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| author | Linus Torvalds <torvalds@linux-foundation.org> | 2025-03-26 09:41:55 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2025-03-26 09:41:55 -0700 |
| commit | e50da555ca4d42b1b98d0f26789db64f26a0919a (patch) | |
| tree | 8335ba0c862679f4294284cba61c48d52abe0665 /Documentation/sound | |
| parent | 1e26c5e28ca5821a824e90dd359556f5e9e7b89f (diff) | |
| parent | 3a949fc08103c0ce3a1d0ef30459c7b3acc6a214 (diff) | |
| download | linux-e50da555ca4d42b1b98d0f26789db64f26a0919a.tar.gz linux-e50da555ca4d42b1b98d0f26789db64f26a0919a.tar.bz2 linux-e50da555ca4d42b1b98d0f26789db64f26a0919a.zip | |
Merge tag 'sound-6.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We've received lots of commits at this time, as a result of various
cleanup and refactoring works as well as a few new drivers and the
generic SoundWire support. Most of changes are device-specific, little
about the core changes. Some highlights below:
Core:
- A couple of (rather minor) race fixes in ALSA sequencer code
- A regression fix in ALSA timer code that may lead to a deadlock
ASoC:
- A large series of code conversion to use modern terminology for the
clocking configuration
- Conversions of PM ops with the modern macros in all ASoC drivers
- Clarification of the control operations
- Prepartory work for more generic SoundWire SCDA controls
- Support for AMD ACP 7.x, AWINC WM88166, Everest ES8388, Intel AVS
PEAKVOL and GAIN DSP modules Mediatek MT8188 DMIC, NXP i.MX95,
nVidia Tegra interconnects, Rockchip RK3588 S/PDIF, Texas
Instruments SN012776 and TAS5770L, and Wolfson WM8904 DMICs
Others:
- Conversions of PM ops with the modern macros in the rest drivers
- USB-audio quirks and fixes for Presonus Studio, DJM-A9, CME
- HD-audio quirks and fixes ASUS, HP, Lenovo, and others"
* tag 'sound-6.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (651 commits)
ALSA: hda: tas2781-i2c: Remove unnecessary NULL check before release_firmware()
ALSA: hda: cs35l56: Remove unnecessary NULL check before release_firmware()
ALSA: hda/realtek: Bass speaker fixup for ASUS UM5606KA
ALSA: hda/realtek: Fix built-in mic assignment on ASUS VivoBook X515UA
ALSA: hda/realtek: Add support for various HP Laptops using CS35L41 HDA
ALSA: timer: Don't take register_mutex with copy_from/to_user()
ASoC: SDCA: Correct handling of selected mode DisCo property
ASoC: amd: yc: update quirk data for new Lenovo model
ALSA: hda/realtek: fix micmute LEDs on HP Laptops with ALC3247
ALSA: hda/realtek: fix micmute LEDs on HP Laptops with ALC3315
ASoC: SOF: mediatek: Commonize duplicated functions
ASoC: dmic: Fix NULL pointer dereference
ASoC: wm8904: add DMIC support
ASoC: wm8904: get platform data from DT
ASoC: dt-bindings: wm8904: Add DMIC, GPIO, MIC and EQ support
ASoC: wm8904: Don't touch GPIO configs set to 0xFFFF
of: Add of_property_read_u16_index
ALSA: oxygen: Fix dependency on CONFIG_PM_SLEEP
ASoC: ops: Apply platform_max after deciding control type
ASoC: ops: Remove some unnecessary local variables
...
