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[PATCH v2 0/2] Add support for QCM6490 and QCS6490
by Mohammad Rafi Shaik 27 May '24

27 May '24

This patchset adds support for sound card on Qualcomm QCM6490 IDP and QCS6490 RB3Gen2 boards. Changes since v1: - Use existing sc8280xp machine driver instead of separate driver. - Modify qcs6490 compatible name as qcs6490-rb3gen2. Mohammad Rafi Shaik (2): ASoC: dt-bindings: qcom,sm8250: Add QCM6490 snd QCS6490 sound card ASoC: qcom: sc8280xp: Add support for QCM6490 and QCS6490 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 2 ++ sound/soc/qcom/sc8280xp.c | 2 ++ 2 files changed, 4 insertions(+) -- 2.25.1

4 6

[PATCH v15 00/16] Add audio support in v4l2 framework
by Shengjiu Wang 27 May '24

27 May '24

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 v15: - update MAINTAINERS for imx-asrc.c and vim2m-audio.c changes in v14: - document the reservation of 'AUXX' fourcc format. - add v4l2_audfmt_to_fourcc() definition. changes in v13 - change 'pixelformat' to 'audioformat' in dev-audio-mem2mem.rst - add more description for clock drift in ext-ctrls-audio-m2m.rst - Add "media: v4l2-ctrls: add support for fraction_bits" from Hans to avoid build issue for kernel test robot changes in v12 - minor changes according to comments - drop min_buffers_needed = 1 and V4L2_CTRL_FLAG_UPDATE flag - drop bus_info changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias 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 Hans Verkuil (1): media: v4l2-ctrls: add support for fraction_bits 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 audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../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 | 59 + .../userspace-api/media/v4l/pixfmt-audio.rst | 100 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 11 +- .../media/videodev2.h.rst.exceptions | 3 + MAINTAINERS | 17 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1256 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-api.c | 1 + drivers/media/v4l2-core/v4l2-ctrls-core.c | 93 +- drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + 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-ctrls.h | 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 | 50 +- 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 +- 43 files changed, 3145 insertions(+), 27 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

11 49

[PATCH] Correct documentation for struct segdist_code
by Abhishek Tamboli 24 May '24

24 May '24

Correct the documentation for struct segdist_code by removing the incorrect reference to segdist_codes and add a note to describe the array. Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202405190241.jIzZxFB5-lkp@intel.com/ Signed-off-by: Abhishek Tamboli <abhishektamboli9(a)gmail.com> --- The documentation for struct segdist_code contained an incorrect reference to segdist_codes as a member. This patch removes that incorrect reference and adds a note to describe the array segdist_codes[] containing all possible Segment Distribution codes. drivers/slimbus/stream.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..d9eaa8989d2e 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -18,7 +18,8 @@ * and the first slot of the next consecutive Segment. * @segdist_code: Segment Distribution Code SD[11:0] * @seg_offset_mask: Segment offset mask in SD[11:0] - * @segdist_codes: List of all possible Segmet Distribution codes. + * + * Note: segdist_codes: List of all possible Segmet Distribution codes. */ static const struct segdist_code { int ratem; base-commit: f3d61438b613b87afb63118bea6fb18c50ba7a6b prerequisite-patch-id: f30868a7fdcfaceb6bab2b6f41402a9aeae858cb -- 2.25.1

1 0

DMA Transfer Synchronization Issue in Out-of-Tree Sound Card Driver
by Louis Chauvet 24 May '24

