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

[PATCH v13 00/16] Add audio support in v4l2 framework
by Shengjiu Wang 11 Mar '24

11 Mar '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 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 | 87 ++ .../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 + .../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 | 44 +- 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 +- 42 files changed, 3109 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

4 26

[PATCH 00/22] ASoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 07 Mar '24

07 Mar '24

This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (22): ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: Commonize ADDA rate transform functions and enums ASoC: mediatek: Assign dummy when codec not specified for a DAI link ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 112 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 112 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 129 +- .../bindings/sound/mt8195-mt6359.yaml | 123 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/common/Makefile | 2 + .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 + .../soc/mediatek/common/mtk-dai-adda-common.h | 45 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 205 ++- .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 94 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 114 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 35 files changed, 1739 insertions(+), 2769 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

4 31

[PATCH] ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()
by Richard Fitzgerald 06 Mar '24

06 Mar '24

snd_soc_card_get_kcontrol() must be holding a read lock on card->controls_rwsem while walking the controls list. Compare with snd_ctl_find_numid(). The existing function is renamed snd_soc_card_get_kcontrol_locked() so that it can be called from contexts that are already holding card->controls_rwsem (for example, control get/put functions). There are few direct or indirect callers of snd_soc_card_get_kcontrol(), and most are safe. Three require changes, which have been included in this patch: codecs/cs35l45.c: cs35l45_activate_ctl() is called from a control put() function so is changed to call snd_soc_card_get_kcontrol_locked(). codecs/cs35l56.c: cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from control get()/put() functions so is changed to call snd_soc_card_get_kcontrol_locked(). fsl/fsl_xcvr.c: fsl_xcvr_activate_ctl() is called from three places, one of which already holds card->controls_rwsem: 1. fsl_xcvr_mode_put(), a control put function, which will already be holding card->controls_rwsem. 2. fsl_xcvr_startup(), a DAI startup function. 3. fsl_xcvr_shutdown(), a DAI shutdown function. To fix this, fsl_xcvr_activate_ctl() has been changed to call snd_soc_card_get_kcontrol_locked() so that it is safe to call directly from fsl_xcvr_mode_put(). The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been changed to take a read lock on card->controls_rsem() around calls to fsl_xcvr_activate_ctl(). While this is not very elegant, it keeps the change small, to avoid this patch creating a large collateral churn in fsl/fsl_xcvr.c. Analysis of other callers of snd_soc_card_get_kcontrol() is that they do not need any changes, they are not holding card->controls_rwsem when they call snd_soc_card_get_kcontrol(). Direct callers of snd_soc_card_get_kcontrol(): fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler Indirect callers via soc_component_notify_control(): codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler codecs/ak4118.c: ak4118_irq_handler() - IRQ handler codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used Indirect callers via snd_soc_limit_volume(): qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function ti/rx51.c: rx51_aic34_init() - DAI init function I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c and ak4118.c changes. Backport note: The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added since the Fixes commit so won't all be present on older kernels. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 209c6cdfd283 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card") --- It would be great if people could test the fsl/, qcom/, ti/rx51 and ak4418 drivers. --- include/sound/soc-card.h | 2 ++ sound/soc/codecs/cs35l45.c | 2 +- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 12 +++++++++++- sound/soc/soc-card.c | 24 ++++++++++++++++++++++-- 5 files changed, 37 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h index ecc02e955279..1f4c39922d82 100644 --- a/include/sound/soc-card.h +++ b/include/sound/soc-card.h @@ -30,6 +30,8 @@ static inline void snd_soc_card_mutex_unlock(struct snd_soc_card *card) struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name); int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, struct snd_soc_jack *jack); int snd_soc_card_jack_new_pins(struct snd_soc_card *card, const char *id, diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 44c221745c3b..2392c6effed8 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -184,7 +184,7 @@ static int cs35l45_activate_ctl(struct snd_soc_component *component, else snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s", ctl_name); - kcontrol = snd_soc_card_get_kcontrol(component->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(component->card, name); if (!kcontrol) { dev_err(component->dev, "Can't find kcontrol %s\n", name); return -EINVAL; diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 2c1313e34cce..6dd0319bc843 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -114,7 +114,7 @@ static int cs35l56_sync_asp1_mixer_widgets_with_firmware(struct cs35l56_private name = full_name; } - kcontrol = snd_soc_card_get_kcontrol(dapm->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(dapm->card, name); if (!kcontrol) { dev_warn(cs35l56->base.dev, "Could not find control %s\n", name); continue; diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index f0fb33d719c2..c46f64557a7f 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -174,7 +174,9 @@ static int fsl_xcvr_activate_ctl(struct snd_soc_dai *dai, const char *name, struct snd_kcontrol *kctl; bool enabled; - kctl = snd_soc_card_get_kcontrol(card, name); + lockdep_assert_held(&card->snd_card->controls_rwsem); + + kctl = snd_soc_card_get_kcontrol_locked(card, name); if (kctl == NULL) return -ENOENT; @@ -576,10 +578,14 @@ static int fsl_xcvr_startup(struct snd_pcm_substream *substream, xcvr->streams |= BIT(substream->stream); if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + /* Disable XCVR controls if there is stream started */ + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, false); + up_read(&card->snd_card->controls_rwsem); } return 0; @@ -598,11 +604,15 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, /* Enable XCVR controls if there is no stream started */ if (!xcvr->streams) { if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, true); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, (xcvr->mode == FSL_XCVR_MODE_ARC)); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, (xcvr->mode == FSL_XCVR_MODE_EARC)); + up_read(&card->snd_card->controls_rwsem); } ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, FSL_XCVR_IRQ_EARC_ALL, 0); diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c index 285ab4c9c716..8a2f163da6bc 100644 --- a/sound/soc/soc-card.c +++ b/sound/soc/soc-card.c @@ -5,6 +5,9 @@ // Copyright (C) 2019 Renesas Electronics Corp. // Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> // + +#include <linux/lockdep.h> +#include <linux/rwsem.h> #include <sound/soc.h> #include <sound/jack.h> @@ -26,12 +29,15 @@ static inline int _soc_card_ret(struct snd_soc_card *card, return ret; } -struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, - const char *name) +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name) { struct snd_card *card = soc_card->snd_card; struct snd_kcontrol *kctl; + /* must be held read or write */ + lockdep_assert_held(&card->controls_rwsem); + if (unlikely(!name)) return NULL; @@ -40,6 +46,20 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, return kctl; return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol_locked); + +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + down_read(&card->controls_rwsem); + kctl = snd_soc_card_get_kcontrol_locked(soc_card, name); + up_read(&card->controls_rwsem); + + return kctl; +} EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); static int jack_new(struct snd_soc_card *card, const char *id, int type, -- 2.30.2

