December 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH V6 0/5] ASoC: codecs: Add Awinic AW883XX audio amplifier driver
by wangweidong.a＠awinic.com 20 Dec '22

20 Dec '22

From: Weidong Wang <wangweidong.a(a)awinic.com> The Awinic AW883XX is an I2S/TDM input, high efficiency digital Smart K audio amplifier with an integrated 10.25V smart boost convert Add a DT schema for describing Awinic AW883xx audio amplifiers. They are controlled using I2C. v5 -> v6: Modify the return value of Kcontrol Delete error prints in Kcontrol Delete the aw883xx_dynamic_create_controls function Release firmware immediately after memcpy() Change the names of the Widget's input and output Delete the alloc and copy for aw_wiget Delete all debug nodes and use regmap's debugfs Use be32_to_cpup instead of custom macros Change inefficient writing Verify the size of the bin file Changing the function name Register directly with the snd_soc_register_component function Delete the aw_dev_reg_dump function Weidong Wang (5): ASoC: codecs: Add i2c and codec registration for aw883xx and their associated operation functions ASoC: codecs: Implementation of aw883xx configuration file parsing function ASoC: codecs: aw883xx chip control logic, such as power on and off ASoC: codecs: Configure aw883xx chip register as well as Kconfig and Makefile ASoC: dt-bindings: Add schema for "awinic,aw883xx" .../bindings/sound/awinic,aw883xx.yaml | 49 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/aw883xx/aw883xx.c | 909 +++++++ sound/soc/codecs/aw883xx/aw883xx.h | 81 + sound/soc/codecs/aw883xx/aw883xx_bin_parse.c | 1324 ++++++++++ sound/soc/codecs/aw883xx/aw883xx_bin_parse.h | 149 ++ sound/soc/codecs/aw883xx/aw883xx_data_type.h | 146 ++ sound/soc/codecs/aw883xx/aw883xx_device.c | 1613 ++++++++++++ sound/soc/codecs/aw883xx/aw883xx_device.h | 537 ++++ sound/soc/codecs/aw883xx/aw883xx_init.c | 615 +++++ .../soc/codecs/aw883xx/aw883xx_pid_2049_reg.h | 2300 +++++++++++++++++ 12 files changed, 7740 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw883xx.yaml create mode 100644 sound/soc/codecs/aw883xx/aw883xx.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_bin_parse.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_bin_parse.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_data_type.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_device.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_device.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_init.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_pid_2049_reg.h base-commit: 479174d402bcf60789106eedc4def3957c060bad -- 2.38.1

4 16

[PATCH] pcm: hw: Do not return error unless there is one
by Jonas Holmberg 20 Dec '22

20 Dec '22

Add a line that was removed by mistake in commit 7f2d6c3aac3505ceee4b0d3e8b3ca423ce29b070. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_hw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 0588ce5e..daa3e1ff 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -356,7 +356,7 @@ static int snd_pcm_hw_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) if (hw->rates.min > 0) { err = _snd_pcm_hw_param_set_minmax(params, SND_PCM_HW_PARAM_RATE, hw->rates.min, 0, hw->rates.max + 1, -1); - + if (err < 0) return err; } -- 2.37.4

