component_hw_params(component, substream, params);
@@ -938,14 +1120,21 @@ out:
component_err:
soc_pcm_components_hw_free(substream, component);
- snd_soc_dai_hw_free(cpu_dai, substream);
- cpu_dai->rate = 0;
+ i = rtd->num_cpus;
interface_err:
+ for_each_rtd_cpu_dais_rollback(rtd, i, cpu_dai) {
+ if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream))
+ continue;
+
+ snd_soc_dai_hw_free(cpu_dai, substream);
+ cpu_dai->rate = 0;
+ }
+
i = rtd->num_codecs;
codec_err:
- for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais_rollback(rtd, i, codec_dai) {
if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
continue;
@@ -965,34 +1154,23 @@ codec_err:
static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
- bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_soc_dai *dai;
int i;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
/* clear the corresponding DAIs parameters when going to be inactive */
- if (cpu_dai->active == 1) {
- cpu_dai->rate = 0;
- cpu_dai->channels = 0;
- cpu_dai->sample_bits = 0;
- }
+ for_each_rtd_dais(rtd, i, dai) {
+ int active = dai->stream_active[substream->stream];
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if (codec_dai->active == 1) {
- codec_dai->rate = 0;
- codec_dai->channels = 0;
- codec_dai->sample_bits = 0;
+ if (dai->active == 1) {
+ dai->rate = 0;
+ dai->channels = 0;
+ dai->sample_bits = 0;
}
- }
- /* apply codec digital mute */
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if ((playback && codec_dai->playback_active == 1) ||
- (!playback && codec_dai->capture_active == 1))
- snd_soc_dai_digital_mute(codec_dai, 1,
- substream->stream);
+ if (active == 1)
+ snd_soc_dai_digital_mute(dai, 1, substream->stream);
}
/* free any machine hw params */
@@ -1002,15 +1180,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
soc_pcm_components_hw_free(substream, NULL);
/* now free hw params for the DAIs */
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+ for_each_rtd_dais(rtd, i, dai) {
+ if (!snd_soc_dai_stream_valid(dai, substream->stream))
continue;
- snd_soc_dai_hw_free(codec_dai, substream);
+ snd_soc_dai_hw_free(dai, substream);
}
- snd_soc_dai_hw_free(cpu_dai, substream);
-
mutex_unlock(&rtd->card->pcm_mutex);
return 0;
}
@@ -1019,8 +1195,7 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
int i, ret;
ret = soc_rtd_trigger(rtd, substream, cmd);
@@ -1033,12 +1208,8 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd)
return ret;
}
- ret = snd_soc_dai_trigger(cpu_dai, substream, cmd);
- if (ret < 0)
- return ret;
-
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- ret = snd_soc_dai_trigger(codec_dai, substream, cmd);
+ for_each_rtd_dais(rtd, i, dai) {
+ ret = snd_soc_dai_trigger(dai, substream, cmd);
if (ret < 0)
return ret;
}
@@ -1050,20 +1221,15 @@ static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
int i, ret;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- ret = snd_soc_dai_trigger(codec_dai, substream, cmd);
+ for_each_rtd_dais(rtd, i, dai) {
+ ret = snd_soc_dai_trigger(dai, substream, cmd);
if (ret < 0)
return ret;
}
- ret = snd_soc_dai_trigger(cpu_dai, substream, cmd);
- if (ret < 0)
- return ret;
-
for_each_rtd_components(rtd, i, component) {
ret = snd_soc_component_trigger(component, substream, cmd);
if (ret < 0)
@@ -1103,20 +1269,15 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
int i, ret;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd);
+ for_each_rtd_dais(rtd, i, dai) {
+ ret = snd_soc_dai_bespoke_trigger(dai, substream, cmd);
if (ret < 0)
return ret;
}
- ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
- if (ret < 0)
- return ret;
-
return 0;
}
/*
@@ -1127,12 +1288,13 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai;
struct snd_soc_dai *codec_dai;
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t offset = 0;
snd_pcm_sframes_t delay = 0;
snd_pcm_sframes_t codec_delay = 0;
+ snd_pcm_sframes_t cpu_delay = 0;
int i;
/* clearing the previous total delay */
@@ -1143,9 +1305,13 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
/* base delay if assigned in pointer callback */
delay = runtime->delay;
- delay += snd_soc_dai_delay(cpu_dai, substream);
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ cpu_delay = max(cpu_delay,
+ snd_soc_dai_delay(cpu_dai, substream));
+ }
+ delay += cpu_delay;
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
codec_delay = max(codec_delay,
snd_soc_dai_delay(codec_dai, substream));
}
@@ -1162,9 +1328,6 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
{
struct snd_soc_dpcm *dpcm;
unsigned long flags;
-#ifdef CONFIG_DEBUG_FS
- char *name;
-#endif
/* only add new dpcms */
for_each_dpcm_be(fe, stream, dpcm) {
@@ -1189,17 +1352,8 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
stream ? "capture" : "playback", fe->dai_link->name,
stream ? "<-" : "->", be->dai_link->name);
-#ifdef CONFIG_DEBUG_FS
- name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name,
- stream ? "capture" : "playback");
- if (name) {
- dpcm->debugfs_state = debugfs_create_dir(name,
- fe->debugfs_dpcm_root);
- debugfs_create_u32("state", 0644, dpcm->debugfs_state,
- &dpcm->state);
- kfree(name);
- }
-#endif
+ dpcm_create_debugfs_state(dpcm, stream);
+
return 1;
}
@@ -1252,9 +1406,8 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
/* BEs still alive need new FE */
dpcm_be_reparent(fe, dpcm->be, stream);
-#ifdef CONFIG_DEBUG_FS
- debugfs_remove_recursive(dpcm->debugfs_state);
-#endif
+ dpcm_remove_debugfs_state(dpcm);
+
spin_lock_irqsave(&fe->card->dpcm_lock, flags);
list_del(&dpcm->list_be);
list_del(&dpcm->list_fe);
@@ -1268,74 +1421,41 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
struct snd_soc_dapm_widget *widget, int stream)
{
struct snd_soc_pcm_runtime *be;
+ struct snd_soc_dapm_widget *w;
struct snd_soc_dai *dai;
int i;
dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- for_each_card_rtds(card, be) {
-
- if (!be->dai_link->no_pcm)
- continue;
-
- dev_dbg(card->dev, "ASoC: try BE : %s\n",
- be->cpu_dai->playback_widget ?
- be->cpu_dai->playback_widget->name : "(not set)");
+ for_each_card_rtds(card, be) {
- if (be->cpu_dai->playback_widget == widget)
- return be;
-
- for_each_rtd_codec_dai(be, i, dai) {
- if (dai->playback_widget == widget)
- return be;
- }
- }
- } else {
-
- for_each_card_rtds(card, be) {
+ if (!be->dai_link->no_pcm)
+ continue;
- if (!be->dai_link->no_pcm)
- continue;
+ for_each_rtd_dais(be, i, dai) {
+ w = snd_soc_dai_get_widget(dai, stream);
- dev_dbg(card->dev, "ASoC: try BE %s\n",
- be->cpu_dai->capture_widget ?
- be->cpu_dai->capture_widget->name : "(not set)");
+ dev_dbg(card->dev, "ASoC: try BE : %s\n",
+ w ? w->name : "(not set)");
- if (be->cpu_dai->capture_widget == widget)
+ if (w == widget)
return be;
-
- for_each_rtd_codec_dai(be, i, dai) {
- if (dai->capture_widget == widget)
- return be;
- }
}
}
- /* dai link name and stream name set correctly ? */
- dev_err(card->dev, "ASoC: can't get %s BE for %s\n",
- stream ? "capture" : "playback", widget->name);
+ /* Widget provided is not a BE */
return NULL;
}
-static inline struct snd_soc_dapm_widget *
- dai_get_widget(struct snd_soc_dai *dai, int stream)
-{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- return dai->playback_widget;
- else
- return dai->capture_widget;
-}
-
static int widget_in_list(struct snd_soc_dapm_widget_list *list,
struct snd_soc_dapm_widget *widget)
{
+ struct snd_soc_dapm_widget *w;
int i;
- for (i = 0; i < list->num_widgets; i++) {
- if (widget == list->widgets[i])
+ for_each_dapm_widgets(list, i, w)
+ if (widget == w)
return 1;
- }
return 0;
}
@@ -1345,36 +1465,17 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
{
struct snd_soc_card *card = widget->dapm->card;
struct snd_soc_pcm_runtime *rtd;
- struct snd_soc_dai *dai;
- int i;
+ int stream;
- if (dir == SND_SOC_DAPM_DIR_OUT) {
- for_each_card_rtds(card, rtd) {
- if (!rtd->dai_link->no_pcm)
- continue;
-
- if (rtd->cpu_dai->playback_widget == widget)
- return true;
-
- for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->playback_widget == widget)
- return true;
- }
- }
- } else { /* SND_SOC_DAPM_DIR_IN */
- for_each_card_rtds(card, rtd) {
- if (!rtd->dai_link->no_pcm)
- continue;
-
- if (rtd->cpu_dai->capture_widget == widget)
- return true;
+ /* adjust dir to stream */
+ if (dir == SND_SOC_DAPM_DIR_OUT)
+ stream = SNDRV_PCM_STREAM_PLAYBACK;
+ else
+ stream = SNDRV_PCM_STREAM_CAPTURE;
- for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->capture_widget == widget)
- return true;
- }
- }
- }
+ rtd = dpcm_get_be(card, widget, stream);
+ if (rtd)
+ return true;
return false;
}
@@ -1385,6 +1486,12 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
struct snd_soc_dai *cpu_dai = fe->cpu_dai;
int paths;
+ if (fe->num_cpus > 1) {
+ dev_err(fe->dev,
+ "%s doesn't support Multi CPU yet\n", __func__);
+ return -EINVAL;
+ }
+
/* get number of valid DAI paths and their widgets */
paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list,
dpcm_end_walk_at_be);
@@ -1395,37 +1502,42 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
return paths;
}
-static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
- struct snd_soc_dapm_widget_list **list_)
+void dpcm_path_put(struct snd_soc_dapm_widget_list **list)
+{
+ snd_soc_dapm_dai_free_widgets(list);
+}
+
+static bool dpcm_be_is_active(struct snd_soc_dpcm *dpcm, int stream,
+ struct snd_soc_dapm_widget_list *list)
{
- struct snd_soc_dpcm *dpcm;
- struct snd_soc_dapm_widget_list *list = *list_;
struct snd_soc_dapm_widget *widget;
struct snd_soc_dai *dai;
- int prune = 0;
- int do_prune;
-
- /* Destroy any old FE <--> BE connections */
- for_each_dpcm_be(fe, stream, dpcm) {
- unsigned int i;
+ unsigned int i;
- /* is there a valid CPU DAI widget for this BE */
- widget = dai_get_widget(dpcm->be->cpu_dai, stream);
+ /* is there a valid DAI widget for this BE */
+ for_each_rtd_dais(dpcm->be, i, dai) {
+ widget = snd_soc_dai_get_widget(dai, stream);
- /* prune the BE if it's no longer in our active list */
+ /*
+ * The BE is pruned only if none of the dai
+ * widgets are in the active list.
+ */
if (widget && widget_in_list(list, widget))
- continue;
+ return true;
+ }
- /* is there a valid CODEC DAI widget for this BE */
- do_prune = 1;
- for_each_rtd_codec_dai(dpcm->be, i, dai) {
- widget = dai_get_widget(dai, stream);
+ return false;
+}
- /* prune the BE if it's no longer in our active list */
- if (widget && widget_in_list(list, widget))
- do_prune = 0;
- }
- if (!do_prune)
+static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
+ struct snd_soc_dapm_widget_list **list_)
+{
+ struct snd_soc_dpcm *dpcm;
+ int prune = 0;
+
+ /* Destroy any old FE <--> BE connections */
+ for_each_dpcm_be(fe, stream, dpcm) {
+ if (dpcm_be_is_active(dpcm, stream, *list_))
continue;
dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n",
@@ -1446,12 +1558,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
struct snd_soc_card *card = fe->card;
struct snd_soc_dapm_widget_list *list = *list_;
struct snd_soc_pcm_runtime *be;
+ struct snd_soc_dapm_widget *widget;
int i, new = 0, err;
/* Create any new FE <--> BE connections */
- for (i = 0; i < list->num_widgets; i++) {
+ for_each_dapm_widgets(list, i, widget) {
- switch (list->widgets[i]->id) {
+ switch (widget->id) {
case snd_soc_dapm_dai_in:
if (stream != SNDRV_PCM_STREAM_PLAYBACK)
continue;
@@ -1465,17 +1578,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
}
/* is there a valid BE rtd for this widget */
- be = dpcm_get_be(card, list->widgets[i], stream);
+ be = dpcm_get_be(card, widget, stream);
if (!be) {
dev_err(fe->dev, "ASoC: no BE found for %s\n",
- list->widgets[i]->name);
+ widget->name);
continue;
}
- /* make sure BE is a real BE */
- if (!be->dai_link->no_pcm)
- continue;
-
/* don't connect if FE is not running */
if (!fe->dpcm[stream].runtime && !fe->fe_compr)
continue;
@@ -1484,7 +1593,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
err = dpcm_be_connect(fe, be, stream);
if (err < 0) {
dev_err(fe->dev, "ASoC: can't connect %s\n",
- list->widgets[i]->name);
+ widget->name);
break;
} else if (err == 0) /* already connected */
continue;
@@ -1671,11 +1780,10 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
- struct snd_soc_dai_driver *codec_dai_drv;
struct snd_soc_pcm_stream *codec_stream;
int i;
- for_each_rtd_codec_dai(be, i, dai) {
+ for_each_rtd_codec_dais(be, i, dai) {
/*
* Skip CODECs which don't support the current stream
* type. See soc_pcm_init_runtime_hw() for more details
@@ -1683,11 +1791,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
if (!snd_soc_dai_stream_valid(dai, stream))
continue;
- codec_dai_drv = dai->driver;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- codec_stream = &codec_dai_drv->playback;
- else
- codec_stream = &codec_dai_drv->capture;
+ codec_stream = snd_soc_dai_get_pcm_stream(dai, stream);
*formats &= codec_stream->formats;
}
@@ -1712,30 +1816,33 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
- struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver;
- struct snd_soc_dai_driver *codec_dai_drv;
struct snd_soc_pcm_stream *codec_stream;
struct snd_soc_pcm_stream *cpu_stream;
+ struct snd_soc_dai *dai;
+ int i;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_stream = &cpu_dai_drv->playback;
- else
- cpu_stream = &cpu_dai_drv->capture;
+ for_each_rtd_cpu_dais(be, i, dai) {
+ /*
+ * Skip CPUs which don't support the current stream
+ * type. See soc_pcm_init_runtime_hw() for more details
+ */
+ if (!snd_soc_dai_stream_valid(dai, stream))
+ continue;
+
+ cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream);
- *channels_min = max(*channels_min, cpu_stream->channels_min);
- *channels_max = min(*channels_max, cpu_stream->channels_max);
+ *channels_min = max(*channels_min,
+ cpu_stream->channels_min);
+ *channels_max = min(*channels_max,
+ cpu_stream->channels_max);
+ }
/*
* chan min/max cannot be enforced if there are multiple CODEC
* DAIs connected to a single CPU DAI, use CPU DAI's directly
*/
if (be->num_codecs == 1) {
- codec_dai_drv = be->codec_dais[0]->driver;
-
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- codec_stream = &codec_dai_drv->playback;
- else
- codec_stream = &codec_dai_drv->capture;
+ codec_stream = snd_soc_dai_get_pcm_stream(be->codec_dais[0], stream);
*channels_min = max(*channels_min,
codec_stream->channels_min);
@@ -1764,41 +1871,23 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be;
- struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver;
- struct snd_soc_dai_driver *codec_dai_drv;
- struct snd_soc_pcm_stream *codec_stream;
- struct snd_soc_pcm_stream *cpu_stream;
+ struct snd_soc_pcm_stream *pcm;
struct snd_soc_dai *dai;
int i;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_stream = &cpu_dai_drv->playback;
- else
- cpu_stream = &cpu_dai_drv->capture;
-
- *rate_min = max(*rate_min, cpu_stream->rate_min);
- *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max);
- *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates);
-
- for_each_rtd_codec_dai(be, i, dai) {
+ for_each_rtd_dais(be, i, dai) {
/*
- * Skip CODECs which don't support the current stream
+ * Skip DAIs which don't support the current stream
* type. See soc_pcm_init_runtime_hw() for more details
*/
if (!snd_soc_dai_stream_valid(dai, stream))
continue;
- codec_dai_drv = dai->driver;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- codec_stream = &codec_dai_drv->playback;
- else
- codec_stream = &codec_dai_drv->capture;
+ pcm = snd_soc_dai_get_pcm_stream(dai, stream);
- *rate_min = max(*rate_min, codec_stream->rate_min);
- *rate_max = min_not_zero(*rate_max,
- codec_stream->rate_max);
- *rates = snd_pcm_rate_mask_intersect(*rates,
- codec_stream->rates);
+ *rate_min = max(*rate_min, pcm->rate_min);
+ *rate_max = min_not_zero(*rate_max, pcm->rate_max);
+ *rates = snd_pcm_rate_mask_intersect(*rates, pcm->rates);
}
}
}
@@ -1807,13 +1896,21 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+ struct snd_soc_dai *cpu_dai;
+ int i;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
- else
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ /*
+ * Skip CPUs which don't support the current stream
+ * type. See soc_pcm_init_runtime_hw() for more details
+ */
+ if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream))
+ continue;
+
+ dpcm_init_runtime_hw(runtime,
+ snd_soc_dai_get_pcm_stream(cpu_dai,
+ substream->stream));
+ }
dpcm_runtime_merge_format(substream, &runtime->hw.formats);
dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min,
@@ -1850,18 +1947,21 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
{
struct snd_soc_dpcm *dpcm;
struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
- struct snd_soc_dai *fe_cpu_dai = fe->cpu_dai;
+ struct snd_soc_dai *fe_cpu_dai;
int err;
+ int i;
/* apply symmetry for FE */
if (soc_pcm_has_symmetry(fe_substream))
fe_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
- /* Symmetry only applies if we've got an active stream. */
- if (fe_cpu_dai->active) {
- err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai);
- if (err < 0)
- return err;
+ for_each_rtd_cpu_dais (fe, i, fe_cpu_dai) {
+ /* Symmetry only applies if we've got an active stream. */
+ if (fe_cpu_dai->active) {
+ err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai);
+ if (err < 0)
+ return err;
+ }
}
/* apply symmetry for BE */
@@ -1870,7 +1970,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
struct snd_pcm_substream *be_substream =
snd_soc_dpcm_get_substream(be, stream);
struct snd_soc_pcm_runtime *rtd;
- struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *dai;
int i;
/* A backend may not have the requested substream */
@@ -1885,17 +1985,9 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
/* Symmetry only applies if we've got an active stream. */
- if (rtd->cpu_dai->active) {
- err = soc_pcm_apply_symmetry(fe_substream,
- rtd->cpu_dai);
- if (err < 0)
- return err;
- }
-
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if (codec_dai->active) {
- err = soc_pcm_apply_symmetry(fe_substream,
- codec_dai);
+ for_each_rtd_dais(rtd, i, dai) {
+ if (dai->active) {
+ err = soc_pcm_apply_symmetry(fe_substream, dai);
if (err < 0)
return err;
}
@@ -1913,7 +2005,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
- ret = dpcm_be_dai_startup(fe, fe_substream->stream);
+ ret = dpcm_be_dai_startup(fe, stream);
if (ret < 0) {
dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret);
goto be_err;
@@ -1934,17 +2026,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
snd_pcm_limit_hw_rates(runtime);
ret = dpcm_apply_symmetry(fe_substream, stream);
- if (ret < 0) {
+ if (ret < 0)
dev_err(fe->dev, "ASoC: failed to apply dpcm symmetry %d\n",
ret);
- goto unwind;
- }
-
- dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
- return 0;
unwind:
- dpcm_be_dai_startup_unwind(fe, fe_substream->stream);
+ if (ret < 0)
+ dpcm_be_dai_startup_unwind(fe, stream);
be_err:
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return ret;
@@ -1998,7 +2086,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
/* shutdown the BEs */
- dpcm_be_dai_shutdown(fe, substream->stream);
+ dpcm_be_dai_shutdown(fe, stream);
dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name);
@@ -2176,9 +2264,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream,
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
- memcpy(&fe->dpcm[substream->stream].hw_params, params,
+ memcpy(&fe->dpcm[stream].hw_params, params,
sizeof(struct snd_pcm_hw_params));
- ret = dpcm_be_dai_hw_params(fe, substream->stream);
+ ret = dpcm_be_dai_hw_params(fe, stream);
if (ret < 0) {
dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret);
goto out;
@@ -2500,7 +2588,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream)
goto out;
}
- ret = dpcm_be_dai_prepare(fe, substream->stream);
+ ret = dpcm_be_dai_prepare(fe, stream);
if (ret < 0)
goto out;
@@ -2652,36 +2740,18 @@ disconnect:
return ret;
}
-static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream)
-{
- int ret;
-
- dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
- ret = dpcm_run_update_startup(fe, stream);
- if (ret < 0)
- dev_err(fe->dev, "ASoC: failed to startup some BEs\n");
- dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
-
- return ret;
-}
-
-static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
-{
- int ret;
-
- dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
- ret = dpcm_run_update_shutdown(fe, stream);
- if (ret < 0)
- dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n");
- dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
-
- return ret;
-}
-
static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
{
struct snd_soc_dapm_widget_list *list;
+ int stream;
int count, paths;
+ int ret;
+
+ if (fe->num_cpus > 1) {
+ dev_err(fe->dev,
+ "%s doesn't support Multi CPU yet\n", __func__);
+ return -EINVAL;
+ }
if (!fe->dai_link->dynamic)
return 0;
@@ -2694,74 +2764,53 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
new ? "new" : "old", fe->dai_link->name);
- /* skip if FE doesn't have playback capability */
- if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) ||
- !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK))
- goto capture;
-
- /* skip if FE isn't currently playing */
- if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active)
- goto capture;
-
- paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
- if (paths < 0) {
- dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
- fe->dai_link->name, "playback");
- return paths;
- }
-
- /* update any playback paths */
- count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
- if (count) {
- if (new)
- dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
- else
- dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
- }
+ for_each_pcm_streams(stream) {
- dpcm_path_put(&list);
+ /* skip if FE doesn't have playback/capture capability */
+ if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) ||
+ !snd_soc_dai_stream_valid(fe->codec_dai, stream))
+ continue;
-capture:
- /* skip if FE doesn't have capture capability */
- if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) ||
- !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE))
- return 0;
+ /* skip if FE isn't currently playing/capturing */
+ if (!fe->cpu_dai->stream_active[stream] ||
+ !fe->codec_dai->stream_active[stream])
+ continue;
- /* skip if FE isn't currently capturing */
- if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active)
- return 0;
+ paths = dpcm_path_get(fe, stream, &list);
+ if (paths < 0) {
+ dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+ fe->dai_link->name,
+ stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ "playback" : "capture");
+ return paths;
+ }
- paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
- if (paths < 0) {
- dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
- fe->dai_link->name, "capture");
- return paths;
- }
+ /* update any playback/capture paths */
+ count = dpcm_process_paths(fe, stream, &list, new);
+ if (count) {
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
+ if (new)
+ ret = dpcm_run_update_startup(fe, stream);
+ else
+ ret = dpcm_run_update_shutdown(fe, stream);
+ if (ret < 0)
+ dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n");
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
- /* update any old capture paths */
- count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
- if (count) {
- if (new)
- dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
- else
- dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
+ dpcm_clear_pending_state(fe, stream);
+ dpcm_be_disconnect(fe, stream);
+ }
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
+ dpcm_path_put(&list);
}
- dpcm_path_put(&list);
-
return 0;
}
/* Called by DAPM mixer/mux changes to update audio routing between PCMs and
* any DAI links.
*/
-int soc_dpcm_runtime_update(struct snd_soc_card *card)
+int snd_soc_dpcm_runtime_update(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *fe;
int ret = 0;
@@ -2785,38 +2834,40 @@ out:
mutex_unlock(&card->mutex);
return ret;
}
-int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_runtime_update);
+
+static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream)
{
+ struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
struct snd_soc_dpcm *dpcm;
- struct snd_soc_dai *dai;
+ int stream = fe_substream->stream;
- for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+ /* mark FE's links ready to prune */
+ for_each_dpcm_be(fe, stream, dpcm)
+ dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
- struct snd_soc_pcm_runtime *be = dpcm->be;
- int i;
+ dpcm_be_disconnect(fe, stream);
- if (be->dai_link->ignore_suspend)
- continue;
+ fe->dpcm[stream].runtime = NULL;
+}
- for_each_rtd_codec_dai(be, i, dai) {
- struct snd_soc_dai_driver *drv = dai->driver;
+static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
+{
+ struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+ int ret;
- dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
- be->dai_link->name);
+ mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+ ret = dpcm_fe_dai_shutdown(fe_substream);
- if (drv->ops && drv->ops->digital_mute &&
- dai->playback_active)
- drv->ops->digital_mute(dai, mute);
- }
- }
+ dpcm_fe_dai_cleanup(fe_substream);
- return 0;
+ mutex_unlock(&fe->card->mutex);
+ return ret;
}
static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
{
struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
- struct snd_soc_dpcm *dpcm;
struct snd_soc_dapm_widget_list *list;
int ret;
int stream = fe_substream->stream;
@@ -2826,8 +2877,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
ret = dpcm_path_get(fe, stream, &list);
if (ret < 0) {
- mutex_unlock(&fe->card->mutex);
- return ret;
+ goto open_end;
} else if (ret == 0) {
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
@@ -2837,37 +2887,12 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
dpcm_process_paths(fe, stream, &list, 1);
ret = dpcm_fe_dai_startup(fe_substream);
- if (ret < 0) {
- /* clean up all links */
- for_each_dpcm_be(fe, stream, dpcm)
- dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
- dpcm_be_disconnect(fe, stream);
- fe->dpcm[stream].runtime = NULL;
- }
+ if (ret < 0)
+ dpcm_fe_dai_cleanup(fe_substream);
dpcm_clear_pending_state(fe, stream);
dpcm_path_put(&list);
- mutex_unlock(&fe->card->mutex);
- return ret;
-}
-
-static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
-{
- struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
- struct snd_soc_dpcm *dpcm;
- int stream = fe_substream->stream, ret;
-
- mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- ret = dpcm_fe_dai_shutdown(fe_substream);
-
- /* mark FE's links ready to prune */
- for_each_dpcm_be(fe, stream, dpcm)
- dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
- dpcm_be_disconnect(fe, stream);
-
- fe->dpcm[stream].runtime = NULL;
+open_end:
mutex_unlock(&fe->card->mutex);
return ret;
}
@@ -2876,7 +2901,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
{
struct snd_soc_dai *codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai;
struct snd_soc_component *component;
struct snd_pcm *pcm;
char new_name[64];
@@ -2888,22 +2913,29 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
capture = rtd->dai_link->dpcm_capture;
} else {
/* Adapt stream for codec2codec links */
- struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ?
