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[PATCH 1/2] ASoC: fsl_rpmsg: add soc specific data structure
by Shengjiu Wang 27 Aug '21

27 Aug '21

Each platform has different supported rates and formats, so add soc specific data for each platform. This soc specific data is attached with compatible string. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_rpmsg.c | 47 +++++++++++++++++++++++++++++++++++---- sound/soc/fsl/fsl_rpmsg.h | 12 ++++++++++ 2 files changed, 55 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index d60f4dac6c1b..bda6cc96071d 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -138,11 +138,42 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-rpmsg", }; +static const struct fsl_rpmsg_soc_data imx7ulp_data = { + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mm_data = { + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U8 | + SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mn_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mp_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + static const struct of_device_id fsl_rpmsg_ids[] = { - { .compatible = "fsl,imx7ulp-rpmsg-audio"}, - { .compatible = "fsl,imx8mm-rpmsg-audio"}, - { .compatible = "fsl,imx8mn-rpmsg-audio"}, - { .compatible = "fsl,imx8mp-rpmsg-audio"}, + { .compatible = "fsl,imx7ulp-rpmsg-audio", .data = &imx7ulp_data}, + { .compatible = "fsl,imx8mm-rpmsg-audio", .data = &imx8mm_data}, + { .compatible = "fsl,imx8mn-rpmsg-audio", .data = &imx8mn_data}, + { .compatible = "fsl,imx8mp-rpmsg-audio", .data = &imx8mp_data}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); @@ -157,6 +188,14 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) if (!rpmsg) return -ENOMEM; + rpmsg->soc_data = of_device_get_match_data(&pdev->dev); + if (rpmsg->soc_data) { + fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats; + fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats; + } + if (of_property_read_bool(np, "fsl,enable-lpa")) { rpmsg->enable_lpa = 1; rpmsg->buffer_size = LPA_LARGE_BUFFER_SIZE; diff --git a/sound/soc/fsl/fsl_rpmsg.h b/sound/soc/fsl/fsl_rpmsg.h index 4f5b49eb18d8..b04086fbf828 100644 --- a/sound/soc/fsl/fsl_rpmsg.h +++ b/sound/soc/fsl/fsl_rpmsg.h @@ -6,6 +6,16 @@ #ifndef __FSL_RPMSG_H #define __FSL_RPMSG_H +/* + * struct fsl_rpmsg_soc_data + * @rates: supported rates + * @formats: supported formats + */ +struct fsl_rpmsg_soc_data { + int rates; + u64 formats; +}; + /* * struct fsl_rpmsg - rpmsg private data * @@ -15,6 +25,7 @@ * @pll8k: parent clock for multiple of 8kHz frequency * @pll11k: parent clock for multiple of 11kHz frequency * @card_pdev: Platform_device pointer to register a sound card + * @soc_data: soc specific data * @mclk_streams: Active streams that are using baudclk * @force_lpa: force enable low power audio routine if condition satisfy * @enable_lpa: enable low power audio routine according to dts setting @@ -27,6 +38,7 @@ struct fsl_rpmsg { struct clk *pll8k; struct clk *pll11k; struct platform_device *card_pdev; + const struct fsl_rpmsg_soc_data *soc_data; unsigned int mclk_streams; int force_lpa; int enable_lpa; -- 2.17.1

3 2

No cards found in 1.2.5
by GitHub issues - opened 27 Aug '21

27 Aug '21

alsa-project/alsa-ucm-conf issue #111 was opened from Tooniis: On 1.2.4: ``` $ alsaucm listcards 0: HDA-Intel 1: SOF 2: bdw-rt5677 3: broadwell-rt286 4: broxton-rt298 5: bytcht-cx2072x 6: bytcht-es8316 7: bytcr-rt5640 8: bytcr-rt5651 9: cht-bsw-rt5672 10: chtmax98090 11: chtnau8824 chtnau8824 internal card 12: chtrt5645 Intel SoC Audio Device 13: chtrt5650 Intel SoC Audio Device 14: hda-dsp 15: kblrt5660 16: skylake-rt286 17: sof-hda-dsp 18: sof-soundwire ``` On 1.2.5: ``` $ alsaucm listcards list is empty ``` My card's config (snd-soc-apq8096, using DB820c config) isn't being loaded on 1.2.5 as a result and I get no sound. Note that in both cases above, alsa-ucm-conf and alsa-lib are of the same version; as mixing versions causes some other errors (lack of conf.virt.d on alsa-lib 1.2.5 alsa-ucm-conf 1.2.4, and invalid syntax 4 on alsa-lib 1.2.4 with alsa-ucm-conf 1.2.5) Using Arch Linux ARM. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/111 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 0/14] Patches to update for rockchip i2s
by Sugar Zhang 27 Aug '21