24 May '24

Hello everyone, I am currently developing an out-of-tree driver for a sound card that utilizes XDMA for sample transfers. I am currently using a kernel 6.5 with the latest xdma driver cherry-picked, but I don't think any changes since 6.5 is addressing my issue. My initial issue pertains to the completion of DMA transfers between a start and a stop command in ALSA. If the interval is too brief, the transfer does not conclude properly, leading to distorted samples. A straightforward solution to this problem was to adequately wait for the transfer to finish upon the stop, ie. sleeping until we know the hardware is done with the transfert (the XDMA controller does not support stopping in the middle of a transfer). To address this DMA issue, I have created a patch [1] that guarantees the completion of the DMA transfer upon the return of xdma_synchronize. This means xdma_synchronize now sleeps, but looking at other drivers around it appears expected to be able to do so. Regarding the audio implementation, the following patch enforces the synchronization: int playback_trigger(struct snd_pcm_substream *substream, int command) { struct my_dev *my_dev = snd_pcm_substream_chip(substream); switch (command) { case SNDRV_PCM_TRIGGER_START: /* Synchronize on start, because the trigger stop is called from an IRQ context */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dmaengine_synchronize(my_dev->playback_dma_chan); pipe_start(&my_dev->playback, substream); break; case SNDRV_PCM_TRIGGER_STOP: pipe_stop(&my_dev->playback, substream); break; default: return -EINVAL; } return 0; } In order for a sleepable function like dmaengine_synchronize() to work in the trigger callbacks, the audio card nonatomic flag had to be set. It basically leads the sound core towards the use of a mutex instead of a spinlock. static int probe([...]) { struct snd_pcm *pcm; [...] /* This flag is needed to be able to sleep in start/stop callbacks */ pcm->nonatomic = true; [...] snd_pcm_set_managed_buffer_all(pcm, [...]); } This approach generally works well, but leads to "scheduling while atomic" errors. Indeed, the IRQ handler from the XDMA driver invokes a function within the sound subsystem, which subsequently acquires a mutex... At the moment, the only solution I've found is to replace the wait_for_completion() in the XDMA driver [2] with a busy wait loop. However, this approach seems incorrect, as all other synchronization functions in other DMA drivers are sleeping, which should not cause an issue. The problem might be related to the sound driver. Should I avoid manually using dmaengine_synchronize? How to achieve the same effect in this case? Perhaps there is a more traditional way to properly clean the stream in the sound subsystem which I overlooked? Could someone please provide guidance on how to resolve this issue? Thanks, Louis Chauvet [1]: https://lore.kernel.org/all/20240327-digigram-xdma-fixes-v1-2-45f4a52c0283@… [2]: https://elixir.bootlin.com/linux/latest/source/drivers/dma/xilinx/xdma.c#L5… -- Louis Chauvet, Bootlin Embedded Linux and Kernel engineering https://bootlin.com

3 3

[PATCH v4 0/3] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 24 May '24

24 May '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v4 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.11 or v6.12 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 12 files changed, 45 insertions(+), 86 deletions(-) -- 2.25.1

2 5

[PATCH v3 0/3] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 24 May '24

24 May '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v3 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. And because of its history, DPCM has been checking CPU side only. But it should check Codec side too same as non-DPCM. Some device/driver has been bypassed this check. It should be error. [2/3] patch indicates warning in such case. Because dpcm_xxx flag is no longer used by [1/3] patch, snd_soc_dai_link_set_capabilities() is no longer needed. [3/3] patch remove it. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.11 or v6.12 ? Please check each driver by that time. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Kuninori Morimoto (3): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: soc-pcm: Indicate warning if CPU / Codec availability mismatch ASoC: remove snd_soc_dai_link_set_capabilities() include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 - sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/meson/axg-card.c | 1 - sound/soc/meson/gx-card.c | 1 - sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 --------- sound/soc/soc-pcm.c | 110 +++++++++++++++++++------- 11 files changed, 81 insertions(+), 81 deletions(-) -- 2.25.1

