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

[PATCH] pcm_rate: fix hw_ptr boundary wrapping issue
by Alan Young 16 Feb '23

16 Feb '23

Wrap the hw_ptr using the total position of the slave hw_ptr, including boundary wraps. Otherwise, small errors can creep in due to residuals (when boundary is not a multiple of period size) and which can accumulate. Signed-off-by: Alan Young <consult.awy(a)gmail.com> Fixes: 7570e5d7 ("pcm: rate: fix the hw_ptr update until the boundary available") --- src/pcm/pcm_rate.c | 54 ++++++++++++++++++++++++++++++---------------- 1 file changed, 36 insertions(+), 18 deletions(-) diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index c8076859..a29fc5a9 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -52,6 +52,7 @@ struct _snd_pcm_rate { snd_pcm_generic_t gen; snd_pcm_uframes_t appl_ptr, hw_ptr, last_slave_hw_ptr; snd_pcm_uframes_t last_commit_ptr; + u_int64_t slave_hw_ptr_wrap; snd_pcm_uframes_t orig_avail_min; snd_pcm_sw_params_t sw_params; snd_pcm_format_t sformat; @@ -562,6 +563,8 @@ static int snd_pcm_rate_init(snd_pcm_t *pcm) if (rate->ops.reset) rate->ops.reset(rate->obj); rate->last_commit_ptr = 0; + rate->last_slave_hw_ptr = 0; + rate->slave_hw_ptr_wrap = 0; rate->start_pending = 0; return 0; } @@ -638,33 +641,48 @@ snd_pcm_rate_read_areas1(snd_pcm_t *pcm, static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr) { snd_pcm_rate_t *rate; - snd_pcm_sframes_t slave_hw_ptr_diff; - snd_pcm_sframes_t last_slave_hw_ptr_frac; + snd_pcm_uframes_t new_hw_ptr; + snd_pcm_uframes_t slave_residual; if (pcm->stream != SND_PCM_STREAM_PLAYBACK) return; rate = pcm->private_data; - slave_hw_ptr_diff = pcm_frame_diff(slave_hw_ptr, rate->last_slave_hw_ptr, rate->gen.slave->boundary); - if (slave_hw_ptr_diff == 0) + if (slave_hw_ptr == rate->last_slave_hw_ptr) return; - last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; - /* While handling fraction part fo slave period, rounded value will be - * introduced by input_frames(). - * To eliminate rounding issue on rate->hw_ptr, subtract last rounded - * value from rate->hw_ptr and add new rounded value of present - * slave_hw_ptr fraction part to rate->hw_ptr. Hence, - * rate->hw_ptr += [ (no. of updated slave periods * pcm rate period size) - - * fractional part of last_slave_hw_ptr rounded value + - * fractional part of updated slave hw ptr's rounded value ] + + /* + * Our hw_ptr may wrap at a different time to that of the slave as the + * number of period / boundary may differ. It is essential that our hw_ptr + * wraps at the correct point, along with our appl_ptr. Otherwise the + * avail calculation will be incorrect. + * + * Track the number of slave hw_ptr boundary wraps and use that to calculate + * our new_hw_ptr from the total number of samples converted. */ - rate->hw_ptr += ( - (((last_slave_hw_ptr_frac + slave_hw_ptr_diff) / rate->gen.slave->period_size) * pcm->period_size) - - rate->ops.input_frames(rate->obj, last_slave_hw_ptr_frac) + - rate->ops.input_frames(rate->obj, (last_slave_hw_ptr_frac + slave_hw_ptr_diff) % rate->gen.slave->period_size)); + if (slave_hw_ptr < rate->last_slave_hw_ptr) { + rate->slave_hw_ptr_wrap += rate->gen.slave->boundary; + } rate->last_slave_hw_ptr = slave_hw_ptr; - rate->hw_ptr %= pcm->boundary; + /* + * Calculate rate->hw_ptr using total number of frames consumed by slave hw_ptr, + * including any boundary wraps. + */ + if (rate->slave_hw_ptr_wrap) { + /* + * Restrict explicit 64-bit calculations to case where rate->slave_hw_ptr_wrap + * is non-zero, which will only happen in 32-bit environments. + */ + u_int64_t wrapped_slave_hw_ptr = slave_hw_ptr + rate->slave_hw_ptr_wrap; + new_hw_ptr = ((wrapped_slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size) % pcm->boundary; + slave_residual = wrapped_slave_hw_ptr % rate->gen.slave->period_size; + } else { + new_hw_ptr = (slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size; + slave_residual = slave_hw_ptr % rate->gen.slave->period_size; + } + new_hw_ptr += rate->ops.input_frames(rate->obj, slave_residual); + rate->hw_ptr = new_hw_ptr % pcm->boundary; } static inline void snd_pcm_rate_sync_hwptr(snd_pcm_t *pcm) -- 2.25.1

