December 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH alsa-ucm-conf 1/2] sof-soundwire: Add basic support for cs42l43
by Charles Keepax 06 Dec '23

06 Dec '23

cs42l43 is a codec device, add basic support for it. Including a dual channel DMIC input, stereo headphones, and a mono headset microphone. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- ucm2/sof-soundwire/cs42l43-dmic.conf | 28 +++++++++++++++++ ucm2/sof-soundwire/cs42l43.conf | 47 ++++++++++++++++++++++++++++ 2 files changed, 75 insertions(+) create mode 100644 ucm2/sof-soundwire/cs42l43-dmic.conf create mode 100644 ucm2/sof-soundwire/cs42l43.conf diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf new file mode 100644 index 0000000..bf59eca --- /dev/null +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -0,0 +1,28 @@ +# Use case Configuration for sof-soundwire card + +SectionDevice."Mic" { + Comment "Microphones" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + cset "name='cs42l43 DP1TX1 Input' 'Decimator 3'" + cset "name='cs42l43 DP1TX2 Input' 'Decimator 4'" + cset "name='cs42l43 Decimator 3 Switch' on" + cset "name='cs42l43 Decimator 4 Switch' on" + ] + + DisableSequence [ + cset "name='cs42l43 Decimator 3 Switch' off" + cset "name='cs42l43 Decimator 4 Switch' off" + cset "name='cs42l43 DP1TX1 Input' 'None'" + cset "name='cs42l43 DP1TX2 Input' 'None'" + ] + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId},4" + } +} diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf new file mode 100644 index 0000000..378dbb2 --- /dev/null +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -0,0 +1,47 @@ +# Use case Configuration for sof-soundwire card + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='cs42l43 Headphone L Input 1' 'DP5RX1'" + cset "name='cs42l43 Headphone R Input 1' 'DP5RX2'" + ] + + DisableSequence [ + cset "name='cs42l43 Headphone L Input 1' 'None'" + cset "name='cs42l43 Headphone R Input 1' 'None'" + ] + + Value { + PlaybackPriority 200 + PlaybackPCM "hw:${CardId},0" + PlaybackVolume "cs42l43 Headphone Digital Volume" + JackControl "Headphone Jack" + } +} + +SectionDevice."Headset" { + Comment "Headset Microphone" + + EnableSequence [ + cset "name='cs42l43 ADC1 Input' 'IN1'" + cset "name='cs42l43 Decimator 1 Mode' 'ADC'" + + cset "name='cs42l43 DP1TX1 Input' 'Decimator 1'" + cset "name='cs42l43 DP1TX2 Input' 'Decimator 1'" + ] + + DisableSequence [ + cset "name='cs42l43 DP1TX1 Input' 'None'" + cset "name='cs42l43 DP1TX2 Input' 'None'" + ] + + Value { + CapturePriority 200 + CapturePCM "hw:${CardId},4" + CaptureVolume "cs42l43 Decimator 1 Volume" + CaptureSwitch "cs42l43 Decimator 1 Switch" + JackControl "Headset Mic Jack" + } +} -- 2.39.2

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default pcm plugin
by GitHub issues - opened 06 Dec '23

06 Dec '23

alsa-project/alsa-lib issue #371 was opened from sylware: Is there a robust programming way to know if the canonical "default" pcm is actually a pipewire/pulseaudio/jack plugin? That at a device/pcm enumeration time, namely before any pcm is actually being configured. I was thinking about the "configuration" API, but I am not that sure it "resolves" everything including user-specific/system-specific configurations (asound rc files and env variables). Issue URL : https://github.com/alsa-project/alsa-lib/issues/371 Repository URL: https://github.com/alsa-project/alsa-lib

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ucm2: sof-soundwire: Use one file for speaker codec initialization
by GitHub pull_request - opened 06 Dec '23

06 Dec '23

alsa-project/alsa-ucm-conf pull request #370 was opened from perexg: Use macros to minimize configuration blocks. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/370 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/370.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: add new framework to quirk chain
by Tim Bosse 06 Dec '23

06 Dec '23

The Framework Laptop 13 (AMD Ryzen 7040Series) has an ALC 295 with a disconnected or faulty headset mic presence detect and works with the same quirk chain as 309d7363ca3d9fcdb92ff2d958be14d7e8707f68. This model has a VID:PID of f111:0006. Signed-off-by: Tim Bosse <flinn(a)timbos.se> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f9ddacfd920e..59f2578839ed 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10271,6 +10271,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. -- 2.43.0