3 17

[GIT PULL] ASoC fixes for v6.10-merge-window
by Mark Brown 23 May '24

23 May '24

The following changes since commit c3c5ac4bd7d7019f2e3ad1720572d53226fe656e: ASoC: Intel: updates for 6.10 - part7 (2024-05-10 12:30:12 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.10-merge-window for you to fetch changes up to 737ce4fb96206f999ddea7530145fc0e8abd5d31: ASoC: dt-bindings: stm32: Ensure compatible pattern matches whole string (2024-05-21 12:35:12 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.10 A bunch of fixes that came in during the merge window, all driver specific and none of them especially remarkable. ---------------------------------------------------------------- Jack Yu (1): ASoC: rt5645: mic-in detection threshold modification Peter Ujfalusi (1): ASoC: Intel: sof_sdw_rt_sdca_jack_common: Use name_prefix for `-sdca` detection Rob Herring (Arm) (1): ASoC: dt-bindings: stm32: Ensure compatible pattern matches whole string Shenghao Ding (2): ASoC: tas2552: Add TX path for capturing AUDIO-OUT data ASoC: tas2781: Fix wrong loading calibrated data sequence Zhang Yi (1): ASoC: codecs: ES8326: solve hp and button detect issue .../devicetree/bindings/sound/st,stm32-sai.yaml | 2 +- include/sound/tas2781-dsp.h | 7 +- sound/soc/codecs/es8326.c | 6 +- sound/soc/codecs/rt5645.c | 1 + sound/soc/codecs/tas2552.c | 15 ++- sound/soc/codecs/tas2781-fmwlib.c | 103 ++++++--------------- sound/soc/codecs/tas2781-i2c.c | 4 +- .../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 2 +- 8 files changed, 51 insertions(+), 89 deletions(-)

2 1

[PATCH v1] ASoc: tas2781: Support still work when only RCA binary loading well without dsp firmware loading.
by Shenghao Ding 23 May '24

23 May '24

In only RCA binary loading case, only default dsp program will be work inside the chip. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Split out the different logical changes into different patches. - rename tasdevice_dsp_fw_state -> tasdevice_fw_state, the fw are not only dsp fw, but also RCA(Reconfigurable data, such as acoustic data and register setting, etc). - Add TASDEVICE_RCA_FW_OK in tasdevice_fw_state to identify the state that only RCA binary file has been download successfully, but dsp fw is not loaded or loading failure. - Add the this strategy into tasdevice_tuning_switch. - If one side of the if/else has a braces both should in tasdevice_tuning_switch. - Identify whehter both RCA and DSP have been loaded or only RCA has been loaded in tasdevice_fw_ready. - Add check fw load status in tasdevice_startup. - remove ret in tasdevice_startup to make th code neater. --- include/sound/tas2781-dsp.h | 3 ++- sound/soc/codecs/tas2781-fmwlib.c | 14 ++++++++++---- sound/soc/codecs/tas2781-i2c.c | 30 ++++++++++++++++++------------ 3 files changed, 30 insertions(+), 17 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 7fba7ea26a4b..92d68ca5eafb 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -117,10 +117,11 @@ struct tasdevice_fw { struct device *dev; }; -enum tasdevice_dsp_fw_state { +enum tasdevice_fw_state { TASDEVICE_DSP_FW_NONE = 0, TASDEVICE_DSP_FW_PENDING, TASDEVICE_DSP_FW_FAIL, + TASDEVICE_RCA_FW_OK, TASDEVICE_DSP_FW_ALL_OK, }; diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 265a8ca25cbb..cfa022ef4a59 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2324,13 +2324,18 @@ void tasdevice_tuning_switch(void *context, int state) struct tasdevice_fw *tas_fmw = tas_priv->fmw; int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + /* + * Only RCA file loaded can still work with default dsp program inside + * the chip? + */ + if (!(tas_priv->fw_state == TASDEVICE_RCA_FW_OK || + tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK)) { + dev_err(tas_priv->dev, "No firmware loaded\n"); return; } if (state == 0) { - if (tas_priv->cur_prog < tas_fmw->nr_programs) { + if (tas_fmw && tas_priv->cur_prog < tas_fmw->nr_programs) { /*dsp mode or tuning mode*/ profile_cfg_id = tas_priv->rcabin.profile_cfg_id; tasdevice_select_tuningprm_cfg(tas_priv, @@ -2340,9 +2345,10 @@ void tasdevice_tuning_switch(void *context, int state) tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_POWER_UP); - } else + } else { tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } } EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, SND_SOC_TAS2781_FMWLIB); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..ccb9313ada9b 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -380,23 +380,30 @@ static void tasdevice_fw_ready(const struct firmware *fmw, mutex_lock(&tas_priv->codec_lock); ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) + if (ret) { + tasdevice_config_info_remove(tas_priv); goto out; + } tasdevice_create_control(tas_priv); tasdevice_dsp_remove(tas_priv); tasdevice_calbin_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", tas_priv->dev_name); + ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; goto out; } - tasdevice_dsp_create_ctrls(tas_priv); + + ret = tasdevice_dsp_create_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dsp controls error\n"); + goto out; + } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; @@ -417,9 +424,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - /*If DSP FW fail, kcontrol won't be created */ - tasdevice_config_info_remove(tas_priv); + if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + /*If DSP FW fail, DSP kcontrol won't be created */ tasdevice_dsp_remove(tas_priv); } mutex_unlock(&tas_priv->codec_lock); @@ -466,14 +472,14 @@ static int tasdevice_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int ret = 0; - if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - ret = -EINVAL; + if (!(tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK || + tas_priv->fw_state == TASDEVICE_RCA_FW_OK)) { + dev_err(tas_priv->dev, "Bin file not loaded\n"); + return -EINVAL; } - return ret; + return 0; } static int tasdevice_hw_params(struct snd_pcm_substream *substream, -- 2.34.1