2 3

[PATCH] ASoC: rsnd: adg: Fix BRG typos
by Geert Uytterhoeven 16 Feb '23

16 Feb '23

"BRG" stands for "Baud Rate Generator", but is frequently misspelled as "RBG". Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/rcar/adg.c | 64 ++++++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 32 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 7bc4421835d7261f..0b8926600d900a68 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -39,10 +39,10 @@ struct rsnd_adg { int clkin_size; int clkout_size; u32 ckr; - u32 rbga; - u32 rbgb; + u32 brga; + u32 brgb; - int rbg_rate[ADG_HZ_SIZE]; /* RBGA / RBGB */ + int brg_rate[ADG_HZ_SIZE]; /* BRGA / BRGB */ }; #define for_each_rsnd_clkin(pos, adg, i) \ @@ -75,7 +75,7 @@ static const char * const clkout_name_gen2[] = { [CLKOUT3] = "audio_clkout3", }; -static u32 rsnd_adg_calculate_rbgx(unsigned long div) +static u32 rsnd_adg_calculate_brgx(unsigned long div) { int i; @@ -131,8 +131,8 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv, adg->clkin_rate[CLKA], /* 0000: CLKA */ adg->clkin_rate[CLKB], /* 0001: CLKB */ adg->clkin_rate[CLKC], /* 0010: CLKC */ - adg->rbg_rate[ADG_HZ_441], /* 0011: RBGA */ - adg->rbg_rate[ADG_HZ_48], /* 0100: RBGB */ + adg->brg_rate[ADG_HZ_441], /* 0011: BRGA */ + adg->brg_rate[ADG_HZ_48], /* 0100: BRGB */ }; min = ~0; @@ -323,10 +323,10 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) /* * find divided clock from BRGA/BRGB */ - if (rate == adg->rbg_rate[ADG_HZ_441]) + if (rate == adg->brg_rate[ADG_HZ_441]) return 0x10; - if (rate == adg->rbg_rate[ADG_HZ_48]) + if (rate == adg->brg_rate[ADG_HZ_48]) return 0x20; return -EIO; @@ -358,13 +358,13 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) ckr = 0x80000000; /* BRGB output = 48kHz */ rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr | ckr); - rsnd_mod_write(adg_mod, BRRA, adg->rbga); - rsnd_mod_write(adg_mod, BRRB, adg->rbgb); + rsnd_mod_write(adg_mod, BRRA, adg->brga); + rsnd_mod_write(adg_mod, BRRB, adg->brgb); dev_dbg(dev, "CLKOUT is based on BRG%c (= %dHz)\n", (ckr) ? 