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

[PATCH v2 0/3] Add audio device nodes for i.MX93 platform
by Chancel Liu 08 Nov '23

08 Nov '23

changes in v2 - Move MQS device node under "/" root node - Convert fsl,mqs.txt to yaml Chancel Liu (3): arm64: dts: imx93: Add audio device nodes arm64: dts: imx93: Configure clock rate for audio PLL ASoC: dt-bindings: fsl,mqs: Convert format to json-schema .../devicetree/bindings/sound/fsl,mqs.txt | 36 ------ .../devicetree/bindings/sound/fsl,mqs.yaml | 113 ++++++++++++++++++ arch/arm64/boot/dts/freescale/imx93.dtsi | 89 ++++++++++++++ 3 files changed, 202 insertions(+), 36 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.yaml -- 2.25.1

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[RFC PATCH v8 00/13] Add audio support in v4l2 framework
by Shengjiu Wang 07 Nov '23

07 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 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 (13): 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: imx-asrc: Add memory to memory driver media: vim2m_audio: add virtual driver for audio memory to memory .../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 | 12 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1186 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 9 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m_audio.c | 680 ++++++++++ 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 | 17 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + 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 +- 35 files changed, 2771 insertions(+), 11 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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[PATCH 1/3] soundwire: qcom: drop unneeded DAI .set_stream callback
by Krzysztof Kozlowski 06 Nov '23

06 Nov '23

Qualcomm Soundwire controller drivers do not support multi-link setups, so DAI .set_stream() callback will not be used. What's more, if called it will overwrite the sdw_stream_runtime runtime set in DAI .startup (qcom_swrm_startup()) causing issues (unsupported multi-link error) when two Soundwire controllers are passed as codec DAIs. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/qcom.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index f1b8d6ac5140..fe65c26c5281 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1267,16 +1267,6 @@ static int qcom_swrm_hw_free(struct snd_pcm_substream *substream, return 0; } -static int qcom_swrm_set_sdw_stream(struct snd_soc_dai *dai, - void *stream, int direction) -{ - struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev); - - ctrl->sruntime[dai->id] = stream; - - return 0; -} - static void *qcom_swrm_get_sdw_stream(struct snd_soc_dai *dai, int direction) { struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev); @@ -1349,7 +1339,6 @@ static const struct snd_soc_dai_ops qcom_swrm_pdm_dai_ops = { .hw_free = qcom_swrm_hw_free, .startup = qcom_swrm_startup, .shutdown = qcom_swrm_shutdown, - .set_stream = qcom_swrm_set_sdw_stream, .get_stream = qcom_swrm_get_sdw_stream, }; -- 2.34.1

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[PATCH 1/3] ASoC: amd: acp: Fix for indentation issue
by Syed Saba Kareem 06 Nov '23

06 Nov '23

Fix indentation issue reported in acp70_pcm_resume() function. Fixes: e84db124cb21 (ASoC: amd: acp: Add pci legacy driver support for acp7.0 platform") Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> --- sound/soc/amd/acp/acp70.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index dd384c966ae9..1d699c96f619 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -229,8 +229,8 @@ static int __maybe_unused acp70_pcm_resume(struct device *dev) } } } - spin_unlock(&adata->acp_lock); - return 0; + spin_unlock(&adata->acp_lock); + return 0; } static const struct dev_pm_ops acp70_dma_pm_ops = { -- 2.25.1

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[PATCH] ASoC: codecs: aw88399: Fix -Wuninitialized in aw_dev_set_vcalb()
by Nathan Chancellor 06 Nov '23

06 Nov '23

Clang warns (or errors with CONFIG_WERROR=y): sound/soc/codecs/aw88399.c:441:18: error: variable 'vsense_select' is uninitialized when used here [-Werror,-Wuninitialized] 441 | vsense_select = vsense_select & (~AW88399_VDSEL_MASK); | ^~~~~~~~~~~~~ sound/soc/codecs/aw88399.c:431:28: note: initialize the variable 'vsense_select' to silence this warning 431 | unsigned int vsense_select, vsense_value; | ^ | = 0 1 error generated. This clearly should have been using the value received from regmap_read(). Use the correct variable to resolve the warning. Closes: https://github.com/ClangBuiltLinux/linux/issues/1952 Fixes: 8ade6cc7e261 ("ASoC: codecs: Add aw88399 amplifier driver") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/codecs/aw88399.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/aw88399.c b/sound/soc/codecs/aw88399.c index ce30bc7cdea9..47fe38d0def5 100644 --- a/sound/soc/codecs/aw88399.c +++ b/sound/soc/codecs/aw88399.c @@ -438,7 +438,7 @@ static int aw_dev_set_vcalb(struct aw88399 *aw88399) if (ret) return ret; - vsense_select = vsense_select & (~AW88399_VDSEL_MASK); + vsense_select = vsense_value & (~AW88399_VDSEL_MASK); ret = aw88399_dev_get_icalk(aw88399, &icalk); if (ret) { --- base-commit: 60781d2d5899d6fc5fd173c1bcffaaec39643e17 change-id: 20231027-asoc-aw88399-fix-wuninitialized-63b554f8fa9a Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

