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authorLinus Torvalds <torvalds@linux-foundation.org>2025-03-26 09:41:55 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2025-03-26 09:41:55 -0700
commite50da555ca4d42b1b98d0f26789db64f26a0919a (patch)
tree8335ba0c862679f4294284cba61c48d52abe0665 /Documentation/sound
parent1e26c5e28ca5821a824e90dd359556f5e9e7b89f (diff)
parent3a949fc08103c0ce3a1d0ef30459c7b3acc6a214 (diff)
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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.rst2
-rw-r--r--Documentation/sound/designs/powersave.rst6
-rw-r--r--Documentation/sound/soc/codec-to-codec.rst4
-rw-r--r--Documentation/sound/soc/dpcm.rst21
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,
}