- &cpu_dai->driver->playback : &cpu_dai->driver->capture;
- struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ?
- &cpu_dai->driver->capture : &cpu_dai->driver->playback;
+ int cpu_capture = rtd->dai_link->params ?
+ SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+ int cpu_playback = rtd->dai_link->params ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+
+ for_each_rtd_codec_dais(rtd, i, codec_dai) {
+ if (rtd->num_cpus == 1) {
+ cpu_dai = rtd->cpu_dais[0];
+ } else if (rtd->num_cpus == rtd->num_codecs) {
+ cpu_dai = rtd->cpu_dais[i];
+ } else {
+ dev_err(rtd->card->dev,
+ "N cpus to M codecs link is not supported yet\n");
+ return -EINVAL;
+ }
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
- snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE))
+ snd_soc_dai_stream_valid(cpu_dai, cpu_playback))
playback = 1;
if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) &&
- snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
+ snd_soc_dai_stream_valid(cpu_dai, cpu_capture))
capture = 1;
}
-
- capture = capture && cpu_capture->channels_min;
- playback = playback && cpu_playback->channels_min;
}
if (rtd->dai_link->playback_only) {
@@ -3017,7 +3049,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
out:
dev_info(rtd->card->dev, "%s <-> %s mapping ok\n",
(rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name,
- cpu_dai->name);
+ (rtd->num_cpus > 1) ? "multicpu" : rtd->cpu_dai->name);
return ret;
}
@@ -3050,33 +3082,17 @@ struct snd_pcm_substream *
}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream);
-/* get the BE runtime state */
-enum snd_soc_dpcm_state
- snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream)
-{
- return be->dpcm[stream].state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state);
-
-/* set the BE runtime state */
-void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be,
- int stream, enum snd_soc_dpcm_state state)
-{
- be->dpcm[stream].state = state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state);
-
-/*
- * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
- * are not running, paused or suspended for the specified stream direction.
- */
-int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
- struct snd_soc_pcm_runtime *be, int stream)
+static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be,
+ int stream,
+ const enum snd_soc_dpcm_state *states,
+ int num_states)
{
struct snd_soc_dpcm *dpcm;
int state;
int ret = 1;
unsigned long flags;
+ int i;
spin_lock_irqsave(&fe->card->dpcm_lock, flags);
for_each_dpcm_fe(be, stream, dpcm) {
@@ -3085,18 +3101,34 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
continue;
state = dpcm->fe->dpcm[stream].state;
- if (state == SND_SOC_DPCM_STATE_START ||
- state == SND_SOC_DPCM_STATE_PAUSED ||
- state == SND_SOC_DPCM_STATE_SUSPEND) {
- ret = 0;
- break;
+ for (i = 0; i < num_states; i++) {
+ if (state == states[i]) {
+ ret = 0;
+ break;
+ }
}
}
spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
- /* it's safe to free/stop this BE DAI */
+ /* it's safe to do this BE DAI */
return ret;
}
+
+/*
+ * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
+ * are not running, paused or suspended for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ const enum snd_soc_dpcm_state state[] = {
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_SUSPEND,
+ };
+
+ return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
+}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
/*
@@ -3106,168 +3138,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be, int stream)
{
- struct snd_soc_dpcm *dpcm;
- int state;
- int ret = 1;
- unsigned long flags;
-
- spin_lock_irqsave(&fe->card->dpcm_lock, flags);
- for_each_dpcm_fe(be, stream, dpcm) {
-
- if (dpcm->fe == fe)
- continue;
+ const enum snd_soc_dpcm_state state[] = {
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_SUSPEND,
+ SND_SOC_DPCM_STATE_PREPARE,
+ };
- state = dpcm->fe->dpcm[stream].state;
- if (state == SND_SOC_DPCM_STATE_START ||
- state == SND_SOC_DPCM_STATE_PAUSED ||
- state == SND_SOC_DPCM_STATE_SUSPEND ||
- state == SND_SOC_DPCM_STATE_PREPARE) {
- ret = 0;
- break;
- }
- }
- spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
-
- /* it's safe to change hw_params */
- return ret;
+ return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
-
-#ifdef CONFIG_DEBUG_FS
-static const char *dpcm_state_string(enum snd_soc_dpcm_state state)
-{
- switch (state) {
- case SND_SOC_DPCM_STATE_NEW:
- return "new";
- case SND_SOC_DPCM_STATE_OPEN:
- return "open";
- case SND_SOC_DPCM_STATE_HW_PARAMS:
- return "hw_params";
- case SND_SOC_DPCM_STATE_PREPARE:
- return "prepare";
- case SND_SOC_DPCM_STATE_START:
- return "start";
- case SND_SOC_DPCM_STATE_STOP:
- return "stop";
- case SND_SOC_DPCM_STATE_SUSPEND:
- return "suspend";
- case SND_SOC_DPCM_STATE_PAUSED:
- return "paused";
- case SND_SOC_DPCM_STATE_HW_FREE:
- return "hw_free";
- case SND_SOC_DPCM_STATE_CLOSE:
- return "close";
- }
-
- return "unknown";
-}
-
-static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
- int stream, char *buf, size_t size)
-{
- struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params;
- struct snd_soc_dpcm *dpcm;
- ssize_t offset = 0;
- unsigned long flags;
-
- /* FE state */
- offset += scnprintf(buf + offset, size - offset,
- "[%s - %s]\n", fe->dai_link->name,
- stream ? "Capture" : "Playback");
-
- offset += scnprintf(buf + offset, size - offset, "State: %s\n",
- dpcm_state_string(fe->dpcm[stream].state));
-
- if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
- (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
- offset += scnprintf(buf + offset, size - offset,
- "Hardware Params: "
- "Format = %s, Channels = %d, Rate = %d\n",
- snd_pcm_format_name(params_format(params)),
- params_channels(params),
- params_rate(params));
-
- /* BEs state */
- offset += scnprintf(buf + offset, size - offset, "Backends:\n");
-
- if (list_empty(&fe->dpcm[stream].be_clients)) {
- offset += scnprintf(buf + offset, size - offset,
- " No active DSP links\n");
- goto out;
- }
-
- spin_lock_irqsave(&fe->card->dpcm_lock, flags);
- for_each_dpcm_be(fe, stream, dpcm) {
- struct snd_soc_pcm_runtime *be = dpcm->be;
- params = &dpcm->hw_params;
-
- offset += scnprintf(buf + offset, size - offset,
- "- %s\n", be->dai_link->name);
-
- offset += scnprintf(buf + offset, size - offset,
- " State: %s\n",
- dpcm_state_string(be->dpcm[stream].state));
-
- if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
- (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
- offset += scnprintf(buf + offset, size - offset,
- " Hardware Params: "
- "Format = %s, Channels = %d, Rate = %d\n",
- snd_pcm_format_name(params_format(params)),
- params_channels(params),
- params_rate(params));
- }
- spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
-out:
- return offset;
-}
-
-static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
- size_t count, loff_t *ppos)
-{
- struct snd_soc_pcm_runtime *fe = file->private_data;
- ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
- char *buf;
-
- buf = kmalloc(out_count, GFP_KERNEL);
- if (!buf)
- return -ENOMEM;
-
- if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
- offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
- buf + offset, out_count - offset);
-
- if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE))
- offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
- buf + offset, out_count - offset);
-
- ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
-
- kfree(buf);
- return ret;
-}
-
-static const struct file_operations dpcm_state_fops = {
- .open = simple_open,
- .read = dpcm_state_read_file,
- .llseek = default_llseek,
-};
-
-void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
-{
- if (!rtd->dai_link)
- return;
-
- if (!rtd->dai_link->dynamic)
- return;
-
- if (!rtd->card->debugfs_card_root)
- return;
-
- rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name,
- rtd->card->debugfs_card_root);
-
- debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root,
- rtd, &dpcm_state_fops);
-}
-#endif
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 575da6aba807..1f81cd2d29cf 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -251,7 +251,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg,
{
int ret = 0;
- if (tplg->comp && tplg->ops && tplg->ops->vendor_load)
+ if (tplg->ops && tplg->ops->vendor_load)
ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr);
else {
dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n",
@@ -283,7 +283,7 @@ static int soc_tplg_vendor_load(struct soc_tplg *tplg,
static int soc_tplg_widget_load(struct soc_tplg *tplg,
struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
{
- if (tplg->comp && tplg->ops && tplg->ops->widget_load)
+ if (tplg->ops && tplg->ops->widget_load)
return tplg->ops->widget_load(tplg->comp, tplg->index, w,
tplg_w);
@@ -295,7 +295,7 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg,
static int soc_tplg_widget_ready(struct soc_tplg *tplg,
struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
{
- if (tplg->comp && tplg->ops && tplg->ops->widget_ready)
+ if (tplg->ops && tplg->ops->widget_ready)
return tplg->ops->widget_ready(tplg->comp, tplg->index, w,
tplg_w);
@@ -307,7 +307,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg,
struct snd_soc_dai_driver *dai_drv,
struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
{
- if (tplg->comp && tplg->ops && tplg->ops->dai_load)
+ if (tplg->ops && tplg->ops->dai_load)
return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv,
pcm, dai);
@@ -318,7 +318,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg,
static int soc_tplg_dai_link_load(struct soc_tplg *tplg,
struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg)
{
- if (tplg->comp && tplg->ops && tplg->ops->link_load)
+ if (tplg->ops && tplg->ops->link_load)
return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg);
return 0;
@@ -327,7 +327,7 @@ static int soc_tplg_dai_link_load(struct soc_tplg *tplg,
/* tell the component driver that all firmware has been loaded in this request */
static void soc_tplg_complete(struct soc_tplg *tplg)
{
- if (tplg->comp && tplg->ops && tplg->ops->complete)
+ if (tplg->ops && tplg->ops->complete)
tplg->ops->complete(tplg->comp);
}
@@ -684,7 +684,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_bind_event);
static int soc_tplg_init_kcontrol(struct soc_tplg *tplg,
struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr)
{
- if (tplg->comp && tplg->ops && tplg->ops->control_load)
+ if (tplg->ops && tplg->ops->control_load)
return tplg->ops->control_load(tplg->comp, tplg->index, k,
hdr);
@@ -1174,7 +1174,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
static int soc_tplg_add_route(struct soc_tplg *tplg,
struct snd_soc_dapm_route *route)
{
- if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load)
+ if (tplg->ops && tplg->ops->dapm_route_load)
return tplg->ops->dapm_route_load(tplg->comp, tplg->index,
route);
@@ -2564,7 +2564,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
}
/* pass control to component driver for optional further init */
- if (tplg->comp && tplg->ops && tplg->ops->manifest)
+ if (tplg->ops && tplg->ops->manifest)
ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
if (!abi_match) /* free the duplicated one */
@@ -2736,6 +2736,10 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
struct soc_tplg tplg;
int ret;
+ /* component needs to exist to keep and reference data while parsing */
+ if (!comp)
+ return -EINVAL;
+
/* setup parsing context */
memset(&tplg, 0, sizeof(tplg));
tplg.fw = fw;
@@ -2774,7 +2778,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm,
{
struct snd_soc_dapm_widget *w, *next_w;
- list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) {
+ for_each_card_widgets_safe(dapm->card, w, next_w) {
/* make sure we are a widget with correct context */
if (w->dobj.type != SND_SOC_DOBJ_WIDGET || w->dapm != dapm)
diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig
index 827b0ec92522..4dda4b62509f 100644
--- a/sound/soc/sof/Kconfig
+++ b/sound/soc/sof/Kconfig
@@ -41,6 +41,15 @@ config SND_SOC_SOF_OF
required to enable i.MX8 devices.
Say Y if you need this option. If unsure select "N".
+config SND_SOC_SOF_DEBUG_PROBES
+ bool "SOF enable data probing"
+ select SND_SOC_COMPRESS
+ help
+ This option enables the data probing feature that can be used to
+ gather data directly from specific points of the audio pipeline.
+ Say Y if you want to enable probes.
+ If unsure, select "N".
+
config SND_SOC_SOF_DEVELOPER_SUPPORT
bool "SOF developer options support"
depends on EXPERT
diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile
index 0a8bc72c28a5..8eca2f85c90e 100644
--- a/sound/soc/sof/Makefile
+++ b/sound/soc/sof/Makefile
@@ -2,6 +2,7 @@
snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\
control.o trace.o utils.o sof-audio.o
+snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o compress.o
snd-sof-pci-objs := sof-pci-dev.o
snd-sof-acpi-objs := sof-acpi-dev.o
diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c
new file mode 100644
index 000000000000..7354dc6a49cf
--- /dev/null
+++ b/sound/soc/sof/compress.c
@@ -0,0 +1,146 @@
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license. When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <cezary.rojewski@intel.com>
+//
+
+#include <sound/soc.h>
+#include "compress.h"
+#include "ops.h"
+#include "probe.h"
+
+struct snd_compr_ops sof_probe_compressed_ops = {
+ .copy = sof_probe_compr_copy,
+};
+EXPORT_SYMBOL(sof_probe_compressed_ops);
+
+int sof_probe_compr_open(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+ int ret;
+
+ ret = snd_sof_probe_compr_assign(sdev, cstream, dai);
+ if (ret < 0) {
+ dev_err(dai->dev, "Failed to assign probe stream: %d\n", ret);
+ return ret;
+ }
+
+ sdev->extractor_stream_tag = ret;
+ return 0;
+}
+EXPORT_SYMBOL(sof_probe_compr_open);
+
+int sof_probe_compr_free(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+ struct sof_probe_point_desc *desc;
+ size_t num_desc;
+ int i, ret;
+
+ /* disconnect all probe points */
+ ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc);
+ if (ret < 0) {
+ dev_err(dai->dev, "Failed to get probe points: %d\n", ret);
+ goto exit;
+ }
+
+ for (i = 0; i < num_desc; i++)
+ sof_ipc_probe_points_remove(sdev, &desc[i].buffer_id, 1);
+ kfree(desc);
+
+exit:
+ ret = sof_ipc_probe_deinit(sdev);
+ if (ret < 0)
+ dev_err(dai->dev, "Failed to deinit probe: %d\n", ret);
+
+ sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID;
+ snd_compr_free_pages(cstream);
+
+ return snd_sof_probe_compr_free(sdev, cstream, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_free);
+
+int sof_probe_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_compr_runtime *rtd = cstream->runtime;
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+ int ret;
+
+ cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG;
+ cstream->dma_buffer.dev.dev = sdev->dev;
+ ret = snd_compr_malloc_pages(cstream, rtd->buffer_size);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_sof_probe_compr_set_params(sdev, cstream, params, dai);
+ if (ret < 0)
+ return ret;
+
+ ret = sof_ipc_probe_init(sdev, sdev->extractor_stream_tag,
+ rtd->dma_bytes);
+ if (ret < 0) {
+ dev_err(dai->dev, "Failed to init probe: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL(sof_probe_compr_set_params);
+
+int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+
+ return snd_sof_probe_compr_trigger(sdev, cstream, cmd, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_trigger);
+
+int sof_probe_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+
+ return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_pointer);
+
+int sof_probe_compr_copy(struct snd_compr_stream *cstream,
+ char __user *buf, size_t count)
+{
+ struct snd_compr_runtime *rtd = cstream->runtime;
+ unsigned int offset, n;
+ void *ptr;
+ int ret;
+
+ if (count > rtd->buffer_size)
+ count = rtd->buffer_size;
+
+ div_u64_rem(rtd->total_bytes_transferred, rtd->buffer_size, &offset);
+ ptr = rtd->dma_area + offset;
+ n = rtd->buffer_size - offset;
+
+ if (count < n) {
+ ret = copy_to_user(buf, ptr, count);
+ } else {
+ ret = copy_to_user(buf, ptr, n);
+ ret += copy_to_user(buf + n, rtd->dma_area, count - n);
+ }
+
+ if (ret)
+ return count - ret;
+ return count;
+}
+EXPORT_SYMBOL(sof_probe_compr_copy);
diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h
new file mode 100644
index 000000000000..800f163603e1
--- /dev/null
+++ b/sound/soc/sof/compress.h
@@ -0,0 +1,31 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/*
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+ *
+ * Author: Cezary Rojewski <cezary.rojewski@intel.com>
+ */
+
+#ifndef __SOF_COMPRESS_H
+#define __SOF_COMPRESS_H
+
+#include <sound/compress_driver.h>
+
+extern struct snd_compr_ops sof_probe_compressed_ops;
+
+int sof_probe_compr_open(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai);
+int sof_probe_compr_free(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai);
+int sof_probe_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params, struct snd_soc_dai *dai);
+int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai);
+int sof_probe_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai);
+int sof_probe_compr_copy(struct snd_compr_stream *cstream,
+ char __user *buf, size_t count);
+
+#endif
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 34cefbaf2d2a..91acfae7935c 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -14,6 +14,9 @@
#include <sound/sof.h>
#include "sof-priv.h"
#include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "probe.h"
+#endif
/* see SOF_DBG_ flags */
int sof_core_debug;
@@ -286,12 +289,15 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data)
/* initialize sof device */
sdev->dev = dev;
- /* initialize default D0 sub-state */
- sdev->d0_substate = SOF_DSP_D0I0;
+ /* initialize default DSP power state */
+ sdev->dsp_power_state.state = SOF_DSP_PM_D0;
sdev->pdata = plat_data;
sdev->first_boot = true;
sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID;
+#endif
dev_set_drvdata(dev, sdev);
/* check all mandatory ops */
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index d2b3b99d3a20..b5c0d6cf72cc 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -17,6 +17,221 @@
#include "sof-priv.h"
#include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "probe.h"
+
+/**
+ * strsplit_u32 - Split string into sequence of u32 tokens
+ * @buf: String to split into tokens.
+ * @delim: String containing delimiter characters.
+ * @tkns: Returned u32 sequence pointer.
+ * @num_tkns: Returned number of tokens obtained.
+ */
+static int
+strsplit_u32(char **buf, const char *delim, u32 **tkns, size_t *num_tkns)
+{
+ char *s;
+ u32 *data, *tmp;
+ size_t count = 0;
+ size_t cap = 32;
+ int ret = 0;
+
+ *tkns = NULL;
+ *num_tkns = 0;
+ data = kcalloc(cap, sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ while ((s = strsep(buf, delim)) != NULL) {
+ ret = kstrtouint(s, 0, data + count);
+ if (ret)
+ goto exit;
+ if (++count >= cap) {
+ cap *= 2;
+ tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL);
+ if (!tmp) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+ data = tmp;
+ }
+ }
+
+ if (!count)
+ goto exit;
+ *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL);
+ if (*tkns == NULL) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+ *num_tkns = count;
+
+exit:
+ kfree(data);
+ return ret;
+}
+
+static int tokenize_input(const char __user *from, size_t count,
+ loff_t *ppos, u32 **tkns, size_t *num_tkns)
+{
+ char *buf;
+ int ret;
+
+ buf = kmalloc(count + 1, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ ret = simple_write_to_buffer(buf, count, ppos, from, count);
+ if (ret != count) {
+ ret = ret >= 0 ? -EIO : ret;
+ goto exit;
+ }
+
+ buf[count] = '\0';
+ ret = strsplit_u32((char **)&buf, ",", tkns, num_tkns);
+exit:
+ kfree(buf);
+ return ret;
+}
+
+static ssize_t probe_points_read(struct file *file,
+ char __user *to, size_t count, loff_t *ppos)
+{
+ struct snd_sof_dfsentry *dfse = file->private_data;
+ struct snd_sof_dev *sdev = dfse->sdev;
+ struct sof_probe_point_desc *desc;
+ size_t num_desc, len = 0;
+ char *buf;
+ int i, ret;
+
+ if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+ dev_warn(sdev->dev, "no extractor stream running\n");
+ return -ENOENT;
+ }
+
+ buf = kzalloc(PAGE_SIZE, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc);
+ if (ret < 0)
+ goto exit;
+
+ for (i = 0; i < num_desc; i++) {
+ ret = snprintf(buf + len, PAGE_SIZE - len,
+ "Id: %#010x Purpose: %d Node id: %#x\n",
+ desc[i].buffer_id, desc[i].purpose, desc[i].stream_tag);
+ if (ret < 0)
+ goto free_desc;
+ len += ret;
+ }
+
+ ret = simple_read_from_buffer(to, count, ppos, buf, len);
+free_desc:
+ kfree(desc);
+exit:
+ kfree(buf);
+ return ret;
+}
+
+static ssize_t probe_points_write(struct file *file,
+ const char __user *from, size_t count, loff_t *ppos)
+{
+ struct snd_sof_dfsentry *dfse = file->private_data;
+ struct snd_sof_dev *sdev = dfse->sdev;
+ struct sof_probe_point_desc *desc;
+ size_t num_tkns, bytes;
+ u32 *tkns;
+ int ret;
+
+ if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+ dev_warn(sdev->dev, "no extractor stream running\n");
+ return -ENOENT;
+ }
+
+ ret = tokenize_input(from, count, ppos, &tkns, &num_tkns);
+ if (ret < 0)
+ return ret;
+ bytes = sizeof(*tkns) * num_tkns;
+ if (!num_tkns || (bytes % sizeof(*desc))) {
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ desc = (struct sof_probe_point_desc *)tkns;
+ ret = sof_ipc_probe_points_add(sdev,
+ desc, bytes / sizeof(*desc));
+ if (!ret)
+ ret = count;
+exit:
+ kfree(tkns);
+ return ret;
+}
+
+static const struct file_operations probe_points_fops = {
+ .open = simple_open,
+ .read = probe_points_read,
+ .write = probe_points_write,
+ .llseek = default_llseek,
+};
+
+static ssize_t probe_points_remove_write(struct file *file,
+ const char __user *from, size_t count, loff_t *ppos)
+{
+ struct snd_sof_dfsentry *dfse = file->private_data;
+ struct snd_sof_dev *sdev = dfse->sdev;
+ size_t num_tkns;
+ u32 *tkns;
+ int ret;
+
+ if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+ dev_warn(sdev->dev, "no extractor stream running\n");
+ return -ENOENT;
+ }
+
+ ret = tokenize_input(from, count, ppos, &tkns, &num_tkns);
+ if (ret < 0)
+ return ret;
+ if (!num_tkns) {
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ ret = sof_ipc_probe_points_remove(sdev, tkns, num_tkns);
+ if (!ret)
+ ret = count;
+exit:
+ kfree(tkns);
+ return ret;
+}
+
+static const struct file_operations probe_points_remove_fops = {
+ .open = simple_open,
+ .write = probe_points_remove_write,
+ .llseek = default_llseek,
+};
+
+static int snd_sof_debugfs_probe_item(struct snd_sof_dev *sdev,
+ const char *name, mode_t mode,
+ const struct file_operations *fops)
+{
+ struct snd_sof_dfsentry *dfse;
+
+ dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL);
+ if (!dfse)
+ return -ENOMEM;
+
+ dfse->type = SOF_DFSENTRY_TYPE_BUF;
+ dfse->sdev = sdev;
+
+ debugfs_create_file(name, mode, sdev->debugfs_root, dfse, fops);
+ /* add to dfsentry list */
+ list_add(&dfse->list, &sdev->dfsentry_list);
+
+ return 0;
+}
+#endif
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
#define MAX_IPC_FLOOD_DURATION_MS 1000
#define MAX_IPC_FLOOD_COUNT 10000
@@ -436,6 +651,17 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev)
return err;
}
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ err = snd_sof_debugfs_probe_item(sdev, "probe_points",
+ 0644, &probe_points_fops);
+ if (err < 0)
+ return err;
+ err = snd_sof_debugfs_probe_item(sdev, "probe_points_remove",
+ 0200, &probe_points_remove_fops);
+ if (err < 0)
+ return err;
+#endif
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
/* create read-write ipc_flood_count debugfs entry */
err = snd_sof_debugfs_buf_item(sdev, NULL, 0,
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index b2556f5e2871..b692752b2178 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -138,7 +138,7 @@ static int imx8_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg)
/*
* DSP control.
*/
-static int imx8_run(struct snd_sof_dev *sdev)
+static int imx8x_run(struct snd_sof_dev *sdev)
{
struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private;
int ret;
@@ -178,6 +178,24 @@ static int imx8_run(struct snd_sof_dev *sdev)
return 0;
}
+static int imx8_run(struct snd_sof_dev *sdev)
+{
+ struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private;
+ int ret;
+
+ ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP,
+ IMX_SC_C_OFS_SEL, 0);
+ if (ret < 0) {
+ dev_err(sdev->dev, "Error system address offset source select\n");
+ return ret;
+ }
+
+ imx_sc_pm_cpu_start(dsp_priv->sc_ipc, IMX_SC_R_DSP, true,
+ RESET_VECTOR_VADDR);
+
+ return 0;
+}
+
static int imx8_probe(struct snd_sof_dev *sdev)
{
struct platform_device *pdev =
@@ -360,7 +378,7 @@ static struct snd_soc_dai_driver imx8_dai[] = {
},
};
-/* i.MX8 ops */
+/* i.MX8 ops */
struct snd_sof_dsp_ops sof_imx8_ops = {
/* probe and remove */
.probe = imx8_probe,
@@ -390,6 +408,39 @@ struct snd_sof_dsp_ops sof_imx8_ops = {
/* DAI drivers */
.drv = imx8_dai,
.num_drv = 1, /* we have only 1 ESAI interface on i.MX8 */
+};
+EXPORT_SYMBOL(sof_imx8_ops);
+
+/* i.MX8X ops */
+struct snd_sof_dsp_ops sof_imx8x_ops = {
+ /* probe and remove */
+ .probe = imx8_probe,
+ .remove = imx8_remove,
+ /* DSP core boot */
+ .run = imx8x_run,
+
+ /* Block IO */
+ .block_read = sof_block_read,
+ .block_write = sof_block_write,
+
+ /* ipc */
+ .send_msg = imx8_send_msg,
+ .fw_ready = sof_fw_ready,
+ .get_mailbox_offset = imx8_get_mailbox_offset,
+ .get_window_offset = imx8_get_window_offset,
+
+ .ipc_msg_data = imx8_ipc_msg_data,
+ .ipc_pcm_params = imx8_ipc_pcm_params,
+
+ /* module loading */
+ .load_module = snd_sof_parse_module_memcpy,
+ .get_bar_index = imx8_get_bar_index,
+ /* firmware loading */
+ .load_firmware = snd_sof_load_firmware_memcpy,
+
+ /* DAI drivers */
+ .drv = imx8_dai,
+ .num_drv = 1, /* we have only 1 ESAI interface on i.MX8 */
/* ALSA HW info flags */
.hw_info = SNDRV_PCM_INFO_MMAP |
@@ -398,6 +449,6 @@ struct snd_sof_dsp_ops sof_imx8_ops = {
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_NO_PERIOD_WAKEUP
};
-EXPORT_SYMBOL(sof_imx8_ops);
+EXPORT_SYMBOL(sof_imx8x_ops);
MODULE_LICENSE("Dual BSD/GPL");
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 56a837d2cb95..c9a2bee4b55c 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -305,6 +305,15 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC
Say Y if you want to enable HDAudio codecs with SOF.
