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

[PATCH 0/7] ASoC: cs43130: Fixes and improvements
by Maciej Strozek 20 Nov '23

20 Nov '23

This patchset aims to add minor fixes (first two patches) and introduce general improvements to the driver (rest of the patches) Maciej Strozek (7): ASoC: cs43130: Fix the position of const qualifier ASoC: cs43130: Fix incorrect frame delay configuration ASoC: cs43130: Allow configuration of bit clock and frame inversion ASoC: cs43130: Store device in private struct and use it more consistently ASoC: cs43130: Add handling of ACPI ASoC: cs43130: Allow driver to work without IRQ thread ASoC:cs43130: Add switch to control normal and alt hp inputs sound/soc/codecs/cs43130.c | 311 ++++++++++++++++++++++++------------- sound/soc/codecs/cs43130.h | 3 + 2 files changed, 204 insertions(+), 110 deletions(-) -- 2.34.1

3 12

[PATCH v9 00/15] Add audio support in v4l2 framework
by Shengjiu Wang 20 Nov '23

20 Nov '23

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add V4L2_CTRL_TYPE_FIXED_POINT media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 9 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 41 + .../userspace-api/media/v4l/pixfmt-audio.rst | 87 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 17 +- .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 9 +- .../media/videodev2.h.rst.exceptions | 4 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 14 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1235 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 11 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 799 +++++++++++ drivers/media/v4l2-core/v4l2-ctrls-api.c | 5 +- drivers/media/v4l2-core/v4l2-ctrls-core.c | 2 + drivers/media/v4l2-core/v4l2-ctrls-defs.c | 16 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 42 + sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 ++++ sound/soc/fsl/fsl_easrc.h | 6 +- 38 files changed, 2967 insertions(+), 15 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

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Fwd: acp3x: Missing product ID for Thinkpad T14s Gen2 AMD Ryzen 7 PRO 5850U
by Bagas Sanjaya 20 Nov '23

20 Nov '23

Hi, I notice a bug report on Bugzilla [1]. Quoting from it: > Hey,sound/soc/amd/renoir/rn-pci-acp3x.c has a hard-coded list of quirks. It seems like some product IDs may be missing. In my case, I am using a Lenovo Thinkpad t14s gen2 with an AMD Ryzen 7 PRO 5850U. > My audio output is fine but my internal microphone cannot be found. > > Running `dmidecode` yields: > > Handle 0x000F, DMI type 1, 27 bytes > System Information > Manufacturer: LENOVO > Product Name: 20XGS1KT02 > Version: ThinkPad T14s Gen 2a > > Could it be that one may have to add the product ID "20XGS1KT02" to the hard-coded list of quirks in rn-pci-acp3x.c? > > > It seems as there have been similar problems in the past, e.g. https://bugzilla.kernel.org/show_bug.cgi?id=216270. See Bugzilla for the full thread. ollisieber, can you send the quirk patch? Thanks. [1]: https://bugzilla.kernel.org/show_bug.cgi?id=218148 -- An old man doll... just what I always wanted! - Clara

3 7

snd_pcm_readn hangs at record & playback on same device
by GitHub issues - edited 20 Nov '23