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[GIT PULL] ASoC fixes for v6.7-rc4
by Mark Brown 06 Dec '23

06 Dec '23

The following changes since commit b85ea95d086471afb4ad062012a4d73cd328fa86: Linux 6.7-rc1 (2023-11-12 16:19:07 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.7-rc4 for you to fetch changes up to 0a10d15280a385e5971fb58a6d2eddbf7c0aa9f3: ASoC: qcom: Limit Digital gains on speaker (2023-12-04 21:42:29 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.7 A crop of fixes for v6.7, one core fix for a merge issue and a bunch of driver specific fixes and new IDs, mostly for x86 platforms. ---------------------------------------------------------------- AngeloGioacchino Del Regno (1): ASoC: SOF: mediatek: mt8186: Add Google Steelix topology compatible Chancel Liu (1): ASoC: imx-rpmsg: SND_SOC_IMX_RPMSG should depend on OF and I2C Charles Keepax (1): ASoC: wm8974: Correct boost mixer inputs David Lin (1): ASoC: nau8822: Fix incorrect type in assignment and cast to restricted __be16 David Rau (1): ASoC: da7219: Support low DC impedance headset Dinghao Liu (1): ASoC: wm_adsp: fix memleak in wm_adsp_buffer_populate Jeremy Soller (1): ASoC: amd: yc: Add DMI entry to support System76 Pangolin 13 Johan Hovold (1): ASoC: soc-pcm: fix up bad merge Kamil Duljas (3): ASoC: Intel: Skylake: Fix mem leak in few functions ASoC: SOF: topology: Fix mem leak in sof_dai_load() ASoC: Intel: Skylake: mem leak in skl register function Maciej Strozek (2): ASoC: cs43130: Fix the position of const qualifier ASoC: cs43130: Fix incorrect frame delay configuration Malcolm Hart (1): ASoC: amd: yc: Fix non-functional mic on ASUS E1504FA Marian Postevca (1): ASoC: amd: acp: Add support for a new Huawei Matebook laptop Mark Brown (3): ASoC: Fixes for cs43130 ASoC: SOF: Extend the enabled DSP core handling ASoC: qcom: Limit Digital gains on speaker Matus Malych (1): ASoC: amd: yc: Add HP 255 G10 into quirk table Neil Armstrong (1): ASoC: codecs: lpass-tx-macro: set active_decimator correct default value Peter Ujfalusi (5): ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available ASoC: Intel: sof_sdw: Always register the HDMI dai links ASoC: hdac_hda: Conditionally register dais for HDMI and Analog ASoC: SOF: ipc4-topology: Correct data structures for the SRC module ASoC: SOF: ipc4-topology: Correct data structures for the GAIN module Ranjani Sridharan (2): ASoC: SOF: ipc4-topology: Add core_mask in struct snd_sof_pipeline ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores Shengjiu Wang (3): ASoC: fsl_sai: Fix no frame sync clock issue on i.MX8MP ASoC: fsl_xcvr: Enable 2 * TX bit clock for spdif only case ASoC: fsl_xcvr: refine the requested phy clock frequency Shuming Fan (1): ASoC: rt5650: add mutex to avoid the jack detection failure Srinivas Kandagatla (2): ASoC: ops: add correct range check for limiting volume ASoC: qcom: sc8280xp: Limit speaker digital volumes sound/soc/amd/acp-config.c | 14 ++++++ sound/soc/amd/yc/acp6x-mach.c | 21 +++++++++ sound/soc/codecs/cs43130.c | 6 +-- sound/soc/codecs/da7219-aad.c | 2 +- sound/soc/codecs/hdac_hda.c | 23 ++++++++-- sound/soc/codecs/lpass-tx-macro.c | 5 +++ sound/soc/codecs/nau8822.c | 9 ++-- sound/soc/codecs/rt5645.c | 10 ++++- sound/soc/codecs/wm8974.c | 6 +-- sound/soc/codecs/wm_adsp.c | 8 +++- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_sai.c | 21 +++++++++ sound/soc/fsl/fsl_xcvr.c | 14 +++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 + sound/soc/intel/boards/sof_sdw.c | 17 ++++---- sound/soc/intel/skylake/skl-pcm.c | 9 +++- sound/soc/intel/skylake/skl-sst-ipc.c | 4 +- sound/soc/qcom/sc8280xp.c | 17 ++++++++ sound/soc/soc-ops.c | 2 +- sound/soc/soc-pcm.c | 11 ++--- sound/soc/sof/ipc3-topology.c | 2 + sound/soc/sof/ipc4-control.c | 20 ++++----- sound/soc/sof/ipc4-topology.c | 61 +++++++++++++++----------- sound/soc/sof/ipc4-topology.h | 34 +++++++++++---- sound/soc/sof/mediatek/mt8186/mt8186.c | 3 ++ sound/soc/sof/sof-audio.c | 65 ++++++++++++++++++---------- sound/soc/sof/sof-audio.h | 2 + sound/soc/sof/topology.c | 4 +- 28 files changed, 286 insertions(+), 107 deletions(-)