If unsure select "N".
+config SND_SOC_SOF_HDA_PROBES
+ bool "SOF enable probes over HDA"
+ depends on SND_SOC_SOF_DEBUG_PROBES
+ help
+ This option enables the data probing for Intel(R).
+ Intel(R) Skylake and newer platforms.
+ Say Y if you want to enable probes.
+ If unsure, select "N".
+
config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1
bool "SOF enable DMI Link L1"
help
@@ -315,17 +324,6 @@ config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1
Say Y if you want to enable DMI Link L1
If unsure, select "N".
-config SND_SOC_SOF_HDA_COMMON_HDMI_CODEC
- bool "SOF common HDA HDMI codec driver"
- depends on SND_SOC_SOF_HDA_LINK
- depends on SND_HDA_CODEC_HDMI
- default SND_HDA_CODEC_HDMI
- help
- This adds support for HDMI audio by using the common HDA
- HDMI/DisplayPort codec driver.
- Say Y if you want to use the common codec driver with SOF.
- If unsure select "Y".
-
endif ## SND_SOC_SOF_HDA_COMMON
config SND_SOC_SOF_HDA_LINK_BASELINE
diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile
index b8f58e006e29..cee02a2e00f4 100644
--- a/sound/soc/sof/intel/Makefile
+++ b/sound/soc/sof/intel/Makefile
@@ -9,6 +9,7 @@ snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \
hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \
hda-dai.o hda-bus.o \
apl.o cnl.o
+snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-compress.o
snd-sof-intel-hda-objs := hda-codec.o
diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c
index 2483b15699e7..02218d22e51f 100644
--- a/sound/soc/sof/intel/apl.c
+++ b/sound/soc/sof/intel/apl.c
@@ -73,6 +73,15 @@ const struct snd_sof_dsp_ops sof_apl_ops = {
.pcm_trigger = hda_dsp_pcm_trigger,
.pcm_pointer = hda_dsp_pcm_pointer,
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+ /* probe callbacks */
+ .probe_assign = hda_probe_compr_assign,
+ .probe_free = hda_probe_compr_free,
+ .probe_set_params = hda_probe_compr_set_params,
+ .probe_trigger = hda_probe_compr_trigger,
+ .probe_pointer = hda_probe_compr_pointer,
+#endif
+
/* firmware loading */
.load_firmware = snd_sof_load_firmware_raw,
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index 9e2d8afe0535..e427d00eca71 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -65,11 +65,6 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context)
hda_dsp_ipc_get_reply(sdev);
snd_sof_ipc_reply(sdev, msg);
- if (sdev->code_loading) {
- sdev->code_loading = 0;
- wake_up(&sdev->waitq);
- }
-
cnl_ipc_dsp_done(sdev);
spin_unlock_irq(&sdev->ipc_lock);
@@ -171,23 +166,48 @@ static bool cnl_compact_ipc_compress(struct snd_sof_ipc_msg *msg,
static int cnl_ipc_send_msg(struct snd_sof_dev *sdev,
struct snd_sof_ipc_msg *msg)
{
+ struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata;
+ struct sof_ipc_cmd_hdr *hdr;
u32 dr = 0;
u32 dd = 0;
+ /*
+ * Currently the only compact IPC supported is the PM_GATE
+ * IPC which is used for transitioning the DSP between the
+ * D0I0 and D0I3 states. And these are sent only during the
+ * set_power_state() op. Therefore, there will never be a case
+ * that a compact IPC results in the DSP exiting D0I3 without
+ * the host and FW being in sync.
+ */
if (cnl_compact_ipc_compress(msg, &dr, &dd)) {
/* send the message via IPC registers */
snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDD,
dd);
snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR,
CNL_DSP_REG_HIPCIDR_BUSY | dr);
- } else {
- /* send the message via mailbox */
- sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data,
- msg->msg_size);
- snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR,
- CNL_DSP_REG_HIPCIDR_BUSY);
+ return 0;
}
+ /* send the message via mailbox */
+ sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data,
+ msg->msg_size);
+ snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR,
+ CNL_DSP_REG_HIPCIDR_BUSY);
+
+ hdr = msg->msg_data;
+
+ /*
+ * Use mod_delayed_work() to schedule the delayed work
+ * to avoid scheduling multiple workqueue items when
+ * IPCs are sent at a high-rate. mod_delayed_work()
+ * modifies the timer if the work is pending.
+ * Also, a new delayed work should not be queued after the
+ * the CTX_SAVE IPC, which is sent before the DSP enters D3.
+ */
+ if (hdr->cmd != (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE))
+ mod_delayed_work(system_wq, &hdev->d0i3_work,
+ msecs_to_jiffies(SOF_HDA_D0I3_WORK_DELAY_MS));
+
return 0;
}
@@ -259,6 +279,15 @@ const struct snd_sof_dsp_ops sof_cnl_ops = {
.pcm_trigger = hda_dsp_pcm_trigger,
.pcm_pointer = hda_dsp_pcm_pointer,
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+ /* probe callbacks */
+ .probe_assign = hda_probe_compr_assign,
+ .probe_free = hda_probe_compr_free,
+ .probe_set_params = hda_probe_compr_set_params,
+ .probe_trigger = hda_probe_compr_trigger,
+ .probe_pointer = hda_probe_compr_pointer,
+#endif
+
/* firmware loading */
.load_firmware = snd_sof_load_firmware_raw,
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index ff45075ef720..3041fbbb010a 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -113,8 +113,14 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
if (ret < 0)
return ret;
- if ((resp & 0xFFFF0000) == IDISP_VID_INTEL)
+ if ((resp & 0xFFFF0000) == IDISP_VID_INTEL) {
+ if (!hdev->bus->audio_component) {
+ dev_dbg(sdev->dev,
+ "iDisp hw present but no driver\n");
+ return -ENOENT;
+ }
hda_priv->need_display_power = true;
+ }
/*
* if common HDMI codec driver is not used, codec load
@@ -203,6 +209,9 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev)
struct hdac_bus *bus = sof_to_bus(sdev);
int ret;
+ if (!bus->audio_component)
+ return 0;
+
/* power down unconditionally */
snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false);
diff --git a/sound/soc/sof/intel/hda-compress.c b/sound/soc/sof/intel/hda-compress.c
new file mode 100644
index 000000000000..38a1ebec8478
--- /dev/null
+++ b/sound/soc/sof/intel/hda-compress.c
@@ -0,0 +1,114 @@
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license. When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <cezary.rojewski@intel.com>
+//
+
+#include <sound/hdaudio_ext.h>
+#include <sound/soc.h>
+#include "../sof-priv.h"
+#include "hda.h"
+
+static inline struct hdac_ext_stream *
+hda_compr_get_stream(struct snd_compr_stream *cstream)
+{
+ return cstream->runtime->private_data;
+}
+
+int hda_probe_compr_assign(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai)
+{
+ struct hdac_ext_stream *stream;
+
+ stream = hda_dsp_stream_get(sdev, cstream->direction);
+ if (!stream)
+ return -EBUSY;
+
+ hdac_stream(stream)->curr_pos = 0;
+ hdac_stream(stream)->cstream = cstream;
+ cstream->runtime->private_data = stream;
+
+ return hdac_stream(stream)->stream_tag;
+}
+
+int hda_probe_compr_free(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai)
+{
+ struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+ int ret;
+
+ ret = hda_dsp_stream_put(sdev, cstream->direction,
+ hdac_stream(stream)->stream_tag);
+ if (ret < 0) {
+ dev_dbg(sdev->dev, "stream put failed: %d\n", ret);
+ return ret;
+ }
+
+ hdac_stream(stream)->cstream = NULL;
+ cstream->runtime->private_data = NULL;
+
+ return 0;
+}
+
+int hda_probe_compr_set_params(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+ struct hdac_stream *hstream = hdac_stream(stream);
+ struct snd_dma_buffer *dmab;
+ u32 bits, rate;
+ int bps, ret;
+
+ dmab = cstream->runtime->dma_buffer_p;
+ /* compr params do not store bit depth, default to S32_LE */
+ bps = snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32_LE);
+ if (bps < 0)
+ return bps;
+ bits = hda_dsp_get_bits(sdev, bps);
+ rate = hda_dsp_get_mult_div(sdev, params->codec.sample_rate);
+
+ hstream->format_val = rate | bits | (params->codec.ch_out - 1);
+ hstream->bufsize = cstream->runtime->buffer_size;
+ hstream->period_bytes = cstream->runtime->fragment_size;
+ hstream->no_period_wakeup = 0;
+
+ ret = hda_dsp_stream_hw_params(sdev, stream, dmab, NULL);
+ if (ret < 0) {
+ dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+int hda_probe_compr_trigger(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+
+ return hda_dsp_stream_trigger(sdev, stream, cmd);
+}
+
+int hda_probe_compr_pointer(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp,
+ struct snd_soc_dai *dai)
+{
+ struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+ struct snd_soc_pcm_stream *pstream;
+
+ pstream = &dai->driver->capture;
+ tstamp->copied_total = hdac_stream(stream)->curr_pos;
+ tstamp->sampling_rate = snd_pcm_rate_bit_to_rate(pstream->rates);
+
+ return 0;
+}
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 871b71a15a63..6288b2f99540 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -18,6 +18,7 @@
#include <linux/module.h>
#include <sound/hdaudio_ext.h>
#include <sound/hda_register.h>
+#include <sound/hda_component.h>
#include "../ops.h"
#include "hda.h"
@@ -64,15 +65,32 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev)
struct hdac_bus *bus = sof_to_bus(sdev);
u32 cap, offset, feature;
int count = 0;
+ int ret;
+
+ /*
+ * On some devices, one reset cycle is necessary before reading
+ * capabilities
+ */
+ ret = hda_dsp_ctrl_link_reset(sdev, true);
+ if (ret < 0)
+ return ret;
+ ret = hda_dsp_ctrl_link_reset(sdev, false);
+ if (ret < 0)
+ return ret;
offset = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_LLCH);
do {
- cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset);
-
dev_dbg(sdev->dev, "checking for capabilities at offset 0x%x\n",
offset & SOF_HDA_CAP_NEXT_MASK);
+ cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset);
+
+ if (cap == -1) {
+ dev_dbg(bus->dev, "Invalid capability reg read\n");
+ break;
+ }
+
feature = (cap & SOF_HDA_CAP_ID_MASK) >> SOF_HDA_CAP_ID_OFF;
switch (feature) {
@@ -105,8 +123,8 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev)
bus->mlcap = bus->remap_addr + offset;
break;
default:
- dev_vdbg(sdev->dev, "found capability %d at 0x%x\n",
- feature, offset);
+ dev_dbg(sdev->dev, "found capability %d at 0x%x\n",
+ feature, offset);
break;
}
@@ -176,6 +194,9 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
if (bus->chip_init)
return 0;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+ snd_hdac_set_codec_wakeup(bus, true);
+#endif
hda_dsp_ctrl_misc_clock_gating(sdev, false);
if (full_reset) {
@@ -183,7 +204,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
ret = hda_dsp_ctrl_link_reset(sdev, true);
if (ret < 0) {
dev_err(sdev->dev, "error: failed to reset HDA controller\n");
- return ret;
+ goto err;
}
usleep_range(500, 1000);
@@ -192,7 +213,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
ret = hda_dsp_ctrl_link_reset(sdev, false);
if (ret < 0) {
dev_err(sdev->dev, "error: failed to exit HDA controller reset\n");
- return ret;
+ goto err;
}
usleep_range(1000, 1200);
@@ -202,7 +223,8 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
/* check to see if controller is ready */
if (!snd_hdac_chip_readb(bus, GCTL)) {
dev_dbg(bus->dev, "controller not ready!\n");
- return -EBUSY;
+ ret = -EBUSY;
+ goto err;
}
/* Accept unsolicited responses */
@@ -268,7 +290,11 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset)
bus->chip_init = true;
+err:
hda_dsp_ctrl_misc_clock_gating(sdev, true);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+ snd_hdac_set_codec_wakeup(bus, false);
+#endif
return ret;
}
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 9c6e3f990ee3..833dc303b394 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -204,7 +204,7 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream,
struct hdac_bus *bus = hstream->bus;
struct hdac_ext_stream *link_dev;
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct sof_intel_hda_stream *hda_stream;
struct hda_pipe_params p_params = {0};
struct hdac_ext_link *link;
@@ -293,7 +293,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
bus = hstream->bus;
rtd = snd_pcm_substream_chip(substream);
- link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name);
+ link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name);
if (!link)
return -EINVAL;
@@ -374,7 +374,7 @@ static int hda_link_hw_free(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name);
+ link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name);
if (!link)
return -EINVAL;
@@ -399,6 +399,19 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = {
.trigger = hda_link_pcm_trigger,
.prepare = hda_link_pcm_prepare,
};
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+#include "../compress.h"
+
+static struct snd_soc_cdai_ops sof_probe_compr_ops = {
+ .startup = sof_probe_compr_open,
+ .shutdown = sof_probe_compr_free,
+ .set_params = sof_probe_compr_set_params,
+ .trigger = sof_probe_compr_trigger,
+ .pointer = sof_probe_compr_pointer,
+};
+
+#endif
#endif
/*
@@ -409,56 +422,167 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = {
struct snd_soc_dai_driver skl_dai[] = {
{
.name = "SSP0 Pin",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "SSP1 Pin",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "SSP2 Pin",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "SSP3 Pin",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "SSP4 Pin",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "SSP5 Pin",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "DMIC01 Pin",
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 4,
+ },
},
{
.name = "DMIC16k Pin",
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 4,
+ },
},
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
{
.name = "iDisp1 Pin",
.ops = &hda_link_dai_ops,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "iDisp2 Pin",
.ops = &hda_link_dai_ops,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "iDisp3 Pin",
.ops = &hda_link_dai_ops,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "iDisp4 Pin",
.ops = &hda_link_dai_ops,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
{
.name = "Analog CPU DAI",
.ops = &hda_link_dai_ops,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 16,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 16,
+ },
},
{
.name = "Digital CPU DAI",
.ops = &hda_link_dai_ops,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 16,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 16,
+ },
},
{
.name = "Alt Analog CPU DAI",
.ops = &hda_link_dai_ops,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 16,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 16,
+ },
+},
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+{
+ .name = "Probe Extraction CPU DAI",
+ .compress_new = snd_soc_new_compress,
+ .cops = &sof_probe_compr_ops,
+ .capture = {
+ .stream_name = "Probe Extraction",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ },
},
#endif
+#endif
};
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index 0848b79967a9..99087b6afb67 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -15,12 +15,21 @@
* Hardware interface for generic Intel audio DSP HDA IP
*/
+#include <linux/module.h>
#include <sound/hdaudio_ext.h>
#include <sound/hda_register.h>
+#include "../sof-audio.h"
#include "../ops.h"
#include "hda.h"
#include "hda-ipc.h"
+static bool hda_enable_trace_D0I3_S0;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG)
+module_param_named(enable_trace_D0I3_S0, hda_enable_trace_D0I3_S0, bool, 0444);
+MODULE_PARM_DESC(enable_trace_D0I3_S0,
+ "SOF HDA enable trace when the DSP is in D0I3 in S0");
+#endif
+
/*
* DSP Core control.
*/
@@ -334,17 +343,15 @@ static int hda_dsp_send_pm_gate_ipc(struct snd_sof_dev *sdev, u32 flags)
pm_gate.flags = flags;
/* send pm_gate ipc to dsp */
- return sof_ipc_tx_message(sdev->ipc, pm_gate.hdr.cmd, &pm_gate,
- sizeof(pm_gate), &reply, sizeof(reply));
+ return sof_ipc_tx_message_no_pm(sdev->ipc, pm_gate.hdr.cmd,
+ &pm_gate, sizeof(pm_gate), &reply,
+ sizeof(reply));
}
-int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
- enum sof_d0_substate d0_substate)
+static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
{
struct hdac_bus *bus = sof_to_bus(sdev);
- u32 flags;
int ret;
- u8 value;
/* Write to D0I3C after Command-In-Progress bit is cleared */
ret = hda_dsp_wait_d0i3c_done(sdev);
@@ -354,7 +361,6 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
}
/* Update D0I3C register */
- value = d0_substate == SOF_DSP_D0I3 ? SOF_HDA_VS_D0I3C_I3 : 0;
snd_hdac_chip_updateb(bus, VS_D0I3C, SOF_HDA_VS_D0I3C_I3, value);
/* Wait for cmd in progress to be cleared before exiting the function */
@@ -367,20 +373,218 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
dev_vdbg(bus->dev, "D0I3C updated, register = 0x%x\n",
snd_hdac_chip_readb(bus, VS_D0I3C));
- if (d0_substate == SOF_DSP_D0I0)
- flags = HDA_PM_PPG;/* prevent power gating in D0 */
- else
- flags = HDA_PM_NO_DMA_TRACE;/* disable DMA trace in D0I3*/
+ return 0;
+}
- /* sending pm_gate IPC */
- ret = hda_dsp_send_pm_gate_ipc(sdev, flags);
+static int hda_dsp_set_D0_state(struct snd_sof_dev *sdev,
+ const struct sof_dsp_power_state *target_state)
+{
+ u32 flags = 0;
+ int ret;
+ u8 value = 0;
+
+ /*
+ * Sanity check for illegal state transitions
+ * The only allowed transitions are:
+ * 1. D3 -> D0I0
+ * 2. D0I0 -> D0I3
+ * 3. D0I3 -> D0I0
+ */
+ switch (sdev->dsp_power_state.state) {
+ case SOF_DSP_PM_D0:
+ /* Follow the sequence below for D0 substate transitions */
+ break;
+ case SOF_DSP_PM_D3:
+ /* Follow regular flow for D3 -> D0 transition */
+ return 0;
+ default:
+ dev_err(sdev->dev, "error: transition from %d to %d not allowed\n",
+ sdev->dsp_power_state.state, target_state->state);
+ return -EINVAL;
+ }
+
+ /* Set flags and register value for D0 target substate */
+ if (target_state->substate == SOF_HDA_DSP_PM_D0I3) {
+ value = SOF_HDA_VS_D0I3C_I3;
+
+ /*
+ * Trace DMA is disabled by default when the DSP enters D0I3.
+ * But it can be kept enabled when the DSP enters D0I3 while the
+ * system is in S0 for debug.
+ */
+ if (hda_enable_trace_D0I3_S0 &&
+ sdev->system_suspend_target != SOF_SUSPEND_NONE)
+ flags = HDA_PM_NO_DMA_TRACE;
+ } else {
+ /* prevent power gating in D0I0 */
+ flags = HDA_PM_PPG;
+ }
+
+ /* update D0I3C register */
+ ret = hda_dsp_update_d0i3c_register(sdev, value);
if (ret < 0)
+ return ret;
+
+ /*
+ * Notify the DSP of the state change.
+ * If this IPC fails, revert the D0I3C register update in order
+ * to prevent partial state change.
+ */
+ ret = hda_dsp_send_pm_gate_ipc(sdev, flags);
+ if (ret < 0) {
dev_err(sdev->dev,
"error: PM_GATE ipc error %d\n", ret);
+ goto revert;
+ }
+
+ return ret;
+
+revert:
+ /* fallback to the previous register value */
+ value = value ? 0 : SOF_HDA_VS_D0I3C_I3;
+
+ /*
+ * This can fail but return the IPC error to signal that
+ * the state change failed.
+ */
+ hda_dsp_update_d0i3c_register(sdev, value);
return ret;
}
+/* helper to log DSP state */
+static void hda_dsp_state_log(struct snd_sof_dev *sdev)
+{
+ switch (sdev->dsp_power_state.state) {
+ case SOF_DSP_PM_D0:
+ switch (sdev->dsp_power_state.substate) {
+ case SOF_HDA_DSP_PM_D0I0:
+ dev_dbg(sdev->dev, "Current DSP power state: D0I0\n");
+ break;
+ case SOF_HDA_DSP_PM_D0I3:
+ dev_dbg(sdev->dev, "Current DSP power state: D0I3\n");
+ break;
+ default:
+ dev_dbg(sdev->dev, "Unknown DSP D0 substate: %d\n",
+ sdev->dsp_power_state.substate);
+ break;
+ }
+ break;
+ case SOF_DSP_PM_D1:
+ dev_dbg(sdev->dev, "Current DSP power state: D1\n");
+ break;
+ case SOF_DSP_PM_D2:
+ dev_dbg(sdev->dev, "Current DSP power state: D2\n");
+ break;
+ case SOF_DSP_PM_D3_HOT:
+ dev_dbg(sdev->dev, "Current DSP power state: D3_HOT\n");
+ break;
+ case SOF_DSP_PM_D3:
+ dev_dbg(sdev->dev, "Current DSP power state: D3\n");
+ break;
+ case SOF_DSP_PM_D3_COLD:
+ dev_dbg(sdev->dev, "Current DSP power state: D3_COLD\n");
+ break;
+ default:
+ dev_dbg(sdev->dev, "Unknown DSP power state: %d\n",
+ sdev->dsp_power_state.state);
+ break;
+ }
+}
+
+/*
+ * All DSP power state transitions are initiated by the driver.
+ * If the requested state change fails, the error is simply returned.
+ * Further state transitions are attempted only when the set_power_save() op
+ * is called again either because of a new IPC sent to the DSP or
+ * during system suspend/resume.
+ */
+int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
+ const struct sof_dsp_power_state *target_state)
+{
+ int ret = 0;
+
+ /*
+ * When the DSP is already in D0I3 and the target state is D0I3,
+ * it could be the case that the DSP is in D0I3 during S0
+ * and the system is suspending to S0Ix. Therefore,
+ * hda_dsp_set_D0_state() must be called to disable trace DMA
+ * by sending the PM_GATE IPC to the FW.
+ */
+ if (target_state->substate == SOF_HDA_DSP_PM_D0I3 &&
+ sdev->system_suspend_target == SOF_SUSPEND_S0IX)
+ goto set_state;
+
+ /*
+ * For all other cases, return without doing anything if
+ * the DSP is already in the target state.
+ */
+ if (target_state->state == sdev->dsp_power_state.state &&
+ target_state->substate == sdev->dsp_power_state.substate)
+ return 0;
+
+set_state:
+ switch (target_state->state) {
+ case SOF_DSP_PM_D0:
+ ret = hda_dsp_set_D0_state(sdev, target_state);
+ break;
+ case SOF_DSP_PM_D3:
+ /* The only allowed transition is: D0I0 -> D3 */
+ if (sdev->dsp_power_state.state == SOF_DSP_PM_D0 &&
+ sdev->dsp_power_state.substate == SOF_HDA_DSP_PM_D0I0)
+ break;
+
+ dev_err(sdev->dev,
+ "error: transition from %d to %d not allowed\n",
+ sdev->dsp_power_state.state, target_state->state);
+ return -EINVAL;
+ default:
+ dev_err(sdev->dev, "error: target state unsupported %d\n",
+ target_state->state);
+ return -EINVAL;
+ }
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "failed to set requested target DSP state %d substate %d\n",
+ target_state->state, target_state->substate);
+ return ret;
+ }
+
+ sdev->dsp_power_state = *target_state;
+ hda_dsp_state_log(sdev);
+ return ret;
+}
+
+/*
+ * Audio DSP states may transform as below:-
+ *
+ * Opportunistic D0I3 in S0
+ * Runtime +---------------------+ Delayed D0i3 work timeout
+ * suspend | +--------------------+
+ * +------------+ D0I0(active) | |
+ * | | <---------------+ |
+ * | +--------> | New IPC | |
+ * | |Runtime +--^--+---------^--+--+ (via mailbox) | |
+ * | |resume | | | | | |
+ * | | | | | | | |
+ * | | System| | | | | |
+ * | | resume| | S3/S0IX | | | |
+ * | | | | suspend | | S0IX | |
+ * | | | | | |suspend | |
+ * | | | | | | | |
+ * | | | | | | | |
+ * +-v---+-----------+--v-------+ | | +------+----v----+
+ * | | | +-----------> |
+ * | D3 (suspended) | | | D0I3 |
+ * | | +--------------+ |
+ * | | System resume | |
+ * +----------------------------+ +----------------+
+ *
+ * S0IX suspend: The DSP is in D0I3 if any D0I3-compatible streams
+ * ignored the suspend trigger. Otherwise the DSP
+ * is in D3.
+ */
+
static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
{
struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
@@ -390,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
#endif
int ret;
+ hda_sdw_int_enable(sdev, false);
+
/* disable IPC interrupts */
hda_dsp_ipc_int_disable(sdev);
@@ -486,10 +692,24 @@ int hda_dsp_resume(struct snd_sof_dev *sdev)
{
struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
struct pci_dev *pci = to_pci_dev(sdev->dev);
+ const struct sof_dsp_power_state target_state = {
+ .state = SOF_DSP_PM_D0,
+ .substate = SOF_HDA_DSP_PM_D0I0,
+ };
+ int ret;
- if (sdev->s0_suspend) {
+ /* resume from D0I3 */
+ if (sdev->dsp_power_state.state == SOF_DSP_PM_D0) {
hda_codec_i915_display_power(sdev, true);
+ /* Set DSP power state */
+ ret = snd_sof_dsp_set_power_state(sdev, &target_state);
+ if (ret < 0) {
+ dev_err(sdev->dev, "error: setting dsp state %d substate %d\n",
+ target_state.state, target_state.substate);
+ return ret;
+ }
+
/* restore L1SEN bit */
if (hda->l1_support_changed)
snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
@@ -503,13 +723,26 @@ int hda_dsp_resume(struct snd_sof_dev *sdev)
}
/* init hda controller. DSP cores will be powered up during fw boot */
- return hda_resume(sdev, false);
+ ret = hda_resume(sdev, false);
+ if (ret < 0)
+ return ret;
+
+ return snd_sof_dsp_set_power_state(sdev, &target_state);
}
int hda_dsp_runtime_resume(struct snd_sof_dev *sdev)
{
+ const struct sof_dsp_power_state target_state = {
+ .state = SOF_DSP_PM_D0,
+ };
+ int ret;
+
/* init hda controller. DSP cores will be powered up during fw boot */
- return hda_resume(sdev, true);
+ ret = hda_resume(sdev, true);
+ if (ret < 0)
+ return ret;
+
+ return snd_sof_dsp_set_power_state(sdev, &target_state);
}
int hda_dsp_runtime_idle(struct snd_sof_dev *sdev)
@@ -527,21 +760,47 @@ int hda_dsp_runtime_idle(struct snd_sof_dev *sdev)
int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev)
{
+ const struct sof_dsp_power_state target_state = {
+ .state = SOF_DSP_PM_D3,
+ };
+ int ret;
+
/* stop hda controller and power dsp off */
- return hda_suspend(sdev, true);
+ ret = hda_suspend(sdev, true);
+ if (ret < 0)
+ return ret;
+
+ return snd_sof_dsp_set_power_state(sdev, &target_state);
}
-int hda_dsp_suspend(struct snd_sof_dev *sdev)
+int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state)
{
struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
struct hdac_bus *bus = sof_to_bus(sdev);
struct pci_dev *pci = to_pci_dev(sdev->dev);
+ const struct sof_dsp_power_state target_dsp_state = {
+ .state = target_state,
+ .substate = target_state == SOF_DSP_PM_D0 ?
