November 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ASoC: amd: Convert to new style DAI format definitions
by Mark Brown 09 Dec '21

09 Dec '21

Convert the AMD machine drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 20 ++++++++++---------- sound/soc/amd/acp-rt5645.c | 4 ++-- sound/soc/amd/acp3x-rt5682-max9836.c | 8 ++++---- 3 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b2065f3fe42c..3bf86c2424ae 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -522,7 +522,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_da7219_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -533,7 +533,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-da7219-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_da7219_cap_ops, @@ -543,7 +543,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_max_play_ops, @@ -554,7 +554,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic0_cap_ops, @@ -565,7 +565,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_dmic1_cap_ops, @@ -578,7 +578,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_rt5682_init, .dpcm_playback = 1, .stop_dma_first = 1, @@ -589,7 +589,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-rt5682-cap", .stream_name = "Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, @@ -599,7 +599,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_playback = 1, .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, @@ -610,7 +610,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic0", .stream_name = "DMIC0 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, @@ -621,7 +621,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .name = "dmic1", .stream_name = "DMIC1 Capture", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .dpcm_capture = 1, .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index d6ba94677ac2..58b75cf399fc 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -111,7 +111,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-play", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = cz_init, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware1, codec, platform), @@ -120,7 +120,7 @@ static struct snd_soc_dai_link cz_dai_rt5650[] = { .name = "amd-rt5645-cap", .stream_name = "RT5645_AIF1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &cz_aif1_ops, SND_SOC_DAILINK_REG(designware2, codec, platform), }, diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e561464f7d60..dad70436d063 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -51,7 +51,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) /* set rt5682 dai fmt */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); + | SND_SOC_DAIFMT_CBP_CFP); if (ret < 0) { dev_err(rtd->card->dev, "Failed to set rt5682 dai fmt: %d\n", ret); @@ -302,7 +302,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .init = acp3x_5682_init, .dpcm_playback = 1, .dpcm_capture = 1, @@ -313,7 +313,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .dpcm_playback = 1, .ops = &acp3x_max_play_ops, .cpus = acp3x_bt, @@ -325,7 +325,7 @@ static struct snd_soc_dai_link acp3x_dai[] = { .name = "acp3x-ec-dmic0-capture", .stream_name = "Capture DMIC0", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, + | SND_SOC_DAIFMT_CBC_CFC, .dpcm_capture = 1, .ops = &acp3x_ec_cap0_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), -- 2.20.1

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Logitech Z10 USB speakers need a volume change before audio works
by Hans de Goede 07 Dec '21

07 Dec '21

Hi, I've a set of Logitech Z10 USB speakers, which act as a USB soundcard. They have this weird glitch where after turning off my PC (and their power-supply as well) and then turning things back on, they are silent until I change the PCM volume control for the speakers inside alsa-mixer. It seems like they need some "set-volume" command to be send over the USB bus to unmute them when initially powered-up / turned on. Is their some existing usb-audio quirk which I can try to work around this? Regards, Hans

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[PATCH 0/2] ASoC: mediatek: support memory-region assignment
by Trevor Wu 06 Dec '21

06 Dec '21

This series of patches adds support for memory-region assignment, so the access region of DMA engine could be restricted. Patches are based on broonie tree "for-next" branch. Trevor Wu (2): ASoC: mediatek: mt8195: support reserved memory assignment dt-bindings: mediatek: mt8195: add memory-region property .../devicetree/bindings/sound/mt8195-afe-pcm.yaml | 8 ++++++++ sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 7 +++++++ 2 files changed, 15 insertions(+) -- 2.18.0

2 3

[PATCH v2 0/4] ASoC: mediatek: Update MT8195 machine driver
by Trevor Wu 06 Dec '21

06 Dec '21

This series of patches adds support for RT5682s headset codec in mt8195 machine drivers, and SOF support on card mt8195-mt6359-rt1019-rt5682 is also included. Patches are based on broonie tree "for-next" branch. Changes since v1: - remove patch3 and patch4 in v1 - add SOF support on card mt8195-mt6359-rt1012-rt5682 - add new propertes to dt-bindings for mt8195-mt6359-rt1019-rt5682 Trevor Wu (4): ASoC: mediatek: mt8195: add headset codec rt5682s support dt-bindings: mediatek: mt8195: add model property ASoC: mediatek: mt8195: add sof support on mt8195-mt6359-rt1019-rt5682 dt-bindings: mediatek: mt8195: add adsp and dai-link property .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 4 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 14 + sound/soc/mediatek/Kconfig | 2 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 29 +- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 347 +++++++++++++++++- 5 files changed, 370 insertions(+), 26 deletions(-) -- 2.18.0