27 Aug '21

These patches fixup or update for rockchip i2s. Changes in v3: - Drop property 'rockchip,playback-only', 'rockchip,capture-only'. Implement it by 'dma-names' of DT instead. Changes in v2: - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - split property trcm into single 'trcm-sync-tx-only' and 'trcm-sync-rx-only' suggested by Nicolas. - drop change-id Sugar Zhang (12): ASoC: rockchip: i2s: Add support for set bclk ratio ASoC: rockchip: i2s: Fixup clk div error ASoC: rockchip: i2s: Improve dma data transfer efficiency ASoC: rockchip: i2s: Fix regmap_ops hang ASoC: rockchip: i2s: Fix concurrency between tx/rx ASoC: rockchip: i2s: Reset the controller if soft reset failed ASoC: dt-bindings: rockchip: Document reset property for i2s ASoC: rockchip: i2s: Make playback/capture optional ASoC: rockchip: i2s: Add compatible for more SoCs ASoC: dt-bindings: rockchip: Add compatible strings for more SoCs ASoC: rockchip: i2s: Add support for frame inversion ASoC: dt-bindings: rockchip: i2s: Document property TRCM Xiaotan Luo (1): ASoC: rockchip: i2s: Fixup config for DAIFMT_DSP_A/B Xing Zheng (1): ASoC: rockchip: i2s: Add support for TRCM property .../devicetree/bindings/sound/rockchip-i2s.yaml | 19 ++ sound/soc/rockchip/rockchip_i2s.c | 278 +++++++++++++++------ sound/soc/rockchip/rockchip_i2s.h | 10 +- 3 files changed, 224 insertions(+), 83 deletions(-) -- 2.7.4

2 13

[PATCH v1 0/7] Patches to update for rockchip pdm
by Sugar Zhang 27 Aug '21

27 Aug '21

These patches fixup or update for rockchip pdm. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 --------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 114 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 ++++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 226 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

2 10

Error in recording and playing at the same time on raspberry pi
by GitHub issues - opened 27 Aug '21

27 Aug '21

alsa-project/alsa-lib issue #174 was opened from amirush786: I am try to record and play at the same time. I have a speaker connected to pi via aux cable. I have to play a sound from the speaker and also re ord it at the same time from the tool which is seeed microphone tool which is connected to pi. I am able to do that but the recorded file has some hissssshhhhhhhh sound in it don't know why it's coming out to that way. But when I do a individual record I can hear a bit proper noise and less of hisshy noise in it. And also how can I check the capture sound is high or not. I followed the approach below. I did Alsamixer command and it's shows me this. So after that I typed in F6 to change my card to seeed card. Then is scroll down to seed card and hit enter. So I see this. Then I press f4 and I see this so I don't know where should I see that capture is high. This is the last thing that messing with me. ![20210822_200619.jpg](https://user-images.githubusercontent.com/31735000/131046816-b77895b8-f6c8-4faa-b624-283100ad00f8.jpg) ![20210822_200626.jpg](https://user-images.githubusercontent.com/31735000/131046829-fab9edb3-d9b2-4938-be84-e7dcccfb4aaf.jpg) ![20210822_200641.jpg](https://user-images.githubusercontent.com/31735000/131046851-631a8e34-33f3-4e88-b971-726b2fdd7756.jpg) Issue URL : https://github.com/alsa-project/alsa-lib/issues/174 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] [v3] ASoC: Intel: sof_rt5682: Add support for max98390 speaker amp
by Mark Hsieh 26 Aug '21