+ SOF_HDA_DSP_PM_D0I3 : 0,
+ };
int ret;
- if (sdev->s0_suspend) {
+ /* cancel any attempt for DSP D0I3 */
+ cancel_delayed_work_sync(&hda->d0i3_work);
+
+ if (target_state == SOF_DSP_PM_D0) {
/* we can't keep a wakeref to display driver at suspend */
hda_codec_i915_display_power(sdev, false);
+ /* Set DSP power state */
+ ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state);
+ if (ret < 0) {
+ dev_err(sdev->dev, "error: setting dsp state %d substate %d\n",
+ target_dsp_state.state,
+ target_dsp_state.substate);
+ return ret;
+ }
+
/* enable L1SEN to make sure the system can enter S0Ix */
hda->l1_support_changed =
snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
@@ -562,7 +821,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev)
return ret;
}
- return 0;
+ return snd_sof_dsp_set_power_state(sdev, &target_dsp_state);
}
int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
@@ -588,7 +847,7 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
*/
if (stream->link_substream) {
rtd = snd_pcm_substream_chip(stream->link_substream);
- name = rtd->codec_dai->component->name;
+ name = asoc_rtd_to_codec(rtd, 0)->component->name;
link = snd_hdac_ext_bus_get_link(bus, name);
if (!link)
return -EINVAL;
@@ -606,3 +865,33 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
#endif
return 0;
}
+
+void hda_dsp_d0i3_work(struct work_struct *work)
+{
+ struct sof_intel_hda_dev *hdev = container_of(work,
+ struct sof_intel_hda_dev,
+ d0i3_work.work);
+ struct hdac_bus *bus = &hdev->hbus.core;
+ struct snd_sof_dev *sdev = dev_get_drvdata(bus->dev);
+ struct sof_dsp_power_state target_state;
+ int ret;
+
+ target_state.state = SOF_DSP_PM_D0;
+
+ /* DSP can enter D0I3 iff only D0I3-compatible streams are active */
+ if (snd_sof_dsp_only_d0i3_compatible_stream_active(sdev))
+ target_state.substate = SOF_HDA_DSP_PM_D0I3;
+ else
+ target_state.substate = SOF_HDA_DSP_PM_D0I0;
+
+ /* remain in D0I0 */
+ if (target_state.substate == SOF_HDA_DSP_PM_D0I0)
+ return;
+
+ /* This can fail but error cannot be propagated */
+ ret = snd_sof_dsp_set_power_state(sdev, &target_state);
+ if (ret < 0)
+ dev_err_ratelimited(sdev->dev,
+ "error: failed to set DSP state %d substate %d\n",
+ target_state.state, target_state.substate);
+}
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 1837f66e361f..6062bb6011fb 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -106,7 +106,9 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev)
ret = reply.error;
} else {
/* reply correct size ? */
- if (reply.hdr.size != msg->reply_size) {
+ if (reply.hdr.size != msg->reply_size &&
+ /* getter payload is never known upfront */
+ !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) {
dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n",
msg->reply_size, reply.hdr.size);
ret = -EINVAL;
@@ -123,12 +125,6 @@ out:
}
-static bool hda_dsp_ipc_is_sof(uint32_t msg)
-{
- return (msg & (HDA_DSP_IPC_PURGE_FW | 0xf << 9)) != msg ||
- (msg & HDA_DSP_IPC_PURGE_FW) != HDA_DSP_IPC_PURGE_FW;
-}
-
/* IPC handler thread */
irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
{
@@ -174,17 +170,9 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
*/
spin_lock_irq(&sdev->ipc_lock);
- /* handle immediate reply from DSP core - ignore ROM messages */
- if (hda_dsp_ipc_is_sof(msg)) {
- hda_dsp_ipc_get_reply(sdev);
- snd_sof_ipc_reply(sdev, msg);
- }
-
- /* wake up sleeper if we are loading code */
- if (sdev->code_loading) {
- sdev->code_loading = 0;
- wake_up(&sdev->waitq);
- }
+ /* handle immediate reply from DSP core */
+ hda_dsp_ipc_get_reply(sdev);
+ snd_sof_ipc_reply(sdev, msg);
/* set the done bit */
hda_dsp_ipc_dsp_done(sdev);
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 8852184a2569..e1550ccd0a49 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -131,6 +131,12 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, const void *fwdata,
goto err;
}
+ /* set DONE bit to clear the reply IPC message */
+ snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR,
+ chip->ipc_ack,
+ chip->ipc_ack_mask,
+ chip->ipc_ack_mask);
+
/* step 5: power down corex */
ret = hda_dsp_core_power_down(sdev,
chip->cores_mask & ~(HDA_DSP_CORE_MASK(0)));
@@ -173,9 +179,6 @@ static int cl_trigger(struct snd_sof_dev *sdev,
/* code loader is special case that reuses stream ops */
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- wait_event_timeout(sdev->waitq, !sdev->code_loading,
- HDA_DSP_CL_TRIGGER_TIMEOUT);
-
snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL,
1 << hstream->index,
1 << hstream->index);
@@ -344,6 +347,24 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev)
}
/*
+ * When a SoundWire link is in clock stop state, a Slave
+ * device may trigger in-band wakes for events such as jack
+ * insertion or acoustic event detection. This event will lead
+ * to a WAKEEN interrupt, handled by the PCI device and routed
+ * to PME if the PCI device is in D3. The resume function in
+ * audio PCI driver will be invoked by ACPI for PME event and
+ * initialize the device and process WAKEEN interrupt.
+ *
+ * The WAKEEN interrupt should be processed ASAP to prevent an
+ * interrupt flood, otherwise other interrupts, such IPC,
+ * cannot work normally. The WAKEEN is handled after the ROM
+ * is initialized successfully, which ensures power rails are
+ * enabled before accessing the SoundWire SHIM registers
+ */
+ if (!sdev->first_boot)
+ hda_sdw_process_wakeen(sdev);
+
+ /*
* at this point DSP ROM has been initialized and
* should be ready for code loading and firmware boot
*/
@@ -396,6 +417,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev)
/* post fw run operations */
int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
{
+ int ret;
+
+ if (sdev->first_boot) {
+ ret = hda_sdw_startup(sdev);
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "error: could not startup SoundWire links\n");
+ return ret;
+ }
+ }
+
+ hda_sdw_int_enable(sdev, true);
+
/* re-enable clock gating and power gating */
return hda_dsp_ctrl_clock_power_gating(sdev, true);
}
diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c
index 23872f6e708d..a46a6baa1c3f 100644
--- a/sound/soc/sof/intel/hda-pcm.c
+++ b/sound/soc/sof/intel/hda-pcm.c
@@ -27,7 +27,7 @@
#define SDnFMT_BITS(x) ((x) << 4)
#define SDnFMT_CHAN(x) ((x) << 0)
-static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate)
+u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate)
{
switch (rate) {
case 8000:
@@ -61,7 +61,7 @@ static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate)
}
};
-static inline u32 get_bits(struct snd_sof_dev *sdev, int sample_bits)
+u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits)
{
switch (sample_bits) {
case 8:
@@ -95,8 +95,8 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev,
u32 size, rate, bits;
size = params_buffer_bytes(params);
- rate = get_mult_div(sdev, params_rate(params));
- bits = get_bits(sdev, params_width(params));
+ rate = hda_dsp_get_mult_div(sdev, params_rate(params));
+ bits = hda_dsp_get_bits(sdev, params_width(params));
hstream->substream = substream;
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index c0ab9bb2a797..5d386956906f 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -547,6 +547,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev,
SOF_HDA_REG_PP_PPCTL, mask, 0);
spin_unlock_irq(&bus->reg_lock);
+ stream->substream = NULL;
+
return 0;
}
@@ -571,6 +573,22 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev)
return ret;
}
+static void
+hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size)
+{
+ u64 prev_pos, pos, num_bytes;
+
+ div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos);
+ pos = snd_hdac_stream_get_pos_posbuf(hstream);
+
+ if (pos < prev_pos)
+ num_bytes = (buffer_size - prev_pos) + pos;
+ else
+ num_bytes = pos - prev_pos;
+
+ hstream->curr_pos += num_bytes;
+}
+
static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status)
{
struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus);
@@ -588,14 +606,19 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status)
snd_hdac_stream_writeb(s, SD_STS, sd_status);
active = true;
- if (!s->substream ||
+ if ((!s->substream && !s->cstream) ||
!s->running ||
(sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0)
continue;
/* Inform ALSA only in case not do that with IPC */
- if (sof_hda->no_ipc_position)
+ if (s->substream && sof_hda->no_ipc_position) {
snd_sof_pcm_period_elapsed(s->substream);
+ } else if (s->cstream) {
+ hda_dsp_set_bytes_transferred(s,
+ s->cstream->runtime->buffer_size);
+ snd_compr_fragment_elapsed(s->cstream);
+ }
}
}
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 25946a1c2822..211e91e79eae 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -18,10 +18,14 @@
#include <sound/hdaudio_ext.h>
#include <sound/hda_register.h>
+#include <linux/acpi.h>
#include <linux/module.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_intel.h>
#include <sound/intel-nhlt.h>
#include <sound/sof.h>
#include <sound/sof/xtensa.h>
+#include "../sof-audio.h"
#include "../ops.h"
#include "hda.h"
@@ -34,6 +38,235 @@
#define EXCEPT_MAX_HDR_SIZE 0x400
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+
+/*
+ * The default for SoundWire clock stop quirks is to power gate the IP
+ * and do a Bus Reset, this will need to be modified when the DSP
+ * needs to remain in D0i3 so that the Master does not lose context
+ * and enumeration is not required on clock restart
+ */
+static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET;
+module_param(sdw_clock_stop_quirks, int, 0444);
+MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks");
+
+static int sdw_params_stream(struct device *dev,
+ struct sdw_intel_stream_params_data *params_data)
+{
+ struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+ struct snd_soc_dai *d = params_data->dai;
+ struct sof_ipc_dai_config config;
+ struct sof_ipc_reply reply;
+ int link_id = params_data->link_id;
+ int alh_stream_id = params_data->alh_stream_id;
+ int ret;
+ u32 size = sizeof(config);
+
+ memset(&config, 0, size);
+ config.hdr.size = size;
+ config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG;
+ config.type = SOF_DAI_INTEL_ALH;
+ config.dai_index = (link_id << 8) | (d->id);
+ config.alh.stream_id = alh_stream_id;
+
+ /* send message to DSP */
+ ret = sof_ipc_tx_message(sdev->ipc,
+ config.hdr.cmd, &config, size, &reply,
+ sizeof(reply));
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n",
+ link_id, d->id, alh_stream_id);
+ }
+
+ return ret;
+}
+
+static int sdw_free_stream(struct device *dev,
+ struct sdw_intel_stream_free_data *free_data)
+{
+ struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+ struct snd_soc_dai *d = free_data->dai;
+ struct sof_ipc_dai_config config;
+ struct sof_ipc_reply reply;
+ int link_id = free_data->link_id;
+ int ret;
+ u32 size = sizeof(config);
+
+ memset(&config, 0, size);
+ config.hdr.size = size;
+ config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG;
+ config.type = SOF_DAI_INTEL_ALH;
+ config.dai_index = (link_id << 8) | d->id;
+ config.alh.stream_id = 0xFFFF; /* invalid value on purpose */
+
+ /* send message to DSP */
+ ret = sof_ipc_tx_message(sdev->ipc,
+ config.hdr.cmd, &config, size, &reply,
+ sizeof(reply));
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "error: failed to free stream for link %d dai->id %d\n",
+ link_id, d->id);
+ }
+
+ return ret;
+}
+
+static const struct sdw_intel_ops sdw_callback = {
+ .params_stream = sdw_params_stream,
+ .free_stream = sdw_free_stream,
+};
+
+void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
+{
+ sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable);
+}
+
+static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ acpi_handle handle;
+ int ret;
+
+ handle = ACPI_HANDLE(sdev->dev);
+
+ /* save ACPI info for the probe step */
+ hdev = sdev->pdata->hw_pdata;
+
+ ret = sdw_intel_acpi_scan(handle, &hdev->info);
+ if (ret < 0) {
+ dev_err(sdev->dev, "%s failed\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int hda_sdw_probe(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ struct sdw_intel_res res;
+ void *sdw;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ memset(&res, 0, sizeof(res));
+
+ res.mmio_base = sdev->bar[HDA_DSP_BAR];
+ res.irq = sdev->ipc_irq;
+ res.handle = hdev->info.handle;
+ res.parent = sdev->dev;
+ res.ops = &sdw_callback;
+ res.dev = sdev->dev;
+ res.clock_stop_quirks = sdw_clock_stop_quirks;
+
+ /*
+ * ops and arg fields are not populated for now,
+ * they will be needed when the DAI callbacks are
+ * provided
+ */
+
+ /* we could filter links here if needed, e.g for quirks */
+ res.count = hdev->info.count;
+ res.link_mask = hdev->info.link_mask;
+
+ sdw = sdw_intel_probe(&res);
+ if (!sdw) {
+ dev_err(sdev->dev, "error: SoundWire probe failed\n");
+ return -EINVAL;
+ }
+
+ /* save context */
+ hdev->sdw = sdw;
+
+ return 0;
+}
+
+int hda_sdw_startup(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ if (!hdev->sdw)
+ return 0;
+
+ return sdw_intel_startup(hdev->sdw);
+}
+
+static int hda_sdw_exit(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ hda_sdw_int_enable(sdev, false);
+
+ if (hdev->sdw)
+ sdw_intel_exit(hdev->sdw);
+ hdev->sdw = NULL;
+
+ return 0;
+}
+
+static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ bool ret = false;
+ u32 irq_status;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ if (!hdev->sdw)
+ return ret;
+
+ /* store status */
+ irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2);
+
+ /* invalid message ? */
+ if (irq_status == 0xffffffff)
+ goto out;
+
+ /* SDW message ? */
+ if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW)
+ ret = true;
+
+out:
+ return ret;
+}
+
+static irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
+{
+ return sdw_intel_thread(irq, context);
+}
+
+static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+ if (hdev->sdw &&
+ snd_sof_dsp_read(sdev, HDA_DSP_BAR,
+ HDA_DSP_REG_SNDW_WAKE_STS))
+ return true;
+
+ return false;
+}
+
+void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+ if (!hdev->sdw)
+ return;
+
+ sdw_intel_process_wakeen_event(hdev->sdw);
+}
+
+#endif
+
/*
* Debug
*/
@@ -54,8 +287,7 @@ static int hda_dmic_num = -1;
module_param_named(dmic_num, hda_dmic_num, int, 0444);
MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number");
-static bool hda_codec_use_common_hdmi =
- IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC);
+static bool hda_codec_use_common_hdmi = IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI);
module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444);
MODULE_PARM_DESC(use_common_hdmi, "SOF HDA use common HDMI codec driver");
#endif
@@ -288,10 +520,8 @@ static int hda_init(struct snd_sof_dev *sdev)
/* init i915 and HDMI codecs */
ret = hda_codec_i915_init(sdev);
- if (ret < 0) {
- dev_err(sdev->dev, "error: init i915 and HDMI codec failed\n");
- return ret;
- }
+ if (ret < 0)
+ dev_warn(sdev->dev, "init of i915 and HDMI codec failed\n");
/* get controller capabilities */
ret = hda_dsp_ctrl_get_caps(sdev);
@@ -349,9 +579,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev,
static int hda_init_caps(struct snd_sof_dev *sdev)
{
struct hdac_bus *bus = sof_to_bus(sdev);
+ struct snd_sof_pdata *pdata = sdev->pdata;
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
struct hdac_ext_link *hlink;
#endif
+ struct sof_intel_hda_dev *hdev = pdata->hw_pdata;
+ u32 link_mask;
int ret = 0;
device_disable_async_suspend(bus->dev);
@@ -365,12 +598,37 @@ static int hda_init_caps(struct snd_sof_dev *sdev)
if (ret < 0) {
dev_err(bus->dev, "error: init chip failed with ret: %d\n",
ret);
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
- hda_codec_i915_exit(sdev);
-#endif
return ret;
}
+ /* scan SoundWire capabilities exposed by DSDT */
+ ret = hda_sdw_acpi_scan(sdev);
+ if (ret < 0) {
+ dev_dbg(sdev->dev, "skipping SoundWire, ACPI scan error\n");
+ goto skip_soundwire;
+ }
+
+ link_mask = hdev->info.link_mask;
+ if (!link_mask) {
+ dev_dbg(sdev->dev, "skipping SoundWire, no links enabled\n");
+ goto skip_soundwire;
+ }
+
+ /*
+ * probe/allocate SoundWire resources.
+ * The hardware configuration takes place in hda_sdw_startup
+ * after power rails are enabled.
+ * It's entirely possible to have a mix of I2S/DMIC/SoundWire
+ * devices, so we allocate the resources in all cases.
+ */
+ ret = hda_sdw_probe(sdev);
+ if (ret < 0) {
+ dev_err(sdev->dev, "error: SoundWire probe error\n");
+ return ret;
+ }
+
+skip_soundwire:
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
if (bus->mlcap)
snd_hdac_ext_bus_get_ml_capabilities(bus);
@@ -379,7 +637,7 @@ static int hda_init_caps(struct snd_sof_dev *sdev)
hda_codec_probe_bus(sdev, hda_codec_use_common_hdmi);
if (!HDA_IDISP_CODEC(bus->codec_mask))
- hda_codec_i915_exit(sdev);
+ hda_codec_i915_display_power(sdev, false);
/*
* we are done probing so decrement link counts
@@ -427,6 +685,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context)
static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context)
{
struct snd_sof_dev *sdev = context;
+ struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata;
/* deal with streams and controller first */
if (hda_dsp_check_stream_irq(sdev))
@@ -435,6 +694,12 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context)
if (hda_dsp_check_ipc_irq(sdev))
sof_ops(sdev)->irq_thread(irq, sdev);
+ if (hda_dsp_check_sdw_irq(sdev))
+ hda_dsp_sdw_thread(irq, hdev->sdw);
+
+ if (hda_sdw_check_wakeen_irq(sdev))
+ hda_sdw_process_wakeen(sdev);
+
/* enable GIE interrupt */
snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
SOF_HDA_INTCTL,
@@ -590,12 +855,11 @@ int hda_dsp_probe(struct snd_sof_dev *sdev)
hda_dsp_ctrl_ppcap_enable(sdev, true);
hda_dsp_ctrl_ppcap_int_enable(sdev, true);
- /* initialize waitq for code loading */
- init_waitqueue_head(&sdev->waitq);
-
/* set default mailbox offset for FW ready message */
sdev->dsp_box.offset = HDA_DSP_MBOX_UPLINK_OFFSET;
+ INIT_DELAYED_WORK(&hdev->d0i3_work, hda_dsp_d0i3_work);
+
return 0;
free_ipc_irq:
@@ -621,11 +885,16 @@ int hda_dsp_remove(struct snd_sof_dev *sdev)
struct pci_dev *pci = to_pci_dev(sdev->dev);
const struct sof_intel_dsp_desc *chip = hda->desc;
+ /* cancel any attempt for DSP D0I3 */
+ cancel_delayed_work_sync(&hda->d0i3_work);
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
/* codec removal, invoke bus_device_remove */
snd_hdac_ext_bus_device_remove(bus);
#endif
+ hda_sdw_exit(sdev);
+
if (!IS_ERR_OR_NULL(hda->dmic_dev))
platform_device_unregister(hda->dmic_dev);
@@ -694,12 +963,11 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev)
/*
* If no machine driver is found, then:
*
- * hda machine driver is used if :
- * 1. there is one HDMI codec and one external HDAudio codec
- * 2. only HDMI codec
+ * generic hda machine driver can handle:
+ * - one HDMI codec, and/or
+ * - one external HDAudio codec
*/
- if (!pdata->machine && codec_num <= 2 &&
- HDA_IDISP_CODEC(bus->codec_mask)) {
+ if (!pdata->machine && codec_num <= 2) {
hda_mach = snd_soc_acpi_intel_hda_machines;
/* topology: use the info from hda_machines */
@@ -709,7 +977,7 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev)
dev_info(bus->dev, "using HDA machine driver %s now\n",
hda_mach->drv_name);
- if (codec_num == 1)
+ if (codec_num == 1 && HDA_IDISP_CODEC(bus->codec_mask))
idisp_str = "-idisp";
else
idisp_str = "";
@@ -763,6 +1031,123 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev)
}
#endif
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+/* Check if all Slaves defined on the link can be found */
+static bool link_slaves_found(struct snd_sof_dev *sdev,
+ const struct snd_soc_acpi_link_adr *link,
+ struct sdw_intel_ctx *sdw)
+{
+ struct hdac_bus *bus = sof_to_bus(sdev);
+ struct sdw_intel_slave_id *ids = sdw->ids;
+ int num_slaves = sdw->num_slaves;
+ unsigned int part_id, link_id, unique_id, mfg_id;
+ int i, j;
+
+ for (i = 0; i < link->num_adr; i++) {
+ u64 adr = link->adr_d[i].adr;
+
+ mfg_id = SDW_MFG_ID(adr);
+ part_id = SDW_PART_ID(adr);
+ link_id = SDW_DISCO_LINK_ID(adr);
+ for (j = 0; j < num_slaves; j++) {
+ if (ids[j].link_id != link_id ||
+ ids[j].id.part_id != part_id ||
+ ids[j].id.mfg_id != mfg_id)
+ continue;
+ /*
+ * we have to check unique id
+ * if there is more than one
+ * Slave on the link
+ */
+ unique_id = SDW_UNIQUE_ID(adr);
+ if (link->num_adr == 1 ||
+ ids[j].id.unique_id == SDW_IGNORED_UNIQUE_ID ||
+ ids[j].id.unique_id == unique_id) {
+ dev_dbg(bus->dev,
+ "found %x at link %d\n",
+ part_id, link_id);
+ break;
+ }
+ }
+ if (j == num_slaves) {
+ dev_dbg(bus->dev,
+ "Slave %x not found\n",
+ part_id);
+ return false;
+ }
+ }
+ return true;
+}
+
+static int hda_sdw_machine_select(struct snd_sof_dev *sdev)
+{
+ struct snd_sof_pdata *pdata = sdev->pdata;
+ const struct snd_soc_acpi_link_adr *link;
+ struct hdac_bus *bus = sof_to_bus(sdev);
+ struct snd_soc_acpi_mach *mach;
+ struct sof_intel_hda_dev *hdev;
+ u32 link_mask;
+ int i;
+
+ hdev = pdata->hw_pdata;
+ link_mask = hdev->info.link_mask;
+
+ /*
+ * Select SoundWire machine driver if needed using the
+ * alternate tables. This case deals with SoundWire-only
+ * machines, for mixed cases with I2C/I2S the detection relies
+ * on the HID list.
+ */
+ if (link_mask && !pdata->machine) {
+ for (mach = pdata->desc->alt_machines;
+ mach && mach->link_mask; mach++) {
+ if (mach->link_mask != link_mask)
+ continue;
+
+ /* No need to match adr if there is no links defined */
+ if (!mach->links)
+ break;
+
+ link = mach->links;
+ for (i = 0; i < hdev->info.count && link->num_adr;
+ i++, link++) {
+ /*
+ * Try next machine if any expected Slaves
+ * are not found on this link.
+ */
+ if (!link_slaves_found(sdev, link, hdev->sdw))
+ break;
+ }
+ /* Found if all Slaves are checked */
+ if (i == hdev->info.count || !link->num_adr)
+ break;
+ }
+ if (mach && mach->link_mask) {
+ dev_dbg(bus->dev,
+ "SoundWire machine driver %s topology %s\n",
+ mach->drv_name,
+ mach->sof_tplg_filename);
+ pdata->machine = mach;
+ mach->mach_params.links = mach->links;
+ mach->mach_params.link_mask = mach->link_mask;
+ mach->mach_params.platform = dev_name(sdev->dev);
+ pdata->fw_filename = mach->sof_fw_filename;
+ pdata->tplg_filename = mach->sof_tplg_filename;
+ } else {
+ dev_info(sdev->dev,
+ "No SoundWire machine driver found\n");
+ }
+ }
+
+ return 0;
+}
+#else
+static int hda_sdw_machine_select(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+#endif
+
void hda_set_mach_params(const struct snd_soc_acpi_mach *mach,
struct device *dev)
{
@@ -782,9 +1167,19 @@ void hda_machine_select(struct snd_sof_dev *sdev)
if (mach) {
sof_pdata->tplg_filename = mach->sof_tplg_filename;
sof_pdata->machine = mach;
+
+ if (mach->link_mask) {
+ mach->mach_params.links = mach->links;
+ mach->mach_params.link_mask = mach->link_mask;
+ }
}
/*
+ * If I2S fails, try SoundWire
+ */
+ hda_sdw_machine_select(sdev);
+
+ /*
* Choose HDA generic machine driver if mach is NULL.
* Otherwise, set certain mach params.
*/
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 6191d9192fae..e9825798de77 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -11,6 +11,9 @@
#ifndef __SOF_INTEL_HDA_H
#define __SOF_INTEL_HDA_H
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_intel.h>
+#include <sound/compress_driver.h>
#include <sound/hda_codec.h>
#include <sound/hdaudio_ext.h>
#include "shim.h"
@@ -174,7 +177,6 @@
* value cannot be read back within the specified time.
*/
#define HDA_DSP_STREAM_RUN_TIMEOUT 300
-#define HDA_DSP_CL_TRIGGER_TIMEOUT 300
#define HDA_DSP_SPIB_ENABLE 1
#define HDA_DSP_SPIB_DISABLE 0
@@ -230,6 +232,9 @@
#define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10)
#define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14)
+#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5)
+#define HDA_DSP_REG_SNDW_WAKE_STS 0x2C192
+
/* Intel HD Audio Inter-Processor Communication Registers */
#define HDA_DSP_IPC_BASE 0x40
#define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00)
@@ -348,7 +353,13 @@
/* Number of DAIs */
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+#define SOF_SKL_NUM_DAIS 16
+#else
#define SOF_SKL_NUM_DAIS 15
+#endif
+
#else
#define SOF_SKL_NUM_DAIS 8
#endif
@@ -392,6 +403,19 @@ struct sof_intel_dsp_bdl {
#define SOF_HDA_PLAYBACK 0
#define SOF_HDA_CAPTURE 1
+/*
+ * Time in ms for opportunistic D0I3 entry delay.
+ * This has been deliberately chosen to be long to avoid race conditions.
+ * Could be optimized in future.