2 5

[RFC patch 0/5] Support BCLK input clock in tlv320aic31xx
by Ariel D'Alessandro 03 Dec '21

03 Dec '21

The tlv320aic31xx codec allows using BCLK as the input clock for PLL, deriving all the frequencies through a set of divisors. In this case, codec sysclk is determined by the hwparams sample rate/format. So its frequency must be updated from the codec itself when these are changed. This patchset modifies the tlv320aic31xx driver to update its sysclk if BCLK is used as the input clock. This allows to be used by the generic fsl-asoc-card, without having to add a specific driver. Ariel D'Alessandro (5): ASoC: tlv320aic31xx: Fix typo in BCLK clock name ASoC: tlv320aic31xx: Add support for pll_r coefficient ASoC: tlv320aic31xx: Add divs for bclk as clk_in ASoC: tlv320aic31xx: Handle BCLK set as PLL input configuration ASoC: fsl-asoc-card: Support fsl,imx-audio-tlv320aic31xx codec sound/soc/codecs/tlv320aic31xx.c | 105 ++++++++++++++++++++----------- sound/soc/codecs/tlv320aic31xx.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 12 ++++ 3 files changed, 83 insertions(+), 36 deletions(-) -- 2.30.2

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[PATCH v6 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 03 Dec '21

03 Dec '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 171 ++++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 299 +++++++++++++++++++++ 4 files changed, 484 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH] Sound: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard
by William Overton 02 Dec '21

02 Dec '21

The kernel already has support for very similar Pioneer djm products and this work is based on that. Added device to quirks-table.h and added control info to mixer_quirks.c. Tested on my hardware and all working. Signed-off-by: William Overton <willovertonuk(a)gmail.com> --- sound/usb/mixer_quirks.c | 34 +++++++++++++++++++++++ sound/usb/quirks-table.h | 58 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 46082dc57be0..d489c1de3bae 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2795,6 +2795,7 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_750_IDX 0x1 #define SND_DJM_850_IDX 0x2 #define SND_DJM_900NXS2_IDX 0x3 +#define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -2984,10 +2985,40 @@ static const struct snd_djm_ctl snd_djm_ctls_900nxs2[] = { SND_DJM_CTL("Ch5 Input", 900nxs2_cap5, 3, SND_DJM_WINDEX_CAP) }; +// DJM-750MK2 +static const u16 snd_djm_opts_750mk2_cap1[] = { + 0x0100, 0x0102, 0x0103, 0x0106, 0x0107, 0x0108, 0x0109, 0x010a }; +static const u16 snd_djm_opts_750mk2_cap2[] = { + 0x0200, 0x0202, 0x0203, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a }; +static const u16 snd_djm_opts_750mk2_cap3[] = { + 0x0300, 0x0302, 0x0303, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a }; +static const u16 snd_djm_opts_750mk2_cap4[] = { + 0x0400, 0x0402, 0x0403, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a }; +static const u16 snd_djm_opts_750mk2_cap5[] = { + 0x0507, 0x0508, 0x0509, 0x050a, 0x0511, 0x0512, 0x0513, 0x0514 }; + +static const u16 snd_djm_opts_750mk2_pb1[] = { 0x0100, 0x0101, 0x0104 }; +static const u16 snd_djm_opts_750mk2_pb2[] = { 0x0200, 0x0201, 0x0204 }; +static const u16 snd_djm_opts_750mk2_pb3[] = { 0x0300, 0x0301, 0x0304 }; + + +static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Ch1 Input", 750mk2_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", 750mk2_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", 750mk2_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", 750mk2_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch5 Input", 750mk2_cap5, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Output", 750mk2_pb1, 0, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch2 Output", 750mk2_pb2, 1, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch3 Output", 750mk2_pb3, 2, SND_DJM_WINDEX_PB) +}; + static const struct snd_djm_device snd_djm_devices[] = { SND_DJM_DEVICE(250mk2), SND_DJM_DEVICE(750), + SND_DJM_DEVICE(750mk2), SND_DJM_DEVICE(850), SND_DJM_DEVICE(900nxs2) }; @@ -3235,6 +3266,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ err = snd_djm_controls_create(mixer, SND_DJM_750_IDX); break; + case USB_ID(0x2b73, 0x001b): /* Pioneer DJ DJM-750MK2 */ + err = snd_djm_controls_create(mixer, SND_DJM_750MK2_IDX); + break; case USB_ID(0x08e4, 0x0163): /* Pioneer DJ DJM-850 */ err = snd_djm_controls_create(mixer, SND_DJM_850_IDX); break; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e03043f7dad3..bc0116273e94 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3850,6 +3850,64 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Pioneer DJ DJM-750MK2 + * 10 channels playback & 12 channels capture @ 48kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x001b), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-850 -- 2.30.2