3 3

ASoC: rockchip_i2s_tdm calibration clocking problem
by Pavel Hofman 04 Mar '24

04 Mar '24

Hi, I am hitting a clock issue with rockchip_i2s_tdm.c + simple-audio-card (+ RK3308). At boot the mclk clk_i2s0_8ch_tx is (somehow) initialized to some (unimportant?) value 50176000 Hz. Note that this frequency is not multiple of either 48kHz or 44.1kHz. Method asoc_simple_parse_clk() reads this value and sets it to simple_dai->sysclk. Subsequently at asoc_simple_dai_init this "random" initial value is stored to i2s_tdm->mclk_tx_freq: 17.839330] rockchip_i2s_tdm_set_sysclk+0x50/0xbc [snd_soc_rockchip_i2s_tdm] [ 17.839367] snd_soc_dai_set_sysclk+0x38/0xb8 [snd_soc_core] [ 17.839596] asoc_simple_init_dai+0x94/0xc0 [snd_soc_simple_card_utils] [ 17.839640] asoc_simple_dai_init+0x130/0x230 [snd_soc_simple_card_utils] [ 17.839672] snd_soc_link_init+0x28/0x90 [snd_soc_core] [ 17.839843] snd_soc_bind_card+0x60c/0xbb4 [snd_soc_core] When starting playback, called by rockchip_i2s_tdm_hw_params(), rockchip_i2s_tdm_calibrate_mclk() correctly switches parent of mclk_parent to correct root pll clock mclk_root0/1 for the given samplerate and correctly configures mclk_parent frequency. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… But right after that, the next line of rockchip_i2s_tdm_hw_params() calls rockchip_i2s_tdm_set_mclk() https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… This method calls clk_set_rate(i2s_tdm->mclk_tx, i2s_tdm->mclk_tx_freq), which resets the clock and its parental chain to the original incorrect value stored in i2s_tdm->mclk_tx_freq from the dai initialization. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… As a result, no matter what sample rate is being played, the i2s mclk clock always ends up configured incorrectly. DTS I2S sets all clocks, therefore the clk calibration in rockchip_i2s_tdm.c should be (and is) used: i2s_8ch_0: i2s@ff300000 { compatible = "rockchip,rk3308-i2s-tdm"; reg = <0x0 0xff300000 0x0 0x1000>; interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; clocks = <&cru SCLK_I2S0_8CH_TX>, <&cru SCLK_I2S0_8CH_RX>, <&cru HCLK_I2S0_8CH>, <&cru SCLK_I2S0_8CH_TX_SRC>, <&cru SCLK_I2S0_8CH_RX_SRC>, <&cru PLL_VPLL0>, <&cru PLL_VPLL1>; clock-names = "mclk_tx", "mclk_rx", "hclk", "mclk_tx_src", "mclk_rx_src", "mclk_root0", "mclk_root1"; ......... It seems to me that the calibration code should also rewrite the initially incorrect i2s_tdm->mclk_tx_freq and i2s_tdm->mclk_rx_freq with correct values corresponding to the momentary hw_params rate, or maybe rockchip_i2s_tdm_set_mclk() should not be called if rockchip_i2s_tdm_calibrate_mclk() is called a line above (i.e. putting the call to rockchip_i2s_tdm_set_mclk() into "else" branch). Thank you very much for help. With regards, Pavel.