20 Nov '23

alsa-project/alsa-lib issue #368 was edited from nodemand: I'm on a RPi5 4GB running Raspberry Pi OS 64bit Bookworm and my application hangs on snd_pcm_readn when I try to read and write to the same device. This occurs on an IQAudio Codec Zero sound card and on a HifiBerry DAC+ADC Pro as well. When I select a USB sound card as the output and one of the others described for input everything works like it should. Here is my code: ``` #include <alsa/asoundlib.h> #include <alsa/control.h> #include <stdio.h> #include <stdlib.h> #include <stddef.h> #include <algorithm> #include <iterator> typedef signed short drwav_int16; typedef signed int drwav_int32; #define PCM_DEVICE_IN "plughw:Zero,0" #define PCM_DEVICE_OUT "plughw:Zero,0" //"plughw:Device,0" int err, result, loops, n, t; unsigned int rate, pcm, tmp; snd_pcm_t *play; snd_pcm_t *record; snd_pcm_hw_params_t *playparams; snd_pcm_hw_params_t *recordparams; snd_pcm_uframes_t frames; snd_pcm_sframes_t recresult; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; int block = 0;//SND_PCM_NONBLOCK; int buffer_frames = 1024; int bits_per_frame = 16; const int bits_per_byte = 8; int channels = 2; int bytesize_per_channel = buffer_frames * bits_per_frame / bits_per_byte; int main(int argc, char **argv) { if (argc < 3) { printf("Usage: %s <sample_rate> <frames>\n", argv[0]); return -1; } rate = atoi(argv[1]); buffer_frames = atoi(argv[2]); bytesize_per_channel = buffer_frames * snd_pcm_format_width(format) / bits_per_byte; drwav_int16 *buffer_in[channels]; /* Open the PCM device in record mode */ if ((pcm = snd_pcm_open(&record, PCM_DEVICE_IN, SND_PCM_STREAM_CAPTURE, block)) < 0) printf("ERROR: Can't open \"%s\" PCM device. %s\n", PCM_DEVICE_IN, snd_strerror(pcm)); /* Allocate parameters object and fill it with default values*/ if ((pcm = snd_pcm_hw_params_malloc(&recordparams)) < 0) printf("ERROR: Can't allocate hardware parameter structure. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_any(record, recordparams)) < 0) printf("ERROR: Can't initialize hardware parameter structure. %s\n", snd_strerror(pcm)); /* Set parameters */ if ((pcm = snd_pcm_hw_params_set_access(record, recordparams, SND_PCM_ACCESS_RW_NONINTERLEAVED )) < 0) printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_format(record, recordparams, format)) < 0) printf("ERROR: Can't set format. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_channels(record, recordparams, channels)) < 0) printf("ERROR: Can't set channels number. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_rate_near(record, recordparams, &rate, 0)) < 0) printf("ERROR: Can't set sample rate. %s\n", snd_strerror(pcm)); /* Write parameters */ if ((pcm = snd_pcm_hw_params(record, recordparams)) < 0) printf("ERROR: Can't set hardware parameters. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_prepare(record)) < 0) printf("ERROR: Can't prepare audio interface for use. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_start(record)) < 0) printf("ERROR: Can't start hardware soundcard. %s\n", snd_strerror(pcm)); /* Resume information */ printf("PCM name: '%s'\n", snd_pcm_name(record)); printf("PCM state: %s\n", snd_pcm_state_name(snd_pcm_state(record))); snd_pcm_hw_params_get_channels(recordparams, &tmp); printf("channels: %i ", tmp); if (tmp == 1) printf("(mono)\n"); else if (tmp == 2) printf("(stereo)\n"); snd_pcm_hw_params_get_rate(recordparams, &tmp, 0); printf("sample rate: %d bps\n", tmp); /* Open the PCM device in playback mode */ if ((pcm = snd_pcm_open(&play, PCM_DEVICE_OUT, SND_PCM_STREAM_PLAYBACK, block)) < 0) printf("ERROR: Can't open \"%s\" PCM device. %s\n", PCM_DEVICE_OUT, snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_malloc(&playparams)) < 0) printf("ERROR: Can't allocate hardware parameter structure. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_any(play, playparams)) < 0) printf("ERROR: Can't initialize hardware parameter structure. %s\n", snd_strerror(pcm)); /* Set parameters */ if ((pcm = snd_pcm_hw_params_set_access(play, playparams, SND_PCM_ACCESS_RW_NONINTERLEAVED )) < 0) printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_format(play, playparams, format)) < 0) printf("ERROR: Can't set format. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_channels(play, playparams, channels)) < 0) printf("ERROR: Can't set channels number. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_rate_near(play, playparams, &rate, 0)) < 0) printf("ERROR: Can't set sample rate. %s\n", snd_strerror(pcm)); /* Write parameters */ if ((pcm = snd_pcm_hw_params(play, playparams)) < 0) printf("ERROR: Can't set hardware parameters. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_prepare(play)) < 0) printf("ERROR: Can't prepare audio interface for use. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_start(play)) < 0) printf("ERROR: Can't start hardware soundcard. %s\n", snd_strerror(pcm)); /* Resume information */ printf("PCM name: '%s'\n", snd_pcm_name(play)); printf("PCM state: %s\n", snd_pcm_state_name(snd_pcm_state(play))); snd_pcm_hw_params_get_channels(playparams, &tmp); printf("channels: %i ", tmp); if (tmp == 1) printf("(mono)\n"); else if (tmp == 2) printf("(stereo)\n"); snd_pcm_hw_params_get_rate(playparams, &tmp, 0); printf("sample rate: %d bps\n", tmp); buffer_in[0] = (drwav_int16*) malloc(bytesize_per_channel); buffer_in[1] = (drwav_int16*) malloc(bytesize_per_channel); printf("buffers allocated: %d bytes per channel\n", bytesize_per_channel); for (loops = 0; loops < 25000; loops++) { if ((recresult = snd_pcm_readn(record, (void**) buffer_in, buffer_frames)) < 0) { printf ("ERROR. Read from audio interface failed. (%s)\n", snd_strerror(recresult)); exit(1); } n = (int) recresult; printf("read %d done, %d frames\n", loops, n); if ((result = snd_pcm_writen(play, (void**) buffer_in, recresult)) == -EPIPE) { printf("XRUN.\n"); snd_pcm_prepare(play); } else if (result < 0) { printf("ERROR. Can't write to playback device. %s\n", snd_strerror(result)); exit(1); } printf("write %d done\n", loops); } snd_pcm_hw_params_free(playparams); snd_pcm_hw_params_free(recordparams); printf("params freed\n"); snd_pcm_drain(play); snd_pcm_close(play); snd_pcm_drain(record); snd_pcm_close(record); printf("audio interface closed\n"); free(buffer_in[0]); free(buffer_in[1]); printf("buffers freed\n"); return 0; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/368 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_readn hangs at record & playback on same device
by GitHub issues - opened 20 Nov '23