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[PATCH] ASoC: Intel: sof_sdw_cs_amp: Connect outputs to a speaker widget
by Richard Fitzgerald 06 Dec '23

06 Dec '23

Hookup the CS35L56 DAPM_OUTPUT widgets to a DAPM_SPK widget so that there is a complete logical path to a speaker. There is no particular reason to use multiple speaker widgets. The CS35L56 are designed to work together as a set so they have all been connected to a single speaker widget. Instead of a hardcoded list of codec widget names, the code walks through all the codecs on the dailink and for every cs35l56 it uses its name prefix to construct the source end of the route. This adds a small amount of overhead during probe but has the benefit that it isn't dependent on every system using the same prefixes. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_sdw_cs_amp.c | 30 +++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index 98f6546f484b..f88c01552a92 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -9,15 +9,24 @@ #include <linux/errno.h> #include <sound/soc.h> #include <sound/soc-acpi.h> +#include <sound/soc-dai.h> #include "sof_sdw_common.h" #define CODEC_NAME_SIZE 8 +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + static int cs_spk_init(struct snd_soc_pcm_runtime *rtd) { const char *dai_name = rtd->dai_link->codecs->dai_name; struct snd_soc_card *card = rtd->card; char codec_name[CODEC_NAME_SIZE]; + char widget_name[16]; + struct snd_soc_dapm_route route = { "Speakers", NULL, widget_name }; + struct snd_soc_dai *codec_dai; + int i, ret; snprintf(codec_name, CODEC_NAME_SIZE, "%s", dai_name); card->components = devm_kasprintf(card->dev, GFP_KERNEL, @@ -26,17 +35,34 @@ static int cs_spk_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; + ret = snd_soc_dapm_new_controls(&card->dapm, sof_widgets, + ARRAY_SIZE(sof_widgets)); + if (ret) { + dev_err(card->dev, "widgets addition failed: %d\n", ret); + return ret; + } + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (!strstr(codec_dai->name, "cs35l56")) + continue; + + snprintf(widget_name, sizeof(widget_name), "%s SPK", + codec_dai->component->name_prefix); + ret = snd_soc_dapm_add_routes(&card->dapm, &route, 1); + if (ret) + return ret; + } + return 0; } - int sof_sdw_cs_amp_init(struct snd_soc_card *card, 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. */ + /* Do init on playback link only. */ if (!playback) return 0; -- 2.30.2

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aplay: Add option for specifying subformat
by GitHub pull_request - edited 06 Dec '23

06 Dec '23

alsa-project/alsa-utils pull request #228 was edited from crojewsk-intel: Make subformat first-class citizen by allowing users to specify it just like it is the case for the format. Default to SND_PCM_SUBFORMAT_STD so they are no surprises. Request URL : https://github.com/alsa-project/alsa-utils/pull/228 Patch URL : https://github.com/alsa-project/alsa-utils/pull/228.patch Repository URL: https://github.com/alsa-project/alsa-utils

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pcm: Add MSBITS subformat options
by GitHub pull_request - edited 06 Dec '23

06 Dec '23

alsa-project/alsa-lib pull request #342 was edited from crojewsk-intel: Improve granularity of format selection for S32/U32 formats by adding masks representing 20, 24 and 32 most significant bits. As a preparation step introduce `snd_pcm_subformat_value()` for easy name-to-subformat conversions. While at it, fix few warnings that pop up in subformat-related code during compilation. Request URL : https://github.com/alsa-project/alsa-lib/pull/342 Patch URL : https://github.com/alsa-project/alsa-lib/pull/342.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2 0/5] ASoC: codecs: add support for WCD939x Codec
by Neil Armstrong 06 Dec '23

