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

[PATCH] ASoC: soc-core: add the driver component name to the component struc
by Kory Maincent 07 Dec '21

07 Dec '21

The function fmt_single_name is using to sanitize namings, and it return the name set to component->name. This function is returning either the "device_name" or the "device_name.driver_name" for i2c devices. There is no use of the component driver name. If a non i2c driver register two components the function will return the same "device_name" for both components. This could cause unexpected issue, in my case it is a debugfs error which tries to create two directory with the same component name. I have fixed it by prefixing the component name with the driver component name. Signed-off-by: Kory Maincent <kory.maincent(a)bootlin.com> --- sound/soc/soc-core.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcf6be4c4aaa..21ff77b231b8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2342,10 +2342,10 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); * Simplify DAI link configuration by removing ".-1" from device names * and sanitizing names. */ -static char *fmt_single_name(struct device *dev, int *id) +static char *fmt_single_name(struct device *dev, const char *snd_drv_name, int *id) { const char *devname = dev_name(dev); - char *found, *name; + char *found, *name, *tmp; unsigned int id1, id2; if (devname == NULL) @@ -2380,6 +2380,14 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } + if (snd_drv_name != NULL) { + /* Add driver component name if present */ + tmp = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", snd_drv_name, name); + devm_kfree(dev, name); + name = devm_kstrdup(dev, tmp, GFP_KERNEL); + devm_kfree(dev, tmp); + } + return name; } @@ -2444,7 +2452,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, */ if (legacy_dai_naming && (dai_drv->id == 0 || dai_drv->name == NULL)) { - dai->name = fmt_single_name(dev, &dai->id); + dai->name = fmt_single_name(dev, dai_drv->name, &dai->id); } else { dai->name = fmt_multiple_name(dev, dai_drv); if (dai_drv->id) @@ -2578,7 +2586,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, driver->name, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.25.1

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[PATCH] sound: ppc: beep: fix clang -Wimplicit-fallthrough
by Anders Roxell 07 Dec '21

07 Dec '21

Clang warns: sound/ppc/beep.c:103:2: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough] case SND_TONE: break; ^ sound/ppc/beep.c:103:2: note: insert 'break;' to avoid fall-through case SND_TONE: break; ^ break; 1 warning generated. Clang is more pedantic than GCC, which does not warn when failing through to a case that is just break or return. Clang's version is more in line with the kernel's own stance in deprecated.rst. Add athe missing break to silence the warning. Reported-by: Naresh Kamboju <naresh.kamboju(a)linaro.org> Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/ppc/beep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index 0f4bce1c0d4f..bf289783eafd 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -99,7 +99,7 @@ static int snd_pmac_beep_event(struct input_dev *dev, unsigned int type, return -1; switch (code) { - case SND_BELL: if (hz) hz = 1000; + case SND_BELL: if (hz) hz = 1000; break; case SND_TONE: break; default: return -1; } -- 2.33.0

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Logitech Z10 USB speakers need a volume change before audio works
by Hans de Goede 07 Dec '21

07 Dec '21

Hi, I've a set of Logitech Z10 USB speakers, which act as a USB soundcard. They have this weird glitch where after turning off my PC (and their power-supply as well) and then turning things back on, they are silent until I change the PCM volume control for the speakers inside alsa-mixer. It seems like they need some "set-volume" command to be send over the USB bus to unmute them when initially powered-up / turned on. Is their some existing usb-audio quirk which I can try to work around this? Regards, Hans

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Add headset Mic supported for Lenovo platform
by Kailang 07 Dec '21

07 Dec '21

Hi Takashi, Attach patch will enable headset Mic support on Lenovo platform. Many Thanks. BR, Kailang

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[PATCH] ALSA: mixart: Reduce size of mixart_timer_notify
by Kees Cook 07 Dec '21