1 1

[PATCH 1/2] ASoC: fsl_sai: Factor out SAI reset
by Marek Vasut 19 Dec '22

19 Dec '22

The driver performs SAI reset in three places, factor the reset handling into single function to avoid duplication. No functional change. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> To: alsa-devel(a)alsa-project.org --- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++------------------ 1 file changed, 20 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 1c9be8a5dcb13..ade5a6b44b66a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -704,6 +704,21 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, return 0; } +static void fsl_sai_reset(struct fsl_sai *sai, bool rx, bool tx) +{ + unsigned int ofs = sai->soc_data->reg_offset; + + if (tx) + regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); + if (rx) + regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); + usleep_range(1000, 2000); + if (tx) + regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0); + if (rx) + regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); +} + static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) { unsigned int ofs = sai->soc_data->reg_offset; @@ -729,12 +744,8 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) * This is a hardware bug, and will be fix in the * next sai version. */ - if (!sai->is_consumer_mode) { - /* Software Reset */ - regmap_write(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_SR); - /* Clear SR bit to finish the reset */ - regmap_write(sai->regmap, FSL_SAI_xCSR(tx, ofs), 0); - } + if (!sai->is_consumer_mode) + fsl_sai_reset(sai, dir == RX, dir == TX); } static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -863,12 +874,7 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); unsigned int ofs = sai->soc_data->reg_offset; - /* Software Reset for both Tx and Rx */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); - /* Clear SR bit to finish the reset */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); + fsl_sai_reset(sai, true, true); regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs), FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), @@ -1632,7 +1638,6 @@ static int fsl_sai_runtime_suspend(struct device *dev) static int fsl_sai_runtime_resume(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); - unsigned int ofs = sai->soc_data->reg_offset; int ret; ret = clk_prepare_enable(sai->bus_clk); @@ -1658,11 +1663,8 @@ static int fsl_sai_runtime_resume(struct device *dev) regcache_cache_only(sai->regmap, false); regcache_mark_dirty(sai->regmap); - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); - usleep_range(1000, 2000); - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); + + fsl_sai_reset(sai, true, true); ret = regcache_sync(sai->regmap); if (ret) -- 2.35.1

3 5

RE: [PATCH 2/2] ASoC: da7213: Add support for mono, set frame width to 32 when possible
by David.Rau.opensource 19 Dec '22

19 Dec '22

Is there anything I should modify or correct to make this commit move forward? David Rau Renesas -----Original Message----- From: Piotr Wojtaszczyk <piotr.wojtaszczyk(a)timesys.com> Sent: Thursday, December 15, 2022 00:25 To: Mark Brown <broonie(a)kernel.org> Cc: David.Rau.opensource <David.Rau.opensource(a)dm.renesas.com>; tiwai(a)suse.com; perex(a)perex.cz; support.opensource(a)diasemi.com; lgirdwood(a)gmail.com; alsa-devel(a)alsa-project.org; linux-kernel(a)vger.kernel.org Subject: Re: [PATCH 2/2] ASoC: da7213: Add support for mono, set frame width to 32 when possible On Wed, Dec 14, 2022 at 3:56 PM Mark Brown <broonie(a)kernel.org> wrote: > > On Wed, Dec 14, 2022 at 03:31:26PM +0100, Piotr Wojtaszczyk wrote: > > It was a single patch, with no deps. > > > > It's flagged as patch 2/2? That's wrong tag, initially I submitted this as a single patch to `support.opensource(a)diasemi.com` and now it comes on LKML after testing by David Rau. -- Piotr Wojtaszczyk Timesys

2 1

[PATCH 00/11] ASoC: dt-bindings: audio-graph-port related update
by Kuninori Morimoto 19 Dec '22