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[PATCH v2 1/2] ALSA: docs: Add Marian M2 driver documentation
by Ivan Orlov 05 Nov '23

05 Nov '23

Add documentation for the new MARIAN Seraph M2 sound card. It covers current development status, available controls of the card and information about the integrated loopback. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Remove redundant documentation fix from the next patch in the series Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/marian-m2.rst | 104 ++++++++++++++++++++++++ 2 files changed, 105 insertions(+) create mode 100644 Documentation/sound/cards/marian-m2.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index e68bbb13c384..e873592d8d00 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -19,3 +19,4 @@ Card-Specific Information serial-u16550 img-spdif-in pcmtest + marian-m2 diff --git a/Documentation/sound/cards/marian-m2.rst b/Documentation/sound/cards/marian-m2.rst new file mode 100644 index 000000000000..bf12445e20d7 --- /dev/null +++ b/Documentation/sound/cards/marian-m2.rst @@ -0,0 +1,104 @@ +.. SPDX-License-Identifier: GPL-2.0 + +======================= +MARIAN Seraph M2 Driver +======================= + +Sep 18, 2023 + +Ivan Orlov <ivan.orlov0322(a)gmail.com> + +STATE OF DEVELOPMENT +==================== + +This driver is based on the driver written by Florian Faber in 2012, which seemed to work fine. +However, the initial code contained multiple issues, which had to be solved before sending the +driver upstream. + +The vendor lost the full documentation, so what we have here was recovered from drafts and found +after experiments with the card. + +What seems to be working fine: +- Playback and capture for all supported rates +- Integrated loopback (with some exceptions, see below) + +MEMORY MODEL +============ + +The hardware requires one huge contiguous DMA space to be allocated. After allocation, the bus address of +this buffer should be written to the hardware register. + +We can split this space into two parts: the first one contains samples for capture, another one contains +play samples: + +CAPTURE_CH_0, CAPTURE_CH_1, ..., CAPTURE_CH_127 | PLAY_CH_0, PLAY_CH_1, ..., PLAY_CH_127 + +The card supports the non-interleaved access mode only, so samples for each channel lay together: + +C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 | C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 + +The count of samples per each channel buffer needs to be set explicitly, so the address of the first +byte of the playback data depends on this value. The playback buffer starts where the capture buffer ends. +It makes the arbitrary period count/buffer size feature impossible to implement, and the driver supports only +2 periods per buffer. + +Controls +======== + +Input 1 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal +Input 2 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal + +Input 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Input 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Input 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Input 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Input 1 Frequency + Measured frequency on Input 1 +Input 2 Frequency + Measured frequency on Input 2 + +Output 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Output 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Output 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Output 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Clock Source + Internal/Sync Bus, Input Port 1, Input Port 2 + +Speed mode + VCO clock range (slow/fast) + +DCO Frequency (Hz) + +DCO Frequency (ms) + +DCO Detune + Range: -200...200 + +Loopback + Enable/Disable integrated loopback + + +Loopback +======== + +The card contains integrated loopback. When it is enabled, it sets the hardware’s DAW-in memory pointer +to the hardware’s DAW-out memory. So, what you play is what you record. + +You can enable the integrated loopback using the corresponding control. + +The loopback seems to work well on lower rates (like 28000). However, when the rate goes higher, I observe +multiple mistakes in recorded byte ordering. -- 2.34.1

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[PATCH AUTOSEL 5.15 13/28] ASoC: codecs: wsa-macro: handle component name prefix
by Sasha Levin 04 Nov '23