07 Dec '21

The mixart_timer_notify structure was larger than could be represented by the mixart_msg_data array storage. Adjust the size to as large as possible to fix the warning seen with -Warray-bounds builds: sound/pci/mixart/mixart_core.c: In function 'snd_mixart_threaded_irq': sound/pci/mixart/mixart_core.c:447:50: error: array subscript 'struct mixart_timer_notify[0]' is partly outside array bounds of 'u32[128]' {aka 'unsigned int[128]'} [-Werror=array-bounds] 447 | for(i=0; i<notify->stream_count; i++) { | ^~ sound/pci/mixart/mixart_core.c:328:12: note: while referencing 'mixart_msg_data' 328 | static u32 mixart_msg_data[MSG_DEFAULT_SIZE / 4]; | ^~~~~~~~~~~~~~~ Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/pci/mixart/mixart_core.c | 3 +-- sound/pci/mixart/mixart_core.h | 10 +++++++++- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index fb8895af0363..853083dd4bad 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -23,8 +23,6 @@ #define MSG_DESCRIPTOR_SIZE 0x24 #define MSG_HEADER_SIZE (MSG_DESCRIPTOR_SIZE + 4) -#define MSG_DEFAULT_SIZE 512 - #define MSG_TYPE_MASK 0x00000003 /* mask for following types */ #define MSG_TYPE_NOTIFY 0 /* embedded -> driver (only notification, do not get_msg() !) */ #define MSG_TYPE_COMMAND 1 /* driver <-> embedded (a command has no answer) */ @@ -444,6 +442,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) struct mixart_timer_notify *notify; notify = (struct mixart_timer_notify *)mixart_msg_data; + BUILD_BUG_ON(sizeof(notify) > sizeof(mixart_msg_data)); for(i=0; i<notify->stream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; diff --git a/sound/pci/mixart/mixart_core.h b/sound/pci/mixart/mixart_core.h index fbf4731a276d..2f0e29ed5d63 100644 --- a/sound/pci/mixart/mixart_core.h +++ b/sound/pci/mixart/mixart_core.h @@ -49,6 +49,7 @@ enum mixart_message_id { MSG_CLOCK_SET_PROPERTIES = 0x200002, }; +#define MSG_DEFAULT_SIZE 512 struct mixart_msg { @@ -251,10 +252,17 @@ struct mixart_sample_pos u32 sample_pos_low_part; } __attribute__((packed)); +/* + * This structure is limited by the size of MSG_DEFAULT_SIZE. Instead of + * having MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS many streams, + * this is capped to have a total size below MSG_DEFAULT_SIZE. + */ +#define MIXART_MAX_TIMER_NOTIFY_STREAMS \ + ((MSG_DEFAULT_SIZE - sizeof(u32)) / sizeof(struct mixart_sample_pos)) struct mixart_timer_notify { u32 stream_count; - struct mixart_sample_pos streams[MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS]; + struct mixart_sample_pos streams[MIXART_MAX_TIMER_NOTIFY_STREAMS]; } __attribute__((packed)); -- 2.30.2

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[GIT PULL] ASoC fixes for v5.16-rc4
by Mark Brown 07 Dec '21

07 Dec '21

The following changes since commit 86f74ba3fef56dd1cee19b7a15ae27fc0da5bb61: ASoC: SOF: hda: reset DAI widget before reconfiguring it (2021-11-24 12:57:11 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.16-rc4 for you to fetch changes up to 3fc27e9a1f619b50700f020e6cd270c1b74755f0: ASoC: codecs: wsa881x: fix return values from kcontrol put (2021-12-01 14:13:53 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.16 A relatively large collection of updates, the size is increased quite a bit by there being some repetitive changes for similar issues that occur multiple times with both notifying control value changes and runtime PM. The Rockchip update looks at first glance like a cleanup but fixes instantiation of the hardware on some systems. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: amd: fix uninitialized variable in snd_acp6x_probe() David Heidelberg (1): ASoC: dt-bindings: wlf,wm8962: add missing interrupt property Hui Wang (1): ASoC: SOF: Intel: Retry codec probing if it fails Mark Brown (1): Suspend related fixes on Tegra Nicolas Frattaroli (1): ASoC: rockchip: i2s_tdm: Dup static DAI template Rob Clark (2): ASoC: rt5682: Fix crash due to out of scope stack vars ASoC: rt5682s: Fix crash due to out of scope stack vars Sameer Pujar (6): ASoC: tegra: Balance runtime PM count ASoC: tegra: Use normal system sleep for SFC ASoC: tegra: Use normal system sleep for MVC ASoC: tegra: Use normal system sleep for Mixer ASoC: tegra: Use normal system sleep for AMX ASoC: tegra: Use normal system sleep for ADX Srinivas Kandagatla (4): ASoC: qdsp6: q6routing: Fix return value from msm_routing_put_audio_mixer ASoC: codecs: wcd934x: handle channel mappping list correctly ASoC: codecs: wcd934x: return correct value from mixer put ASoC: codecs: wsa881x: fix return values from kcontrol put .../devicetree/bindings/sound/wlf,wm8962.yaml | 3 + sound/soc/amd/yc/pci-acp6x.c | 3 +- sound/soc/codecs/rt5682.c | 10 +- sound/soc/codecs/rt5682s.c | 10 +- sound/soc/codecs/wcd934x.c | 126 +++++++++++++++------ sound/soc/codecs/wsa881x.c | 16 ++- sound/soc/qcom/qdsp6/q6routing.c | 8 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 52 +++++---- sound/soc/sof/intel/hda-codec.c | 14 ++- sound/soc/tegra/tegra210_adx.c | 4 +- sound/soc/tegra/tegra210_amx.c | 4 +- sound/soc/tegra/tegra210_mixer.c | 4 +- sound/soc/tegra/tegra210_mvc.c | 8 +- sound/soc/tegra/tegra210_sfc.c | 4 +- 14 files changed, 179 insertions(+), 87 deletions(-)