'B' : 'A', - (ckr) ? adg->rbg_rate[ADG_HZ_48] : - adg->rbg_rate[ADG_HZ_441]); + (ckr) ? adg->brg_rate[ADG_HZ_48] : + adg->brg_rate[ADG_HZ_441]); return 0; } @@ -484,7 +484,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; struct property *prop; - u32 ckr, rbgx, rbga, rbgb; + u32 ckr, brgx, brga, brgb; u32 rate, div; u32 req_rate[ADG_HZ_SIZE] = {}; uint32_t count = 0; @@ -501,8 +501,8 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) }; ckr = 0; - rbga = 2; /* default 1/6 */ - rbgb = 2; /* default 1/6 */ + brga = 2; /* default 1/6 */ + brgb = 2; /* default 1/6 */ /* * ADG supports BRRA/BRRB output only @@ -543,30 +543,30 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) if (0 == rate) /* not used */ continue; - /* RBGA */ - if (!adg->rbg_rate[ADG_HZ_441] && (0 == rate % 44100)) { + /* BRGA */ + if (!adg->brg_rate[ADG_HZ_441] && (0 == rate % 44100)) { div = 6; if (req_Hz[ADG_HZ_441]) div = rate / req_Hz[ADG_HZ_441]; - rbgx = rsnd_adg_calculate_rbgx(div); - if (BRRx_MASK(rbgx) == rbgx) { - rbga = rbgx; - adg->rbg_rate[ADG_HZ_441] = rate / div; + brgx = rsnd_adg_calculate_brgx(div); + if (BRRx_MASK(brgx) == brgx) { + brga = brgx; + adg->brg_rate[ADG_HZ_441] = rate / div; ckr |= brg_table[i] << 20; if (req_Hz[ADG_HZ_441]) parent_clk_name = __clk_get_name(clk); } } - /* RBGB */ - if (!adg->rbg_rate[ADG_HZ_48] && (0 == rate % 48000)) { + /* BRGB */ + if (!adg->brg_rate[ADG_HZ_48] && (0 == rate % 48000)) { div = 6; if (req_Hz[ADG_HZ_48]) div = rate / req_Hz[ADG_HZ_48]; - rbgx = rsnd_adg_calculate_rbgx(div); - if (BRRx_MASK(rbgx) == rbgx) { - rbgb = rbgx; - adg->rbg_rate[ADG_HZ_48] = rate / div; + brgx = rsnd_adg_calculate_brgx(div); + if (BRRx_MASK(brgx) == brgx) { + brgb = brgx; + adg->brg_rate[ADG_HZ_48] = rate / div; ckr |= brg_table[i] << 16; if (req_Hz[ADG_HZ_48]) parent_clk_name = __clk_get_name(clk); @@ -620,8 +620,8 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) rsnd_adg_get_clkout_end: adg->ckr = ckr; - adg->rbga = rbga; - adg->rbgb = rbgb; + adg->brga = brga; + adg->brgb = brgb; return 0; @@ -663,9 +663,9 @@ void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct seq_file *m) __clk_get_name(clk), clk, clk_get_rate(clk)); dbg_msg(dev, m, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", - adg->ckr, adg->rbga, adg->rbgb); - dbg_msg(dev, m, "BRGA (for 44100 base) = %d\n", adg->rbg_rate[ADG_HZ_441]); - dbg_msg(dev, m, "BRGB (for 48000 base) = %d\n", adg->rbg_rate[ADG_HZ_48]); + adg->ckr, adg->brga, adg->brgb); + dbg_msg(dev, m, "BRGA (for 44100 base) = %d\n", adg->brg_rate[ADG_HZ_441]); + dbg_msg(dev, m, "BRGB (for 48000 base) = %d\n", adg->brg_rate[ADG_HZ_48]); /* * Actual CLKOUT will be exchanged in rsnd_adg_ssi_clk_try_start() -- 2.34.1

