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Broken microphone switch on mtk-rt5650
by GitHub issues - edited 20 Jul '23

20 Jul '23

alsa-project/alsa-ucm-conf issue #336 was edited from jenneron: The problem is that both headphones and mic use `JackControl "Headset Jack"` When there are headphones without mic connected it switches to external mic which is not present. In kernel there appears to be only one switch https://github.com/torvalds/linux/blob/3c8b5861850c734add65233e538d4a8c2dff… responsible for handling both headphone and microphone. So, how is userspace supposed to distinguish connection of headset and headphones? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/336 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Broken microphone switch on mtk-rt5650
by GitHub issues - opened 20 Jul '23

20 Jul '23

alsa-project/alsa-ucm-conf issue #336 was opened from jenneron: The problem is that both headphones and mic use `JackControl "Headset Jack"` When there are headphones without mic connected it switches to external mic which is not present. In kernel there appears to be only one switch https://github.com/torvalds/linux/blob/3c8b5861850c734add65233e538d4a8c2dff… responsible for handling both headphone and microphone. So, how is userspace supposed to distinguish connection of headset and headphones without mic? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/336 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: fsl_spdif: Add support for 22.05 kHz sample rate
by Matus Gajdos 19 Jul '23

19 Jul '23

Add support for 22.05 kHz sample rate for TX. Signed-off-by: Matus Gajdos <matuszpd(a)gmail.com> --- sound/soc/fsl/fsl_spdif.c | 8 ++++++-- sound/soc/fsl/fsl_spdif.h | 6 ++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 015c3708aa04..95e639711eba 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -514,6 +514,10 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, int ret; switch (sample_rate) { + case 22050: + rate = SPDIF_TXRATE_22050; + csfs = IEC958_AES3_CON_FS_22050; + break; case 32000: rate = SPDIF_TXRATE_32000; csfs = IEC958_AES3_CON_FS_32000; @@ -1422,7 +1426,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, struct clk *clk, u64 savesub, enum spdif_txrate index, bool round) { - static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + static const u32 rate[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, }; bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; @@ -1483,7 +1487,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index) { - static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + static const u32 rate[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, }; struct platform_device *pdev = spdif_priv->pdev; struct device *dev = &pdev->dev; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 75b42a692c90..2bc1b10c17d4 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -175,7 +175,8 @@ enum spdif_gainsel { /* SPDIF tx rate */ enum spdif_txrate { - SPDIF_TXRATE_32000 = 0, + SPDIF_TXRATE_22050 = 0, + SPDIF_TXRATE_32000, SPDIF_TXRATE_44100, SPDIF_TXRATE_48000, SPDIF_TXRATE_88200, @@ -191,7 +192,8 @@ enum spdif_txrate { #define SPDIF_QSUB_SIZE (SPDIF_UBITS_SIZE / 8) -#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ +#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | \ SNDRV_PCM_RATE_88200 | \ -- 2.25.1

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[PATCH] ASoC: mediatek: mt8188: DPCM used FE and BE merged parameters
by Trevor Wu 19 Jul '23

19 Jul '23

To ensure that DPCM takes into account the backend hardware limitations when user space queries the hw_params of a device, we need to add dpcm_merged_format, dpcm_merged_chan, and dpcm_merged_rate to the FE dai_links. This patch includes only stereo FE dai_links, since multi-channel FEs may be reserved for specific purposes. Therefore, it may not be appropriate to consider BE conditions. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index ac69c23e0da1..7c9e08e6a4f5 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -723,6 +723,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(playback2), }, [DAI_LINK_DL3_FE] = { @@ -734,6 +737,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(playback3), }, [DAI_LINK_DL6_FE] = { @@ -745,6 +751,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(playback6), }, [DAI_LINK_DL7_FE] = { @@ -833,6 +842,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(capture4), }, [DAI_LINK_UL5_FE] = { @@ -844,6 +856,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(capture5), }, [DAI_LINK_UL6_FE] = { -- 2.18.0

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[PATCH] ASoC: amd: acp: fix SND_SOC_AMD_ACP_PCI depdenencies
by Arnd Bergmann 19 Jul '23

19 Jul '23

From: Arnd Bergmann <arnd(a)arndb.de> The new PM functions require code that is part of the snd-acp-legacy-common module: x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `snd_acp_resume': acp-pci.c:(.text+0x23): undefined reference to `acp_init' x86_64-linux-ld: acp-pci.c:(.text+0x58): undefined reference to `acp_enable_interrupts' x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `snd_acp_suspend': acp-pci.c:(.text+0x89): undefined reference to `acp_deinit' x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `acp_pci_remove': acp-pci.c:(.text+0xec): undefined reference to `acp_deinit' x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `acp_pci_probe': acp-pci.c:(.text+0x26b): undefined reference to `acp_init' Select that Kconfig symbol as is done for the other frontends. Fixes: 088a40980efbc ("ASoC: amd: acp: add pm ops support for acp pci driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/amd/acp/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 9e31b5d167908..631cdf96d6376 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -30,6 +30,7 @@ config SND_SOC_AMD_ACP_PCM config SND_SOC_AMD_ACP_PCI tristate "AMD ACP PCI Driver Support" + select SND_SOC_AMD_ACP_LEGACY_COMMON depends on X86 && PCI help This options enables generic PCI driver for ACP device. -- 2.39.2

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[PATCH 0/7] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 19 Jul '23