19 Dec '22

Hi ASoC ML, DT ML Cc Geert, Nvidia member These patches fixups audio-graph-port, and its related DT schema. Audio-Graph-Card and Simple-Audio-Card are similar Card and are sharing same utils. Thus we can also sharing same schema. Renesas and Nvidia are main user of Audio-Graph-Card. This patch-set fixup some Renesas's "make dtbs_check". Geert Uytterhoeven (1): ASoC: dt-bindings: ti,pcm3168a: Convert to json-schema Kuninori Morimoto (10): ASoC: dt-bindings: audio-graph-port: use definitions for port/endpoint ASoC: dt-bindings: audio-graph-port: add definitions/ports ASoC: dt-bindings: audio-graph-port: add missing mclk-fs ASoC: dt-bindings: audio-graph-port: add clocks on endpoint ASoC: dt-bindings: audio-graph-port: remove prefix ASoC: dt-bindings: nvidia: use audio-graph-port.yaml ports ASoC: dt-bindings: ak4613: enable Of-graph (Audio-Graph-Card) style ASoC: dt-bindings: renesas,rsnd: add missing playback/capture ASoC: dt-bindings: renesas,rsnd: #sound-dai-cells is not mandatory ASoC: dt-bindings: simple-card: add missing #address-cells/#size-cells .../devicetree/bindings/sound/ak4613.yaml | 7 ++ .../bindings/sound/audio-graph-port.yaml | 65 ++++++++---- .../bindings/sound/nvidia,tegra186-asrc.yaml | 14 +-- .../bindings/sound/nvidia,tegra186-dspk.yaml | 21 ++-- .../sound/nvidia,tegra210-admaif.yaml | 7 +- .../bindings/sound/nvidia,tegra210-adx.yaml | 14 +-- .../bindings/sound/nvidia,tegra210-ahub.yaml | 7 +- .../bindings/sound/nvidia,tegra210-amx.yaml | 16 +-- .../bindings/sound/nvidia,tegra210-dmic.yaml | 21 ++-- .../bindings/sound/nvidia,tegra210-i2s.yaml | 21 ++-- .../bindings/sound/nvidia,tegra210-mixer.yaml | 14 +-- .../bindings/sound/nvidia,tegra210-mvc.yaml | 21 ++-- .../bindings/sound/nvidia,tegra210-ope.yaml | 21 ++-- .../bindings/sound/nvidia,tegra210-sfc.yaml | 21 ++-- .../bindings/sound/renesas,rsnd.yaml | 29 ++++- .../bindings/sound/simple-card.yaml | 4 + .../devicetree/bindings/sound/ti,pcm3168a.txt | 56 ---------- .../bindings/sound/ti,pcm3168a.yaml | 100 ++++++++++++++++++ 18 files changed, 228 insertions(+), 231 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -- 2.25.1

5 26

Add rate limitation in a SectionDevice
by GitHub issues - edited 19 Dec '22

19 Dec '22

alsa-project/alsa-ucm-conf issue #253 was edited from fchiby: Hello @perexg, I created an alsa-ucm configuration to my board i350-evk (mt8365-evk). Here my configuration files: [mt8365-evk.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/10… [HiFi.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/10247429… However pulseaudio initialization failed because of hw:mt8365evk,2, this pcm device is connected to jack_mic and Amic. ``` D: [pulseaudio] alsa-util.c: _buffer_size:4408 , _period_size:1102 I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) D: [pulseaudio] alsa-util.c: _buffer_size:4408 , _period_size:1102 I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) D: [pulseaudio] alsa-util.c: _buffer_size:4408 , _period_size:1102 I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) D: [pulseaudio] alsa-util.c: Set neither period nor buffer size. I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying plug:SLAVE='_ucm0006.hw:mt8365evk,2' with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] pcm.c: Unknown PCM _ucm0006.hw:mt8365evk,2 I: [pulseaudio] alsa-util.c: Error opening PCM device plug:SLAVE='_ucm0006.hw:mt8365evk,2': No such file or directory D: [pulseaudio] alsa-mixer.c: Profile set 0xaaab0a13a700, auto_profiles=no, probed=yes, n_mappings=0, n_profiles=0, n_decibel_fixes=0 E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="platform-sound" card_name="alsa_card.platform-sound" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` Whith dmesg I got more details about pulseaudio failure: ``` [ 160.295717] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_fe_hw_params AWB period = 1102 rate = 44100 channels = 1 [ 160.295865] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_int_adda_prepare 'Capture' rate = 44100 [ 160.295881] mt8365-afe-pcm 11220000.audio-controller: ASoC: error at snd_soc_pcm_dai_prepare on INT ADDA: -22 [ 160.295891] MTK Codec: ASoC: soc_pcm_prepare() failed (-22) [ 160.295900] AWB_FE: ASoC: dpcm_be_dai_prepare() failed (-22) [ 160.295908] AWB_FE: ASoC: dpcm_fe_dai_prepare() failed (-22) [ 160.296395] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_fe_hw_params AWB period = 512 rate = 44100 channels = 1 [ 160.296543] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_int_adda_prepare 'Capture' rate = 44100 [ 160.296559] mt8365-afe-pcm 11220000.audio-controller: ASoC: error at snd_soc_pcm_dai_prepare on INT ADDA: -22 [ 160.296569] MTK Codec: ASoC: soc_pcm_prepare() failed (-22) [ 160.296577] AWB_FE: ASoC: dpcm_be_dai_prepare() failed (-22) [ 160.296585] AWB_FE: ASoC: dpcm_fe_dai_prepare() failed (-22) ``` When I checked the machine driver, hw:mt8365evk,2 FE (AWB) has 8000_192000, while hw:mt8365evk,2 BE (INT_ADDA) has only [8000, 16000, 32000, 48000] as sample rates. Actually, the rate limitation is not on afe dai, it's on PMIC. So we prefer to not restrict hw:mt8365evk,2 FE AWB rates. My question is, **is it possible to specify the rate limitation in a** `SectionDevice` ? I checked the alsa-ucm-conf documentation and did not find any section for this :/ [alsa-ucm-conf docs](https://www.alsa-project.org/alsa-doc/alsa-lib/group__ucm__conf.htmll) Thanks a bunch ! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/253 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add rate limitation in a SectionDevice
by GitHub issues - opened 19 Dec '22