+ */
+#define SOF_HDA_D0I3_WORK_DELAY_MS 5000
+
+/* HDA DSP D0 substate */
+enum sof_hda_D0_substate {
+ SOF_HDA_DSP_PM_D0I0, /* default D0 substate */
+ SOF_HDA_DSP_PM_D0I3, /* low power D0 substate */
+};
+
/* represents DSP HDA controller frontend - i.e. host facing control */
struct sof_intel_hda_dev {
@@ -414,6 +438,15 @@ struct sof_intel_hda_dev {
/* DMIC device */
struct platform_device *dmic_dev;
+
+ /* delayed work to enter D0I3 opportunistically */
+ struct delayed_work d0i3_work;
+
+ /* ACPI information stored between scan and probe steps */
+ struct sdw_intel_acpi_info info;
+
+ /* sdw context allocated by SoundWire driver */
+ struct sdw_intel_ctx *sdw;
};
static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s)
@@ -469,9 +502,9 @@ void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev);
void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev);
int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
- enum sof_d0_substate d0_substate);
+ const struct sof_dsp_power_state *target_state);
-int hda_dsp_suspend(struct snd_sof_dev *sdev);
+int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state);
int hda_dsp_resume(struct snd_sof_dev *sdev);
int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev);
int hda_dsp_runtime_resume(struct snd_sof_dev *sdev);
@@ -481,10 +514,13 @@ void hda_dsp_dump_skl(struct snd_sof_dev *sdev, u32 flags);
void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags);
void hda_ipc_dump(struct snd_sof_dev *sdev);
void hda_ipc_irq_dump(struct snd_sof_dev *sdev);
+void hda_dsp_d0i3_work(struct work_struct *work);
/*
* DSP PCM Operations.
*/
+u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate);
+u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits);
int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream);
int hda_dsp_pcm_close(struct snd_sof_dev *sdev,
@@ -533,6 +569,29 @@ int hda_ipc_pcm_params(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream,
const struct sof_ipc_pcm_params_reply *reply);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+/*
+ * Probe Compress Operations.
+ */
+int hda_probe_compr_assign(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai);
+int hda_probe_compr_free(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai);
+int hda_probe_compr_set_params(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_params *params,
+ struct snd_soc_dai *dai);
+int hda_probe_compr_trigger(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai);
+int hda_probe_compr_pointer(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp,
+ struct snd_soc_dai *dai);
+#endif
+
/*
* DSP IPC Operations.
*/
@@ -606,6 +665,61 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag);
int hda_dsp_trace_release(struct snd_sof_dev *sdev);
int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd);
+/*
+ * SoundWire support
+ */
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+
+int hda_sdw_startup(struct snd_sof_dev *sdev);
+void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable);
+void hda_sdw_process_wakeen(struct snd_sof_dev *sdev);
+
+#else
+
+static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline int hda_sdw_probe(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline int hda_sdw_startup(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline int hda_sdw_exit(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
+{
+}
+
+static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
+{
+ return false;
+}
+
+static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
+{
+ return IRQ_HANDLED;
+}
+
+static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
+{
+ return false;
+}
+
+static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+{
+}
+#endif
+
/* common dai driver */
extern struct snd_soc_dai_driver skl_dai[];
diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c
index 78aa1da7c7a9..1c6794918cbb 100644
--- a/sound/soc/sof/ipc.c
+++ b/sound/soc/sof/ipc.c
@@ -214,15 +214,17 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg,
snd_sof_handle_fw_exception(ipc->sdev);
ret = -ETIMEDOUT;
} else {
- /* copy the data returned from DSP */
ret = msg->reply_error;
- if (msg->reply_size)
- memcpy(reply_data, msg->reply_data, msg->reply_size);
- if (ret < 0)
+ if (ret < 0) {
dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n",
hdr->cmd, msg->reply_size);
- else
+ } else {
ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd);
+ if (msg->reply_size)
+ /* copy the data returned from DSP */
+ memcpy(reply_data, msg->reply_data,
+ msg->reply_size);
+ }
}
return ret;
@@ -268,7 +270,6 @@ static int sof_ipc_tx_message_unlocked(struct snd_sof_ipc *ipc, u32 header,
spin_unlock_irq(&sdev->ipc_lock);
if (ret < 0) {
- /* So far IPC TX never fails, consider making the above void */
dev_err_ratelimited(sdev->dev,
"error: ipc tx failed with error %d\n",
ret);
@@ -289,6 +290,32 @@ int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header,
void *msg_data, size_t msg_bytes, void *reply_data,
size_t reply_bytes)
{
+ const struct sof_dsp_power_state target_state = {
+ .state = SOF_DSP_PM_D0,
+ };
+ int ret;
+
+ /* ensure the DSP is in D0 before sending a new IPC */
+ ret = snd_sof_dsp_set_power_state(ipc->sdev, &target_state);
+ if (ret < 0) {
+ dev_err(ipc->sdev->dev, "error: resuming DSP %d\n", ret);
+ return ret;
+ }
+
+ return sof_ipc_tx_message_no_pm(ipc, header, msg_data, msg_bytes,
+ reply_data, reply_bytes);
+}
+EXPORT_SYMBOL(sof_ipc_tx_message);
+
+/*
+ * send IPC message from host to DSP without modifying the DSP state.
+ * This will be used for IPC's that can be handled by the DSP
+ * even in a low-power D0 substate.
+ */
+int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header,
+ void *msg_data, size_t msg_bytes,
+ void *reply_data, size_t reply_bytes)
+{
int ret;
if (msg_bytes > SOF_IPC_MSG_MAX_SIZE ||
@@ -305,7 +332,7 @@ int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header,
return ret;
}
-EXPORT_SYMBOL(sof_ipc_tx_message);
+EXPORT_SYMBOL(sof_ipc_tx_message_no_pm);
/* handle reply message from DSP */
int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id)
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index fc4ab51bacf4..1f2e0be812bd 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -95,9 +95,6 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
/* process structure data */
switch (ext_hdr->type) {
- case SOF_IPC_EXT_DMA_BUFFER:
- ret = 0;
- break;
case SOF_IPC_EXT_WINDOW:
ret = get_ext_windows(sdev, ext_hdr);
break;
@@ -469,9 +466,6 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev)
const char *fw_filename;
int ret;
- /* set code loading condition to true */
- sdev->code_loading = 1;
-
/* Don't request firmware again if firmware is already requested */
if (plat_data->fw)
return 0;
diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index e929a6e0058f..a771500ac442 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -146,10 +146,11 @@ static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev)
return 0;
}
-static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev)
+static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev,
+ u32 target_state)
{
if (sof_ops(sdev)->suspend)
- return sof_ops(sdev)->suspend(sdev);
+ return sof_ops(sdev)->suspend(sdev, target_state);
return 0;
}
@@ -193,14 +194,15 @@ static inline int snd_sof_dsp_set_clk(struct snd_sof_dev *sdev, u32 freq)
return 0;
}
-static inline int snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev,
- enum sof_d0_substate substate)
+static inline int
+snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev,
+ const struct sof_dsp_power_state *target_state)
{
if (sof_ops(sdev)->set_power_state)
- return sof_ops(sdev)->set_power_state(sdev, substate);
+ return sof_ops(sdev)->set_power_state(sdev, target_state);
- /* D0 substate is not supported */
- return -ENOTSUPP;
+ /* D0 substate is not supported, do nothing here. */
+ return 0;
}
/* debug */
@@ -391,6 +393,49 @@ snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev,
return 0;
}
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+static inline int
+snd_sof_probe_compr_assign(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_assign(sdev, cstream, dai);
+}
+
+static inline int
+snd_sof_probe_compr_free(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_free(sdev, cstream, dai);
+}
+
+static inline int
+snd_sof_probe_compr_set_params(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_params *params, struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_set_params(sdev, cstream, params, dai);
+}
+
+static inline int
+snd_sof_probe_compr_trigger(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_trigger(sdev, cstream, cmd, dai);
+}
+
+static inline int
+snd_sof_probe_compr_pointer(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai)
+{
+ if (sof_ops(sdev) && sof_ops(sdev)->probe_pointer)
+ return sof_ops(sdev)->probe_pointer(sdev, cstream, tstamp, dai);
+
+ return 0;
+}
+#endif
+
/* machine driver */
static inline int
snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata)
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 29435ba2d329..47cd741f2a8c 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -16,6 +16,9 @@
#include "sof-priv.h"
#include "sof-audio.h"
#include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "compress.h"
+#endif
/* Create DMA buffer page table for DSP */
static int create_page_table(struct snd_soc_component *component,
@@ -54,7 +57,7 @@ static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream
/*
* sof pcm period elapse work
*/
-static void sof_pcm_period_elapsed_work(struct work_struct *work)
+void snd_sof_pcm_period_elapsed_work(struct work_struct *work)
{
struct snd_sof_pcm_stream *sps =
container_of(work, struct snd_sof_pcm_stream,
@@ -372,7 +375,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START;
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
- if (sdev->s0_suspend &&
+ if (sdev->system_suspend_target == SOF_SUSPEND_S0IX &&
spcm->stream[substream->stream].d0i3_compatible) {
/*
* trap the event, not sending trigger stop to
@@ -472,8 +475,6 @@ static int sof_pcm_open(struct snd_soc_component *component,
dev_dbg(component->dev, "pcm: open stream %d dir %d\n",
spcm->pcm.pcm_id, substream->stream);
- INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work,
- sof_pcm_period_elapsed_work);
caps = &spcm->pcm.caps[substream->stream];
@@ -598,8 +599,7 @@ static int sof_pcm_new(struct snd_soc_component *component,
snd_pcm_set_managed_buffer(pcm->streams[stream].substream,
SNDRV_DMA_TYPE_DEV_SG, sdev->dev,
- le32_to_cpu(caps->buffer_size_min),
- le32_to_cpu(caps->buffer_size_max));
+ 0, le32_to_cpu(caps->buffer_size_max));
capture:
stream = SNDRV_PCM_STREAM_CAPTURE;
@@ -621,8 +621,7 @@ capture:
snd_pcm_set_managed_buffer(pcm->streams[stream].substream,
SNDRV_DMA_TYPE_DEV_SG, sdev->dev,
- le32_to_cpu(caps->buffer_size_min),
- le32_to_cpu(caps->buffer_size_max));
+ 0, le32_to_cpu(caps->buffer_size_max));
return 0;
}
@@ -788,6 +787,10 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev)
#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS)
pd->compr_ops = &sof_compressed_ops;
#endif
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ /* override cops when probe support is enabled */
+ pd->compr_ops = &sof_probe_compressed_ops;
+#endif
pd->pcm_construct = sof_pcm_new;
pd->ignore_machine = drv_name;
pd->be_hw_params_fixup = sof_pcm_dai_link_fixup;
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index a0cde053b61a..c410822d9920 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -12,6 +12,42 @@
#include "sof-priv.h"
#include "sof-audio.h"
+/*
+ * Helper function to determine the target DSP state during
+ * system suspend. This function only cares about the device
+ * D-states. Platform-specific substates, if any, should be
+ * handled by the platform-specific parts.
+ */
+static u32 snd_sof_dsp_power_target(struct snd_sof_dev *sdev)
+{
+ u32 target_dsp_state;
+
+ switch (sdev->system_suspend_target) {
+ case SOF_SUSPEND_S3:
+ /* DSP should be in D3 if the system is suspending to S3 */
+ target_dsp_state = SOF_DSP_PM_D3;
+ break;
+ case SOF_SUSPEND_S0IX:
+ /*
+ * Currently, the only criterion for retaining the DSP in D0
+ * is that there are streams that ignored the suspend trigger.
+ * Additional criteria such Soundwire clock-stop mode and
+ * device suspend latency considerations will be added later.
+ */
+ if (snd_sof_stream_suspend_ignored(sdev))
+ target_dsp_state = SOF_DSP_PM_D0;
+ else
+ target_dsp_state = SOF_DSP_PM_D3;
+ break;
+ default:
+ /* This case would be during runtime suspend */
+ target_dsp_state = SOF_DSP_PM_D3;
+ break;
+ }
+
+ return target_dsp_state;
+}
+
static int sof_send_pm_ctx_ipc(struct snd_sof_dev *sdev, int cmd)
{
struct sof_ipc_pm_ctx pm_ctx;
@@ -50,6 +86,7 @@ static void sof_cache_debugfs(struct snd_sof_dev *sdev)
static int sof_resume(struct device *dev, bool runtime_resume)
{
struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+ u32 old_state = sdev->dsp_power_state.state;
int ret;
/* do nothing if dsp resume callbacks are not set */
@@ -74,6 +111,10 @@ static int sof_resume(struct device *dev, bool runtime_resume)
return ret;
}
+ /* Nothing further to do if resuming from a low-power D0 substate */
+ if (!runtime_resume && old_state == SOF_DSP_PM_D0)
+ return 0;
+
sdev->fw_state = SOF_FW_BOOT_PREPARE;
/* load the firmware */
@@ -124,15 +165,13 @@ static int sof_resume(struct device *dev, bool runtime_resume)
"error: ctx_restore ipc error during resume %d\n",
ret);
- /* initialize default D0 sub-state */
- sdev->d0_substate = SOF_DSP_D0I0;
-
return ret;
}
static int sof_suspend(struct device *dev, bool runtime_suspend)
{
struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+ u32 target_state = 0;
int ret;
/* do nothing if dsp suspend callback is not set */
@@ -140,10 +179,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
return 0;
if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)
- goto power_down;
-
- /* release trace */
- snd_sof_release_trace(sdev);
+ goto suspend;
/* set restore_stream for all streams during system suspend */
if (!runtime_suspend) {
@@ -156,6 +192,15 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
}
}
+ target_state = snd_sof_dsp_power_target(sdev);
+
+ /* Skip to platform-specific suspend if DSP is entering D0 */
+ if (target_state == SOF_DSP_PM_D0)
+ goto suspend;
+
+ /* release trace */
+ snd_sof_release_trace(sdev);
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE)
/* cache debugfs contents during runtime suspend */
if (runtime_suspend)
@@ -179,22 +224,26 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
ret);
}
-power_down:
+suspend:
/* return if the DSP was not probed successfully */
if (sdev->fw_state == SOF_FW_BOOT_NOT_STARTED)
return 0;
- /* power down all DSP cores */
+ /* platform-specific suspend */
if (runtime_suspend)
ret = snd_sof_dsp_runtime_suspend(sdev);
else
- ret = snd_sof_dsp_suspend(sdev);
+ ret = snd_sof_dsp_suspend(sdev, target_state);
if (ret < 0)
dev_err(sdev->dev,
"error: failed to power down DSP during suspend %d\n",
ret);
+ /* Do not reset FW state if DSP is in D0 */
+ if (target_state == SOF_DSP_PM_D0)
+ return ret;
+
/* reset FW state */
sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
@@ -221,112 +270,14 @@ int snd_sof_runtime_resume(struct device *dev)
}
EXPORT_SYMBOL(snd_sof_runtime_resume);
-int snd_sof_set_d0_substate(struct snd_sof_dev *sdev,
- enum sof_d0_substate d0_substate)
-{
- int ret;
-
- if (sdev->d0_substate == d0_substate)
- return 0;
-
- /* do platform specific set_state */
- ret = snd_sof_dsp_set_power_state(sdev, d0_substate);
- if (ret < 0)
- return ret;
-
- /* update dsp D0 sub-state */
- sdev->d0_substate = d0_substate;
-
- return 0;
-}
-EXPORT_SYMBOL(snd_sof_set_d0_substate);
-
-/*
- * Audio DSP states may transform as below:-
- *
- * D0I3 compatible stream
- * Runtime +---------------------+ opened only, timeout
- * suspend | +--------------------+
- * +------------+ D0(active) | |
- * | | <---------------+ |
- * | +--------> | | |
- * | |Runtime +--^--+---------^--+--+ The last | |
- * | |resume | | | | opened D0I3 | |
- * | | | | | | compatible | |
- * | | resume| | | | stream closed | |
- * | | from | | D3 | | | |
- * | | D3 | |suspend | | d0i3 | |
- * | | | | | |suspend | |
- * | | | | | | | |
- * | | | | | | | |
- * +-v---+-----------+--v-------+ | | +------+----v----+
- * | | | +-----------> |
- * | D3 (suspended) | | | D0I3 +-----+
- * | | +--------------+ | |
- * | | resume from | | |
- * +-------------------^--------+ d0i3 suspend +----------------+ |
- * | |
- * | D3 suspend |
- * +------------------------------------------------+
- *
- * d0i3_suspend = s0_suspend && D0I3 stream opened,
- * D3 suspend = !d0i3_suspend,
- */
-
int snd_sof_resume(struct device *dev)
{
- struct snd_sof_dev *sdev = dev_get_drvdata(dev);
- int ret;
-
- if (snd_sof_dsp_d0i3_on_suspend(sdev)) {
- /* resume from D0I3 */
- dev_dbg(sdev->dev, "DSP will exit from D0i3...\n");
- ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I0);
- if (ret == -ENOTSUPP) {
- /* fallback to resume from D3 */
- dev_dbg(sdev->dev, "D0i3 not supported, fall back to resume from D3...\n");
- goto d3_resume;
- } else if (ret < 0) {
- dev_err(sdev->dev, "error: failed to exit from D0I3 %d\n",
- ret);
- return ret;
- }
-
- /* platform-specific resume from D0i3 */
- return snd_sof_dsp_resume(sdev);
- }
-
-d3_resume:
- /* resume from D3 */
return sof_resume(dev, false);
}
EXPORT_SYMBOL(snd_sof_resume);
int snd_sof_suspend(struct device *dev)
{
- struct snd_sof_dev *sdev = dev_get_drvdata(dev);
- int ret;
-
- if (snd_sof_dsp_d0i3_on_suspend(sdev)) {
- /* suspend to D0i3 */
- dev_dbg(sdev->dev, "DSP is trying to enter D0i3...\n");
- ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3);
- if (ret == -ENOTSUPP) {
- /* fallback to D3 suspend */
- dev_dbg(sdev->dev, "D0i3 not supported, fall back to D3...\n");
- goto d3_suspend;
- } else if (ret < 0) {
- dev_err(sdev->dev, "error: failed to enter D0I3, %d\n",
- ret);
- return ret;
- }
-
- /* platform-specific suspend to D0i3 */
- return snd_sof_dsp_suspend(sdev);
- }
-
-d3_suspend:
- /* suspend to D3 */
return sof_suspend(dev, false);
}
EXPORT_SYMBOL(snd_sof_suspend);
@@ -336,10 +287,13 @@ int snd_sof_prepare(struct device *dev)
struct snd_sof_dev *sdev = dev_get_drvdata(dev);
#if defined(CONFIG_ACPI)
- sdev->s0_suspend = acpi_target_system_state() == ACPI_STATE_S0;
+ if (acpi_target_system_state() == ACPI_STATE_S0)
+ sdev->system_suspend_target = SOF_SUSPEND_S0IX;
+ else
+ sdev->system_suspend_target = SOF_SUSPEND_S3;
#else
/* will suspend to S3 by default */
- sdev->s0_suspend = false;
+ sdev->system_suspend_target = SOF_SUSPEND_S3;
#endif
return 0;
@@ -350,6 +304,6 @@ void snd_sof_complete(struct device *dev)
{
struct snd_sof_dev *sdev = dev_get_drvdata(dev);
- sdev->s0_suspend = false;
+ sdev->system_suspend_target = SOF_SUSPEND_NONE;
}
EXPORT_SYMBOL(snd_sof_complete);
diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c
new file mode 100644
index 000000000000..c38169fe00c5
--- /dev/null
+++ b/sound/soc/sof/probe.c
@@ -0,0 +1,290 @@
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license. When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <cezary.rojewski@intel.com>
+//
+
+#include "sof-priv.h"
+#include "probe.h"
+
+/**
+ * sof_ipc_probe_init - initialize data probing
+ * @sdev: SOF sound device
+ * @stream_tag: Extractor stream tag
+ * @buffer_size: DMA buffer size to set for extractor
+ *
+ * Host chooses whether extraction is supported or not by providing
+ * valid stream tag to DSP. Once specified, stream described by that
+ * tag will be tied to DSP for extraction for the entire lifetime of
+ * probe.
+ *
+ * Probing is initialized only once and each INIT request must be
+ * matched by DEINIT call.
+ */
+int sof_ipc_probe_init(struct snd_sof_dev *sdev,
+ u32 stream_tag, size_t buffer_size)
+{
+ struct sof_ipc_probe_dma_add_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, dma, 1);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_INIT;
+ msg->num_elems = 1;
+ msg->dma[0].stream_tag = stream_tag;
+ msg->dma[0].dma_buffer_size = buffer_size;
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_init);
+
+/**
+ * sof_ipc_probe_deinit - cleanup after data probing
+ * @sdev: SOF sound device
+ *
+ * Host sends DEINIT request to free previously initialized probe
+ * on DSP side once it is no longer needed. DEINIT only when there
+ * are no probes connected and with all injectors detached.
+ */
+int sof_ipc_probe_deinit(struct snd_sof_dev *sdev)
+{
+ struct sof_ipc_cmd_hdr msg;
+ struct sof_ipc_reply reply;
+
+ msg.size = sizeof(msg);
+ msg.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DEINIT;
+
+ return sof_ipc_tx_message(sdev->ipc, msg.cmd, &msg, msg.size,
+ &reply, sizeof(reply));
+}
+EXPORT_SYMBOL(sof_ipc_probe_deinit);
+
+static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd,
+ void **params, size_t *num_params)
+{
+ struct sof_ipc_probe_info_params msg = {{{0}}};
+ struct sof_ipc_probe_info_params *reply;
+ size_t bytes;
+ int ret;
+
+ *params = NULL;
+ *num_params = 0;
+
+ reply = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL);
+ if (!reply)
+ return -ENOMEM;
+ msg.rhdr.hdr.size = sizeof(msg);
+ msg.rhdr.hdr.cmd = SOF_IPC_GLB_PROBE | cmd;
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg.rhdr.hdr.cmd, &msg,
+ msg.rhdr.hdr.size, reply, SOF_IPC_MSG_MAX_SIZE);
+ if (ret < 0 || reply->rhdr.error < 0)
+ goto exit;
+
+ if (!reply->num_elems)
+ goto exit;
+
+ if (cmd == SOF_IPC_PROBE_DMA_INFO)
+ bytes = sizeof(reply->dma[0]);
+ else
+ bytes = sizeof(reply->desc[0]);
+ bytes *= reply->num_elems;
+ *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL);
+ if (!*params) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+ *num_params = reply->num_elems;
+
+exit:
+ kfree(reply);
+ return ret;
+}
+
+/**
+ * sof_ipc_probe_dma_info - retrieve list of active injection dmas
+ * @sdev: SOF sound device
+ * @dma: Returned list of active dmas
+ * @num_dma: Returned count of active dmas
+ *
+ * Host sends DMA_INFO request to obtain list of injection dmas it
+ * can use to transfer data over with.
+ *
+ * Note that list contains only injection dmas as there is only one
+ * extractor (dma) and it is always assigned on probing init.
+ * DSP knows exactly where data from extraction probes is going to,
+ * which is not the case for injection where multiple streams
+ * could be engaged.
+ */
+int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev,
+ struct sof_probe_dma **dma, size_t *num_dma)
+{
+ return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_DMA_INFO,
+ (void **)dma, num_dma);
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_info);
+
+/**
+ * sof_ipc_probe_dma_add - attach to specified dmas
+ * @sdev: SOF sound device
+ * @dma: List of streams (dmas) to attach to
+ * @num_dma: Number of elements in @dma
+ *
+ * Contrary to extraction, injection streams are never assigned
+ * on init. Before attempting any data injection, host is responsible
+ * for specifying streams which will be later used to transfer data
+ * to connected probe points.
+ */
+int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev,
+ struct sof_probe_dma *dma, size_t num_dma)
+{
+ struct sof_ipc_probe_dma_add_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, dma, num_dma);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_dma;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_ADD;
+ memcpy(&msg->dma[0], dma, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_add);
+
+/**
+ * sof_ipc_probe_dma_remove - detach from specified dmas
+ * @sdev: SOF sound device
+ * @stream_tag: List of stream tags to detach from
+ * @num_stream_tag: Number of elements in @stream_tag
+ *
+ * Host sends DMA_REMOVE request to free previously attached stream
+ * from being occupied for injection. Each detach operation should
+ * match equivalent DMA_ADD. Detach only when all probes tied to
+ * given stream have been disconnected.
+ */
+int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev,
+ unsigned int *stream_tag, size_t num_stream_tag)
+{
+ struct sof_ipc_probe_dma_remove_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, stream_tag, num_stream_tag);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_stream_tag;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_REMOVE;
+ memcpy(&msg->stream_tag[0], stream_tag, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_remove);
+
+/**
+ * sof_ipc_probe_points_info - retrieve list of active probe points
+ * @sdev: SOF sound device
+ * @desc: Returned list of active probes
+ * @num_desc: Returned count of active probes
+ *
+ * Host sends PROBE_POINT_INFO request to obtain list of active probe
+ * points, valid for disconnection when given probe is no longer
+ * required.
+ */
+int sof_ipc_probe_points_info(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc **desc, size_t *num_desc)
+{
+ return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_POINT_INFO,
+ (void **)desc, num_desc);
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_info);
+
+/**
+ * sof_ipc_probe_points_add - connect specified probes
+ * @sdev: SOF sound device
+ * @desc: List of probe points to connect
+ * @num_desc: Number of elements in @desc
+ *
+ * Dynamically connects to provided set of endpoints. Immediately
+ * after connection is established, host must be prepared to
+ * transfer data from or to target stream given the probing purpose.
+ *
+ * Each probe point should be removed using PROBE_POINT_REMOVE
+ * request when no longer needed.
+ */
+int sof_ipc_probe_points_add(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc *desc, size_t num_desc)
+{
+ struct sof_ipc_probe_point_add_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, desc, num_desc);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_desc;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_ADD;
+ memcpy(&msg->desc[0], desc, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_add);
+
+/**
+ * sof_ipc_probe_points_remove - disconnect specified probes
+ * @sdev: SOF sound device
+ * @buffer_id: List of probe points to disconnect
+ * @num_buffer_id: Number of elements in @desc
+ *
+ * Removes previously connected probes from list of active probe
+ * points and frees all resources on DSP side.