19 Jul '23

Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. I suspect the avs and skylake drivers used snd_hdac_i915_init purely for the modprobe, but I don't have the hardware to test if it can be safely removed. It can still be done easily in a followup patch to simplify probing. Maarten Lankhorst (7): ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Remove deferred probe for SOF ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org sound/hda/hdac_i915.c | 15 +++------ sound/pci/hda/hda_intel.c | 58 +++++++++++++++++++---------------- sound/soc/intel/avs/core.c | 13 +++++--- sound/soc/intel/skylake/skl.c | 31 ++++++------------- sound/soc/sof/Kconfig | 19 ------------ sound/soc/sof/core.c | 38 ++--------------------- sound/soc/sof/intel/Kconfig | 1 - sound/soc/sof/intel/hda.c | 32 +++++++++++-------- sound/soc/sof/sof-pci-dev.c | 3 +- sound/soc/sof/sof-priv.h | 5 --- 10 files changed, 75 insertions(+), 140 deletions(-) -- 2.39.2

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[PATCH v2 0/5] ASoC: Improve coverage in default KUnit runs
by Mark Brown 19 Jul '23

19 Jul '23

We have some KUnit tests for ASoC but they're not being run as much as they should be since ASoC isn't enabled in the configs used by default with KUnit and in the case of the topology tests there is no way to enable them without enabling drivers that use them. This series provides a Kconfig option which KUnit can use directly rather than worry about drivers. Further, since KUnit is typically run in UML but ALSA prevents build with UML we need to remove that Kconfig conflict. As far as I can tell the motiviation for this is that many ALSA drivers use iomem APIs which are not available under UML and it's more trouble than it's worth to go through and add per driver dependencies. In order to avoid these issues we also provide stubs for these APIs so there are no build time issues if a driver relies on iomem but does not depend on it. With these stubs I am able to build all the sound drivers available in a UML defconfig (UML allmodconfig appears to have substantial other issues in a quick test). With this series I am able to run the topology KUnit tests as part of a kunit --alltests run. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Changes in v2: - Add support for building ALSA with UML. - Link to v1: https://lore.kernel.org/r/20230712-asoc-topology-kunit-enable-v1-0-b9f2da9d… --- Mark Brown (5): driver core: Provide stubs for !IOMEM builds platform: Provide stubs for !HAS_IOMEM builds ALSA: Enable build with UML kunit: Enable ASoC in all_tests.config ASoC: topology: Add explicit build option include/linux/device.h | 26 ++++++++++++++++++++++++++ include/linux/platform_device.h | 28 ++++++++++++++++++++++++++++ sound/Kconfig | 4 ---- sound/soc/Kconfig | 11 +++++++++++ tools/testing/kunit/configs/all_tests.config | 5 +++++ 5 files changed, 70 insertions(+), 4 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230701-asoc-topology-kunit-enable-5e8dd50d0ed7 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/2] ASoC: Improve coverage in default KUnit runs
by Mark Brown 19 Jul '23

19 Jul '23

We have some KUnit tests for ASoC but they're not being run as much as they should be since ASoC isn't enabled in the configs used by default with KUnit and in the case of the topolofy tests there is no way to enable them without enabling drivers that use them. Let's improve that. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (2): kunit: Enable ASoC in all_tests.config ASoC: topology: Add explicit build option sound/soc/Kconfig | 11 +++++++++++ tools/testing/kunit/configs/all_tests.config | 5 +++++ 2 files changed, 16 insertions(+) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230701-asoc-topology-kunit-enable-5e8dd50d0ed7 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/3] regmap: Add interface for checking if a register is cached
by Mark Brown 19 Jul '23

19 Jul '23

HDA has a use case for checking if a register is present in the cache which it awkwardly open codes with use of _cache_only() and a read, provide a direct API for this. --- Mark Brown (3): regmap: Let users check if a register is cached regmap: Provide test for regcache_reg_present() ALSA: hda: Use regcache_reg_cached() rather than open coding drivers/base/regmap/regcache.c | 23 ++++++++++++++++++++++ drivers/base/regmap/regmap-kunit.c | 40 ++++++++++++++++++++++++++++++++++++++ include/linux/regmap.h | 1 + sound/hda/hdac_regmap.c | 9 +++------ 4 files changed, 67 insertions(+), 6 deletions(-) --- base-commit: 3953d5c79c21defa716624a8623c4157c0f2fee0 change-id: 20230716-regmap-cache-check-6f6939f41ed5 Best regards, -- Mark Brown <broonie(a)kernel.org>

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Fwd: Odd issue with ALSA connecting directly to HDMI audio output
by Ico Bukvic 19 Jul '23

19 Jul '23

Hi all, I have a consumer-level setup audio question. My research project uses a Raspberry Pi in conjunction with a small portable monitor that has built-in loudspeakers that are to be used via HDMI connection. The goal here is portability and minimal cost. Curiously, when trying to connect to monitor's loudspeakers (via HDMI) using ALSA, no matter what settings I pick, at worst I cannot get audio out, and at best, get a buzzy sound whose waveform is in the attached image. The sound is clearly periodic and impulse-like. It happens every 59 samples. Changing sample rate increases pitch, suggesting 59 samples remain constant regardless of the sampling rate. When playing the actual audio, you can hear it in the background, but this buzz is easily overpowering it. Changing sample rate or bit depth, or buffering, makes no difference. Selecting 8 channels for output (one of HDMI's standards, as far as I can tell), sometimes manages to output a tiny chunk of a good audio sample, and then goes silent. Now, even more curiously, when using the portaudio backend and connecting to the default pulseaudio sink, works just fine. What could be causing this discrepancy? Best, Ico -- Ivica Ico Bukvic, D.M.A. Director, Creativity + Innovation Director, Human-Centered Design iPhD Institute for Creativity, Arts, and Technology Virginia Tech Creative Technologies in Music School of Performing Arts – 0141 Blacksburg, VA 24061 (540) 231-6139 ico(a)vt.edu ci.icat.vt.edu l2ork.icat.vt.edu ico.bukvic.net

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