06 Dec '23

Add the main WCD9390/WCD9395 Audio Codec driver to support: - 4 ADC inputs for up to 5 Analog Microphones - 4 DMIC inputs for up to 8 Digital Microphones - 4 Microphone BIAS - Stereo Headphone output - Mono EAR output - MBHC engine for Headset Detection This adds: - bindings - MBHC changes to support Type-C muc - Soundwire Slave driver - Code driver The USB Mux subsystem support will be submitted separalety since it's a functionally separate subsystem connected over I2C with it's own reset signal. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Changes in v2: - Bindings: - Dropped all references to "Slave" or "Host" terminology when possible - Collected review for first patch - Added wcd9395 as fallback of wcd9390 - Fixes typos errors - MBHC: - Dropped all references to "Slave" or "Host" terminology when possible - Fixed EXPORT_SYMBOL into EXPORT_SYMBOL_GPL - Fixed typo in commit message - Soundwire Devices driver - Dropped all references to "Slave" or "Host" terminology when possible - Dropped comments and unused code - Reworked wcd939x_swr_get_current_bank() - Added comments to wcd9390_interrupt_callback() - Reworked regmap's wcd939x_readonly/volatile_register checks - Added comments explaining while bind/unbind are empty - Added comment on SDW_SLAVE_ENTRY meaning - Added more register fields defines - Style fixes - Codec driver - Dropped all references to "Slave" or "Host" terminology when possible - Added MICB_BIAS_ values enum and used them in the code - Moved zdet_param to the top - Added TLV data for ear_pa_gain and used it - Defined as much as possible every bit field used on/from register - Replaced 1/0 to true/false when writing to single bit fields - Replaced for loop on all bits with ffs(), simplified code - Simplified MICB fields handling code - Reworked and simplified wcd939x_get/set_compander and other kcontrol get/setters - Reworked and simplified MHGC impedance/zdet/qval code, dropped dead code - Added comments on wcd939x_wd_handle_irq() utility - Added comment on the interrupt handling architecture - I've not moved common code yet, I'll probably do later since it would alter wcd939x code - Link to v1: https://lore.kernel.org/r/20231123-topic-sm8650-upstream-wcd939x-codec-v1-0… --- Neil Armstrong (5): ASoC: dt-bindings: qcom,wcd938x: move out common properties ASoC: dt-bindings: document WCD939x Audio Codec ASoC: codec: wcd-mbhc-v2: add support when connected behind an USB-C audio mux ASoC: codecs: Add WCD939x Soundwire devices driver ASoC: codecs: Add WCD939x Codec driver .../devicetree/bindings/sound/qcom,wcd938x.yaml | 81 +- .../bindings/sound/qcom,wcd939x-sdw.yaml | 69 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 96 + .../bindings/sound/qcom,wcd93xx-common.yaml | 95 + sound/soc/codecs/Kconfig | 18 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd-clsh-v2.h | 1 + sound/soc/codecs/wcd-mbhc-v2.c | 95 +- sound/soc/codecs/wcd-mbhc-v2.h | 3 + sound/soc/codecs/wcd939x-sdw.c | 1584 +++++++++ sound/soc/codecs/wcd939x.c | 3655 ++++++++++++++++++++ sound/soc/codecs/wcd939x.h | 983 ++++++ 12 files changed, 6592 insertions(+), 95 deletions(-) --- base-commit: 07b677953b9dca02928be323e2db853511305fa9 change-id: 20231123-topic-sm8650-upstream-wcd939x-codec-c46f621b55d4 Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

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BUG hda/cs8904: Headphones not detected and no speaker sound after system suspend
by Andrés Pedraza 05 Dec '23

05 Dec '23

Hello my dudes. I've been having this bug for a few months now. Sound on my laptop works just fine until the system suspends or I close the lid. If my headphones are connected when I suspend the system, then there's no sound on headphones or speakers on wake and PulseAudio claims there's never any headphones connected. If my headphones aren't connected on suspend, then speakers work just fine, but headphones are still never detected and make no sound. Additionally, I recently found out that after waking from suspend, if I run `sudo dmesg | grep snd` I find two timeout warnings that weren't there before, and more showing up as I keep running the command. They go: > snd_hda_codec_cs8409 hdaudioC1D0: Timeout waiting for PDN_DONE for > CS42L42 I'm on a Dell Vostro 3405 laptop, which I know has a Cirrus sound card. I recently compiled and tested this bug on the Linux 6.6.2 stable kernel. No dice. I attach the outputs of `pactl list sinks`, `sudo dmesg | grep snd` and `sudo lspci -v | grep snd -B 10`. Please do tell if the issue might be caused by some other module or BIOS configuration, or if you need more data.

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