+ */
+int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev,
+ unsigned int *buffer_id, size_t num_buffer_id)
+{
+ struct sof_ipc_probe_point_remove_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, buffer_id, num_buffer_id);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_buffer_id;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_REMOVE;
+ memcpy(&msg->buffer_id[0], buffer_id, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_remove);
diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h
new file mode 100644
index 000000000000..45daa5552834
--- /dev/null
+++ b/sound/soc/sof/probe.h
@@ -0,0 +1,85 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/*
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+ *
+ * Author: Cezary Rojewski <cezary.rojewski@intel.com>
+ */
+
+#ifndef __SOF_PROBE_H
+#define __SOF_PROBE_H
+
+#include <sound/sof/header.h>
+
+struct snd_sof_dev;
+
+#define SOF_PROBE_INVALID_NODE_ID UINT_MAX
+
+struct sof_probe_dma {
+ unsigned int stream_tag;
+ unsigned int dma_buffer_size;
+} __packed;
+
+enum sof_connection_purpose {
+ SOF_CONNECTION_PURPOSE_EXTRACT = 1,
+ SOF_CONNECTION_PURPOSE_INJECT,
+};
+
+struct sof_probe_point_desc {
+ unsigned int buffer_id;
+ unsigned int purpose;
+ unsigned int stream_tag;
+} __packed;
+
+struct sof_ipc_probe_dma_add_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ struct sof_probe_dma dma[0];
+} __packed;
+
+struct sof_ipc_probe_info_params {
+ struct sof_ipc_reply rhdr;
+ unsigned int num_elems;
+ union {
+ struct sof_probe_dma dma[0];
+ struct sof_probe_point_desc desc[0];
+ };
+} __packed;
+
+struct sof_ipc_probe_dma_remove_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ unsigned int stream_tag[0];
+} __packed;
+
+struct sof_ipc_probe_point_add_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ struct sof_probe_point_desc desc[0];
+} __packed;
+
+struct sof_ipc_probe_point_remove_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ unsigned int buffer_id[0];
+} __packed;
+
+int sof_ipc_probe_init(struct snd_sof_dev *sdev,
+ u32 stream_tag, size_t buffer_size);
+int sof_ipc_probe_deinit(struct snd_sof_dev *sdev);
+int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev,
+ struct sof_probe_dma **dma, size_t *num_dma);
+int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev,
+ struct sof_probe_dma *dma, size_t num_dma);
+int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev,
+ unsigned int *stream_tag, size_t num_stream_tag);
+int sof_ipc_probe_points_info(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc **desc, size_t *num_desc);
+int sof_ipc_probe_points_add(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc *desc, size_t num_desc);
+int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev,
+ unsigned int *buffer_id, size_t num_buffer_id);
+
+#endif
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 0d8f65b9ae25..fc4ed2a8a914 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -11,7 +11,40 @@
#include "sof-audio.h"
#include "ops.h"
-bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev)
+/*
+ * helper to determine if there are only D0i3 compatible
+ * streams active
+ */
+bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_sof_pcm *spcm;
+ bool d0i3_compatible_active = false;
+ int dir;
+
+ list_for_each_entry(spcm, &sdev->pcm_list, list) {
+ for_each_pcm_streams(dir) {
+ substream = spcm->stream[dir].substream;
+ if (!substream || !substream->runtime)
+ continue;
+
+ /*
+ * substream->runtime being not NULL indicates that
+ * that the stream is open. No need to check the
+ * stream state.
+ */
+ if (!spcm->stream[dir].d0i3_compatible)
+ return false;
+
+ d0i3_compatible_active = true;
+ }
+ }
+
+ return d0i3_compatible_active;
+}
+EXPORT_SYMBOL(snd_sof_dsp_only_d0i3_compatible_stream_active);
+
+bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev)
{
struct snd_sof_pcm *spcm;
@@ -38,7 +71,14 @@ int sof_set_hw_params_upon_resume(struct device *dev)
* have been suspended.
*/
list_for_each_entry(spcm, &sdev->pcm_list, list) {
- for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
+ for_each_pcm_streams(dir) {
+ /*
+ * do not reset hw_params upon resume for streams that
+ * were kept running during suspend
+ */
+ if (spcm->stream[dir].suspend_ignored)
+ continue;
+
substream = spcm->stream[dir].substream;
if (!substream || !substream->runtime)
continue;
@@ -279,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp,
int dir;
list_for_each_entry(spcm, &sdev->pcm_list, list) {
- dir = SNDRV_PCM_STREAM_PLAYBACK;
- if (spcm->stream[dir].comp_id == comp_id) {
- *direction = dir;
- return spcm;
- }
-
- dir = SNDRV_PCM_STREAM_CAPTURE;
- if (spcm->stream[dir].comp_id == comp_id) {
- *direction = dir;
- return spcm;
+ for_each_pcm_streams(dir) {
+ if (spcm->stream[dir].comp_id == comp_id) {
+ *direction = dir;
+ return spcm;
+ }
}
}
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index a62fb2da6a6e..bf65f31af858 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -11,6 +11,8 @@
#ifndef __SOUND_SOC_SOF_AUDIO_H
#define __SOUND_SOC_SOF_AUDIO_H
+#include <linux/workqueue.h>
+
#include <sound/soc.h>
#include <sound/control.h>
#include <sound/sof/stream.h> /* needs to be included before control.h */
@@ -189,6 +191,7 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp,
struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_soc_component *scomp,
unsigned int pcm_id);
void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream);
+void snd_sof_pcm_period_elapsed_work(struct work_struct *work);
/*
* Mixer IPC
@@ -202,7 +205,8 @@ int snd_sof_ipc_set_get_comp_data(struct snd_sof_control *scontrol,
/* PM */
int sof_restore_pipelines(struct device *dev);
int sof_set_hw_params_upon_resume(struct device *dev);
-bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev);
+bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev);
+bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev);
/* Machine driver enumeration */
int sof_machine_register(struct snd_sof_dev *sdev, void *pdata);
diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c
index 39ea8af6213f..16e49f2ee629 100644
--- a/sound/soc/sof/sof-of-dev.c
+++ b/sound/soc/sof/sof-of-dev.c
@@ -13,12 +13,21 @@
#include "ops.h"
extern struct snd_sof_dsp_ops sof_imx8_ops;
+extern struct snd_sof_dsp_ops sof_imx8x_ops;
/* platform specific devices */
#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8)
static struct sof_dev_desc sof_of_imx8qxp_desc = {
.default_fw_path = "imx/sof",
.default_tplg_path = "imx/sof-tplg",
+ .default_fw_filename = "sof-imx8x.ri",
+ .nocodec_tplg_filename = "sof-imx8-nocodec.tplg",
+ .ops = &sof_imx8x_ops,
+};
+
+static struct sof_dev_desc sof_of_imx8qm_desc = {
+ .default_fw_path = "imx/sof",
+ .default_tplg_path = "imx/sof-tplg",
.default_fw_filename = "sof-imx8.ri",
.nocodec_tplg_filename = "sof-imx8-nocodec.tplg",
.ops = &sof_imx8_ops,
@@ -103,6 +112,7 @@ static int sof_of_remove(struct platform_device *pdev)
static const struct of_device_id sof_of_ids[] = {
#if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8)
{ .compatible = "fsl,imx8qxp-dsp", .data = &sof_of_imx8qxp_desc},
+ { .compatible = "fsl,imx8qm-dsp", .data = &sof_of_imx8qm_desc},
#endif
{ }
};
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index bc2337cf1142..a4b297c842df 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -54,10 +54,26 @@ extern int sof_core_debug;
(IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) || \
IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST))
-/* DSP D0ix sub-state */
-enum sof_d0_substate {
- SOF_DSP_D0I0 = 0, /* DSP default D0 substate */
- SOF_DSP_D0I3, /* DSP D0i3(low power) substate*/
+/* DSP power state */
+enum sof_dsp_power_states {
+ SOF_DSP_PM_D0,
+ SOF_DSP_PM_D1,
+ SOF_DSP_PM_D2,
+ SOF_DSP_PM_D3_HOT,
+ SOF_DSP_PM_D3,
+ SOF_DSP_PM_D3_COLD,
+};
+
+struct sof_dsp_power_state {
+ u32 state;
+ u32 substate; /* platform-specific */
+};
+
+/* System suspend target state */
+enum sof_system_suspend_state {
+ SOF_SUSPEND_NONE = 0,
+ SOF_SUSPEND_S0IX,
+ SOF_SUSPEND_S3,
};
struct snd_sof_dev;
@@ -154,6 +170,27 @@ struct snd_sof_dsp_ops {
snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream); /* optional */
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ /* Except for probe_pointer, all probe ops are mandatory */
+ int (*probe_assign)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_free)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_set_params)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_params *params,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_trigger)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_pointer)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp,
+ struct snd_soc_dai *dai); /* optional */
+#endif
+
/* host read DSP stream data */
void (*ipc_msg_data)(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream,
@@ -169,14 +206,15 @@ struct snd_sof_dsp_ops {
int (*post_fw_run)(struct snd_sof_dev *sof_dev); /* optional */
/* DSP PM */
- int (*suspend)(struct snd_sof_dev *sof_dev); /* optional */
+ int (*suspend)(struct snd_sof_dev *sof_dev,
+ u32 target_state); /* optional */
int (*resume)(struct snd_sof_dev *sof_dev); /* optional */
int (*runtime_suspend)(struct snd_sof_dev *sof_dev); /* optional */
int (*runtime_resume)(struct snd_sof_dev *sof_dev); /* optional */
int (*runtime_idle)(struct snd_sof_dev *sof_dev); /* optional */
int (*set_hw_params_upon_resume)(struct snd_sof_dev *sdev); /* optional */
int (*set_power_state)(struct snd_sof_dev *sdev,
- enum sof_d0_substate d0_substate); /* optional */
+ const struct sof_dsp_power_state *target_state); /* optional */
/* DSP clocking */
int (*set_clk)(struct snd_sof_dev *sof_dev, u32 freq); /* optional */
@@ -323,10 +361,11 @@ struct snd_sof_dev {
*/
struct snd_soc_component_driver plat_drv;
- /* power states related */
- enum sof_d0_substate d0_substate;
- /* flag to track if the intended power target of suspend is S0ix */
- bool s0_suspend;
+ /* current DSP power state */
+ struct sof_dsp_power_state dsp_power_state;
+
+ /* Intended power target of system suspend */
+ enum sof_system_suspend_state system_suspend_target;
/* DSP firmware boot */
wait_queue_head_t boot_wait;
@@ -376,16 +415,15 @@ struct snd_sof_dev {
u32 enabled_cores_mask; /* keep track of enabled cores */
/* FW configuration */
- struct sof_ipc_dma_buffer_data *info_buffer;
struct sof_ipc_window *info_window;
/* IPC timeouts in ms */
int ipc_timeout;
int boot_timeout;
- /* Wait queue for code loading */
- wait_queue_head_t waitq;
- int code_loading;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ unsigned int extractor_stream_tag;
+#endif
/* DMA for Trace */
struct snd_dma_buffer dmatb;
@@ -417,8 +455,6 @@ int snd_sof_resume(struct device *dev);
int snd_sof_suspend(struct device *dev);
int snd_sof_prepare(struct device *dev);
void snd_sof_complete(struct device *dev);
-int snd_sof_set_d0_substate(struct snd_sof_dev *sdev,
- enum sof_d0_substate d0_substate);
void snd_sof_new_platform_drv(struct snd_sof_dev *sdev);
@@ -454,6 +490,9 @@ int snd_sof_ipc_valid(struct snd_sof_dev *sdev);
int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header,
void *msg_data, size_t msg_bytes, void *reply_data,
size_t reply_bytes);
+int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header,
+ void *msg_data, size_t msg_bytes,
+ void *reply_data, size_t reply_bytes);
/*
* Trace/debug
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 9f4f8868b386..fe8ba3e05e08 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -9,6 +9,7 @@
//
#include <linux/firmware.h>
+#include <linux/workqueue.h>
#include <sound/tlv.h>
#include <sound/pcm_params.h>
#include <uapi/sound/sof/tokens.h>
@@ -1240,6 +1241,8 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp,
{
struct snd_soc_card *card = scomp->card;
struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dai *cpu_dai;
+ int i;
list_for_each_entry(rtd, &card->rtd_list, list) {
dev_vdbg(scomp->dev, "tplg: check widget: %s stream: %s dai stream: %s\n",
@@ -1254,13 +1257,15 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp,
switch (w->id) {
case snd_soc_dapm_dai_out:
- rtd->cpu_dai->capture_widget = w;
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+ cpu_dai->capture_widget = w;
dai->name = rtd->dai_link->name;
dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n",
w->name, rtd->dai_link->name);
break;
case snd_soc_dapm_dai_in:
- rtd->cpu_dai->playback_widget = w;
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+ cpu_dai->playback_widget = w;
dai->name = rtd->dai_link->name;
dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n",
w->name, rtd->dai_link->name);
@@ -2444,7 +2449,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
struct snd_soc_tplg_stream_caps *caps;
struct snd_soc_tplg_private *private = &pcm->priv;
struct snd_sof_pcm *spcm;
- int stream = SNDRV_PCM_STREAM_PLAYBACK;
+ int stream;
int ret = 0;
/* nothing to do for BEs atm */
@@ -2456,8 +2461,12 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
return -ENOMEM;
spcm->scomp = scomp;
- spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED;
- spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED;
+
+ for_each_pcm_streams(stream) {
+ spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED;
+ INIT_WORK(&spcm->stream[stream].period_elapsed_work,
+ snd_sof_pcm_period_elapsed_work);
+ }
spcm->pcm = *pcm;
dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name);
@@ -2478,8 +2487,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
if (!spcm->pcm.playback)
goto capture;
+ stream = SNDRV_PCM_STREAM_PLAYBACK;
+
dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: playback d0i3:%d\n",
- spcm->pcm.pcm_name, spcm->stream[0].d0i3_compatible);
+ spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible);
caps = &spcm->pcm.caps[stream];
@@ -2509,7 +2520,7 @@ capture:
return ret;
dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: capture d0i3:%d\n",
- spcm->pcm.pcm_name, spcm->stream[1].d0i3_compatible);
+ spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible);
caps = &spcm->pcm.caps[stream];
diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig
index 5474fd3de8c0..5e0ac8278572 100644
--- a/sound/soc/sprd/Kconfig
+++ b/sound/soc/sprd/Kconfig
@@ -8,7 +8,7 @@ config SND_SOC_SPRD
the Spreadtrum SoCs' Audio interfaces.
config SND_SOC_SPRD_MCDT
- bool "Spreadtrum multi-channel data transfer support"
+ tristate "Spreadtrum multi-channel data transfer support"
depends on SND_SOC_SPRD
help
Say y here to enable multi-channel data transfer support. It
diff --git a/sound/soc/sprd/sprd-mcdt.h b/sound/soc/sprd/sprd-mcdt.h
index 9cc7e207ac76..679e3af3baad 100644
--- a/sound/soc/sprd/sprd-mcdt.h
+++ b/sound/soc/sprd/sprd-mcdt.h
@@ -48,7 +48,7 @@ struct sprd_mcdt_chan {
struct list_head list;
};
-#ifdef CONFIG_SND_SOC_SPRD_MCDT
+#if IS_ENABLED(CONFIG_SND_SOC_SPRD_MCDT)
struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel,
enum sprd_mcdt_channel_type type);
void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan);
diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c
index 6cddf551bc11..74d48340cade 100644
--- a/sound/soc/sprd/sprd-pcm-compress.c
+++ b/sound/soc/sprd/sprd-pcm-compress.c
@@ -135,7 +135,7 @@ static int sprd_platform_compr_dma_config(struct snd_compr_stream *cstream,
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct device *dev = component->dev;
- struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct sprd_pcm_dma_params *dma_params = data->dma_params;
struct sprd_compr_dma *dma = &stream->dma[channel];
struct dma_slave_config config = { };
@@ -321,7 +321,7 @@ static int sprd_platform_compr_open(struct snd_compr_stream *cstream)
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct device *dev = component->dev;
- struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct sprd_compr_stream *stream;
struct sprd_compr_callback cb;
int stream_id = cstream->direction, ret;
diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c
index 2284558684bc..d12d3cad8cbd 100644
--- a/sound/soc/sprd/sprd-pcm-dma.c
+++ b/sound/soc/sprd/sprd-pcm-dma.c
@@ -200,7 +200,7 @@ static int sprd_pcm_hw_params(struct snd_soc_component *component,
unsigned long flags;
int ret, i, j, sg_num;
- dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
if (!dma_params) {
dev_warn(component->dev, "no dma parameters setting\n");
dma_private->params = NULL;
diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c
index 51407a21c440..16ff02953015 100644
--- a/sound/soc/stm/stm32_adfsdm.c
+++ b/sound/soc/stm/stm32_adfsdm.c
@@ -215,7 +215,7 @@ static int stm32_adfsdm_trigger(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct stm32_adfsdm_priv *priv =
- snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -235,7 +235,7 @@ static int stm32_adfsdm_pcm_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
int ret;
ret = snd_soc_set_runtime_hwparams(substream, &stm32_adfsdm_pcm_hw);
@@ -250,7 +250,7 @@ static int stm32_adfsdm_pcm_close(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct stm32_adfsdm_priv *priv =
- snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
priv->substream = NULL;
@@ -263,7 +263,7 @@ static snd_pcm_uframes_t stm32_adfsdm_pcm_pointer(
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct stm32_adfsdm_priv *priv =
- snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
return bytes_to_frames(substream->runtime, priv->pos);
}
@@ -274,7 +274,7 @@ static int stm32_adfsdm_pcm_hw_params(struct snd_soc_component *component,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct stm32_adfsdm_priv *priv =
- snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
priv->pcm_buff = substream->runtime->dma_area;
@@ -287,7 +287,7 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_component *component,
{
struct snd_pcm *pcm = rtd->pcm;
struct stm32_adfsdm_priv *priv =
- snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
unsigned int size = DFSDM_MAX_PERIODS * DFSDM_MAX_PERIOD_SIZE;
snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c
index 3e7226a53e53..7c4d63c33f15 100644
--- a/sound/soc/stm/stm32_i2s.c
+++ b/sound/soc/stm/stm32_i2s.c
@@ -831,25 +831,33 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev,
/* Get clocks */
i2s->pclk = devm_clk_get(&pdev->dev, "pclk");
if (IS_ERR(i2s->pclk)) {
- dev_err(&pdev->dev, "Could not get pclk\n");
+ if (PTR_ERR(i2s->pclk) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Could not get pclk: %ld\n",
+ PTR_ERR(i2s->pclk));
return PTR_ERR(i2s->pclk);
}
i2s->i2sclk = devm_clk_get(&pdev->dev, "i2sclk");
if (IS_ERR(i2s->i2sclk)) {
- dev_err(&pdev->dev, "Could not get i2sclk\n");
+ if (PTR_ERR(i2s->i2sclk) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Could not get i2sclk: %ld\n",
+ PTR_ERR(i2s->i2sclk));
return PTR_ERR(i2s->i2sclk);
}
i2s->x8kclk = devm_clk_get(&pdev->dev, "x8k");
if (IS_ERR(i2s->x8kclk)) {
- dev_err(&pdev->dev, "missing x8k parent clock\n");
+ if (PTR_ERR(i2s->x8kclk) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Could not get x8k parent clock: %ld\n",
+ PTR_ERR(i2s->x8kclk));
return PTR_ERR(i2s->x8kclk);
}
i2s->x11kclk = devm_clk_get(&pdev->dev, "x11k");
if (IS_ERR(i2s->x11kclk)) {
- dev_err(&pdev->dev, "missing x11k parent clock\n");
+ if (PTR_ERR(i2s->x11kclk) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Could not get x11k parent clock: %ld\n",
+ PTR_ERR(i2s->x11kclk));
return PTR_ERR(i2s->x11kclk);
}
@@ -866,12 +874,24 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev,
}
/* Reset */
- rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
- if (!IS_ERR(rst)) {
- reset_control_assert(rst);
- udelay(2);
- reset_control_deassert(rst);
+ rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL);
+ if (IS_ERR(rst)) {
+ if (PTR_ERR(rst) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Reset controller error %ld\n",
+ PTR_ERR(rst));
+ return PTR_ERR(rst);
}
+ reset_control_assert(rst);
+ udelay(2);
+ reset_control_deassert(rst);
+
+ return 0;
+}
+
+static int stm32_i2s_remove(struct platform_device *pdev)
+{
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
@@ -903,42 +923,51 @@ static int stm32_i2s_probe(struct platform_device *pdev)
i2s->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "pclk",
i2s->base, i2s->regmap_conf);
if (IS_ERR(i2s->regmap)) {
- dev_err(&pdev->dev, "regmap init failed\n");
+ if (PTR_ERR(i2s->regmap) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Regmap init error %ld\n",
+ PTR_ERR(i2s->regmap));
return PTR_ERR(i2s->regmap);
}
- ret = devm_snd_soc_register_component(&pdev->dev, &stm32_i2s_component,
- i2s->dai_drv, 1);
- if (ret)
+ ret = snd_dmaengine_pcm_register(&pdev->dev, &stm32_i2s_pcm_config, 0);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "PCM DMA register error %d\n", ret);
return ret;
+ }
- ret = devm_snd_dmaengine_pcm_register(&pdev->dev,
- &stm32_i2s_pcm_config, 0);
- if (ret)
+ ret = snd_soc_register_component(&pdev->dev, &stm32_i2s_component,
+ i2s->dai_drv, 1);
+ if (ret) {
+ snd_dmaengine_pcm_unregister(&pdev->dev);
return ret;
+ }
/* Set SPI/I2S in i2s mode */
ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG,
I2S_CGFR_I2SMOD, I2S_CGFR_I2SMOD);
if (ret)
- return ret;
+ goto error;
ret = regmap_read(i2s->regmap, STM32_I2S_IPIDR_REG, &val);
if (ret)
- return ret;
+ goto error;
if (val == I2S_IPIDR_NUMBER) {
ret = regmap_read(i2s->regmap, STM32_I2S_HWCFGR_REG, &val);
if (ret)
- return ret;
+ goto error;
if (!FIELD_GET(I2S_HWCFGR_I2S_SUPPORT_MASK, val)) {
dev_err(&pdev->dev,
"Device does not support i2s mode\n");
- return -EPERM;
+ ret = -EPERM;
+ goto error;
}
ret = regmap_read(i2s->regmap, STM32_I2S_VERR_REG, &val);
+ if (ret)
+ goto error;
dev_dbg(&pdev->dev, "I2S version: %lu.%lu registered\n",
FIELD_GET(I2S_VERR_MAJ_MASK, val),
@@ -946,6 +975,11 @@ static int stm32_i2s_probe(struct platform_device *pdev)
}
return ret;
+
+error:
+ stm32_i2s_remove(pdev);
+
+ return ret;
}
MODULE_DEVICE_TABLE(of, stm32_i2s_ids);
@@ -981,6 +1015,7 @@ static struct platform_driver stm32_i2s_driver = {
.pm = &stm32_i2s_pm_ops,
},
.probe = stm32_i2s_probe,
+ .remove = stm32_i2s_remove,
};
module_platform_driver(stm32_i2s_driver);
diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index e20267504b16..058757c721f0 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -174,20 +174,26 @@ static int stm32_sai_probe(struct platform_device *pdev)
if (!STM_SAI_IS_F4(sai)) {
sai->pclk = devm_clk_get(&pdev->dev, "pclk");
if (IS_ERR(sai->pclk)) {
- dev_err(&pdev->dev, "missing bus clock pclk\n");
+ if (PTR_ERR(sai->pclk) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "missing bus clock pclk: %ld\n",
+ PTR_ERR(sai->pclk));
return PTR_ERR(sai->pclk);
}
}
sai->clk_x8k = devm_clk_get(&pdev->dev, "x8k");
if (IS_ERR(sai->clk_x8k)) {
- dev_err(&pdev->dev, "missing x8k parent clock\n");
+ if (PTR_ERR(sai->clk_x8k) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "missing x8k parent clock: %ld\n",
+ PTR_ERR(sai->clk_x8k));
return PTR_ERR(sai->clk_x8k);
}
sai->clk_x11k = devm_clk_get(&pdev->dev, "x11k");
if (IS_ERR(sai->clk_x11k)) {
- dev_err(&pdev->dev, "missing x11k parent clock\n");
+ if (PTR_ERR(sai->clk_x11k) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "missing x11k parent clock: %ld\n",
+ PTR_ERR(sai->clk_x11k));
return PTR_ERR(sai->clk_x11k);
}
@@ -197,12 +203,16 @@ static int stm32_sai_probe(struct platform_device *pdev)
return sai->irq;
/* reset */
- rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
- if (!IS_ERR(rst)) {
- reset_control_assert(rst);
- udelay(2);
- reset_control_deassert(rst);
+ rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL);
+ if (IS_ERR(rst)) {
+ if (PTR_ERR(rst) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Reset controller error %ld\n",
+ PTR_ERR(rst));
+ return PTR_ERR(rst);
}
+ reset_control_assert(rst);
+ udelay(2);
+ reset_control_deassert(rst);
/* Enable peripheral clock to allow register access */
ret = clk_prepare_enable(sai->pclk);
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 10eb4b8e8e7e..2bd280c01c33 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -1238,7 +1238,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
int *ptr = (int *)(runtime->dma_area + hwoff +
channel * (runtime->dma_bytes / runtime->channels));
@@ -1380,7 +1380,9 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
sai->regmap = devm_regmap_init_mmio(&pdev->dev, base,
sai->regmap_config);
if (IS_ERR(sai->regmap)) {
- dev_err(&pdev->dev, "Failed to initialize MMIO\n");
+ if (PTR_ERR(sai->regmap) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Regmap init error %ld\n",
+ PTR_ERR(sai->regmap));
return PTR_ERR(sai->regmap);
}
@@ -1471,7 +1473,9 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
of_node_put(args.np);
sai->sai_ck = devm_clk_get(&pdev->dev, "sai_ck");
if (IS_ERR(sai->sai_ck)) {
- dev_err(&pdev->dev, "Missing kernel clock sai_ck\n");
+ if (PTR_ERR(sai->sai_ck) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Missing kernel clock sai_ck: %ld\n",
+ PTR_ERR(sai->sai_ck));
return PTR_ERR(sai->sai_ck);
}
@@ -1545,7 +1549,8 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
if (ret) {
- dev_err(&pdev->dev, "Could not register pcm dma\n");
+ if (ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Could not register pcm dma\n");
return ret;
}
diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c
index 3769d9ce5dbe..1bfa3b2ba974 100644
--- a/sound/soc/stm/stm32_spdifrx.c
+++ b/sound/soc/stm/stm32_spdifrx.c
@@ -406,7 +406,9 @@ static int stm32_spdifrx_dma_ctrl_register(struct device *dev,
spdifrx->ctrl_chan = dma_request_chan(dev, "rx-ctrl");
if (IS_ERR(spdifrx->ctrl_chan)) {
- dev_err(dev, "dma_request_slave_channel failed\n");
+ if (PTR_ERR(spdifrx->ctrl_chan) != -EPROBE_DEFER)
+ dev_err(dev, "dma_request_slave_channel error %ld\n",
+ PTR_ERR(spdifrx->ctrl_chan));
return PTR_ERR(spdifrx->ctrl_chan);
}
@@ -929,7 +931,9 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev,
spdifrx->kclk = devm_clk_get(&pdev->dev, "kclk");
if (IS_ERR(spdifrx->kclk)) {
- dev_err(&pdev->dev, "Could not get kclk\n");
+ if (PTR_ERR(spdifrx->kclk) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Could not get kclk: %ld\n",
+ PTR_ERR(spdifrx->kclk));
return PTR_ERR(spdifrx->kclk);
}
@@ -940,6 +944,22 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev,
return 0;
}
+static int stm32_spdifrx_remove(struct platform_device *pdev)
+{
+ struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev);
+
+ if (spdifrx->ctrl_chan)
+ dma_release_channel(spdifrx->ctrl_chan);
+
+ if (spdifrx->dmab)
+ snd_dma_free_pages(spdifrx->dmab);
+
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+
+ return 0;
+}
+
static int stm32_spdifrx_probe(struct platform_device *pdev)
{
struct stm32_spdifrx_data *spdifrx;
@@ -967,7 +987,9 @@ static int stm32_spdifrx_probe(struct platform_device *pdev)
spdifrx->base,
spdifrx->regmap_conf);
if (IS_ERR(spdifrx->regmap)) {
- dev_err(&pdev->dev, "Regmap init failed\n");
+ if (PTR_ERR(spdifrx->regmap) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Regmap init error %ld\n",
+ PTR_ERR(spdifrx->regmap));
return PTR_ERR(spdifrx->regmap);
}
@@ -978,37 +1000,46 @@ static int stm32_spdifrx_probe(struct platform_device *pdev)
return ret;
}
- rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
- if (!IS_ERR(rst)) {
- reset_control_assert(rst);
- udelay(2);
- reset_control_deassert(rst);
+ rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL);
+ if (IS_ERR(rst)) {
+ if (PTR_ERR(rst) != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "Reset controller error %ld\n",
+ PTR_ERR(rst));
+ return PTR_ERR(rst);
}
+ reset_control_assert(rst);
+ udelay(2);
+ reset_control_deassert(rst);