3 2

[PATCH 1/2] ALSA: pcm: generate default audio timestamp
by Alan Young 16 Feb '23

16 Feb '23

If there is no get_time_info() available or the requested audio timestamp type is AUDIO_TSTAMP_TYPE_DEFAULT then need to set audio_tstamp_report.actual_type = AUDIO_TSTAMP_TYPE_DEFAULT (in the else clause near the start of snd_pcm_update_hw_ptr0()) in order for update_audio_tstamp() to generate the audio timestamp from the number of frames. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Signed-off-by: consult.awy(a)gmail.com Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_lib.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8b6aeb8a78f7..943f5396fc60 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -289,8 +289,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */ if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT) snd_pcm_gettime(runtime, &curr_tstamp); - } else + } else { + runtime->audio_tstamp_report.actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; snd_pcm_gettime(runtime, &curr_tstamp); + } } if (pos == SNDRV_PCM_POS_XRUN) { -- 2.31.1

1 0

[PATCH 2/2] ALSA: Change calculation of audio time from frames
by Alan Young 16 Feb '23

16 Feb '23

Introduce snd_pcm_lib_frames_to_timespec64() to replace the calculation audio_nsecs = div_u64(audio_frames * 1000000000LL, runtime->rate); which overflows after a playback duration of 4.8 days @ 44100 HZ or 0.56 days @ 384000 Hz. The implementation of snd_pcm_lib_frames_to_timespec64() extends these limits, to 1.5 million years (@ 384000 Hz), or the 68 year limit of struct timespec on 32-bit systems. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Fixes: 4eeaaeaea1ce ("ALSA: core: add hooks for audio timestamps") Signed-off-by: consult.awy(a)gmail.com Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_lib.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 943f5396fc60..e0a6556d33ee 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -205,12 +205,19 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return 0; } +static inline void snd_pcm_lib_frames_to_timespec64(u64 frames, unsigned int rate, struct timespec64 *audio_tstamp) +{ + u32 remainder; + audio_tstamp->tv_sec = div_u64_rem(frames, rate, &remainder); + audio_tstamp->tv_nsec = div_u64(mul_u32_u32(remainder, NSEC_PER_SEC), rate); +} + static void update_audio_tstamp(struct snd_pcm_substream *substream, struct timespec64 *curr_tstamp, struct timespec64 *audio_tstamp) { struct snd_pcm_runtime *runtime = substream->runtime; - u64 audio_frames, audio_nsecs; + u64 audio_frames; struct timespec64 driver_tstamp; if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE) @@ -233,9 +240,7 @@ static void update_audio_tstamp(struct snd_pcm_substream *substream, else audio_frames += runtime->delay; } - audio_nsecs = div_u64(audio_frames * 1000000000LL, - runtime->rate); - *audio_tstamp = ns_to_timespec64(audio_nsecs); + snd_pcm_lib_frames_to_timespec64(audio_frames, runtime->rate, audio_tstamp); } if (runtime->status->audio_tstamp.tv_sec != audio_tstamp->tv_sec || -- 2.31.1

1 0

[PATCH 1/2] ALSA: pcm: generate default audio timestamp
by Alan Young 16 Feb '23

16 Feb '23

If there is no get_time_info() available or the requested audio timestamp type is AUDIO_TSTAMP_TYPE_DEFAULT then need to set audio_tstamp_report.actual_type = AUDIO_TSTAMP_TYPE_DEFAULT (in the else clause near the start of snd_pcm_update_hw_ptr0()) in order for update_audio_tstamp() to generate the audio timestamp from the number of frames. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Signed-off-by: consult.awy(a)gmail.com --- sound/core/pcm_lib.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8b6aeb8a78f7..943f5396fc60 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -289,8 +289,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */ if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT) snd_pcm_gettime(runtime, &curr_tstamp); - } else + } else { + runtime->audio_tstamp_report.actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; snd_pcm_gettime(runtime, &curr_tstamp); + } } if (pos == SNDRV_PCM_POS_XRUN) { -- 2.31.1

2 1

[PATCH 2/2] ALSA: Change calculation of audio time from frames
by Alan Young 16 Feb '23