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Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - edited 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was edited from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DYNAMIC_PIPELINE + 2]", I get the error: ALSA lib confeval.c:263:(snd_config_evaluate_string) wrong expression '$[$DYNAMIC_PIPELINE + 2]' ALSA lib conf.c:5632:(snd_config_expand_custom) Expand error (walk): Invalid argument Failed to expand pre-processor definitions in input config Is it possible to add support for some non-alphanumeric characters in the variable names , esp '_'? I tried this change it seems to suffice: ``` diff --git a/src/confeval.c b/src/confeval.c index a971bf38..7330d67e 100644 --- a/src/confeval.c +++ b/src/confeval.c @@ -190,7 +190,7 @@ int _snd_eval_string(snd_config_t **dst, const char *s, } else { e = s + 1; while (*e) { - if (!isalnum(*e)) + if (!isalnum(*e) && (*e != '_')) break; e++; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

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Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - edited 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was edited from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DYNAMIC_PIPELINE + 2]", I get the error: ALSA lib confeval.c:263:(snd_config_evaluate_string) wrong expression '$[$DYNAMIC_PIPELINE + 77]' ALSA lib conf.c:5632:(snd_config_expand_custom) Expand error (walk): Invalid argument Failed to expand pre-processor definitions in input config Is it possible to add support for some non-alphanumeric characters in the variable names , esp '_'? I tried this change it seems to suffice: ``` diff --git a/src/confeval.c b/src/confeval.c index a971bf38..7330d67e 100644 --- a/src/confeval.c +++ b/src/confeval.c @@ -190,7 +190,7 @@ int _snd_eval_string(snd_config_t **dst, const char *s, } else { e = s + 1; while (*e) { - if (!isalnum(*e)) + if (!isalnum(*e) && (*e != '_')) break; e++; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

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Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - opened 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was opened from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DMIC_CH_COUNT + 2]", I get the error: Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

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Issues with plughw working, however raw hw not on MAX98360A
by Dylanger Daly 07 Dec '21

07 Dec '21

Hi all, I'm currently having issues with my speakers on a Chromebook running the MAX98360A DAC. director@DRIFTWOOD:~$ aplay -D hw:sc7180rt5682max,1 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav XDG_RUNTIME_DIR (/run/user/0) is not owned by us (uid 1000), but by uid 0! (This could e.g. happen if you try to connect to a non-root PulseAudio as a root u) Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "hw:sc7180rt5682max,1": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 64 CHANNELS: 4 RATE: 48000 PERIOD_TIME: 64000 PERIOD_SIZE: 3072 PERIOD_BYTES: 24576 PERIODS: 2 BUFFER_TIME: 128000 BUFFER_SIZE: 6144 BUFFER_BYTES: 49152 TICK_TIME: ALL -------------------- aplay: set_params:1358: Channels count non available director@DRIFTWOOD:~$ aplay -D plughw:sc7180rt5682max,1 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav XDG_RUNTIME_DIR (/run/user/0) is not owned by us (uid 1000), but by uid 0! (This could e.g. happen if you try to connect to a non-root PulseAudio as a root u) Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "plughw:sc7180rt5682max,1": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED MMAP_COMPLEX RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE FLOAT64_LE FLOAT64_BE MU_LAW A_LAW IMA_AE SUBFORMAT: STD SAMPLE_BITS: [4 64] FRAME_BITS: [4 640000] CHANNELS: [1 10000] RATE: [4000 4294967295) PERIOD_TIME: 64000 PERIOD_SIZE: [256 274877907) PERIOD_BYTES: [128 4294967295) PERIODS: (0 2147484) BUFFER_TIME: [1 4294967295] BUFFER_SIZE: [512 549755813] BUFFER_BYTES: [256 4294967295] TICK_TIME: ALL -------------------- When I specify plughw, meaning "add software processing", I can hear perfect output, however if I attempt to use the raw hw handle, I get significantly less format support and the channel count isn't even known, and there is no sound output. I believe this is happening because the driver isn't being picked up, and plughw simply adds what it can to the device making it work. The end result is neither PipeWire or PluseAudio detect the speaker device, I assume this is because they're expecting to use the raw hw device handle Here's my ucm2 config: https://github.com/hexdump0815/imagebuilder/blob/d02a71ace7b93b8efbff842662… Any help would be appreciated, thank you

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