- ret = devm_snd_soc_register_component(&pdev->dev,
- &stm32_spdifrx_component,
- stm32_spdifrx_dai,
- ARRAY_SIZE(stm32_spdifrx_dai));
- if (ret)
+ pcm_config = &stm32_spdifrx_pcm_config;
+ ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0);
+ if (ret) {
+ if (ret != -EPROBE_DEFER)
+ dev_err(&pdev->dev, "PCM DMA register error %d\n", ret);
return ret;
+ }
+
+ ret = snd_soc_register_component(&pdev->dev,
+ &stm32_spdifrx_component,
+ stm32_spdifrx_dai,
+ ARRAY_SIZE(stm32_spdifrx_dai));
+ if (ret) {
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+ return ret;
+ }
ret = stm32_spdifrx_dma_ctrl_register(&pdev->dev, spdifrx);
if (ret)
goto error;
- pcm_config = &stm32_spdifrx_pcm_config;
- ret = devm_snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0);
- if (ret) {
- dev_err(&pdev->dev, "PCM DMA register returned %d\n", ret);
- goto error;
- }
-
ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_IDR, &idr);
if (ret)
goto error;
if (idr == SPDIFRX_IPIDR_NUMBER) {
ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver);
+ if (ret)
+ goto error;
dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n",
FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver),
@@ -1018,27 +1049,11 @@ static int stm32_spdifrx_probe(struct platform_device *pdev)
return ret;
error:
- if (!IS_ERR(spdifrx->ctrl_chan))
- dma_release_channel(spdifrx->ctrl_chan);
- if (spdifrx->dmab)
- snd_dma_free_pages(spdifrx->dmab);
+ stm32_spdifrx_remove(pdev);
return ret;
}
-static int stm32_spdifrx_remove(struct platform_device *pdev)
-{
- struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev);
-
- if (spdifrx->ctrl_chan)
- dma_release_channel(spdifrx->ctrl_chan);
-
- if (spdifrx->dmab)
- snd_dma_free_pages(spdifrx->dmab);
-
- return 0;
-}
-
MODULE_DEVICE_TABLE(of, stm32_spdifrx_ids);
#ifdef CONFIG_PM_SLEEP
diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c
index 98a9fe645521..86779a99df75 100644
--- a/sound/soc/sunxi/sun4i-spdif.c
+++ b/sound/soc/sunxi/sun4i-spdif.c
@@ -244,7 +244,7 @@ static int sun4i_spdif_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
return -EINVAL;
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 686561df8e13..ca51af114419 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -86,7 +86,6 @@
#define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK GENMASK(12, 9)
struct sun8i_codec {
- struct device *dev;
struct regmap *regmap;
struct clk *clk_module;
struct clk *clk_bus;
@@ -542,8 +541,6 @@ static int sun8i_codec_probe(struct platform_device *pdev)
if (!scodec)
return -ENOMEM;
- scodec->dev = &pdev->dev;
-
scodec->clk_module = devm_clk_get(&pdev->dev, "mod");
if (IS_ERR(scodec->clk_module)) {
dev_err(&pdev->dev, "Failed to get the module clock\n");
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 9e8b1497efd3..ec39ecba1e8b 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -37,7 +37,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
int srate, mclk;
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
index 4954a33ff46b..d800b62b36f8 100644
--- a/sound/soc/tegra/tegra_max98090.c
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -38,7 +38,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index d46915a3ec4c..9878bc3eb89e 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -40,7 +40,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index 81cb6cc6236e..5821313db977 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -42,7 +42,7 @@ static int tegra_rt5677_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk, err;
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index e13b81d29cf3..dc411ba2e36d 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -36,7 +36,7 @@ static int tegra_sgtl5000_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_sgtl5000 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index f6dd790dad71..0d653a605358 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -40,7 +40,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index f08d3489c3cf..9b5651502f12 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -45,7 +45,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
int srate, mclk;
@@ -143,19 +143,37 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w,
return 0;
}
+static int tegra_wm8903_event_int_mic(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
+
+ if (!gpio_is_valid(machine->gpio_int_mic_en))
+ return 0;
+
+ gpio_set_value_cansleep(machine->gpio_int_mic_en,
+ SND_SOC_DAPM_EVENT_ON(event));
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Int Spk", tegra_wm8903_event_int_spk),
SND_SOC_DAPM_HP("Headphone Jack", tegra_wm8903_event_hp),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", tegra_wm8903_event_int_mic),
};
static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
SOC_DAPM_PIN_SWITCH("Int Spk"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
};
static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_component *component = codec_dai->component;
struct snd_soc_card *card = rtd->card;
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
@@ -187,7 +205,7 @@ static int tegra_wm8903_remove(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd =
snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_component *component = codec_dai->component;
wm8903_mic_detect(component, NULL, 0, 0);
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 3f67ddd13674..f9834afaa2e8 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -35,7 +35,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card);
int srate, mclk;
diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index 29f61053ab62..c5408c129f34 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0-only
menu "Audio support for Texas Instruments SoCs"
-depends on DMA_OMAP || TI_EDMA || COMPILE_TEST
+depends on DMA_OMAP || TI_EDMA || TI_K3_UDMA || COMPILE_TEST
config SND_SOC_TI_EDMA_PCM
tristate
@@ -10,6 +10,10 @@ config SND_SOC_TI_SDMA_PCM
tristate
select SND_SOC_GENERIC_DMAENGINE_PCM
+config SND_SOC_TI_UDMA_PCM
+ tristate
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+
comment "Texas Instruments DAI support for:"
config SND_SOC_DAVINCI_ASP
tristate "daVinci Audio Serial Port (ASP) or McBSP support"
@@ -24,6 +28,7 @@ config SND_SOC_DAVINCI_MCASP
tristate "Multichannel Audio Serial Port (McASP) support"
select SND_SOC_TI_EDMA_PCM
select SND_SOC_TI_SDMA_PCM
+ select SND_SOC_TI_UDMA_PCM
help
Say Y or M here if you want to have support for McASP IP found in
various Texas Instruments SoCs like:
@@ -31,6 +36,7 @@ config SND_SOC_DAVINCI_MCASP
- Sitara line of SoCs (AM335x, AM438x, etc)
- DRA7x devices
- Keystone devices
+ - K3 devices (am654, j721e)
config SND_SOC_DAVINCI_VCIF
tristate "daVinci Voice Interface (VCIF) support"
diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile
index 08c44d56ef3e..ea48c6679cc7 100644
--- a/sound/soc/ti/Makefile
+++ b/sound/soc/ti/Makefile
@@ -3,9 +3,11 @@
# Platform drivers
snd-soc-ti-edma-objs := edma-pcm.o
snd-soc-ti-sdma-objs := sdma-pcm.o
+snd-soc-ti-udma-objs := udma-pcm.o
obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o
obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o
+obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o
# CPU DAI drivers
snd-soc-davinci-asp-objs := davinci-i2s.o
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index 8e2fb81ad05c..e17cd5e939f0 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -460,14 +460,14 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream)
static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct snd_soc_dapm_context *dapm = &card->dapm;
int ret;
/* Codec is ready, now add/activate board specific controls */
/* Store a pointer to the codec structure for tty ldisc use */
- cx20442_codec = rtd->codec_dai->component;
+ cx20442_codec = asoc_rtd_to_codec(rtd, 0)->component;
/* Add hook switch - can be used to control the codec from userspace
* even if line discipline fails */
diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
index 686b23d7a90d..2cfbeebdfb41 100644
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -54,8 +54,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_soc_card *soc_card = rtd->card;
int ret = 0;
unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *)
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index e1e937eb1dc1..734ffe925c4d 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -38,6 +38,7 @@
#include "edma-pcm.h"
#include "sdma-pcm.h"
+#include "udma-pcm.h"
#include "davinci-mcasp.h"
#define MCASP_MAX_AFIFO_DEPTH 64
@@ -1764,10 +1765,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of(
} else if (match) {
pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata),
GFP_KERNEL);
- if (!pdata) {
- ret = -ENOMEM;
- return pdata;
- }
+ if (!pdata)
+ return NULL;
} else {
/* control shouldn't reach here. something is wrong */
ret = -EINVAL;
@@ -1875,6 +1874,7 @@ nodata:
enum {
PCM_EDMA,
PCM_SDMA,
+ PCM_UDMA,
};
static const char *sdma_prefix = "ti,omap";
@@ -1912,6 +1912,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp);
if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix)))
return PCM_SDMA;
+ else if (strstr(tmp, "udmap"))
+ return PCM_UDMA;
return PCM_EDMA;
}
@@ -2371,6 +2373,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
case PCM_SDMA:
ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
break;
+ case PCM_UDMA:
+ ret = udma_pcm_platform_register(&pdev->dev);
+ break;
default:
dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret);
case -EPROBE_DEFER:
diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c
index c84650e4a7aa..ee4d3ef821a1 100644
--- a/sound/soc/ti/davinci-vcif.c
+++ b/sound/soc/ti/davinci-vcif.c
@@ -43,7 +43,7 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct davinci_vcif_dev *davinci_vcif_dev =
- snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
u32 w;
@@ -62,7 +62,7 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct davinci_vcif_dev *davinci_vcif_dev =
- snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
u32 w;
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 3ad2b6daf31e..a1672b479cb7 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -101,7 +101,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int err;
/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c
index 6d564ab5e437..61e45fea5dd8 100644
--- a/sound/soc/ti/omap-abe-twl6040.c
+++ b/sound/soc/ti/omap-abe-twl6040.c
@@ -46,7 +46,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
struct snd_soc_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int clk_id, freq;
@@ -78,7 +78,7 @@ static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret = 0;
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
@@ -166,7 +166,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_component *component = rtd->codec_dai->component;
+ struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
struct snd_soc_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
index 1a3fe854e856..5a32b54bbf3b 100644
--- a/sound/soc/ti/omap-mcbsp-st.c
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -489,7 +489,7 @@ OMAP_MCBSP_ST_CONTROLS(3);
int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
if (!mcbsp->st_data) {
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 302d5c493c29..3d41ca2238d4 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -737,7 +737,7 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream,
unsigned int packet_size)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
int words;
@@ -902,7 +902,7 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay(
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
u16 fifo_use;
snd_pcm_sframes_t delay;
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index d7ac4df6f2d9..f2dbadea33bb 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -532,7 +532,7 @@ static const struct snd_soc_component_driver omap_mcpdm_component = {
void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
u8 rx1, u8 rx2)
{
- struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+ struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2);
}
diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c
index 545f8dac9bd5..b04146311b31 100644
--- a/sound/soc/ti/omap3pandora.c
+++ b/sound/soc/ti/omap3pandora.c
@@ -32,8 +32,8 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret;
/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c
index 1ca466bc4025..e01485cc51a1 100644
--- a/sound/soc/ti/osk5912.c
+++ b/sound/soc/ti/osk5912.c
@@ -39,7 +39,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int err;
/* Set the codec system clock for DAC and ADC */
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index fdb0dc85fe67..2a714a004163 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -103,7 +103,7 @@ static int rx51_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
/* Set the codec system clock for DAC and ADC */
return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c
new file mode 100644
index 000000000000..39830caaaf7c
--- /dev/null
+++ b/sound/soc/ti/udma-pcm.c
@@ -0,0 +1,43 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "udma-pcm.h"
+
+static const struct snd_pcm_hardware udma_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .buffer_bytes_max = SIZE_MAX,
+ .period_bytes_min = 32,
+ .period_bytes_max = SZ_64K,
+ .periods_min = 2,
+ .periods_max = UINT_MAX,
+};
+
+static const struct snd_dmaengine_pcm_config udma_dmaengine_pcm_config = {
+ .pcm_hardware = &udma_pcm_hardware,
+ .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+};
+
+int udma_pcm_platform_register(struct device *dev)
+{
+ return devm_snd_dmaengine_pcm_register(dev, &udma_dmaengine_pcm_config,
+ 0);
+}
+EXPORT_SYMBOL_GPL(udma_pcm_platform_register);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("UDMA PCM ASoC platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h
new file mode 100644
index 000000000000..54111e7312c1
--- /dev/null
+++ b/sound/soc/ti/udma-pcm.h
@@ -0,0 +1,18 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ */
+
+#ifndef __UDMA_PCM_H__
+#define __UDMA_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_SOC_TI_UDMA_PCM)
+int udma_pcm_platform_register(struct device *dev);
+#else
+static inline int udma_pcm_platform_register(struct device *dev)
+{
+ return 0;
+}
+#endif /* CONFIG_SND_SOC_TI_UDMA_PCM */
+
+#endif /* __UDMA_PCM_H__ */
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 985487cc3a55..4b1cd4da3e36 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -269,7 +269,7 @@ static int txx9aclc_pcm_new(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
- struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_pcm *pcm = rtd->pcm;
struct platform_device *pdev = to_platform_device(component->dev);
struct txx9aclc_soc_device *dev;
diff --git a/sound/soc/uniphier/aio-compress.c b/sound/soc/uniphier/aio-compress.c
index 17f773ac5ca1..232d3cc5bce0 100644
--- a/sound/soc/uniphier/aio-compress.c
+++ b/sound/soc/uniphier/aio-compress.c
@@ -23,7 +23,7 @@ static int uniphier_aio_comprdma_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_compr *compr = rtd->compr;
struct device *dev = compr->card->dev;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[compr->direction];
size_t size = AUD_RING_SIZE;
int dma_dir = DMA_FROM_DEVICE, ret;
@@ -56,7 +56,7 @@ static int uniphier_aio_comprdma_free(struct snd_soc_pcm_runtime *rtd)
{
struct snd_compr *compr = rtd->compr;
struct device *dev = compr->card->dev;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[compr->direction];
int dma_dir = DMA_FROM_DEVICE;
@@ -73,7 +73,7 @@ static int uniphier_aio_comprdma_free(struct snd_soc_pcm_runtime *rtd)
static int uniphier_aio_compr_open(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
int ret;
@@ -98,7 +98,7 @@ static int uniphier_aio_compr_open(struct snd_compr_stream *cstream)
static int uniphier_aio_compr_free(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
int ret;
@@ -118,7 +118,7 @@ static int uniphier_aio_compr_get_params(struct snd_compr_stream *cstream,
struct snd_codec *params)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
*params = sub->cparams.codec;
@@ -130,7 +130,7 @@ static int uniphier_aio_compr_set_params(struct snd_compr_stream *cstream,
struct snd_compr_params *params)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
struct device *dev = &aio->chip->pdev->dev;
int ret;
@@ -165,7 +165,7 @@ static int uniphier_aio_compr_set_params(struct snd_compr_stream *cstream,
static int uniphier_aio_compr_hw_free(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
sub->setting = 0;
@@ -177,7 +177,7 @@ static int uniphier_aio_compr_prepare(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_compr_runtime *runtime = cstream->runtime;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
int bytes = runtime->fragment_size;
unsigned long flags;
@@ -215,7 +215,7 @@ static int uniphier_aio_compr_trigger(struct snd_compr_stream *cstream,
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_compr_runtime *runtime = cstream->runtime;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
struct device *dev = &aio->chip->pdev->dev;
int bytes = runtime->fragment_size, ret = 0;
@@ -248,7 +248,7 @@ static int uniphier_aio_compr_pointer(struct snd_compr_stream *cstream,
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_compr_runtime *runtime = cstream->runtime;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
int bytes = runtime->fragment_size;
unsigned long flags;
@@ -322,7 +322,7 @@ static int uniphier_aio_compr_copy(struct snd_compr_stream *cstream,
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_compr_runtime *runtime = cstream->runtime;
struct device *carddev = rtd->compr->card->dev;
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
size_t cnt = min_t(size_t, count, aio_rb_space_to_end(sub) / 2);
int bytes = runtime->fragment_size;
diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c
index da83423c52e2..4bbcb007df41 100644
--- a/sound/soc/uniphier/aio-dma.c
+++ b/sound/soc/uniphier/aio-dma.c
@@ -109,7 +109,7 @@ static int uniphier_aiodma_prepare(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[substream->stream];
int bytes = runtime->period_size *
runtime->channels * samples_to_bytes(runtime, 1);
@@ -136,7 +136,7 @@ static int uniphier_aiodma_trigger(struct snd_soc_component *component,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[substream->stream];
struct device *dev = &aio->chip->pdev->dev;
int bytes = runtime->period_size *
@@ -172,7 +172,7 @@ static snd_pcm_uframes_t uniphier_aiodma_pointer(
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
- struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+ struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
struct uniphier_aio_sub *sub = &aio->sub[substream->stream];
int bytes = runtime->period_size *
runtime->channels * samples_to_bytes(runtime, 1);
diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c
index 77655084bbde..6aaa19829a73 100644
--- a/sound/soc/ux500/mop500_ab8500.c
+++ b/sound/soc/ux500/mop500_ab8500.c
@@ -215,8 +215,8 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct device *dev = rtd->card->dev;
unsigned int fmt;
int channels, ret = 0, driver_mode, slots;
@@ -339,7 +339,7 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream,
static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
mutex_lock(&mop500_ab8500_params_lock);
__clear_bit(cpu_dai->id, &mop500_ab8500_usage);
diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index 9445dbe8e039..39b96c132bc8 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -46,7 +46,7 @@ static const struct snd_pcm_hardware ux500_pcm_hw = {
static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_substream *substream)
{
- struct snd_soc_dai *dai = rtd->cpu_dai;
+ struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
u16 per_data_width, mem_data_width;
struct stedma40_chan_cfg *dma_cfg;
struct ux500_msp_dma_params *dma_params;
@@ -86,7 +86,7 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
struct dma_slave_config *slave_config)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct msp_i2s_platform_data *pdata = rtd->cpu_dai->dev->platform_data;
+ struct msp_i2s_platform_data *pdata = asoc_rtd_to_cpu(rtd, 0)->dev->platform_data;
struct snd_dmaengine_dai_dma_data *snd_dma_params;
struct ux500_msp_dma_params *ste_dma_params;
dma_addr_t dma_addr;
@@ -94,11 +94,11 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
if (pdata) {
ste_dma_params =
- snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
dma_addr = ste_dma_params->tx_rx_addr;
} else {
snd_dma_params =
- snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
dma_addr = snd_dma_params->addr;
}
diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c
index bcf442faff7c..68af2176b19c 100644
--- a/sound/soc/xtensa/xtfpga-i2s.c
+++ b/sound/soc/xtensa/xtfpga-i2s.c
@@ -373,7 +373,7 @@ static int xtfpga_pcm_open(struct snd_soc_component *component,
void *p;
snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware);
- p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ p = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
runtime->private_data = p;
return 0;
diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c
index 60382ec23832..a3a07c0730e6 100644
--- a/sound/soc/zte/zx-spdif.c
+++ b/sound/soc/zte/zx-spdif.c
@@ -322,7 +322,6 @@ static int zx_spdif_probe(struct platform_device *pdev)
zx_spdif->mapbase = res->start;
zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(zx_spdif->reg_base)) {
- dev_err(&pdev->dev, "ioremap failed!\n");
return PTR_ERR(zx_spdif->reg_base);
}
diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c
index 0e5a05b25a77..4f787185d630 100644
--- a/sound/soc/zte/zx-tdm.c
+++ b/sound/soc/zte/zx-tdm.c
@@ -371,7 +371,6 @@ static struct snd_soc_dai_driver zx_tdm_dai = {
static int zx_tdm_probe(struct platform_device *pdev)
{
- struct device *dev = &pdev->dev;
struct of_phandle_args out_args;
unsigned int dma_reg_offset;
struct zx_tdm_info *zx_tdm;
@@ -384,7 +383,7 @@ static int zx_tdm_probe(struct platform_device *pdev)
if (!zx_tdm)
return -ENOMEM;
- zx_tdm->dev = dev;
+ zx_tdm->dev = &pdev->dev;
zx_tdm->dai_wclk = devm_clk_get(&pdev->dev, "wclk");
if (IS_ERR(zx_tdm->dai_wclk)) {
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index 78edd7d2f418..56031026b113 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \
mixer_scarlett.o \
mixer_scarlett_gen2.o \
mixer_us16x08.o \
+ mixer_s1810c.o \
pcm.o \
power.o \
proc.o \
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 827fb0bc8b56..fd6fd1726ea0 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -72,6 +72,7 @@ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */
static bool ignore_ctl_error;
static bool autoclock = true;
static char *quirk_alias[SNDRV_CARDS];
+static char *delayed_register[SNDRV_CARDS];
bool snd_usb_use_vmalloc = true;
bool snd_usb_skip_validation;
@@ -95,6 +96,8 @@ module_param(autoclock, bool, 0444);
MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes).");
module_param_array(quirk_alias, charp, NULL, 0444);
MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef.");
+module_param_array(delayed_register, charp, NULL, 0444);
+MODULE_PARM_DESC(delayed_register, "Quirk for delayed registration, given by id:iface, e.g. 0123abcd:4.");
module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444);
MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes).");
module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444);
@@ -525,6 +528,21 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id)
return false;
}
+static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface)
+{
+ int i;
+ unsigned int id, inum;
+
+ for (i = 0; i < ARRAY_SIZE(delayed_register); i++) {
+ if (delayed_register[i] &&
+ sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 &&
+ id == chip->usb_id)
+ return inum != iface;
+ }
+
+ return false;
+}
+
static const struct usb_device_id usb_audio_ids[]; /* defined below */
/* look for the corresponding quirk */
@@ -662,10 +680,22 @@ static int usb_audio_probe(struct usb_interface *intf,
goto __error;
}
- /* we are allowed to call snd_card_register() many times */
- err = snd_card_register(chip->card);
- if (err < 0)
- goto __error;
+ if (chip->need_delayed_register) {
+ dev_info(&dev->dev,
+ "Found post-registration device assignment: %08x:%02x\n",
+ chip->usb_id, ifnum);
+ chip->need_delayed_register = false; /* clear again */
+ }
+
+ /* we are allowed to call snd_card_register() many times, but first
+ * check to see if a device needs to skip it or do anything special
+ */
+ if (!snd_usb_registration_quirk(chip, ifnum) &&
+ !check_delayed_register_option(chip, ifnum)) {
+ err = snd_card_register(chip->card);
+ if (err < 0)
+ goto __error;
+ }
if (quirk && quirk->shares_media_device) {
/* don't want to fail when snd_media_device_create() fails */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index a48313dfa967..b118cf97607f 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i
return ret;
}
-/*
- * Assume the clock is valid if clock source supports only one single sample
- * rate, the terminal is connected directly to it (there is no clock selector)
- * and clock type is internal. This is to deal with some Denon DJ controllers
- * that always reports that clock is invalid.
- */
static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
struct audioformat *fmt,
int source_id)
{
+ bool ret = false;
+ int count;
+ unsigned char data;
+ struct usb_device *dev = chip->dev;
+
if (fmt->protocol == UAC_VERSION_2) {
struct uac_clock_source_descriptor *cs_desc =
snd_usb_find_clock_source(chip->ctrl_intf, source_id);
@@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
if (!cs_desc)
return false;
- return (fmt->nr_rates == 1 &&
- (fmt->clock & 0xff) == cs_desc->bClockID &&
- (cs_desc->bmAttributes & 0x3) !=
- UAC_CLOCK_SOURCE_TYPE_EXT);
+ /*
+ * Assume the clock is valid if clock source supports only one
+ * single sample rate, the terminal is connected directly to it
+ * (there is no clock selector) and clock type is internal.
+ * This is to deal with some Denon DJ controllers that always
+ * reports that clock is invalid.
+ */
+ if (fmt->nr_rates == 1 &&
+ (fmt->clock & 0xff) == cs_desc->bClockID &&
+ (cs_desc->bmAttributes & 0x3) !=
+ UAC_CLOCK_SOURCE_TYPE_EXT)
+ return true;
+ }
+
+ /*
+ * MOTU MicroBook IIc
+ * Sample rate changes takes more than 2 seconds for this device. Clock
+ * validity request returns false during that period.
+ */
+ if (chip->usb_id == USB_ID(0x07fd, 0x0004)) {
+ count = 0;
+
+ while ((!ret) && (count < 50)) {
+ int err;
+
+ msleep(100);
+
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+ USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+ UAC2_CS_CONTROL_CLOCK_VALID << 8,
+ snd_usb_ctrl_intf(chip) | (source_id << 8),
+ &data, sizeof(data));
+ if (err < 0) {
+ dev_warn(&dev->dev,
+ "%s(): cannot get clock validity for id %d\n",
+ __func__, source_id);
+ return false;
+ }
+
+ ret = !!data;
+ count++;
+ }
}
- return false;
+ return ret;
}
static bool uac_clock_source_is_valid(struct snd_usb_audio *chip,
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 9f5cb4ed3a0c..50e1874c847c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -247,6 +247,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
return 0;
}
+
+/*
+ * Presonus Studio 1810c supports a limited set of sampling
+ * rates per altsetting but reports the full set each time.
+ * If we don't filter out the unsupported rates and attempt
+ * to configure the card, it will hang refusing to do any
+ * further audio I/O until a hard reset is performed.
+ *
+ * The list of supported rates per altsetting (set of available
+ * I/O channels) is described in the owner's manual, section 2.2.