20 Nov '23

alsa-project/alsa-lib issue #368 was opened from nodemand: I'm on a RPi5 4GB running Raspberry Pi OS 64bit Bookworm and my application hangs on snd_pcm_readn when I try to read and write to the same device. This occurs on an IQAudio Codec Zero sound card and on a HifiBerry DAC+ADC Pro as well. When I select a USB sound card as the output and one of the others described for input everything works like it should. Here is my code: ` #include <alsa/asoundlib.h> #include <alsa/control.h> #include <stdio.h> #include <stdlib.h> #include <stddef.h> #include <algorithm> #include <iterator> typedef signed short drwav_int16; typedef signed int drwav_int32; #define PCM_DEVICE_IN "plughw:Zero,0" #define PCM_DEVICE_OUT "plughw:Zero,0" //"plughw:Device,0" int err, result, loops, n, t; unsigned int rate, pcm, tmp; snd_pcm_t *play; snd_pcm_t *record; snd_pcm_hw_params_t *playparams; snd_pcm_hw_params_t *recordparams; snd_pcm_uframes_t frames; snd_pcm_sframes_t recresult; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; int block = 0;//SND_PCM_NONBLOCK; int buffer_frames = 1024; int bits_per_frame = 16; const int bits_per_byte = 8; int channels = 2; int bytesize_per_channel = buffer_frames * bits_per_frame / bits_per_byte; int main(int argc, char **argv) { if (argc < 3) { printf("Usage: %s <sample_rate> <frames>\n", argv[0]); return -1; } rate = atoi(argv[1]); buffer_frames = atoi(argv[2]); bytesize_per_channel = buffer_frames * snd_pcm_format_width(format) / bits_per_byte; drwav_int16 *buffer_in[channels]; /* Open the PCM device in record mode */ if ((pcm = snd_pcm_open(&record, PCM_DEVICE_IN, SND_PCM_STREAM_CAPTURE, block)) < 0) printf("ERROR: Can't open \"%s\" PCM device. %s\n", PCM_DEVICE_IN, snd_strerror(pcm)); /* Allocate parameters object and fill it with default values*/ if ((pcm = snd_pcm_hw_params_malloc(&recordparams)) < 0) printf("ERROR: Can't allocate hardware parameter structure. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_any(record, recordparams)) < 0) printf("ERROR: Can't initialize hardware parameter structure. %s\n", snd_strerror(pcm)); /* Set parameters */ if ((pcm = snd_pcm_hw_params_set_access(record, recordparams, SND_PCM_ACCESS_RW_NONINTERLEAVED )) < 0) printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_format(record, recordparams, format)) < 0) printf("ERROR: Can't set format. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_channels(record, recordparams, channels)) < 0) printf("ERROR: Can't set channels number. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_rate_near(record, recordparams, &rate, 0)) < 0) printf("ERROR: Can't set sample rate. %s\n", snd_strerror(pcm)); /* Write parameters */ if ((pcm = snd_pcm_hw_params(record, recordparams)) < 0) printf("ERROR: Can't set hardware parameters. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_prepare(record)) < 0) printf("ERROR: Can't prepare audio interface for use. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_start(record)) < 0) printf("ERROR: Can't start hardware soundcard. %s\n", snd_strerror(pcm)); /* Resume information */ printf("PCM name: '%s'\n", snd_pcm_name(record)); printf("PCM state: %s\n", snd_pcm_state_name(snd_pcm_state(record))); snd_pcm_hw_params_get_channels(recordparams, &tmp); printf("channels: %i ", tmp); if (tmp == 1) printf("(mono)\n"); else if (tmp == 2) printf("(stereo)\n"); snd_pcm_hw_params_get_rate(recordparams, &tmp, 0); printf("sample rate: %d bps\n", tmp); /* Open the PCM device in playback mode */ if ((pcm = snd_pcm_open(&play, PCM_DEVICE_OUT, SND_PCM_STREAM_PLAYBACK, block)) < 0) printf("ERROR: Can't open \"%s\" PCM device. %s\n", PCM_DEVICE_OUT, snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_malloc(&playparams)) < 0) printf("ERROR: Can't allocate hardware parameter structure. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_any(play, playparams)) < 0) printf("ERROR: Can't initialize hardware parameter structure. %s\n", snd_strerror(pcm)); /* Set parameters */ if ((pcm = snd_pcm_hw_params_set_access(play, playparams, SND_PCM_ACCESS_RW_NONINTERLEAVED )) < 0) printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_format(play, playparams, format)) < 0) printf("ERROR: Can't set format. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_channels(play, playparams, channels)) < 0) printf("ERROR: Can't set channels number. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_hw_params_set_rate_near(play, playparams, &rate, 0)) < 0) printf("ERROR: Can't set sample rate. %s\n", snd_strerror(pcm)); /* Write parameters */ if ((pcm = snd_pcm_hw_params(play, playparams)) < 0) printf("ERROR: Can't set hardware parameters. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_prepare(play)) < 0) printf("ERROR: Can't prepare audio interface for use. %s\n", snd_strerror(pcm)); if ((pcm = snd_pcm_start(play)) < 0) printf("ERROR: Can't start hardware soundcard. %s\n", snd_strerror(pcm)); /* Resume information */ printf("PCM name: '%s'\n", snd_pcm_name(play)); printf("PCM state: %s\n", snd_pcm_state_name(snd_pcm_state(play))); snd_pcm_hw_params_get_channels(playparams, &tmp); printf("channels: %i ", tmp); if (tmp == 1) printf("(mono)\n"); else if (tmp == 2) printf("(stereo)\n"); snd_pcm_hw_params_get_rate(playparams, &tmp, 0); printf("sample rate: %d bps\n", tmp); buffer_in[0] = (drwav_int16*) malloc(bytesize_per_channel); buffer_in[1] = (drwav_int16*) malloc(bytesize_per_channel); printf("buffers allocated: %d bytes per channel\n", bytesize_per_channel); for (loops = 0; loops < 25000; loops++) { if ((recresult = snd_pcm_readn(record, (void**) buffer_in, buffer_frames)) < 0) { printf ("ERROR. Read from audio interface failed. (%s)\n", snd_strerror(recresult)); exit(1); } n = (int) recresult; printf("read %d done, %d frames\n", loops, n); if ((result = snd_pcm_writen(play, (void**) buffer_in, recresult)) == -EPIPE) { printf("XRUN.\n"); snd_pcm_prepare(play); } else if (result < 0) { printf("ERROR. Can't write to playback device. %s\n", snd_strerror(result)); exit(1); } printf("write %d done\n", loops); } snd_pcm_hw_params_free(playparams); snd_pcm_hw_params_free(recordparams); printf("params freed\n"); snd_pcm_drain(play); snd_pcm_close(play); snd_pcm_drain(record); snd_pcm_close(record); printf("audio interface closed\n"); free(buffer_in[0]); free(buffer_in[1]); printf("buffers freed\n"); return 0; } ` Issue URL : https://github.com/alsa-project/alsa-lib/issues/368 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Problem with headphone mic input
by Bhavya Kapoor 19 Nov '23