26 Aug '21

Configure adl_max98390_rt5682 to support the rt5682 headset codec and max98390 speaker. Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 57 +++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 12 ++++ sound/soc/intel/boards/sof_rt5682.c | 19 +++++++ .../intel/common/soc-acpi-intel-adl-match.c | 13 +++++ 5 files changed, 102 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 7e29b0d911e2..0017c08c5a62 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -470,6 +470,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1015 select SND_SOC_RT1015P select SND_SOC_RT5682_I2C + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e9c52f8b6428..a6fcf6e08996 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,63 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { /* For Right */ + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { /* For Left */ + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + /* DEV0 tdm slot configuration */ + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 2, 16); + /* DEV1 tdm slot configuration */ + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 2, 16); + } + return 0; +} + +const struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2674f1e373ef..bad7b5303280 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,18 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern const struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39217223d50c..18b6ef311462 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,7 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -781,6 +782,13 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; } else { max_98357a_dai_link(&links[id]); } @@ -917,6 +925,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682); if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1043,6 +1053,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.17.1

3 2

[PATCH 0/3] ASoC: wcd9335: Firx some resources leak in the probe and remove function
by Christophe JAILLET 26 Aug '21

26 Aug '21

The first 2 patches are sraightforward and look logical to me. However, the 3rd one in purely speculative. It is based on the fact that a comment states that we enable some irqs on some slave ports. That said, it writes 0xFF in some registers. So, I guess that we should disable these irqs when the driver is removed. That said, writing 0x00 at the same place looks logical to me. This cis untested and NOT based on any documentation. Just a blind fix. Review with care. You'll be warned :) Christophe JAILLET (3): ASoC: wcd9335: Fix a double irq free in the remove function ASoC: wcd9335: Fix a memory leak in the error handling path of the probe function ASoC: wcd9335: Disable irq on slave ports in the remove function sound/soc/codecs/wcd9335.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) -- 2.30.2

2 4

[PATCH v2] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 26 Aug '21

26 Aug '21

The document was sent in [1] and merged to ASoC for-next branch as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [2] revealed that the commit 1afc60e00de3 breaks dt_binding_check in linux-next due to dt-bindings/clock/mt8192-clk.h doesn't exist. [3] reverted the commit 1afc60e00de3 and merged as commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document"). dt-bindings/clock/mt8192-clk.h now can be found in linux-next per [4], re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176577.… [2]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… [3]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176897.… [4]: https://git.kernel.org/pub/scm/linux/kernel/git/clk/linux.git/commit/?h=clk… Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from v1 (https://patchwork.kernel.org/project/alsa-devel/list/?series=537817): - Add more context to the commit message. .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..0170c93802d2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.rc2.250.ged5fa647cd-goog

2 1

[PATCH] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 26 Aug '21

26 Aug '21

The document was removed temporarily by commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") because the dependency was not merged at that time. The dependency has merged. Re-adds back the document. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- More context: - [1] reported `dt_binding_check` was broken. - [2] reverted. - The dependency has merged in [3] and it can be found from linux-next. [1]: https://patchwork.kernel.org/project/alsa-devel/patch/1604390378-23993-8-gi… [2]: https://patchwork.kernel.org/project/alsa-devel/patch/1605081920-11848-1-gi… [3]: https://git.kernel.org/pub/scm/linux/kernel/git/clk/linux.git/commit/?h=clk… .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..0170c93802d2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.rc2.250.ged5fa647cd-goog

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[PATCH] ALSA: hda: Disable runtime resume at shutdown
by Takashi Iwai 26 Aug '21

26 Aug '21

Although we modified the codec shutdown callback to perform runtime-suspend, it's still not fully effective, as this may be resumed again at any time later. For fixing such an unwanted resume, this patch replaces pm_runtime_suspend() with pm_runtime_force_suspend(), and call pm_runtime_disable() afterward. It assures to keep the device suspended. Also for code simplification, we apply the code unconditionally; when it's been already suspended, nothing would happen by calls of snd_pcm_suspend_all() and pm_runtime_force_suspend(), just proceed to pm_runtime_disable(). Fixes: b98444ed597d ("ALSA: hda: Suspend codec at shutdown") Reported-and-tested-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c91c92c9ab2..a9ebefd60cf6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2986,13 +2986,11 @@ void snd_hda_codec_shutdown(struct hda_codec *codec) { struct hda_pcm *cpcm; - if (pm_runtime_suspended(hda_codec_dev(codec))) - return; - list_for_each_entry(cpcm, &codec->pcm_list_head, list) snd_pcm_suspend_all(cpcm->pcm); - pm_runtime_suspend(hda_codec_dev(codec)); + pm_runtime_force_suspend(hda_codec_dev(codec)); + pm_runtime_disable(hda_codec_dev(codec)); } /* -- 2.31.1

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