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

[PATCH v2 6/7] ASoC: cs43130: Allow driver to work without IRQ thread
by Maciej Strozek 20 Nov '23

20 Nov '23

This is to support systems without physical IRQ connection. The device only requires the IRQ for a couple of internal delays, this polling mechanism is a fallback when no IRQ is specified. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- V1 -> V2: Add changelog message sound/soc/codecs/cs43130.c | 56 +++++++++++++++++++++++++++++++------- sound/soc/codecs/cs43130.h | 1 + 2 files changed, 47 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index fd39328579fb..1e7c32eedc7b 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -326,6 +326,43 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int return ret; } +static int cs43130_wait_for_completion(struct cs43130_private *cs43130, struct completion *to_poll, + int time) +{ + int stickies, offset, flag; + int ret = 0; + + if (cs43130->has_irq_line) { + ret = wait_for_completion_timeout(to_poll, msecs_to_jiffies(time)); + } else { + if (to_poll == &cs43130->xtal_rdy) { + offset = 0; + flag = CS43130_XTAL_RDY_INT; + } else if (to_poll == &cs43130->pll_rdy) { + offset = 0; + flag = CS43130_PLL_RDY_INT; + } else if (to_poll == &cs43130->hpload_evt) { + offset = 3; + flag = CS43130_HPLOAD_NO_DC_INT | CS43130_HPLOAD_UNPLUG_INT | + CS43130_HPLOAD_OOR_INT | CS43130_HPLOAD_AC_INT | + CS43130_HPLOAD_DC_INT | CS43130_HPLOAD_ON_INT | + CS43130_HPLOAD_OFF_INT; + } else { + return 0; + } + + ret = regmap_read_poll_timeout(cs43130->regmap, CS43130_INT_STATUS_1 + offset, + stickies, (stickies & flag), + 1000, time * 1000); + + /* + * Return 0 for an timeout/error to be consistent with wait_for_completion_timeout + */ + ret = !ret; + } + return ret; +} + static int cs43130_change_clksrc(struct snd_soc_component *component, enum cs43130_mclk_src_sel src) { @@ -364,8 +401,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); @@ -400,8 +436,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); @@ -416,8 +451,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_PLL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_PLL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->pll_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->pll_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_PLL_RDY_INT_MASK, 1 << CS43130_PLL_RDY_INT_SHIFT); @@ -2040,8 +2074,8 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, regmap_multi_reg_write(cs43130->regmap, seq, seq_size); - ret = wait_for_completion_timeout(&cs43130->hpload_evt, - msecs_to_jiffies(1000)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->hpload_evt, 1000); + regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); if (!ret) { dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt\n"); @@ -2545,8 +2579,10 @@ static int cs43130_i2c_probe(struct i2c_client *client) IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs43130", cs43130); if (ret != 0) { - dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); - goto err; + dev_dbg(cs43130->dev, "Failed to request IRQ: %d, will poll instead\n", ret); + cs43130->has_irq_line = 0; + } else { + cs43130->has_irq_line = 1; } cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index d3f595bbd3ba..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -508,6 +508,7 @@ struct cs43130_private { struct gpio_desc *reset_gpio; unsigned int dev_id; /* codec device ID */ int xtal_ibias; + bool has_irq_line; /* shared by both DAIs */ struct mutex clk_mutex; -- 2.34.1

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[PATCH v2 0/2] ALSA: cs35l41: prevent old firmwares using unsupported commands
by Stefan Binding 20 Nov '23

20 Nov '23

Some systems use older firmware which does not support newer commands which are used to enable external boost. For those systems, we can workaround this by writing the registers directly. We can use the firmware version, stored inside cs_dsp, to determine whether or not the command is supported. To achieve this, it requires a cleanup in the api, to pass the cs_dsp struct into the function. We can also remove the redundant boolean firmware_running from the HDA driver, and use the equivalent state inside cs_dsp. Changes since V1: - Added fixes tag to all patches - patches rebased after v6.7 release Stefan Binding (2): ALSA: hda: cs35l41: Remove unnecessary boolean state variable firmware_running ALSA: cs35l41: Fix for old systems which do not support command include/sound/cs35l41.h | 2 +- sound/pci/hda/cs35l41_hda.c | 28 ++++++++++++---------------- sound/soc/codecs/cs35l41-lib.c | 6 ++++-- sound/soc/codecs/cs35l41.c | 4 ++-- 4 files changed, 19 insertions(+), 21 deletions(-) -- 2.34.1

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[PATCH] ALSA: hda - Fix speaker and headset mic pin config for CHUWI CoreBook XPro
by kovalev＠altlinux.org 20 Nov '23

20 Nov '23

From: Vasiliy Kovalev <kovalev(a)altlinux.org> This patch corrected the speaker and headset mic pin config to the more appropriate values. Signed-off-by: Vasiliy Kovalev <kovalev(a)altlinux.org> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0163d4c7fdda83..19828cf4b60b57 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7096,6 +7096,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -7442,6 +7443,14 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x03a19020 }, /* headset mic */ + { 0x1b, 0x90170150 }, /* speaker */ + { } + }, + }, [ALC269_FIXUP_AMIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10007,6 +10016,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), + SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), -- 2.33.8

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Re: [PATCH 6/7] ASoC: cs43130: Allow driver to work without IRQ thread
by ckeepax＠opensource.cirrus.com 20 Nov '23

20 Nov '23

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

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[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

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[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

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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

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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

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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

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