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author | Linus Torvalds <torvalds@linux-foundation.org> | 2020-10-15 21:07:44 +0300 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2020-10-15 21:07:44 +0300 |
commit | c48b75b7271db23c1b2d1204d6e8496d91f27711 (patch) | |
tree | 83c95f082e0605257b8af3ebd70b2c448262fd88 /sound/soc/intel/boards/sof_sdw_rt1316.c | |
parent | 93b694d096cc10994c817730d4d50288f9ae3d66 (diff) | |
parent | ce1558c285f9ad04c03b46833a028230771cc0a7 (diff) | |
download | linux-c48b75b7271db23c1b2d1204d6e8496d91f27711.tar.xz |
Merge tag 'sound-5.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"The amount of changes is smaller at this round (what a surprise), but
lots of activity is seen. Most of changes are about ASoC driver
development, especially Intel platforms. Here are some highlights:
General:
- Replace all tasklet usages with other alternatives
- Cleanup of the ASoC error unwinding code
- Fixes for trivial issues caught by static checker
- Spell fixes allover the places
ALSA Core:
- Lockdep fix for control devices
- Fix for potential OSS sequencer mutex stalls
HD-audio and USB-audio:
- SoundBlaster AE-7 support
- Changes in quirk table for the rename handling
- Quirks for HP and ASUS machines, Pioneer DJ DJM-250MK2.
ASoC:
- Lots of updates for Intel SOF and SoundWire enablement
- Replacement of the DSP driver for some older x86 systems; the new
code was written from scratch, better maintenance expected
- Helpers for parsing auxiluary devices from the device tree
- New support for AllWinner A64, Cirrus Logic CS4234, Mediatek MT6359
Microchip S/PDIF TX and RX controllers, Realtek RT1015P, and Texas
Instruments J721E, TAS2110, TAS2564 and TAS2764"
* tag 'sound-5.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (498 commits)
ALSA: hda/hdmi: fix incorrect locking in hdmi_pcm_close
ALSA: hda: fix jack detection with Realtek codecs when in D3
ALSA: fireworks: use semicolons rather than commas to separate statements
ALSA: hda: use semicolons rather than commas to separate statements
ALSA: hda/i915 - fix list corruption with concurrent probes
ASoC: dmaengine: Document support for TX only or RX only streams
ASoC: mchp-spdiftx: remove 'TX' from playback stream name
ASoC: ti: davinci-mcasp: Use &pdev->dev for early dev_warn
ASoC: tas2764: Add the driver for the TAS2764
dt-bindings: tas2764: Add the TAS2764 binding doc
ASoC: Intel: catpt: Add explicit DMADEVICES kconfig dependency
ASoC: Intel: catpt: Fix compilation when CONFIG_MODULES is disabled
ASoC: stm32: dfsdm: add actual resolution trace
ASoC: stm32: dfsdm: change rate limits
ASoC: qcom: sc7180: Add support for audio over DP
Asoc: qcom: lpass-platform : Increase buffer size
ASoC: qcom: Add support for lpass hdmi driver
Asoc: qcom: lpass:Update lpaif_dmactl members order
Asoc:qcom:lpass-cpu:Update dts property read API
ASoC: dt-bindings: Add dt binding for lpass hdmi
...
Diffstat (limited to 'sound/soc/intel/boards/sof_sdw_rt1316.c')
-rw-r--r-- | sound/soc/intel/boards/sof_sdw_rt1316.c | 119 |
1 files changed, 119 insertions, 0 deletions
diff --git a/sound/soc/intel/boards/sof_sdw_rt1316.c b/sound/soc/intel/boards/sof_sdw_rt1316.c new file mode 100644 index 000000000000..d6e1ebf18d57 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt1316.c @@ -0,0 +1,119 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt1316 - Helpers to handle RT1316 from generic machine driver + */ + +#include <linux/device.h> +#include <linux/errno.h> +#include <sound/control.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget rt1316_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +/* + * dapm routes for rt1316 will be registered dynamically according + * to the number of rt1316 used. The first two entries will be registered + * for one codec case, and the last two entries are also registered + * if two 1316s are used. + */ +static const struct snd_soc_dapm_route rt1316_map[] = { + { "Speaker", NULL, "rt1316-1 SPOL" }, + { "Speaker", NULL, "rt1316-1 SPOR" }, + { "Speaker", NULL, "rt1316-2 SPOL" }, + { "Speaker", NULL, "rt1316-2 SPOR" }, +}; + +static const struct snd_kcontrol_new rt1316_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:rt1316", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt1316_controls, + ARRAY_SIZE(rt1316_controls)); + if (ret) { + dev_err(card->dev, "rt1316 controls addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt1316_widgets, + ARRAY_SIZE(rt1316_widgets)); + if (ret) { + dev_err(card->dev, "rt1316 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1316_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, rt1316_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); +} + +int sof_sdw_rt1316_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* Count amp number and do init on playback link only. */ + if (!playback) + return 0; + + info->amp_num++; + if (info->amp_num == 1) + dai_links->init = first_spk_init; + + if (info->amp_num == 2) { + /* + * if two 1316s 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; +} |