16 Feb '23

Introduce snd_pcm_lib_frames_to_timespec64() to replace the calculation audio_nsecs = div_u64(audio_frames * 1000000000LL, runtime->rate); which overflows after a playback duration of 4.8 days @ 44100 HZ or 0.56 days @ 384000 Hz. The implementation of snd_pcm_lib_frames_to_timespec64() extends these limits, to 1.5 million years (@ 384000 Hz), or the 68 year limit of struct timespec on 32-bit systems. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Fixes: 4eeaaeaea1ce ("ALSA: core: add hooks for audio timestamps") Signed-off-by: consult.awy(a)gmail.com --- sound/core/pcm_lib.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 943f5396fc60..e0a6556d33ee 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -205,12 +205,19 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return 0; } +static inline void snd_pcm_lib_frames_to_timespec64(u64 frames, unsigned int rate, struct timespec64 *audio_tstamp) +{ + u32 remainder; + audio_tstamp->tv_sec = div_u64_rem(frames, rate, &remainder); + audio_tstamp->tv_nsec = div_u64(mul_u32_u32(remainder, NSEC_PER_SEC), rate); +} + static void update_audio_tstamp(struct snd_pcm_substream *substream, struct timespec64 *curr_tstamp, struct timespec64 *audio_tstamp) { struct snd_pcm_runtime *runtime = substream->runtime; - u64 audio_frames, audio_nsecs; + u64 audio_frames; struct timespec64 driver_tstamp; if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE) @@ -233,9 +240,7 @@ static void update_audio_tstamp(struct snd_pcm_substream *substream, else audio_frames += runtime->delay; } - audio_nsecs = div_u64(audio_frames * 1000000000LL, - runtime->rate); - *audio_tstamp = ns_to_timespec64(audio_nsecs); + snd_pcm_lib_frames_to_timespec64(audio_frames, runtime->rate, audio_tstamp); } if (runtime->status->audio_tstamp.tv_sec != audio_tstamp->tv_sec || -- 2.31.1

