April 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: soc-pcm: fixup soc_pcm_params_symmetry()
by Kuninori Morimoto 16 Apr '21

16 Apr '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 3a9067211122 ("ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()") cleanups soc_pcm_params_symmetry() by addig new __soc_pcm_params_symmetry() macro. It checks symmetry first, and checks each DAI settings if symmetry was true. But original code checked symmetric_rate : DAI_Link / CPU (A) symmetric_channels : DAI_Link / CPU / Codec (B) symmetric_sample_bits : DAI_Link / CPU / Codec (B) (A) was using for_each_rtd_cpu_dais() (B) was using for_each_rtd_dais() Current code is using (B) for all symmetric_xxx. This is bug. One note is that we can use .be_hw_params_fixup in DPCM case. This means, BE settings might be fixuped/updated by FE. It can be happen not only for rate, but for channels/sample_bits too. And also, DPCM uses dummy-DAI which will be used for all DPCM connectoin. ALSA SoC will clean DAI params (a) if it was last user (b). But dummy-DAI is used from many place, it might not be cleaned. static int soc_pcm_hw_clean(...) { ... for_each_rtd_dais(rtd, i, dai) { ... (b) if (snd_soc_dai_active(dai) == 1) (a) soc_pcm_set_dai_params(dai, NULL); ... } ... } The solution maybe (A1) Symmetric checks CPU only (A2) Symmetric checks both CPU / Codec, but ignores dummy-DAI This patch selects A1. For example, if Sound Card is exchanging all rate to 48kHz by using .be_hw_params_fixup() on DPCM, below can be happen. It is using 44100 Hz, but has mismatch between dummy-DAI which is keeping 48kHz from 1st aplay. # aplay 44100.wav # aplay 44100.wav => [kernel] be.ak4613-hifi: ASoC: unmatched rate symmetry: 44100 - 48000 [kernel] be.ak4613-hifi: ASoC: hw_params BE failed -22 [kernel] fe.rsnd-dai.0: ASoC: hw_params BE failed -22 aplay: set_params:1407: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (23219 23220) PERIOD_SIZE: 1024 PERIOD_BYTES: 4096 PERIODS: 4 BUFFER_TIME: (92879 92880) BUFFER_SIZE: 4096 BUFFER_BYTES: 16384 TICK_TIME: 0 Fixes: 3a9067211122 ("ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Hi Mark This patch is needed for latest linus tree (= for v5.12). Please let me know if you want (A2). sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 02968a4e52b4..92e95e4aef9f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -388,7 +388,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, #define __soc_pcm_params_symmetry(name) \ symmetry = rtd->dai_link->symmetric_##name; \ - for_each_rtd_dais(rtd, i, dai) \ + for_each_rtd_cpu_dais(rtd, i, dai) \ symmetry |= dai->driver->symmetric_##name; \ \ if (symmetry) \ -- 2.25.1

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[PATCH 0/1] ALSA: hda/realtek: fix mic boost on Intel NUC 8
by Phil Calvin 16 Apr '21

16 Apr '21

I realized that I mistakenly sent this patch only to the listserv, and not directly to the relevant maintainers as I should have. I'm sorry for my error and the resulting list noise! A big thank you to everybody that helps maintain the Intel HDA subsystem. It is a well-documented and clear codebase to enter as a first-time kernel hacker. Thanks also to Dr. Iwai for `hda-emu`, which saved me a lot of rebooting while testing. I have tested the patch on this hardware both by recompiling the relevant modules and running them with the 5.10 kernel distributed in Debian `testing`, and atop the 5.12-rc6 kernel built with `make deb-pkg`. I could not decipher the ordering of the fixup tables, so I am of course happy to move these around to fit the organizational scheme as needed. I'm not an EE, but I am inclined to think the upper two boost values can probably be made to work by some kind of software workaround, given that they do produce analog noise coming from something. If you have any hunches about a possible workaround based on experience with other similar codecs, please point me to those fixups on- or off-list and I can try them out on my hardware. Phil Phil Calvin (1): ALSA: hda/realtek: fix mic boost on Intel NUC 8 sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) -- 2.30.2

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[PATCH v3 0/2] clean up dt_binding_check and dtbs_check warnings of renesas, rsnd.yaml
by Zhen Lei 15 Apr '21

15 Apr '21

v2 --> v3: Modified based on Rob's review comments. For example, "ports" and "port" are not allowed at the same time. For the details, please refer: https://www.mail-archive.com/linux-kernel@vger.kernel.org/msg2539753.html v1 --> v2: The analysis was not clear in the previous version, "$ref: audio-graph.yaml#" is not required. It was then adjusted based on the following patch series: [1] https://lore.kernel.org/r/20210323163634.877511-1-robh@kernel.org/ v1: When I do dt_binding_check for all YAML files, below warning is reported: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.example.dt.yaml: sound@ec500000: 'dais' is a required property From schema: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml make dt_binding_check -j24 2>err.txt Before: cat err.txt | wc -l 2 After: cat err.txt | wc -l 0 make dtbs_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -j24 2>err.txt Before: cat err.txt | wc -l 130 After: cat err.txt | wc -l 0 Zhen Lei (2): ASoC: dt-bindings: renesas, rsnd: Clear warning 'dais' is a required property ASoC: dt-bindings: renesas, rsnd: Clear warning 'ports' does not match any of the regexes .../devicetree/bindings/sound/renesas,rsnd.yaml | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) -- 2.26.0.106.g9fadedd

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[PATCH v2] ASoC: SOF: use current DAI config during resume
by Pierre-Louis Bossart 15 Apr '21

15 Apr '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Recently, the sof_pcm_dai_link_fixup() function was updated to match SSP config with the PCM hw_params and set the current_config for the DAI widget. But the sof_restore_pipelines() function still chooses the default config for the DAI widget upon resuming. Fix this to use the last used config when setting up the DAI widget during resume. Fixes: c943a586f6e49 ("ASoC: SOF: match SSP config with pcm hw params") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: used SHA1 from Mark's tree, not the SOF one. Resending on behalf of Ranjani -> added Signed-off-by sound/soc/sof/sof-audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0d0d47dc0246..510883cd9107 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -267,7 +267,7 @@ int sof_restore_pipelines(struct device *dev) /* restore dai links */ list_for_each_entry_reverse(dai, &sdev->dai_list, list) { struct sof_ipc_reply reply; - struct sof_ipc_dai_config *config = dai->dai_config; + struct sof_ipc_dai_config *config = &dai->dai_config[dai->current_config]; if (!config) { dev_err(dev, "error: no config for DAI %s\n", -- 2.25.1

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[PATCH v1] ASoC: Intel: kbl_da7219_max98927: Fix kabylake_ssp_fixup function
by Lukasz Majczak 15 Apr '21

15 Apr '21

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Tested for all use cases of the driver. Based on similar fix in kbl_rt5663_rt5514_max98927.c from Harsha Priya <harshapriya.n(a)intel.com> and Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Cc: <stable(a)vger.kernel.org> # 5.4+ Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- Hi, This is basically a cherry-pick of this change: https://patchwork.kernel.org/project/alsa-devel/patch/1595432147-11166-1-gi… just applied to the kbl_da7219_max98927. Best regards, Lukasz sound/soc/intel/boards/kbl_da7219_max98927.c | 38 +++++++++++++++----- 1 file changed, 30 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 9dfe5bd9180d..4b7b4a044f81 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -284,11 +284,33 @@ 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. @@ -311,9 +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); @@ -324,7 +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; -- 2.25.1

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[PATCH 0/5] ASoC: simple-card/audio-graph: adjust to multi CPU/Codec
by Kuninori Morimoto 15 Apr '21

15 Apr '21

Hi Mark These patches adjusts to multi CPU/Codec on simple-card / audio-graph. This is part of prepare for new audio-graph-card2. Kuninori Morimoto (5): ASoC: simple-card: remove unused variable from simple_parse_of() ASoC: simple-card: use asoc_link_to_xxx() macro ASoC: simple-card: use simple_props_to_xxx() macro ASoC: audio-graph: use asoc_link_to_xxx() macro ASoC: audio-graph: use simple_props_to_xxx() macro sound/soc/generic/audio-graph-card.c | 43 +++++++++++++----------- sound/soc/generic/simple-card.c | 50 +++++++++++++--------------- 2 files changed, 47 insertions(+), 46 deletions(-) -- 2.25.1

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[PATCH] ASoC: codecs: wcd934x: Fix missing IRQF_ONESHOT as only threaded handler
by zhuguangqing83＠gmail.com 15 Apr '21

15 Apr '21

From: Guangqing Zhu <zhuguangqing83(a)gmail.com> Coccinelle noticed: sound/soc/codecs/wcd934x.c:5041:7-32: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT Signed-off-by: Guangqing Zhu <zhuguangqing83(a)gmail.com> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 5fe403307b72..cddc49bbb7f6 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -5040,7 +5040,7 @@ static int wcd934x_codec_probe(struct platform_device *pdev) ret = devm_request_threaded_irq(dev, irq, NULL, wcd934x_slim_irq_handler, - IRQF_TRIGGER_RISING, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "slim", wcd); if (ret) { dev_err(dev, "Failed to request slimbus irq\n"); -- 2.17.1

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[PATCH] ASoC: SOF: use current DAI config during resume
by Ranjani Sridharan 15 Apr '21

15 Apr '21

Recently, the sof_pcm_dai_link_fixup() function was updated to match SSP config with the PCM hw_params and set the current_config for the DAI widget. But the sof_restore_pipelines() function still chooses the default config for the DAI widget upon resuming. Fix this to use the last used config when setting up the DAI widget during resume. Fixes: a1e1f10a6ed6 ("ASoC: SOF: match SSP config with pcm hw params") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0d0d47dc0246..510883cd9107 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -267,7 +267,7 @@ int sof_restore_pipelines(struct device *dev) /* restore dai links */ list_for_each_entry_reverse(dai, &sdev->dai_list, list) { struct sof_ipc_reply reply; - struct sof_ipc_dai_config *config = dai->dai_config; + struct sof_ipc_dai_config *config = &dai->dai_config[dai->current_config]; if (!config) { dev_err(dev, "error: no config for DAI %s\n", -- 2.25.1

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[PATCH 00/14] ASoC: simple-card-utils: prepare for multi support
by Kuninori Morimoto 15 Apr '21

15 Apr '21

Hi Mark I want to add new audio-graph-card2 driver which can support not only DPCM, but also Multi-CPU/Codec, and Codec2Codec. And it is also supporting audio-graph-card2 base custom driver. But before supporting such driver, we need to cleanup existing simple-card / audio-graph, because these and new driver are sharing code. This patch-set are for Multi-CPU/Codec support, and some cleanups. Kuninori Morimoto (14): ASoC: simple-card-utils: enable flexible CPU/Codec/Platform ASoC: simple-card-utils: share dummy DAI and reduce memory ASoC: simple-card-utils: setup dai_props cpu_dai/codec_dai at initial timing ASoC: simple-card-utils: remove li->dais/li->conf ASoC: simple-card-utils: use for_each_prop_xxx() ASoC: simple-card-utils: remove asoc_simple_parse_xxx() ASoC: simple-card-utils: care multi DAI at asoc_simple_clean_reference() ASoC: simple-card-utils: indicate dai_fmt if exist ASoC: simple-card-utils: indicate missing CPU/Codec numbers for debug ASoC: simple-card-utils: add simple_props_to_xxx() macro ASoC: simple-card-utils: multi support at asoc_simple_canonicalize_cpu/platform() ASoC: simple-card-utils: tidyup debug info for clock ASoC: simple-card-utils: tidyup dev_dbg() to use 1 line ASoC: simple-card-utils: tidyup asoc_simple_parse_convert() include/sound/simple_card_utils.h | 116 +++++++---- sound/soc/generic/audio-graph-card.c | 90 ++++----- sound/soc/generic/simple-card-utils.c | 272 +++++++++++++++++--------- sound/soc/generic/simple-card.c | 104 +++++----- 4 files changed, 352 insertions(+), 230 deletions(-) -- 2.25.1

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[RFC PATCH 0/2] ASoC: remove cppchecks warnings on lm49453 and da732x
by Pierre-Louis Bossart 15 Apr '21

15 Apr '21

There are the last two patches in the cleanups, this time I am not sure what the code does and what the proper fix might be. Feedback welcome. Pierre-Louis Bossart (2): ASoC: lm49453: fix useless assignment before return ASoC: da732x: simplify code sound/soc/codecs/da732x.c | 17 ++++++----------- sound/soc/codecs/da732x.h | 12 ++++-------- sound/soc/codecs/lm49453.c | 2 -- 3 files changed, 10 insertions(+), 21 deletions(-) -- 2.25.1

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