04 Nov '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit c29e5263d32a6d0ec094d425ae7fef3fa8d4da1c ] When comparing widget names in wsa_macro_spk_boost_event(), consider also the component's name prefix. Otherwise the WSA codec won't have proper mixer setup resulting in no sound playback through speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://lore.kernel.org/r/20231003155710.821315-3-krzysztof.kozlowski@linar… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-wsa-macro.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index dd1a8b7bc794c..643ddf7dd6c5d 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1668,12 +1668,12 @@ static int wsa_macro_spk_boost_event(struct snd_soc_dapm_widget *w, u16 boost_path_ctl, boost_path_cfg1; u16 reg, reg_mix; - if (!strcmp(w->name, "WSA_RX INT0 CHAIN")) { + if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT0 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST0_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX0_RX_PATH_CFG1; reg = CDC_WSA_RX0_RX_PATH_CTL; reg_mix = CDC_WSA_RX0_RX_PATH_MIX_CTL; - } else if (!strcmp(w->name, "WSA_RX INT1 CHAIN")) { + } else if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT1 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST1_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX1_RX_PATH_CFG1; reg = CDC_WSA_RX1_RX_PATH_CTL; -- 2.42.0

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[PATCH] ASoC: rt5682s: Add LDO control for dacref
by Jack Yu 01 Nov '23

01 Nov '23

Add LDO control for dacref. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 12 ++++++++++++ sound/soc/codecs/rt5682s.h | 3 +++ 2 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index c261c33c4be7..c05996e8fc5a 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1017,6 +1017,15 @@ static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); +/* LDO output select */ +static const char * const rt5682s_ldo_output_select[] = { + "1.607V", "1.5V", "1.406V", "1.731V" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_ldo_output_enum, RT5682S_BIAS_CUR_CTRL_7, + RT5682S_DVO_LDO_DACREF_SFT, rt5682s_ldo_output_select); + + static const struct snd_kcontrol_new rt5682s_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682S_DAC1_DIG_VOL, @@ -1035,6 +1044,9 @@ static const struct snd_kcontrol_new rt5682s_snd_controls[] = { /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682S_STO1_ADC_BOOST, RT5682S_STO1_ADC_L_BST_SFT, RT5682S_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + + /* LDO Output Select Control */ + SOC_ENUM("LDO Output Select", rt5682s_ldo_output_enum), }; /** diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 1d79d432d0d8..0ae5a35825f4 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1263,6 +1263,9 @@ #define RT5682S_JDH_NO_PLUG (0x1 << 4) #define RT5682S_JDH_PLUG (0x0 << 4) +/* Bias current control 7 (0x0110) */ +#define RT5682S_DVO_LDO_DACREF_SFT 4 + /* Charge Pump Internal Register1 (0x0125) */ #define RT5682S_CP_CLK_HP_MASK (0x3 << 4) #define RT5682S_CP_CLK_HP_100KHZ (0x0 << 4) -- 2.34.1

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[PATCH 0/3]
by Zhu Ning 01 Nov '23

01 Nov '23

*** BLURB HERE *** Hi , 3 patches here for the es8326 driver... We developed a new version of the chip. 3 Three patches are used for compatibility with the old and new versions of the chip. The test results from the test department met our expectations Thanks, Zhuning. Zhu Ning (3): ASoC: codecs: ES8326: Add chip version flag ASoC: codecs: ES8326: Changing initialisation and broadcasting ASoC: codecs: ES8326: Changing the headset detection time sound/soc/codecs/es8326.c | 219 ++++++++++++++++++++++---------------- 1 file changed, 127 insertions(+), 92 deletions(-) mode change 100644 => 100755 sound/soc/codecs/es8326.c -- 2.17.1

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[PATCH next] ALSA: hda: cs35l41: Fix missing error code in cs35l41_smart_amp()
by Harshit Mogalapalli 31 Oct '23

31 Oct '23

When firmware status is invalid, assign -EINVAL to ret as ret is '0' at that point and returning success is incorrect when firmware status is invalid. Fixes: a51d8ba03a4f ("ALSA: hda: cs35l41: Check CSPL state after loading firmware") Signed-off-by: Harshit Mogalapalli <harshit.m.mogalapalli(a)oracle.com> --- This is found using smatch and only compile tested. --- sound/pci/hda/cs35l41_hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 496ff6a9d300..09f8d8fa4c71 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1042,6 +1042,7 @@ static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) default: dev_err(cs35l41->dev, "Firmware status is invalid: %u\n", fw_status); + ret = -EINVAL; goto clean_dsp; } -- 2.39.3

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