19 Dec '22

alsa-project/alsa-ucm-conf issue #253 was opened from fchiby: Hello, I created an alsa-ucm configuration to my board i350-evk (mt8365-evk). Here my configuration files: [mt8365-evk.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/10… [HiFi.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/10247429… However pulseaudio initialization failed because of hw:mt8365evk,2, this pcm device is connected to jack_mic and Amic. ``` D: [pulseaudio] alsa-util.c: _buffer_size:4408 , _period_size:1102 I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) D: [pulseaudio] alsa-util.c: _buffer_size:4408 , _period_size:1102 I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) D: [pulseaudio] alsa-util.c: _buffer_size:4408 , _period_size:1102 I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) D: [pulseaudio] alsa-util.c: Set neither period nor buffer size. I: [pulseaudio] pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying plug:SLAVE='_ucm0006.hw:mt8365evk,2' with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] pcm.c: Unknown PCM _ucm0006.hw:mt8365evk,2 I: [pulseaudio] alsa-util.c: Error opening PCM device plug:SLAVE='_ucm0006.hw:mt8365evk,2': No such file or directory D: [pulseaudio] alsa-mixer.c: Profile set 0xaaab0a13a700, auto_profiles=no, probed=yes, n_mappings=0, n_profiles=0, n_decibel_fixes=0 E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="platform-sound" card_name="alsa_card.platform-sound" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` Whith dmesg I got more details about pulseaudio failure: ``` [ 160.295717] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_fe_hw_params AWB period = 1102 rate = 44100 channels = 1 [ 160.295865] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_int_adda_prepare 'Capture' rate = 44100 [ 160.295881] mt8365-afe-pcm 11220000.audio-controller: ASoC: error at snd_soc_pcm_dai_prepare on INT ADDA: -22 [ 160.295891] MTK Codec: ASoC: soc_pcm_prepare() failed (-22) [ 160.295900] AWB_FE: ASoC: dpcm_be_dai_prepare() failed (-22) [ 160.295908] AWB_FE: ASoC: dpcm_fe_dai_prepare() failed (-22) [ 160.296395] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_fe_hw_params AWB period = 512 rate = 44100 channels = 1 [ 160.296543] mt8365-afe-pcm 11220000.audio-controller: mt8365_afe_int_adda_prepare 'Capture' rate = 44100 [ 160.296559] mt8365-afe-pcm 11220000.audio-controller: ASoC: error at snd_soc_pcm_dai_prepare on INT ADDA: -22 [ 160.296569] MTK Codec: ASoC: soc_pcm_prepare() failed (-22) [ 160.296577] AWB_FE: ASoC: dpcm_be_dai_prepare() failed (-22) [ 160.296585] AWB_FE: ASoC: dpcm_fe_dai_prepare() failed (-22) ``` When I checked the machine driver, hw:mt8365evk,2 FE (AWB) has 8000_192000, while hw:mt8365evk,2 BE (INT_ADDA) has only [8000, 16000, 32000, 48000] as sample rates. Actually, the rate limitation is not on afe dai, it's on PMIC. So we prefer to not restrict hw:mt8365evk,2 FE AWB rates. My question is, **is it possible to specify the rate limitation in a** `SectionDevice` ? I checked the alsa-ucm-conf documentation and did not find any section for this :/ [alsa-ucm-conf docs](https://www.alsa-project.org/alsa-doc/alsa-lib/group__ucm__conf.htmll) Thanks a bunch ! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/253 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 00/12] ASoC: mediatek: Add support for MT8188 SoC
by Trevor Wu 19 Dec '22