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[PATCH] [alsa-lib][PATCH] conf: USB-Audio: Add Corsair HS60 Pro to the IEC958 blacklist
by David Senoner 04 Mar '24

04 Mar '24

This device is just an external USB soundcard with a female 3.5mm jack for a headset, no S/PDIF | IEC958 connector. Add it to the blacklist to prevent it being opened. Signed-off-by: David Senoner <seda18(a)rolmail.net> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 80631b2e..08346a1b 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -53,6 +53,7 @@ USB-Audio.pcm.iec958_device { "Blue Snowball" 999 "C-Media USB Headphone Set" 999 "Cmedia Audio" 999 + "Corsair HS60 PRO Surround USB S" 999 "DELL PROFESSIONAL SOUND BAR AE5" 999 "HP Digital Stereo Headset" 999 "GN 9330" 999 -- 2.43.2

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[PATCH v3 0/5] ALSA/ASoC: Conditionally skip i915 init and cleanups
by Cezary Rojewski 04 Mar '24

04 Mar '24

A small set of changes to improve initialization of the audio stack on HDAudio devices and pair of cleanups. As the first change is the most important one here, following is the technical background for it: Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends removed support for i915 for all CNL-based platforms. HDAudio library, however, still treats such platforms as valid candidates for i915 binding. Update query mechanism to reflect changes made in drm tree. At the same time, i915 support for LKF-based platforms has not been provided so remove them from valid binding candidates. The snd_soc_hda change is a follow up for the above and the cleanup patches do not bring any functional changes. Changes in v3: - snd_soc_hda_codec now returns -ENODEV on attach() if i915 is not present - denylist now const - added new patch for the avs-driver to address -ENODEV during probe_codec() - note: retained reviewed-by for patch 1/4 as changes are minimal Changes in v2: - list of problematic VGA devices is now declared locally, no more touching drm stuff Cezary Rojewski (5): ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing ASoC: Intel: avs: Ignore codecs with no suppoting driver ASoC: codecs: hda: Cleanup error messages ALSA: hda: Reuse for_each_pcm_streams() sound/hda/hdac_i915.c | 32 +++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.c | 2 +- sound/soc/codecs/hda.c | 15 ++++++++++----- sound/soc/intel/avs/core.c | 9 +++++---- 4 files changed, 45 insertions(+), 13 deletions(-) -- 2.25.1

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[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Cezary Rojewski 03 Mar '24

03 Mar '24

Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f9c0adc0738d..4e9e7d2a942d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1718,7 +1718,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1771,7 +1771,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1815,7 +1815,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.25.1

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[PATCH] soundwire: constify the struct device_type usage
by Ricardo B. Marliere 03 Mar '24