Diffstat (limited to 'Documentation/sound')
| -rw-r--r-- | Documentation/sound/alsa-configuration.rst | 2 | ||||
| -rw-r--r-- | Documentation/sound/designs/powersave.rst | 6 | ||||
| -rw-r--r-- | Documentation/sound/soc/codec-to-codec.rst | 4 | ||||
| -rw-r--r-- | Documentation/sound/soc/dpcm.rst | 21 |
4 files changed, 18 insertions, 15 deletions
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 04254474fa04..a45174d165eb 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -58,7 +58,7 @@ debug 2 = verbose debug messages); This option appears only when ``CONFIG_SND_DEBUG=y``. This option can be dynamically changed via sysfs - /sys/modules/snd/parameters/debug file. + /sys/module/snd/parameters/debug file. Module snd-pcm-oss ------------------ diff --git a/Documentation/sound/designs/powersave.rst b/Documentation/sound/designs/powersave.rst index 138157452eb9..ca7d1e838b4d 100644 --- a/Documentation/sound/designs/powersave.rst +++ b/Documentation/sound/designs/powersave.rst @@ -25,15 +25,15 @@ operations. The ``power_save`` option is exported as writable. This means you can adjust the value via sysfs on the fly. For example, to turn on the automatic power-save mode with 10 seconds, write to -``/sys/modules/snd_ac97_codec/parameters/power_save`` (usually as root): +``/sys/module/snd_ac97_codec/parameters/power_save`` (usually as root): :: - # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save + # echo 10 > /sys/module/snd_ac97_codec/parameters/power_save Note that you might hear click noise/pop when changing the power state. Also, it often takes certain time to wake up from the -power-down to the active state. These are often hardly to fix, so +power-down to the active state. These are often hard to fix, so don't report extra bug reports unless you have a fix patch ;-) For HD-audio interface, there is another module option, diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst index 0418521b6e03..973c147d9d82 100644 --- a/Documentation/sound/soc/codec-to-codec.rst +++ b/Documentation/sound/soc/codec-to-codec.rst @@ -68,7 +68,7 @@ file: .codec_dai_name = "codec-2-dai_name", .platform_name = "samsung-i2s.0", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ignore_suspend = 1, .c2c_params = &dsp_codec_params, .num_c2c_params = 1, @@ -80,7 +80,7 @@ file: .codec_name = "codec-3, .codec_dai_name = "codec-3-dai_name", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ignore_suspend = 1, .c2c_params = &dsp_codec_params, .num_c2c_params = 1, diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst index 02419a6f8213..7b6aeab3c207 100644 --- a/Documentation/sound/soc/dpcm.rst +++ b/Documentation/sound/soc/dpcm.rst @@ -147,14 +147,16 @@ For the example above we have to define 4 FE DAI links and 6 BE DAI links. The FE DAI links are defined as follows :- :: + SND_SOC_DAILINK_DEFS(pcm0, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_PLATFORM("dsp-audio"))); + static struct snd_soc_dai_link machine_dais[] = { { .name = "PCM0 System", .stream_name = "System Playback", - .cpu_dai_name = "System Pin", - .platform_name = "dsp-audio", - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", + SND_SOC_DAILINK_REG(pcm0), .dynamic = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, }, @@ -174,15 +176,16 @@ dynamic and will change depending on runtime config. The BE DAIs are configured as follows :- :: + SND_SOC_DAILINK_DEFS(headset, + DAILINK_COMP_ARRAY(COMP_CPU("ssp-dai.0")), + DAILINK_COMP_ARRAY(COMP_CODEC("rt5640.0-001c", "rt5640-aif1"))); + static struct snd_soc_dai_link machine_dais[] = { .....< FE DAI links here > { .name = "Codec Headset", - .cpu_dai_name = "ssp-dai.0", - .platform_name = "snd-soc-dummy", + SND_SOC_DAILINK_REG(headset), .no_pcm = 1, - .codec_name = "rt5640.0-001c", - .codec_dai_name = "rt5640-aif1", .ignore_suspend = 1, .ignore_pmdown_time = 1, .be_hw_params_fixup = hswult_ssp0_fixup, @@ -362,7 +365,7 @@ The machine driver sets some additional parameters to the DAI link i.e. .codec_dai_name = "modem-aif1", .codec_name = "modem", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .c2c_params = &dai_params, .num_c2c_params = 1, } |
