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author | Linus Torvalds <torvalds@linux-foundation.org> | 2021-04-30 22:48:14 +0300 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2021-04-30 22:48:14 +0300 |
commit | b71428d7ab333a157216a1d73c8c82a178efada9 (patch) | |
tree | 94d268210d84948d5984f2fbe7d890c4aed1fabe /sound/soc/intel/boards/kbl_da7219_max98927.c | |
parent | 95275402f66e88c56144a2d859c13594b651b29b (diff) | |
parent | 2e6a731296be9d356fdccee9fb6ae345dad96438 (diff) | |
download | linux-b71428d7ab333a157216a1d73c8c82a178efada9.tar.xz |
Merge tag 'sound-5.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"No surprises in this development cycle, and most of work is about the
fixes and the improvements of the existing code, while a new LED
control layer and a few new drivers have been introduced.
Here are some highlights:
Core:
- A common mute-LED framework was introduced. It is used by HD-audio
for now, more adaption will follow later. The former "Mic Mute-LED
Mode" mixer control has been replaced with the corresponding sysfs
now.
- User-control management was changed to count consumed bytes instead
of capping by number of elements; this will allow more controls in
the normal usage pattern while avoiding the possible memory
exhaustion DoS
ASoC:
- Continued refactoring and cleanups in ASoC core and generic card
drivers
- Wide range of small cppcheck and warning fixes
- New drivers for Freescale i.MX DMA over rpmsg, Mediatek MT6358
accessory detection, and Realtek RT1019, RT1316, RT711 and RT715
USB-audio:
- Continued improvements and fixes of the implicit feedback mode,
including better support for Pioneer and Roland/BOSS devices
HD-audio:
- Default back to non-buffer preallocation on x86
- Cirrus codec improvements, more quirks for Realtek codecs
Others:
- New virtio sound driver
- FireWire Bebob updates"
* tag 'sound-5.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (587 commits)
ALSA: hda/conexant: Re-order CX5066 quirk table entries
ALSA: hda/realtek: Remove redundant entry for ALC861 Haier/Uniwill devices
ALSA: hda/realtek: Re-order ALC662 quirk table entries
ALSA: hda/realtek: Re-order remaining ALC269 quirk table entries
ALSA: hda/realtek: Re-order ALC269 Lenovo quirk table entries
ALSA: hda/realtek: Re-order ALC269 Sony quirk table entries
ALSA: hda/realtek: Re-order ALC269 ASUS quirk table entries
ALSA: hda/realtek: Re-order ALC269 Dell quirk table entries
ALSA: hda/realtek: Re-order ALC269 Acer quirk table entries
ALSA: hda/realtek: Re-order ALC269 HP quirk table entries
ALSA: hda/realtek: Re-order ALC882 Clevo quirk table entries
ALSA: hda/realtek: Re-order ALC882 Sony quirk table entries
ALSA: hda/realtek: Re-order ALC882 Acer quirk table entries
ALSA: usb-audio: Remove redundant assignment to len
ALSA: hda/realtek: Add quirk for Intel Clevo PCx0Dx
ALSA: virtio: fix kernel-doc
ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye
ALSA: hda/cirrus: Set Initial DMIC volume for Bullseye to -26 dB
ALSA: sb: Fix two use after free in snd_sb_qsound_build
ALSA: emu8000: Fix a use after free in snd_emu8000_create_mixer
...
Diffstat (limited to 'sound/soc/intel/boards/kbl_da7219_max98927.c')
-rw-r--r-- | sound/soc/intel/boards/kbl_da7219_max98927.c | 48 |
1 files changed, 36 insertions, 12 deletions
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index cc9a2509ace2..4b7b4a044f81 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -111,8 +111,9 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_SPK("Left Spk", NULL), SND_SOC_DAPM_SPK("Right Spk", NULL), SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("DP", NULL), - SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), @@ -126,8 +127,9 @@ static const struct snd_soc_dapm_route kabylake_map[] = { /* other jacks */ { "DMic", NULL, "SoC DMIC" }, - { "HDMI", NULL, "hif5 Output" }, - { "DP", NULL, "hif6 Output" }, + {"HDMI1", NULL, "hif5-0 Output"}, + {"HDMI2", NULL, "hif6-0 Output"}, + {"HDMI3", NULL, "hif7-0 Output"}, /* CODEC BE connections */ { "Left HiFi Playback", NULL, "ssp0 Tx" }, @@ -282,12 +284,34 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ + /* * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE, * where as kblda7219m98927 & kblmax98927 supports S16_LE by default. * Skipping the port wise FE and BE configuration for kblda7219m98373 & @@ -309,9 +333,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -322,7 +346,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; |