03 Mar '24

Since commit aed65af1cc2f ("drivers: make device_type const"), the driver core can properly handle constant struct device_type. Move the sdw_master_type and sdw_slave_type variables to be constant structures as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ricardo B. Marliere <ricardo(a)marliere.net> --- drivers/soundwire/master.c | 2 +- drivers/soundwire/slave.c | 2 +- include/linux/soundwire/sdw_type.h | 4 ++-- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 51abedbbaa66..b2c64512739d 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -112,7 +112,7 @@ static const struct dev_pm_ops master_dev_pm = { pm_generic_runtime_resume, NULL) }; -struct device_type sdw_master_type = { +const struct device_type sdw_master_type = { .name = "soundwire_master", .release = sdw_master_device_release, .pm = &master_dev_pm, diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 060c2982e26b..9963b92eb505 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -16,7 +16,7 @@ static void sdw_slave_release(struct device *dev) kfree(slave); } -struct device_type sdw_slave_type = { +const struct device_type sdw_slave_type = { .name = "sdw_slave", .release = sdw_slave_release, .uevent = sdw_slave_uevent, diff --git a/include/linux/soundwire/sdw_type.h b/include/linux/soundwire/sdw_type.h index b445f7200f06..693320b4f5c2 100644 --- a/include/linux/soundwire/sdw_type.h +++ b/include/linux/soundwire/sdw_type.h @@ -5,8 +5,8 @@ #define __SOUNDWIRE_TYPES_H extern const struct bus_type sdw_bus_type; -extern struct device_type sdw_slave_type; -extern struct device_type sdw_master_type; +extern const struct device_type sdw_slave_type; +extern const struct device_type sdw_master_type; static inline int is_sdw_slave(const struct device *dev) { --- base-commit: 81a7d0c4d059cb5c122110acbeec7bedfb91a741 change-id: 20240219-device_cleanup-soundwire-c037fffa2746 Best regards, -- Ricardo B. Marliere <ricardo(a)marliere.net>

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[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Amadeusz Sławiński 03 Mar '24

03 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 69719b335bcb..652aae4e221d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1717,7 +1717,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1770,7 +1770,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1814,7 +1814,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.34.1

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[PATCH V3 00/13] soundwire/SOF: add SoundWire Interface support for AMD SOF stack
by Vijendar Mukunda 02 Mar '24

02 Mar '24

This patch series is to redesign existing platform device creation logic for SoundWire managers and Implement generic functions for SoundWire manager probe, start and exit sequence which are common for both Legacy (NO DSP enabled) and SOF stack, and add SoundWire Interface support for AMD SOF stack (ACP 6.3 based platform). Below patch series is reviewed in https://github.com/thesofproject/linux/pull/4699 Changes since V1: - add new commit to update license for amd soundwire manager driver files. - wrap the commit message with in 75 characters. - add amd tag in commit title for soundwire patches Changes since V2: - drop "drivers" tag from commit title for SoundWire patches. This patch series has dependencies. "drivers: soundwire: amd: refactor amd soundwire manager device node creation" patch has dependency on below patch. "ASoC/soundwire: implement generic api for scanning amd soundwire controller" Similarly SOF patches has dependency on Soundwire tree patches. This patch series has to be reviewed and merged in one go. Vijendar Mukunda (13): ASoC/soundwire: implement generic api for scanning amd soundwire controller soundwire: amd: update license soundwire: amd: refactor amd soundwire manager device node creation soundwire: amd: implement function to extract slave information soundwire: amd: refactor soundwire pads enable soundwire: amd: refactor register mask structure ASoC: SOF: amd: add code for invoking soundwire manager helper functions ASoC: SOF: amd: add interrupt handling for SoundWire manager devices ASoC: SOF: amd: Add Soundwire DAI configuration support for AMD platforms ASoC: SOF: amd: add machine select logic for soundwire based platforms ASoC: SOF: amd: update descriptor fields for acp6.3 based platform ASoC: SOF: amd: select soundwire dependency flag for acp6.3 based platform ASoC: SOF: amd: refactor acp driver pm ops drivers/soundwire/Makefile | 2 +- drivers/soundwire/amd_init.c | 235 +++++++++++++++++++++++++++++ drivers/soundwire/amd_init.h | 13 ++ drivers/soundwire/amd_manager.c | 45 ++---- drivers/soundwire/amd_manager.h | 14 +- include/linux/soundwire/sdw_amd.h | 81 ++++++++-- include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai.h | 2 + include/uapi/sound/sof/tokens.h | 4 + sound/soc/amd/acp/Kconfig | 7 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/amd-sdw-acpi.c | 62 ++++++++ sound/soc/sof/amd/Kconfig | 18 +++ sound/soc/sof/amd/acp-common.c | 65 +++++++- sound/soc/sof/amd/acp-dsp-offset.h | 10 ++ sound/soc/sof/amd/acp.c | 202 ++++++++++++++++++++++++- sound/soc/sof/amd/acp.h | 26 +++- sound/soc/sof/amd/pci-acp63.c | 7 + sound/soc/sof/ipc3-pcm.c | 25 +++ sound/soc/sof/ipc3-topology.c | 40 +++++ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 5 + 22 files changed, 802 insertions(+), 71 deletions(-) create mode 100644 drivers/soundwire/amd_init.c create mode 100644 drivers/soundwire/amd_init.h create mode 100644 sound/soc/amd/acp/amd-sdw-acpi.c -- 2.34.1

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