2 1

[GIT PULL] ASoC updates for v6.3
by Mark Brown 16 Feb '23

16 Feb '23

The following changes since commit c1619ea22d9509cfbcf5c2c1cc65563179e9dc8e: ASoC: SOF: sof-audio: Fixes for widget prepare and (2023-01-18 15:28:17 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.3 for you to fetch changes up to 76f5aaabce492aa6991c28c96bb78b00b05d06c5: ASoC: soc-ac97: Return correct error codes (2023-02-15 16:09:07 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.3 There's been quite a lot of activity this release, but not really one big feature - lots of new devices, plus a lot of cleanup and modernisation work spread throughout the subsystem: - More factoring out of common operations into helper functions by Morimoto-san. - DT schema conversons and stylistic nits. - Continued work on building out the new SOF IPC4 scheme. - Support for Awinc AT88395, Infineon PEB2466, Iron Device SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034, Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1. ---------------------------------------------------------------- Ajye Huang (2): ASoC: mediatek: mt8186-rt5682: primary_codec_init() warn: missing error code? 'ret' ASoC: nau8315: remove dependency on GPIOLIB Alexander Stein (2): ASoC: fsl_sai: Use dev_err_probe ASoC: imx-hdmi: Use dev_err_probe Alexander Sverdlin (1): ASoC: ep93xx: ignore 0 Hz sysclk Alexandru Ardelean (1): ASoC: codecs: tas5720: split a tas5720_mute_soc_component() function Amadeusz Sławiński (18): ASoC: Intel: avs: Add peakvol runtime-parameter requests ASoC: Intel: avs: Add control volume operations ASoC: Intel: avs: Parse control tuples ASoC: Intel: avs: Peakvol module configuration ASoC: Intel: avs: Correctly access topology fields ASoC: Intel: avs: Use min_t instead of min with cast ASoC: topology: Properly access value coming from topology file ASoC: topology: Remove unused SOC_TPLG_PASS_PINS constant ASoC: topology: Fix typo in functions name ASoC: topology: Fix function name ASoC: topology: Rename remove_ handlers ASoC: topology: Remove unnecessary forward declarations ASoC: topology: Pass correct pointer instead of casting ASoC: topology: Return an error on complete() failure ASoC: topology: Remove unnecessary check for EOF ASoC: topology: Unify kcontrol removal code ASoC: topology: Use unload() op directly ASoC: Intel: Skylake: Fix struct definition Andy Shevchenko (7): ASoC: Intel: bytcht_cx2072x: Replace open coded acpi_dev_put() ASoC: Intel: bytcht_da7213: Replace open coded acpi_dev_put() ASoC: Intel: cht_bsw_rt5645: Replace open coded acpi_dev_put() ASoC: Intel: cht_bsw_rt5672: Replace open coded acpi_dev_put() ASoC: Intel: sof-wm8804: Replace open coded acpi_dev_put() ASoC: soc-ac97: Convert to agnostic GPIO API ASoC: soc-ac97: Return correct error codes Arnd Bergmann (4): ASoC: PXA: make SND_PXA2XX_SOC_AC97 user-selectable ASoC: ux500: remove platform_data support ASoC: ux500: remove stedma40 references ASoC: SOF: fix intel-soundwire link failure Astrid Rost (3): ASoC: soc-component: add get_jack_type ASoC: simple-card-utils: create jack inputs for aux_devs ASoC: ts3a227e: add set_jack and get_jack_type Bard Liao (3): ASoC: SOF: ipc4-topology: add buffer type support ASoC: SOF: ipc4-topology: use different channel mask for each sdw amp feedback ASoC: SOF: ipc4-topology: set copier sink format Brent Lu (1): ASoC: Intel: sof_ssp_amp: remove unused variable Cezary Rojewski (2): ASoC: Intel: avs: Use asoc_substream_to_rtd() to obtain rtd ASoC: Intel: avs: Simplify probe-component implementation Chancel Liu (3): ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX93 platform ASoC: fsl_xcvr: Add support for i.MX93 platform ASoC: fsl_xcvr: Add constraints of period size while using eDMA Charles Keepax (2): ASoC: cs35l45: Remove separate tables module ASoC: cs35l45: Remove separate namespace for tables Christophe Branchereau (1): ASoC: codecs/jz4760: add digital gain controls Chunxu Li (1): ASoC: mediatek: mt8186: add AFE_DAC_CON0 to volatile register list Claudiu Beznea (12): ASoC: dt-bindings: microchip: use proper naming syntax ASoC: mchp-pdmc: use runtime pm for clock power saving ASoC: mchp-pdmc: add support for suspend to RAM ASoC: mchp-spdiftx: use