1 0

[PATCH v2 0/2] Add PDM controller for StarFive JH8100 SoC
by Xingyu Wu 23 May '24

23 May '24

The Pulse Density Modulation (PDM) controller is a digital PDM microphone interface controller and decoder that supports both mono/stereo PDM format, and a Inter-IC Sound (I2S) transmitter that outputs standard stereo audio data to another device. The PDM controller includes two PDM blocks, each PDM block can drive one bitstream sampling clock and two bitstream coming data (mono/stereo) with sampling clock rising and falling edge. The first patch adds documentation to describe PDM controller bindings. And the second patch adds PDM driver for the StarFive JH8100 SoC. The DTS patch of PDM controller will be submitted after the patchs of JH8100 DTS are accepted. Changes since v1: - Changed the subject prefix of the bindings. - Dropped the confused property in the bindings. - Added the $ref to dai-common. - Changed the kcontrol names and put it into ops. - Added the reset when device resume. - Modified two blocks into a whole PDM which supports up to 4 channels of recorder. v1: https://lore.kernel.org/all/20240307033708.139535-1-xingyu.wu@starfivetech.… Xingyu Wu (2): ASoC: dt-bindings: Add PDM controller for the StarFive JH8100 SoC ASoC: starfive: Add PDM controller support .../bindings/sound/starfive,jh8100-pdm.yaml | 82 ++++ MAINTAINERS | 7 + sound/soc/starfive/Kconfig | 7 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh8100_pdm.c | 447 ++++++++++++++++++ 5 files changed, 544 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh8100-pdm.yaml create mode 100644 sound/soc/starfive/jh8100_pdm.c -- 2.25.1

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topology: pre-processor: Introduce extends/overrides keywords for classes
by GitHub pull_request - edited 23 May '24

23 May '24

alsa-project/alsa-utils pull request #268 was edited from ranj063: Introduce a new keyword "Subtreecopy" for extending existing conf nodes with additional nodes. This feature is useful for extending previous pipeline class definitions with the addition of one or more widgets without having to duplicate everything in the new class definition. For example: Consider a pipeline class definition as below. Note that only the widgets & routes are shown here. ``` Class.Pipeline.mixout-gain-dai-copier-playback { Object.Widget { mixout."1" {} dai-copier."1" {} gain."1" {} pipeline."1" {} } Object.Base { !route [ { source mixout.$index.1 sink gain.$index.1 } ] } } ``` If we want to extend this pipeline with the addition of an eqiir/eqfir, we can create a Subtreecopy node with type extend as follows: ``` Subtreecopy.mixout-gain-eqiir-eqfir-dai-copier-playback { source "Class.Pipeline.mixout-gain-dai-copier-playback" target "Class.Pipeline" type extend Object.Widget { eqiir.1 {} eqfir.1 {} } Object.Base { !route [ { source gain.$index.1 sink eqiir.$index.1 } { source eqiir.$index.1 sink eqfir.$index.1 } ] } } ``` But if we want to modify an existing pipeline class while modifying the order of widgets and/or inserting new widgets, we should use the type "override" instead. This allows for adding new widgets to the list of widgets in the base class definition while also allowing overriding the routes to allow inserting the new widgets and reordering the widgets in the base class. For example, if we want to add a drc widget between the gain and the eqiir modules in the above class, we can do the following: ``` Subtreecopy.mixout-efx-dai-copier-playback { source "Class.Pipeline.mixout-gain-eqiir-eqfir-dai-copier-playback" target "Class.Pipeline" type override Object.Widget { drc.1 {} } Object.Base { !route [ { source mixout.$index.1 sink gain.$index.1 } { source gain.$index.1 sink drc.$index.1 } { source drc.$index.1 sink eqiir.$index.1 } { source eqiir.$index.1 sink eqfir.$index.1 } ] } } ``` Note that the routes array contains all the routes in the new subtreecopy definition. This subtreecopy feature can also be used to copy normal config blocks without having the need to duplicate information. For example, if all the widgets in a pipeline support the same audio formats, we can define it as follows: ``` Class.Pipeline.dai-copier-eqiir-module-copier-capture { formats { num_input_audio_formats 2 num_output_audio_formats 2 Object.Base.input_audio_format [ { in_bit_depth 32 in_valid_bit_depth 32 } { in_channels 4 in_bit_depth 32 in_valid_bit_depth 32 in_ch_cfg $CHANNEL_CONFIG_3_POINT_1 in_ch_map $CHANNEL_MAP_3_POINT_1 } ] Object.Base.output_audio_format [ { out_bit_depth 32 out_valid_bit_depth 32 } { out_channels 4 out_bit_depth 32 out_valid_bit_depth 32 out_ch_cfg $CHANNEL_CONFIG_3_POINT_1 out_ch_map $CHANNEL_MAP_3_POINT_1 } ] } Object.Widget { dai-copier."1" { type dai_out num_output_pins 1 Subtreecopy.formats { source "Class.Pipeline.dai-copier-eqiir-module-copier-capture.formats" type override } } eqiir."1" { Subtreecopy.formats { source "Class.Pipeline.dai-copier-eqiir-module-copier-capture.formats" type override } Object.Control.bytes."1" { IncludeByKey.DMIC0_DAI_EQIIR { "passthrough" "include/components/eqiir/passthrough.conf" "highpass_40hz_0db" "include/components/eqiir/highpass_40hz_0db_48khz.conf" "highpass_40hz_20db" "include/components/eqiir/highpass_40hz_20db_48khz.conf" } } } module-copier."2" { Subtreecopy.formats { source "Class.Pipeline.dai-copier-eqiir-module-copier-capture.formats" type override } } pipeline."1" { priority 0 lp_mode 0 } } } ``` The subtreecopy node in each widget ensures that the pipeline formats are applied to all widgets without the need for duplicating them for each widget. Link: https://github.com/thesofproject/sof/issues/9082 Request URL : https://github.com/alsa-project/alsa-utils/pull/268 Patch URL : https://github.com/alsa-project/alsa-utils/pull/268.patch Repository URL: https://github.com/alsa-project/alsa-utils

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