19 Dec '22

This series of patches adds support for Mediatek AFE of MT8188 SoC. Patches are based on broonie tree "for-next" branch. Changes since v2: - drop CLK_IGNORE_UNUSED flag - include bitfield.h to reslove the issue reported by kernel test robot - rename mt8188-afe-pcm.yaml to mt8188-afe.yaml - refine dt-binding files based on reviewer's suggestions Changes since v1: - remove bus protection functions in case of unmerged dependency problem - replace some bit operation macro with FIELD_PREP - simplify register control by regmap_set_bits and regmap_clear_bits - fix dt-binding errors - rename compatible string for recognition Trevor Wu (12): ASoC: mediatek: common: add SMC ops and SMC CMD ASoC: mediatek: mt8188: add common header ASoC: mediatek: mt8188: support audsys clock ASoC: mediatek: mt8188: support adda in platform driver ASoC: mediatek: mt8188: support etdm in platform driver ASoC: mediatek: mt8188: support pcmif in platform driver ASoC: mediatek: mt8188: support audio clock control ASoC: mediatek: mt8188: add platform driver ASoC: mediatek: mt8188: add control for timing select dt-bindings: mediatek: mt8188: add audio afe document ASoC: mediatek: mt8188: add machine driver with mt6359 dt-bindings: mediatek: mt8188: add mt8188-mt6359 document .../devicetree/bindings/sound/mt8188-afe.yaml | 196 + .../bindings/sound/mt8188-mt6359.yaml | 60 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-base-afe.h | 19 + sound/soc/mediatek/mt8188/Makefile | 15 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 656 ++++ sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 114 + sound/soc/mediatek/mt8188/mt8188-afe-common.h | 151 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 3359 +++++++++++++++++ sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 205 + sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 15 + .../soc/mediatek/mt8188/mt8188-audsys-clkid.h | 83 + sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 632 ++++ sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 2591 +++++++++++++ sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 367 ++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 808 ++++ sound/soc/mediatek/mt8188/mt8188-reg.h | 3180 ++++++++++++++++ 18 files changed, 12475 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8188-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8188-mt6359.yaml create mode 100644 sound/soc/mediatek/mt8188/Makefile create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-common.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-adda.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-mt6359.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-reg.h -- 2.18.0

5 32

[PATCH AUTOSEL 4.14 03/23] ASoC: codecs: rt298: Add quirk for KBL-R RVP platform
by Sasha Levin 18 Dec '22

18 Dec '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 953dbd1cef18ce9ac0d69c1bd735b929fe52a17e ] KBL-R RVP platforms also use combojack, so we need to enable that configuration for them. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20221010121955.718168-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt298.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index ce963768449f..041955eb7922 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1170,6 +1170,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Geminilake") } }, + { + .ident = "Intel Kabylake R RVP", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") + } + }, { } }; -- 2.35.1

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[PATCH AUTOSEL 4.19 03/26] ASoC: codecs: rt298: Add quirk for KBL-R RVP platform
by Sasha Levin 18 Dec '22

18 Dec '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 953dbd1cef18ce9ac0d69c1bd735b929fe52a17e ] KBL-R RVP platforms also use combojack, so we need to enable that configuration for them. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20221010121955.718168-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt298.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 06cdba4edfe2..3181b91a025b 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1169,6 +1169,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Geminilake") } }, + { + .ident = "Intel Kabylake R RVP", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") + } + }, { } }; -- 2.35.1

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