+ */
+static bool s1810c_valid_sample_rate(struct audioformat *fp,
+ unsigned int rate)
+{
+ switch (fp->altsetting) {
+ case 1:
+ /* All ADAT ports available */
+ return rate <= 48000;
+ case 2:
+ /* Half of ADAT ports available */
+ return (rate == 88200 || rate == 96000);
+ case 3:
+ /* Analog I/O only (no S/PDIF nor ADAT) */
+ return rate >= 176400;
+ default:
+ return false;
+ }
+ return false;
+}
+
/*
* Helper function to walk the array of sample rate triplets reported by
* the device. The problem is that we need to parse whole array first to
@@ -283,6 +313,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
}
for (rate = min; rate <= max; rate += res) {
+
+ /* Filter out invalid rates on Presonus Studio 1810c */
+ if (chip->usb_id == USB_ID(0x0194f, 0x010c) &&
+ !s1810c_valid_sample_rate(fp, rate))
+ goto skip_rate;
+
if (fp->rate_table)
fp->rate_table[nr_rates] = rate;
if (!fp->rate_min || rate < fp->rate_min)
@@ -297,6 +333,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
break;
}
+skip_rate:
/* avoid endless loop */
if (res == 0)
break;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 392e5fda680c..047b90595d65 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -91,7 +91,7 @@ struct usb_ms_endpoint_descriptor {
__u8 bDescriptorType;
__u8 bDescriptorSubtype;
__u8 bNumEmbMIDIJack;
- __u8 baAssocJackID[0];
+ __u8 baAssocJackID[];
} __attribute__ ((packed));
struct snd_usb_midi_in_endpoint;
@@ -1826,6 +1826,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi,
return 0;
}
+static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor(
+ struct usb_host_endpoint *hostep)
+{
+ unsigned char *extra = hostep->extra;
+ int extralen = hostep->extralen;
+
+ while (extralen > 3) {
+ struct usb_ms_endpoint_descriptor *ms_ep =
+ (struct usb_ms_endpoint_descriptor *)extra;
+
+ if (ms_ep->bLength > 3 &&
+ ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT &&
+ ms_ep->bDescriptorSubtype == UAC_MS_GENERAL)
+ return ms_ep;
+ if (!extra[0])
+ break;
+ extralen -= extra[0];
+ extra += extra[0];
+ }
+ return NULL;
+}
+
/*
* Returns MIDIStreaming device capabilities.
*/
@@ -1863,11 +1885,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi,
ep = get_ep_desc(hostep);
if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep))
continue;
- ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra;
- if (hostep->extralen < 4 ||
- ms_ep->bLength < 4 ||
- ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
- ms_ep->bDescriptorSubtype != UAC_MS_GENERAL)
+ ms_ep = find_usb_ms_endpoint_descriptor(hostep);
+ if (!ms_ep)
continue;
if (usb_endpoint_dir_out(ep)) {
if (endpoints[epidx].out_ep) {
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 81b2db0edd5f..721d12130d0c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -292,6 +292,11 @@ static int uac2_ctl_value_size(int val_type)
* retrieve a mixer value
*/
+static inline int mixer_ctrl_intf(struct usb_mixer_interface *mixer)
+{
+ return get_iface_desc(mixer->hostif)->bInterfaceNumber;
+}
+
static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request,
int validx, int *value_ret)
{
@@ -306,7 +311,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request,
return -EIO;
while (timeout-- > 0) {
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
validx, idx, buf, val_len);
@@ -354,7 +359,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request,
if (ret)
goto error;
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
validx, idx, buf, size);
@@ -479,7 +484,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
return -EIO;
while (timeout-- > 0) {
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
err = snd_usb_ctl_msg(chip->dev,
usb_sndctrlpipe(chip->dev, 0), request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
@@ -901,6 +906,12 @@ static int parse_term_effect_unit(struct mixer_build *state,
struct usb_audio_term *term,
void *p1, int id)
{
+ struct uac2_effect_unit_descriptor *d = p1;
+ int err;
+
+ err = __check_input_term(state, d->bSourceID, term);
+ if (err < 0)
+ return err;
term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */
term->id = id;
return 0;
@@ -1203,7 +1214,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
usb_audio_err(cval->head.mixer->chip,
"%d:%d: cannot get min/max values for control %d (id %d)\n",
- cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip),
+ cval->head.id, mixer_ctrl_intf(cval->head.mixer),
cval->control, cval->head.id);
return -EINVAL;
}
@@ -1422,7 +1433,7 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol,
if (ret)
goto error;
- idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+ idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
if (cval->head.mixer->protocol == UAC_VERSION_2) {
struct uac2_connectors_ctl_blk uac2_conn;
@@ -1674,6 +1685,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
/* get min/max values */
get_min_max_with_quirks(cval, 0, kctl);
+ /* skip a bogus volume range */
+ if (cval->max <= cval->min) {
+ usb_audio_dbg(mixer->chip,
+ "[%d] FU [%s] skipped due to invalid volume\n",
+ cval->head.id, kctl->id.name);
+ snd_ctl_free_one(kctl);
+ return;
+ }
+
+
if (control == UAC_FU_VOLUME) {
check_mapped_dB(map, cval);
if (cval->dBmin < cval->dBmax || !cval->initialized) {
@@ -3203,7 +3224,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry,
list_for_each_entry(mixer, &chip->mixer_list, list) {
snd_iprintf(buffer,
"USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n",
- chip->usb_id, snd_usb_ctrl_intf(chip),
+ chip->usb_id, mixer_ctrl_intf(mixer),
mixer->ignore_ctl_error);
snd_iprintf(buffer, "Card: %s\n", chip->card->longname);
for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) {
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index c237e24f08d9..02b036b2aefb 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -34,6 +34,7 @@
#include "mixer_scarlett.h"
#include "mixer_scarlett_gen2.h"
#include "mixer_us16x08.h"
+#include "mixer_s1810c.h"
#include "helper.h"
struct std_mono_table {
@@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
case USB_ID(0x2a39, 0x3fd4): /* RME */
err = snd_rme_controls_create(mixer);
break;
+
+ case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */
+ err = snd_sc1810_init_mixer(mixer);
+ break;
}
return err;
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
new file mode 100644
index 000000000000..6483e47bafd0
--- /dev/null
+++ b/sound/usb/mixer_s1810c.c
@@ -0,0 +1,595 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com>
+ *
+ * Based on reverse engineering of the communication protocol
+ * between the windows driver / Univeral Control (UC) program
+ * and the device, through usbmon.
+ *
+ * For now this bypasses the mixer, with all channels split,
+ * so that the software can mix with greater flexibility.
+ * It also adds controls for the 4 buttons on the front of
+ * the device.
+ */
+
+#include <linux/usb.h>
+#include <linux/usb/audio-v2.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+
+#include "usbaudio.h"
+#include "mixer.h"
+#include "mixer_quirks.h"
+#include "helper.h"
+#include "mixer_s1810c.h"
+
+#define SC1810C_CMD_REQ 160
+#define SC1810C_CMD_REQTYPE \
+ (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT)
+#define SC1810C_CMD_F1 0x50617269
+#define SC1810C_CMD_F2 0x14
+
+/*
+ * DISCLAIMER: These are just guesses based on the
+ * dumps I got.
+ *
+ * It seems like a selects between
+ * device (0), mixer (0x64) and output (0x65)
+ *
+ * For mixer (0x64):
+ * * b selects an input channel (see below).
+ * * c selects an output channel pair (see below).
+ * * d selects left (0) or right (1) of that pair.
+ * * e 0-> disconnect, 0x01000000-> connect,
+ * 0x0109-> used for stereo-linking channels,
+ * e is also used for setting volume levels
+ * in which case b is also set so I guess
+ * this way it is possible to set the volume
+ * level from the specified input to the
+ * specified output.
+ *
+ * IN Channels:
+ * 0 - 7 Mic/Inst/Line (Analog inputs)
+ * 8 - 9 S/PDIF
+ * 10 - 17 ADAT
+ * 18 - 35 DAW (Inputs from the host)
+ *
+ * OUT Channels (pairs):
+ * 0 -> Main out
+ * 1 -> Line1/2
+ * 2 -> Line3/4
+ * 3 -> S/PDIF
+ * 4 -> ADAT?
+ *
+ * For device (0):
+ * * b and c are not used, at least not on the
+ * dumps I got.
+ * * d sets the control id to be modified
+ * (see below).
+ * * e sets the setting for that control.
+ * (so for the switches I was interested
+ * in it's 0/1)
+ *
+ * For output (0x65):
+ * * b is the output channel (see above).
+ * * c is zero.
+ * * e I guess the same as with mixer except 0x0109
+ * which I didn't see in my dumps.
+ *
+ * The two fixed fields have the same values for
+ * mixer and output but a different set for device.
+ */
+struct s1810c_ctl_packet {
+ u32 a;
+ u32 b;
+ u32 fixed1;
+ u32 fixed2;
+ u32 c;
+ u32 d;
+ u32 e;
+};
+
+#define SC1810C_CTL_LINE_SW 0
+#define SC1810C_CTL_MUTE_SW 1
+#define SC1810C_CTL_AB_SW 3
+#define SC1810C_CTL_48V_SW 4
+
+#define SC1810C_SET_STATE_REQ 161
+#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE
+#define SC1810C_SET_STATE_F1 0x64656D73
+#define SC1810C_SET_STATE_F2 0xF4
+
+#define SC1810C_GET_STATE_REQ 162
+#define SC1810C_GET_STATE_REQTYPE \
+ (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN)
+#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1
+#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2
+
+#define SC1810C_STATE_F1_IDX 2
+#define SC1810C_STATE_F2_IDX 3
+
+/*
+ * This packet includes mixer volumes and
+ * various other fields, it's an extended
+ * version of ctl_packet, with a and b
+ * being zero and different f1/f2.
+ */
+struct s1810c_state_packet {
+ u32 fields[63];
+};
+
+#define SC1810C_STATE_48V_SW 58
+#define SC1810C_STATE_LINE_SW 59
+#define SC1810C_STATE_MUTE_SW 60
+#define SC1810C_STATE_AB_SW 62
+
+struct s1810_mixer_state {
+ uint16_t seqnum;
+ struct mutex usb_mutex;
+ struct mutex data_mutex;
+};
+
+static int
+snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a,
+ u32 b, u32 c, u32 d, u32 e)
+{
+ struct s1810c_ctl_packet pkt = { 0 };
+ int ret = 0;
+
+ pkt.fixed1 = SC1810C_CMD_F1;
+ pkt.fixed2 = SC1810C_CMD_F2;
+
+ pkt.a = a;
+ pkt.b = b;
+ pkt.c = c;
+ pkt.d = d;
+ /*
+ * Value for settings 0/1 for this
+ * output channel is always 0 (probably because
+ * there is no ADAT output on 1810c)
+ */
+ pkt.e = (c == 4) ? 0 : e;
+
+ ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
+ SC1810C_CMD_REQ,
+ SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt));
+ if (ret < 0) {
+ dev_warn(&dev->dev, "could not send ctl packet\n");
+ return ret;
+ }
+ return 0;
+}
+
+/*
+ * When opening Universal Control the program periodicaly
+ * sends and receives state packets for syncinc state between
+ * the device and the host.
+ *
+ * Note that if we send only the request to get data back we'll
+ * get an error, we need to first send an empty state packet and
+ * then ask to receive a filled. Their seqnumbers must also match.
+ */
+static int
+snd_sc1810c_get_status_field(struct usb_device *dev,
+ u32 *field, int field_idx, uint16_t *seqnum)
+{
+ struct s1810c_state_packet pkt_out = { { 0 } };
+ struct s1810c_state_packet pkt_in = { { 0 } };
+ int ret = 0;
+
+ pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1;
+ pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2;
+ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+ SC1810C_SET_STATE_REQ,
+ SC1810C_SET_STATE_REQTYPE,
+ (*seqnum), 0, &pkt_out, sizeof(pkt_out));
+ if (ret < 0) {
+ dev_warn(&dev->dev, "could not send state packet (%d)\n", ret);
+ return ret;
+ }
+
+ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+ SC1810C_GET_STATE_REQ,
+ SC1810C_GET_STATE_REQTYPE,
+ (*seqnum), 0, &pkt_in, sizeof(pkt_in));
+ if (ret < 0) {
+ dev_warn(&dev->dev, "could not get state field %u (%d)\n",
+ field_idx, ret);
+ return ret;
+ }
+
+ (*field) = pkt_in.fields[field_idx];
+ (*seqnum)++;
+ return 0;
+}
+
+/*
+ * This is what I got when bypassing the mixer with
+ * all channels split. I'm not 100% sure of what's going
+ * on, I could probably clean this up based on my observations
+ * but I prefer to keep the same behavior as the windows driver.
+ */
+static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip)
+{
+ u32 a, b, c, e, n, off;
+ struct usb_device *dev = chip->dev;
+
+ /* Set initial volume levels ? */
+ a = 0x64;
+ e = 0xbc;
+ for (n = 0; n < 2; n++) {
+ off = n * 18;
+ for (b = off, c = 0; b < 18 + off; b++) {
+ /* This channel to all outputs ? */
+ for (c = 0; c <= 8; c++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ }
+ /* This channel to main output (again) */
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+ }
+ /*
+ * I noticed on UC that DAW channels have different
+ * initial volumes, so this makes sense.
+ */
+ e = 0xb53bf0;
+ }
+
+ /* Connect analog outputs ? */
+ a = 0x65;
+ e = 0x01000000;
+ for (b = 1; b < 3; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+ }
+ snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e);
+
+ /* Set initial volume levels for S/PDIF mappings ? */
+ a = 0x64;
+ e = 0xbc;
+ c = 3;
+ for (n = 0; n < 2; n++) {
+ off = n * 18;
+ for (b = off; b < 18 + off; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ }
+ e = 0xb53bf0;
+ }
+
+ /* Connect S/PDIF output ? */
+ a = 0x65;
+ e = 0x01000000;
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+ /* Connect all outputs (again) ? */
+ a = 0x65;
+ e = 0x01000000;
+ for (b = 0; b < 4; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+ }
+
+ /* Basic routing to get sound out of the device */
+ a = 0x64;
+ e = 0x01000000;
+ for (c = 0; c < 4; c++) {
+ for (b = 0; b < 36; b++) {
+ if ((c == 0 && b == 18) || /* DAW1/2 -> Main */
+ (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */
+ (c == 2 && b == 22) || /* DAW4/5 -> Line5/6 */
+ (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */
+ /* Left */
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+ b++;
+ /* Right */
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ } else {
+ /* Leave the rest disconnected */
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+ }
+ }
+ }
+
+ /* Set initial volume levels for S/PDIF (again) ? */
+ a = 0x64;
+ e = 0xbc;
+ c = 3;
+ for (n = 0; n < 2; n++) {
+ off = n * 18;
+ for (b = off; b < 18 + off; b++) {
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+ }
+ e = 0xb53bf0;
+ }
+
+ /* Connect S/PDIF outputs (again) ? */
+ a = 0x65;
+ e = 0x01000000;
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+ /* Again ? */
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+ snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+ return 0;
+}
+
+/*
+ * Sync state with the device and retrieve the requested field,
+ * whose index is specified in (kctl->private_value & 0xFF),
+ * from the received fields array.
+ */
+static int
+snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer,
+ struct snd_kcontrol *kctl, u32 *state)
+{
+ struct snd_usb_audio *chip = mixer->chip;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 field = 0;
+ u32 ctl_idx = (u32) (kctl->private_value & 0xFF);
+ int ret = 0;
+
+ mutex_lock(&private->usb_mutex);
+ ret = snd_sc1810c_get_status_field(chip->dev, &field,
+ ctl_idx, &private->seqnum);
+ if (ret < 0)
+ goto unlock;
+
+ *state = field;
+ unlock:
+ mutex_unlock(&private->usb_mutex);
+ return ret ? ret : 0;
+}
+
+/*
+ * Send a control packet to the device for the control id
+ * specified in (kctl->private_value >> 8) with value
+ * specified in (kctl->private_value >> 16).
+ */
+static int
+snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer,
+ struct snd_kcontrol *kctl)
+{
+ struct snd_usb_audio *chip = mixer->chip;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 pval = (u32) kctl->private_value;
+ u32 ctl_id = (pval >> 8) & 0xFF;
+ u32 ctl_val = (pval >> 16) & 0x1;
+ int ret = 0;
+
+ mutex_lock(&private->usb_mutex);
+ ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val);
+ mutex_unlock(&private->usb_mutex);
+ return ret;
+}
+
+/* Generic get/set/init functions for switch controls */
+
+static int
+snd_s1810c_switch_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ctl_elem)
+{
+ struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+ struct usb_mixer_interface *mixer = list->mixer;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 pval = (u32) kctl->private_value;
+ u32 ctl_idx = pval & 0xFF;
+ u32 state = 0;
+ int ret = 0;
+
+ mutex_lock(&private->data_mutex);
+ ret = snd_s1810c_get_switch_state(mixer, kctl, &state);
+ if (ret < 0)
+ goto unlock;
+
+ switch (ctl_idx) {
+ case SC1810C_STATE_LINE_SW:
+ case SC1810C_STATE_AB_SW:
+ ctl_elem->value.enumerated.item[0] = (int)state;
+ break;
+ default:
+ ctl_elem->value.integer.value[0] = (long)state;
+ }
+
+ unlock:
+ mutex_unlock(&private->data_mutex);
+ return (ret < 0) ? ret : 0;
+}
+
+static int
+snd_s1810c_switch_set(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ctl_elem)
+{
+ struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+ struct usb_mixer_interface *mixer = list->mixer;
+ struct s1810_mixer_state *private = mixer->private_data;
+ u32 pval = (u32) kctl->private_value;
+ u32 ctl_idx = pval & 0xFF;
+ u32 curval = 0;
+ u32 newval = 0;
+ int ret = 0;
+
+ mutex_lock(&private->data_mutex);
+ ret = snd_s1810c_get_switch_state(mixer, kctl, &curval);
+ if (ret < 0)
+ goto unlock;
+
+ switch (ctl_idx) {
+ case SC1810C_STATE_LINE_SW:
+ case SC1810C_STATE_AB_SW:
+ newval = (u32) ctl_elem->value.enumerated.item[0];
+ break;
+ default:
+ newval = (u32) ctl_elem->value.integer.value[0];
+ }
+
+ if (curval == newval)
+ goto unlock;
+
+ kctl->private_value &= ~(0x1 << 16);
+ kctl->private_value |= (unsigned int)(newval & 0x1) << 16;
+ ret = snd_s1810c_set_switch_state(mixer, kctl);
+
+ unlock:
+ mutex_unlock(&private->data_mutex);
+ return (ret < 0) ? 0 : 1;
+}
+
+static int
+snd_s1810c_switch_init(struct usb_mixer_interface *mixer,
+ const struct snd_kcontrol_new *new_kctl)
+{
+ struct snd_kcontrol *kctl;
+ struct usb_mixer_elem_info *elem;
+
+ elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL);
+ if (!elem)
+ return -ENOMEM;
+
+ elem->head.mixer = mixer;
+ elem->control = 0;
+ elem->head.id = 0;
+ elem->channels = 1;
+
+ kctl = snd_ctl_new1(new_kctl, elem);
+ if (!kctl) {
+ kfree(elem);
+ return -ENOMEM;
+ }
+ kctl->private_free = snd_usb_mixer_elem_free;
+
+ return snd_usb_mixer_add_control(&elem->head, kctl);
+}
+
+static int
+snd_s1810c_line_sw_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const texts[2] = {
+ "Preamp On (Mic/Inst)",
+ "Preamp Off (Line in)"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_line_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line 1/2 Source Type",
+ .info = snd_s1810c_line_sw_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_mute_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mute Main Out Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_48v_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "48V Phantom Power On Mic Inputs Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8)
+};
+
+static int
+snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *const texts[2] = {
+ "1/2",
+ "3/4"
+ };
+
+ return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_ab_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone 1 Source Route",
+ .info = snd_s1810c_ab_sw_info,
+ .get = snd_s1810c_switch_get,
+ .put = snd_s1810c_switch_set,
+ .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8)
+};
+
+static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer)
+{
+ struct s1810_mixer_state *private = mixer->private_data;
+ kfree(private);
+ mixer->private_data = NULL;
+}
+
+/* Entry point, called from mixer_quirks.c */
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
+{
+ struct s1810_mixer_state *private = NULL;
+ struct snd_usb_audio *chip = mixer->chip;
+ struct usb_device *dev = chip->dev;
+ int ret = 0;
+
+ /* Run this only once */
+ if (!list_empty(&chip->mixer_list))
+ return 0;
+
+ dev_info(&dev->dev,
+ "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
+ if (chip->setup == 1)
+ dev_info(&dev->dev, "(8out/18in @ 48KHz)\n");
+ else if (chip->setup == 2)
+ dev_info(&dev->dev, "(6out/8in @ 192KHz)\n");
+ else
+ dev_info(&dev->dev, "(8out/14in @ 96KHz)\n");
+
+ ret = snd_s1810c_init_mixer_maps(chip);
+ if (ret < 0)
+ return ret;
+
+ private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL);
+ if (!private)
+ return -ENOMEM;
+
+ mutex_init(&private->usb_mutex);
+ mutex_init(&private->data_mutex);
+
+ mixer->private_data = private;
+ mixer->private_free = snd_sc1810_mixer_state_free;
+
+ private->seqnum = 1;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw);
+ if (ret < 0)
+ return ret;
+ return ret;
+}
diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h
new file mode 100644
index 000000000000..a79a3743cff3
--- /dev/null
+++ b/sound/usb/mixer_s1810c.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com>
+ */
+
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index bd258f1ec2dd..a4e4064f9aee 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -357,7 +357,12 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
ep = 0x81;
ifnum = 1;
goto add_sync_ep_from_ifnum;
- case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
+ case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */
+ /* MicroBook IIc */
+ if (altsd->bInterfaceClass == USB_CLASS_AUDIO)
+ return 0;
+
+ /* MicroBook II */
ep = 0x84;
ifnum = 0;
goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index ffbf4bd9208c..4174ad11fca6 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -70,7 +70,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
snd_iprintf(buffer, " Interface %d\n", fp->iface);
snd_iprintf(buffer, " Altset %d\n", fp->altsetting);
snd_iprintf(buffer, " Format:");
- for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt)
+ pcm_for_each_format(fmt)
if (fp->formats & pcm_format_to_bits(fmt))
snd_iprintf(buffer, " %s",
snd_pcm_format_name(fmt));
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d187aa6d50db..1c8719292eee 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3472,7 +3472,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
},
/* MOTU Microbook II */
{
- USB_DEVICE(0x07fd, 0x0004),
+ USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004),
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "MOTU",
.product_name = "MicroBookII",
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 7f558f4b4520..86f192a3043d 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1252,6 +1252,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip,
return 0; /* keep this altsetting */
}
+static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno)
+{
+ /*
+ * Altno settings:
+ *
+ * Playback (Interface 1):
+ * 1: 6 Analog + 2 S/PDIF
+ * 2: 6 Analog + 2 S/PDIF
+ * 3: 6 Analog
+ *
+ * Capture (Interface 2):
+ * 1: 8 Analog + 2 S/PDIF + 8 ADAT
+ * 2: 8 Analog + 2 S/PDIF + 4 ADAT
+ * 3: 8 Analog
+ */
+
+ /*
+ * I'll leave 2 as the default one and
+ * use device_setup to switch to the
+ * other two.
+ */
+ if ((chip->setup == 0 || chip->setup > 2) && altno != 2)
+ return 1;
+ else if (chip->setup == 1 && altno != 1)
+ return 1;
+ else if (chip->setup == 2 && altno != 3)
+ return 1;
+
+ return 0;
+}
+
int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
int iface,
int altno)
@@ -1265,6 +1297,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
/* fasttrackpro usb: skip altsets incompatible with device_setup */
if (chip->usb_id == USB_ID(0x0763, 0x2012))
return fasttrackpro_skip_setting_quirk(chip, iface, altno);
+ /* presonus studio 1810c: skip altsets incompatible with device_setup */
+ if (chip->usb_id == USB_ID(0x0194f, 0x010c))
+ return s1810c_skip_setting_quirk(chip, iface, altno);
+
return 0;
}
@@ -1316,7 +1352,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
return snd_usb_axefx3_boot_quirk(dev);
case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
- return snd_usb_motu_microbookii_boot_quirk(dev);
+ /*
+ * For some reason interface 3 with vendor-spec class is
+ * detected on MicroBook IIc.
+ */
+ if (get_iface_desc(intf->altsetting)->bInterfaceClass ==
+ USB_CLASS_VENDOR_SPEC &&
+ get_iface_desc(intf->altsetting)->bInterfaceNumber < 3)
+ return snd_usb_motu_microbookii_boot_quirk(dev);
+ break;
}
return 0;
@@ -1754,5 +1798,47 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
else
fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
break;
+ case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook IIc */
+ /*
+ * MaxPacketsOnly attribute is erroneously set in endpoint
+ * descriptors. As a result this card produces noise with
+ * all sample rates other than 96 KHz.
+ */
+ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
+ break;
}
}
+
+/*
+ * registration quirk:
+ * the registration is skipped if a device matches with the given ID,
+ * unless the interface reaches to the defined one. This is for delaying
+ * the registration until the last known interface, so that the card and
+ * devices appear at the same time.
+ */
+
+struct registration_quirk {
+ unsigned int usb_id; /* composed via USB_ID() */
+ unsigned int interface; /* the interface to trigger register */
+};
+
+#define REG_QUIRK_ENTRY(vendor, product, iface) \
+ { .usb_id = USB_ID(vendor, product), .interface = (iface) }
+
+static const struct registration_quirk registration_quirks[] = {
+ REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */
+ { 0 } /* terminator */
+};
+
+/* return true if skipping registration */
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface)
+{
+ const struct registration_quirk *q;
+
+ for (q = registration_quirks; q->usb_id; q++)
+ if (chip->usb_id == q->usb_id)
+ return iface != q->interface;
+
+ /* Register as normal */
+ return false;
+}
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index df0355843a4c..c76cf24a640a 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -51,4 +51,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
struct audioformat *fp,
int stream);
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface);
+
#endif /* __USBAUDIO_QUIRKS_H */
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index afd5aa574611..15296f2c902c 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -502,6 +502,9 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip,
subs = &as->substream[stream];
if (subs->ep_num)
continue;
+ if (snd_device_get_state(chip->card, as->pcm) !=
+ SNDRV_DEV_BUILD)
+ chip->need_delayed_register = true;
err = snd_pcm_new_stream(as->pcm, stream, 1);
if (err < 0)
return err;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 6fe3ab582ec6..1c892c7f14d7 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -34,6 +34,7 @@ struct snd_usb_audio {
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */
+ unsigned int need_delayed_register:1; /* warn for delayed registration */
int num_interfaces;
int num_suspended_intf;
int sample_rate_read_error;
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 772f6f3ccbb1..37d290fe9d43 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops =
*/
static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream)
{
- kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
- usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
+ int stream;
- kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]);
- usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL;
+ for_each_pcm_streams(stream) {
+ kfree(usX2Y_substream[stream]);
+ usX2Y_substream[stream] = NULL;
+ }
}
static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm)