January 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ASoC: ti: j721e-evm: Use devm_kcalloc() instead of devm_kzalloc()
by Erick Archer 22 Jan '24

22 Jan '24

Use 2-factor multiplication argument form devm_kcalloc() instead of devm_kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Reviewed-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Erick Archer <erick.archer(a)gmx.com> --- Changes in v2: - Fix the commit title adding "j721e-evm" --- sound/soc/ti/j721e-evm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index b4b158dc736e..d9d1e021f5b2 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -649,7 +649,7 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, * Link 2: McASP10 <- pcm3168a_1 ADC */ comp_count = 6; - compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), + compnent = devm_kcalloc(priv->dev, comp_count, sizeof(*compnent), GFP_KERNEL); if (!compnent) { ret = -ENOMEM; @@ -763,7 +763,7 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, * \ pcm3168a_b ADC */ comp_count = 8; - compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), + compnent = devm_kcalloc(priv->dev, comp_count, sizeof(*compnent), GFP_KERNEL); if (!compnent) { ret = -ENOMEM; -- 2.25.1

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[PATCH] ASoC: qcom: Use devm_kcalloc() instead of devm_kzalloc()
by Erick Archer 22 Jan '24

22 Jan '24

Use 2-factor multiplication argument form devm_kcalloc() instead of devm_kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Erick Archer <erick.archer(a)gmx.com> --- sound/soc/qcom/common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 756706d5b493..747041fa7866 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -73,7 +73,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) link = card->dai_link; for_each_available_child_of_node(dev->of_node, np) { - dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + dlc = devm_kcalloc(dev, 2, sizeof(*dlc), GFP_KERNEL); if (!dlc) { ret = -ENOMEM; goto err_put_np; -- 2.25.1

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[PATCH] ASoC: Intel: mtl-match: Add cs42l43_l0 cs35l56_l23 for MTL
by Maciej Strozek 22 Jan '24

22 Jan '24

Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- .../intel/common/soc-acpi-intel-mtl-match.c | 56 +++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index feb12c6c85d1..625f991b1ad4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -377,6 +377,36 @@ static const struct snd_soc_acpi_adr_device cs35l56_2_adr[] = { } }; +static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { + { + .adr = 0x00023301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00023201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_l_adr[] = { + { + .adr = 0x00033001fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033101fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP2" + } +}; + static const struct snd_soc_acpi_link_adr rt5682_link2_max98373_link0[] = { /* Expected order: jack -> amp */ { @@ -554,6 +584,26 @@ static const struct snd_soc_acpi_link_adr mtl_cs42l43_cs35l56[] = { {} }; +static const struct snd_soc_acpi_link_adr cs42l43_link0_cs35l56_link2_link3[] = { + /* Expected order: jack -> amp */ + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_r_adr), + .adr_d = cs35l56_2_r_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_l_adr), + .adr_d = cs35l56_3_l_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -599,6 +649,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt1318-l12-rt714-l0.tplg" }, + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = cs42l43_link0_cs35l56_link2_link3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-cs42l43-l0-cs35l56-l23.tplg", + }, { .link_mask = GENMASK(2, 0), .links = mtl_cs42l43_cs35l56, -- 2.34.1

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ucm2: Qualcomm: Lenovo-X13s: reduce default headphones volume further
by GitHub pull_request - opened 22 Jan '24

22 Jan '24

alsa-project/alsa-ucm-conf pull request #387 was opened from jhovold: A recent change reduced the default headphones volume from setting 20 (0 dB) to 10 (-15 dB) but also this setting can be quite loud and cause distortion with some headphones. Reduce the default volume further to setting 2 (-27 dB), which results in more comfortable volume levels with such headphones. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/387 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/387.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] soundwire: stream: add missing const to Documentation
by Krzysztof Kozlowski 22 Jan '24

22 Jan '24

Commit 21f4c443731f ("soundwire: stream: constify sdw_port_config when adding devices") added const to sdw_port_config argument, but forgot documentation. Fixes: 21f4c443731f ("soundwire: stream: constify sdw_port_config when adding devices") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/driver-api/soundwire/stream.rst | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/driver-api/soundwire/stream.rst b/Documentation/driver-api/soundwire/stream.rst index b432a2de45d3..2a794484f62c 100644 --- a/Documentation/driver-api/soundwire/stream.rst +++ b/Documentation/driver-api/soundwire/stream.rst @@ -324,12 +324,12 @@ framework, this stream state is linked to .hw_params() operation. int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, - struct sdw_ports_config * ports_config, + const struct sdw_ports_config * ports_config, struct sdw_stream_runtime * stream); int sdw_stream_add_slave(struct sdw_slave * slave, struct sdw_stream_config * stream_config, - struct sdw_ports_config * ports_config, + const struct sdw_ports_config * ports_config, struct sdw_stream_runtime * stream); -- 2.34.1

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ucm2: conf.d: mt8370-evk: Fix the type of mt8370-evk.conf
by GitHub pull_request - opened 22 Jan '24

22 Jan '24

alsa-project/alsa-ucm-conf pull request #386 was opened from zoran055: Fix the type of mt8370-evk.conf at /ucm2/conf.d, to let the symbolic link can correctly point to the conf. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/386 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/386.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Allow for an optional output in plugin multi
by GitHub issues - opened 22 Jan '24

22 Jan '24

alsa-project/alsa-lib issue #379 was opened from wonbinbk: Let's say I have 2 sound cards, card 0 and card 1. Card 1 is unplug-able from the system so it might not exist when applications read asound.conf. I think it makes sense to add ability to specify an optional pcm device which if it doesn't exist won't break alsa. ``` ALSA lib ../../alsa-lib-1.2.6.1/src/confmisc.c:165:(snd_config_get_card) Cannot get card index for 1 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/379 Repository URL: https://github.com/alsa-project/alsa-lib

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How to configure a pcm to write to it periodically?
by Philipp Stanner 21 Jan '24

21 Jan '24

Hi all, hope this is the right list. [1] stated that it's also intended for "application developers", so I ask here. I've got a program that uses timers to periodically read samples from a microphone and write to a sound device. The timer's period is, for example, 20ms. So I have a sample rate of r = 16000, frame sizes of 20ms, a bit depth of 16 and therefore 320 samples per 20ms. I try to write those 320 samples timer-controled every 20ms to the pcm device. That works fine when reading samples. When writing, however, it seems that snd_pcm_writei() blocks longer than the 20ms. Which it shouldn't do, as far as I understand snd_pcm_set_params() delay-parameter. I'll attach the example below. So, I'd have a number of questions: 1. Am I even using ALSA as it's intended? As far as I saw in the API, there's the possiblity to get poll()able file descriptors from the API. So should I, instead of using my own timers, configure the ALSA interface to become readable every 20ms? 2. Why does my example not work as I intended? What does the "latency" parameter in snd_pcm_set_params() actually do? 3. Is the 'size' parameter in the write functions [2] the size in bytes or in samples? So if I configured SND_PCM_S16_LE and pass size = 10, will it read 10 bytes or 160 bit? It seems to be that it's the number of samples, whose type has been set with snd_pcm_set_params(). The docu isn't super clear to me, though. It seems that the size parameter has changed a bit. At least the examples here [3] are faulting when I build them with -fsanitize=address Thx a lot! Philipp [1] https://www.alsa-project.org/wiki/Mailing-lists [2] https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gabc748a5… [3] http://equalarea.com/paul/alsa-audio.html Example: /* * This is a mini-example of the code I took for my draft ofa broader application. * The main difference is that I call snd_pcm_writei() when my timer fires. */ #include <alsa/asoundlib.h> static char *device = "default"; /* playback device */ unsigned char buffer[16*1024]; /* some random data */ int main(void) { int err; unsigned int i; snd_pcm_t *handle; snd_pcm_sframes_t frames; for (i = 0; i < sizeof(buffer); i++) buffer[i] = random() & 0xff; if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if ((err = snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 1, 16000, 1, 20000)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } for (i = 0; i < 16; i++) { frames = snd_pcm_writei(handle, buffer, sizeof(buffer) / sizeof(short)); if (frames < 0) frames = snd_pcm_recover(handle, frames, 0); if (frames < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(frames)); break; } if (frames > 0 && frames < (long)(sizeof(buffer) / sizeof(short))) printf("Short write (expected %li, wrote %li)\n", (long)(sizeof(buffer) / sizeof(short)), frames); } /* pass the remaining samples, otherwise they're dropped in close */ err = snd_pcm_drain(handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); snd_pcm_close(handle); return 0; }

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RE: [PATCH 3/5] ASoC: codecs: ES8326: Adding new volume kcontrols
by Zhu Ning 20 Jan '24

20 Jan '24

> Volumes should be standard volume controls with TLV information rather > than enumerations. I see that you can't just use a standard volume > control here since there's an invalid value that needs to be skipped > over, and it doesn't help that there's no ordering to the values either. > I think the best thing would probably be to open code a custom volume > control - this feels weird enough that it's probably not worrying about > trying for reuse. Just have put and get functions that map 0..6 into > the register values for the volumes in ascending order, then you can > have a control that looks normal to userspace. I'll try your mentioned method in a new version of the same patch.

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RE: [PATCH 3/5] ASoC: codecs: ES8326: Adding new volume kcontrols
by Zhu Ning 20 Jan '24

20 Jan '24

> kernel test robot noticed the following build errors: > [auto build test ERROR on broonie-sound/for-next] > [also build test ERROR on tiwai-sound/for-next tiwai-sound/for-linus linus/master next-20240119] > [cannot apply to v6.7] > [If your patch is applied to the wrong git tree, kindly drop us a note. > And when submitting patch, we suggest to use '--base' as documented in > https://git-scm.com/docs/git-format-patch#_base_tree_information] > url: https://github.com/intel-lab-lkp/linux/commits/Zhu-Ning/ASoC-codecs-ES8326-… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > patch link: https://lore.kernel.org/r/20240119112858.2982-4-zhuning0077%40gmail.com > patch subject: [PATCH 3/5] ASoC: codecs: ES8326: Adding new volume kcontrols > config: m68k-allmodconfig (https://download.01.org/0day-ci/archive/20240120/202401201033.SJS8Mtlr-lkp@…) > compiler: m68k-linux-gcc (GCC) 13.2.0 > reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240120/202401201033.SJS8Mtlr-lkp@…) > If you fix the issue in a separate patch/commit (i.e. not just a new version of > the same patch/commit), kindly add following tags > | Reported-by: kernel test robot <lkp(a)intel.com> > | Closes: https://lore.kernel.org/oe-kbuild-all/202401201033.SJS8Mtlr-lkp@intel.com/ I'm so sorry for my mistake.I'll fix it in the new version of the same patch/commit.

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