19 Nov '23

After an update , my kernel stopped taking mic input from headphones but it works in windows on same laptop. I tried everything but nothing seems to work. Below is alsa-info output https://alsa-project.org/db/?f=e504a5004074ef1fa82be42666f5aa88ed913a5f Regards

1 0

[PATCH 00/34] biops: add atomig find_bit() operations
by Yury Norov 18 Nov '23

18 Nov '23

Add helpers around test_and_{set,clear}_bit() that allow to search for clear or set bits and flip them atomically. The target patterns may look like this: for (idx = 0; idx < nbits; idx++) if (test_and_clear_bit(idx, bitmap)) do_something(idx); Or like this: do { bit = find_first_bit(bitmap, nbits); if (bit >= nbits) return nbits; } while (!test_and_clear_bit(bit, bitmap)); return bit; In both cases, the opencoded loop may be converted to a single function or iterator call. Correspondingly: for_each_test_and_clear_bit(idx, bitmap, nbits) do_something(idx); Or: return find_and_clear_bit(bitmap, nbits); Obviously, the less routine code people have write themself, the less probability to make a mistake. Patch #31 of this series fixes one such error in perf/m1 codebase. Those are not only handy helpers but also resolve a non-trivial issue of using non-atomic find_bit() together with atomic test_and_{set,clear)_bit(). The trick is that find_bit() implies that the bitmap is a regular non-volatile piece of memory, and compiler is allowed to use such optimization techniques like re-fetching memory instead of caching it. For example, find_first_bit() is implemented like this: for (idx = 0; idx * BITS_PER_LONG < sz; idx++) { val = addr[idx]; if (val) { sz = min(idx * BITS_PER_LONG + __ffs(val), sz); break; } } On register-memory architectures, like x86, compiler may decide to access memory twice - first time to compare against 0, and second time to fetch its value to pass it to __ffs(). When running find_first_bit() on volatile memory, the memory may get changed in-between, and for instance, it may lead to passing 0 to __ffs(), which is undefined. This is a potentially dangerous call. find_and_clear_bit() as a wrapper around test_and_clear_bit() naturally treats underlying bitmap as a volatile memory and prevents compiler from such optimizations. Now that KCSAN is catching exactly this type of situations and warns on undercover memory modifications. We can use it to reveal improper usage of find_bit(), and convert it to atomic find_and_*_bit() as appropriate. The 1st patch of the series adds the following atomic primitives: find_and_set_bit(addr, nbits); find_and_set_next_bit(addr, nbits, start); ... Here find_and_{set,clear} part refers to the corresponding test_and_{set,clear}_bit function, and suffixes like _wrap or _lock derive semantics from corresponding find() or test() functions. For brevity, the naming omits the fact that we search for zero bit in find_and_set, and correspondingly, search for set bit in find_and_clear functions. The patch also adds iterators with atomic semantics, like for_each_test_and_set_bit(). Here, the naming rule is to simply prefix corresponding atomic operation with 'for_each'. This series is a result of discussion [1]. All find_bit() functions imply exclusive access to the bitmaps. However, KCSAN reports quite a number of warnings related to find_bit() API. Some of them are not pointing to real bugs because in many situations people intentionally allow concurrent bitmap operations. If so, find_bit() can be annotated such that KCSAN will ignore it: bit = data_race(find_first_bit(bitmap, nbits)); This series addresses the other important case where people really need atomic find ops. As the following patches show, the resulting code looks safer and more verbose comparing to opencoded loops followed by atomic bit flips. In [1] Mirsad reported 2% slowdown in a single-thread search test when switching find_bit() function to treat bitmaps as volatile arrays. On the other hand, kernel robot in the same thread reported +3.7% to the performance of will-it-scale.per_thread_ops test. Assuming that our compilers are sane and generate better code against properly annotated data, the above discrepancy doesn't look weird. When running on non-volatile bitmaps, plain find_bit() outperforms atomic find_and_bit(), and vice-versa. So, all users of find_bit() API, where heavy concurrency is expected, are encouraged to switch to atomic find_and_bit() as appropriate. 