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[PATCH v6 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 02 Dec '21

02 Dec '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 ++++++++++ sound/soc/qcom/lpass-cpu.c | 244 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 578 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 434 ++++++++++++++++ sound/soc/qcom/lpass.h | 151 ++++++ 9 files changed, 1784 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v2 00/20] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 02 Dec '21

02 Dec '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. I based S/PDIF patches on Arnd's Bergmann patch from a separate series [1] that removes obsolete slave_id. This eases merging of the patches by removing the merge conflict. [1] https://patchwork.ozlabs.org/project/linux-tegra/list/?submitter=80402 Changelog: v1: - Corrected I2S yaml problem that was reported by the DT bot for v1 by removing the non-existent required clock-names property. - Removed assigned-clocks property from S/PDIF yaml since this property is now inherited from the clocks property. - Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making it the first sound/soc patch in the series, like it was suggested by Mark Brown in the comment to v1. Also reworded commit message of this patch to *not* make it looks like it should be backported to stable kernels. Arnd Bergmann (1): ASoC: tegra20-spdif: stop setting slave_id Dmitry Osipenko (19): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property dt-bindings: host1x: Document optional HDMI sound-dai-cells ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates drm/tegra: hdmi: Unwind tegra_hdmi_init() errors drm/tegra: hdmi: Register audio CODEC on Tegra20 ARM: tegra_defconfig: Enable S/PDIF driver ARM: tegra: Add S/PDIF node to Tegra20 device-tree ARM: tegra: Add HDMI audio graph to Tegra20 device-tree ARM: tegra: acer-a500: Enable S/PDIF and HDMI audio ARM: tegra: paz00: Enable S/PDIF and HDMI audio .../display/tegra/nvidia,tegra20-host1x.txt | 1 + .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 77 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 85 ++++++++ .../boot/dts/tegra20-acer-a500-picasso.dts | 8 + arch/arm/boot/dts/tegra20-paz00.dts | 8 + arch/arm/boot/dts/tegra20.dtsi | 40 +++- arch/arm/configs/tegra_defconfig | 1 + drivers/gpu/drm/tegra/Kconfig | 3 + drivers/gpu/drm/tegra/hdmi.c | 168 +++++++++++++-- sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 195 +++++++++++++----- sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 15 files changed, 571 insertions(+), 102 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

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[PATCH v2 0/4] ARM: dts: stm32: merge spi and i2s nodes
by Olivier Moysan 02 Dec '21

02 Dec '21

When a STM32 SPI instance offers I2S feature, two nodes are defined in STM32 SoC device tree to support both SPI and I2S. Merge SPI node and I2S nodes into a single node, to avoid hardware description duplication and compilation warnings. Changes in v2: - Squash some patches in the serie to avoid dtbs check warnings on intermediate compilation steps. Olivier Moysan (4): ASoC: dt-bindings: stm32: i2s: add audio-graph-card port ASoC: dt-bindings: stm32: i2s: allow additional properties. ASoC: dt-bindings: stm32: i2s: update example ARM: dts: stm32: merge spi and i2s nodes .../bindings/sound/st,stm32-i2s.yaml | 9 ++++- arch/arm/boot/dts/stm32mp151.dtsi | 39 ++----------------- arch/arm/boot/dts/stm32mp157c-ev1.dts | 2 +- .../arm/boot/dts/stm32mp15xx-dhcom-drc02.dtsi | 2 +- .../boot/dts/stm32mp15xx-dhcor-avenger96.dtsi | 2 +- arch/arm/boot/dts/stm32mp15xx-dkx.dtsi | 7 +++- 6 files changed, 18 insertions(+), 43 deletions(-) -- 2.17.1

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