1st patch of this series adds atomic find_and_bit() API, and all the following patches spread it over the kernel. They can be applied separately from each other on per-subsystems basis, or I can pull them in bitmap tree, as appropriate. [1] https://lore.kernel.org/lkml/634f5fdf-e236-42cf-be8d-48a581c21660@alu.unizg… Yury Norov (34): lib/find: add atomic find_bit() primitives lib/sbitmap; make __sbitmap_get_word() using find_and_set_bit() watch_queue: use atomic find_bit() in post_one_notification() sched: add cpumask_find_and_set() and use it in __mm_cid_get() mips: sgi-ip30: rework heart_alloc_int() sparc: fix opencoded find_and_set_bit() in alloc_msi() perf/arm: optimize opencoded atomic find_bit() API drivers/perf: optimize ali_drw_get_counter_idx() by using find_bit() dmaengine: idxd: optimize perfmon_assign_event() ath10k: optimize ath10k_snoc_napi_poll() by using find_bit() wifi: rtw88: optimize rtw_pci_tx_kick_off() by using find_bit() wifi: intel: use atomic find_bit() API where appropriate KVM: x86: hyper-v: optimize and cleanup kvm_hv_process_stimers() PCI: hv: switch hv_get_dom_num() to use atomic find_bit() scsi: use atomic find_bit() API where appropriate powerpc: use atomic find_bit() API where appropriate iommu: use atomic find_bit() API where appropriate media: radio-shark: use atomic find_bit() API where appropriate sfc: switch to using atomic find_bit() API where appropriate tty: nozomi: optimize interrupt_handler() usb: cdc-acm: optimize acm_softint() block: null_blk: fix opencoded find_and_set_bit() in get_tag() RDMA/rtrs: fix opencoded find_and_set_bit_lock() in __rtrs_get_permit() mISDN: optimize get_free_devid() media: em28xx: cx231xx: fix opencoded find_and_set_bit() ethernet: rocker: optimize ofdpa_port_internal_vlan_id_get() serial: sc12is7xx: optimize sc16is7xx_alloc_line() bluetooth: optimize cmtp_alloc_block_id() net: smc: fix opencoded find_and_set_bit() in smc_wr_tx_get_free_slot_index() ALSA: use atomic find_bit() functions where applicable drivers/perf: optimize m1_pmu_get_event_idx() by using find_bit() API m68k: rework get_mmu_context() microblaze: rework get_mmu_context() sh: rework ilsel_enable() arch/m68k/include/asm/mmu_context.h | 11 +- arch/microblaze/include/asm/mmu_context_mm.h | 11 +- arch/mips/sgi-ip30/ip30-irq.c | 12 +- arch/powerpc/mm/book3s32/mmu_context.c | 10 +- arch/powerpc/platforms/pasemi/dma_lib.c | 45 +-- arch/powerpc/platforms/powernv/pci-sriov.c | 12 +- arch/sh/boards/mach-x3proto/ilsel.c | 4 +- arch/sparc/kernel/pci_msi.c | 9 +- arch/x86/kvm/hyperv.c | 39 ++- drivers/block/null_blk/main.c | 41 +-- drivers/dma/idxd/perfmon.c | 8 +- drivers/infiniband/ulp/rtrs/rtrs-clt.c | 15 +- drivers/iommu/arm/arm-smmu/arm-smmu.h | 10 +- drivers/iommu/msm_iommu.c | 18 +- drivers/isdn/mISDN/core.c | 9 +- drivers/media/radio/radio-shark.c | 5 +- drivers/media/radio/radio-shark2.c | 5 +- drivers/media/usb/cx231xx/cx231xx-cards.c | 16 +- drivers/media/usb/em28xx/em28xx-cards.c | 37 +-- drivers/net/ethernet/rocker/rocker_ofdpa.c | 11 +- drivers/net/ethernet/sfc/rx_common.c | 4 +- drivers/net/ethernet/sfc/siena/rx_common.c | 4 +- drivers/net/ethernet/sfc/siena/siena_sriov.c | 14 +- drivers/net/wireless/ath/ath10k/snoc.c | 9 +- .../net/wireless/intel/iwlegacy/4965-mac.c | 7 +- drivers/net/wireless/intel/iwlegacy/common.c | 8 +- drivers/net/wireless/intel/iwlwifi/dvm/sta.c | 8 +- drivers/net/wireless/intel/iwlwifi/dvm/tx.c | 19 +- drivers/net/wireless/realtek/rtw88/pci.c | 5 +- drivers/net/wireless/realtek/rtw89/pci.c | 5 +- drivers/pci/controller/pci-hyperv.c | 7 +- drivers/perf/alibaba_uncore_drw_pmu.c | 10 +- drivers/perf/apple_m1_cpu_pmu.c | 8 +- drivers/perf/arm-cci.c | 23 +- drivers/perf/arm-ccn.c | 10 +- drivers/perf/arm_dmc620_pmu.c | 9 +- drivers/perf/arm_pmuv3.c | 8 +- drivers/scsi/mpi3mr/mpi3mr_os.c | 21 +- drivers/scsi/qedi/qedi_main.c | 9 +- drivers/scsi/scsi_lib.c | 5 +- drivers/tty/nozomi.c | 5 +- drivers/tty/serial/sc16is7xx.c | 8 +- drivers/usb/class/cdc-acm.c | 5 +- include/linux/cpumask.h | 12 + include/linux/find.h | 289 ++++++++++++++++++ kernel/sched/sched.h | 52 +--- kernel/watch_queue.c | 6 +- lib/find_bit.c | 85 ++++++ lib/sbitmap.c | 46 +-- net/bluetooth/cmtp/core.c | 10 +- net/smc/smc_wr.c | 10 +- sound/pci/hda/hda_codec.c | 7 +- sound/usb/caiaq/audio.c | 13 +- 53 files changed, 588 insertions(+), 481 deletions(-) -- 2.39.2

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[PATCH 0/4] ASoC: Intel: bytcht_es8316: Determine quirks/routing through ACPI DSM
by Hans de Goede 18 Nov '23

18 Nov '23

Hi All, This takes some of the work done to auto-configure quirks/routing for ESS83xx codecs by getting the info from ACPI from: https://github.com/thesofproject/linux/pull/4112 And then builds on top of this to add auto-configuration to the bytcht_es8316 board driver. Note compared to the pull-request, which deals with the ES8336, this series deals with the ES8316 (for which I have several devices to test on) and this moves all handling from the codec driver to the board driver where this IMHO belongs. After this series audio now works properly on a CHT Chuwi Hi12 tablet without needing to add an extra quirk for that model. This has also been tested on the following devices, where things are unchanged from before (the ACPI autoconfiguration gives the same results as the old defaults) : Onda V80 plus (CHT) GP-electronic T701 (BYT) I also tested this on a Nanote UMPC-01, here the _DSM result for PLATFORM_SPK_TYPE_ARG wrongly returns 1 (mono) while the device actually has 2 speakers, so this model needs to keep its DMI quirk. Regards, Hans Hans de Goede (2): ASoC: Intel: bytcht_es8316: Add is_bytcr helper variable ASoC: Intel: bytcht_es8316: Determine quirks/routing with codec-dev ACPI DSM Pierre-Louis Bossart (2): ASoC: Intel: es83xx: add ACPI DSM helper module ASoC: Intel: bytcht_es8316: Dump basic _DSM information sound/soc/intel/boards/Kconfig | 5 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_es8316.c | 71 +++- sound/soc/intel/boards/es83xx-dsm-common.c | 89 +++++ sound/soc/intel/boards/es83xx-dsm-common.h | 393 +++++++++++++++++++++ 5 files changed, 557 insertions(+), 3 deletions(-) create mode 100644 sound/soc/intel/boards/es83xx-dsm-common.c create mode 100644 sound/soc/intel/boards/es83xx-dsm-common.h -- 2.41.0

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[PATCH] ASoC: nau8822: Fix incorrect type in assignment and cast to restricted __be16
by David Lin 17 Nov '23

17 Nov '23

This issue is reproduced when W=1 build in compiler gcc-12. The following are sparse warnings: sound/soc/codecs/nau8822.c:199:25: sparse: sparse: incorrect type in assignment sound/soc/codecs/nau8822.c:199:25: sparse: expected unsigned short sound/soc/codecs/nau8822.c:199:25: sparse: got restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202311122320.T1opZVkP-lkp@intel.com/ Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8822.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index ff3024899f45..7199d734c79f 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -184,6 +184,7 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *params = (void *)kcontrol->private_value; int i, reg; u16 reg_val, *val; + __be16 tmp; val = (u16 *)ucontrol->value.bytes.data; reg = NAU8822_REG_EQ1; @@ -192,8 +193,8 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - reg_val = cpu_to_be16(reg_val); - memcpy(val + i, &reg_val, sizeof(reg_val)); + tmp = cpu_to_be16(reg_val); + memcpy(val + i, &tmp, sizeof(tmp)); } return 0; @@ -216,6 +217,7 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol, void *data; u16 *val, value; int i, reg, ret; + __be16 *tmp; data = kmemdup(ucontrol->value.bytes.data, params->max, GFP_KERNEL | GFP_DMA); @@ -228,7 +230,8 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - value = be16_to_cpu(*(val + i)); + tmp = (__be16 *)(val + i); + value = be16_to_cpup(tmp); ret = snd_soc_component_write(component, reg + i, value); if (ret) { dev_err(component->dev, -- 2.25.1

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[PATCH] ASoC: Intel: Skylake: mem leak in skl register function
by Kamil Duljas 17 Nov '23

17 Nov '23

skl_platform_register() uses krealloc. When krealloc is fail, then previous memory is not freed. The leak is also when soc component registration failed. Signed-off-by: Kamil Duljas <kamil.duljas(a)gmail.com> --- sound/soc/intel/skylake/skl-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index d0c02e8a6785..e2578ee001ff 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1462,6 +1462,7 @@ int skl_platform_register(struct device *dev) dais = krealloc(skl->dais, sizeof(skl_fe_dai) + sizeof(skl_platform_dai), GFP_KERNEL); if (!dais) { + kfree(skl->dais); ret = -ENOMEM; goto err; } @@ -1474,8 +1475,10 @@ int skl_platform_register(struct device *dev) ret = devm_snd_soc_register_component(dev, &skl_component, skl->dais, num_dais); - if (ret) + if (ret) { + kfree(skl->dais); dev_err(dev, "soc component registration failed %d\n", ret); + } err: return ret; } -- 2.42.0.windows.2

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