FIELD_PREP() where possible ASoC: mchp-spdifrx: fix controls which rely on rsr register ASoC: mchp-spdifrx: fix return value in case completion times out ASoC: mchp-spdifrx: fix controls that works with completion mechanism ASoC: mchp-spdifrx: disable all interrupts in mchp_spdifrx_dai_remove() ASoC: mchp-spdifrx: use unsigned long to store clk_get_rate() value ASoC: mchp-spdifrx: remove struct mchp_spdifrx_dev::fmt member ASoC: mchp-spdifrx: add runtime pm support ASoC: mchp-spdifrx: document data structures Colin Ian King (2): ASoC: SMA1303: Fix spelling mistake "Invald" -> "Invalid" ASoC: codecs: aw88395: Fix spelling mistake "cersion" -> "version" Dan Carpenter (1): ASoC: mediatek: mt8188: remove some dead code Daniel Baluta (4): ASoC: SOF: Prepare ipc_msg_data to be used with compress API ASoC: SOF: Prepare set_stream_data_offset for compress API ASoC: SOF: Add support for compress API for stream data/offset ASoC: SOF: compress: Set compress data offset David Rau (1): ASoC: da7213: Add support for mono, set frame width to 32 when possible Deepak R Varma (2): ASoC: imx-pcm-rpmsg: Remove unused variable ASoC: codecs: wcd934x: Use min macro for comparison and assignment Emanuele Ghidoli (3): ASoC: dt-bindings: nau8822: convert to the dtschema ASoC: nau8822: add speaker Bridge Tied Output configuration ASoC: dt-bindings: nau8822: add nuvoton,spk-btl property to dtschema Faiz Abbas (1): ASoC: dt-bindings: simple-card: Document simple-audio-card,plat Geert Uytterhoeven (1): ASoC: dt-bindings: ti,pcm3168a: Convert to json-schema Herve Codina (6): ASoC: Add Renesas IDT821034 codec bindings ASoC: codecs: Add support for the Renesas IDT821034 codec MAINTAINERS: add the Renesas IDT821034 codec entry ASoC: dt-bindings: Add the Infineon PEB2466 codec ASoC: codecs: Add support for the Infineon PEB2466 codec MAINTAINERS: add the Infineon PEB2466 codec entry Jai Luthra (2): dt-bindings: sound: tlv320aic3x: Convert to dtschema ASoC: tlv320aic3x: Add optional clock and port properties Jerome Brunet (9): ASoC: dt-bindings: create component common schema ASoC: dt-bindings: meson: fix gx-card codec node regex ASoC: dt-bindings: meson: convert axg tdm interface to schema ASoC: dt-bindings: meson: convert axg tdm formatters to schema ASoC: dt-bindings: meson: convert axg pdm to schema ASoC: dt-bindings: meson: convert axg fifo to schema ASoC: dt-bindings: meson: convert axg spdif input to schema ASoC: dt-bindings: meson: convert axg spdif output to schema ASoC: dt-bindings: meson: convert axg sound card control to schema Jiapeng Chong (1): ASoC: codecs: Fix unsigned comparison with less than zero Judy Hsiao (1): ASoC: qcom: lpass-platform: Use SNDRV_DMA_TYPE_NONCOHERENT page allocation Kees Cook (1): ASoC: kirkwood: Iterate over array indexes instead of using pointer math Kiseok Jo (7): ASoC: Modified the schema binding and added the vendor prefixes. MAINTAINERS: add IRON DEVICE AUDIO CODEC DRIVERS ASoC: dt-bindings: irondevice,sma1303.yaml: Fix about breaking the checks ASoC: SMA1303: Remove the I2C Retry property in devicetree ASoC: SMA1303: Convert the TDM slot properties in devicetree to mixer ASoC: dt-bindings: irondevice,sma1303: Rework binding and add missing properties ASoC: SMA1303: Remove the sysclk setting in devicetree KiseokJo (1): ASoC: The Iron Device SMA1303 is a boosted Class-D audio amplifier. Krzysztof Kozlowski (28): ASoC: dt-bindings: adi,adau7002: Convert to DT schema ASoC: dt-bindings: cirrus,cs35l41: add interrupts ASoC: dt-bindings: cirrus,cs35l41: cleanup $ref and example ASoC: dt-bindings: qcom,lpass-cpu: Document required-opps ASoC: dt-bindings: qcom,lpass-cpu: Fix DAI children pattern ASoC: dt-bindings: qcom,lpass-cpu: Correct and constrain clocks, interrupts, reg ASoC: dt-bindings: simple-card: correct DAI subnodes pattern ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250 ASoC: dt-bindings: qcom,lpass-tx-macro: correct clocks on SC7280 ASoC: codecs: wsa883x: Simplify &pdev->dev in probe ASoC: codecs: wsa881x: Simplify &pdev->dev in probe ASoC: codecs: wsa881x: Simplify with dev_err_probe ASoC: codecs: wsa881x: Use proper shutdown GPIO polarity ASoC: codecs: constify static sdw_slave_ops struct ASoC: codecs: constify static sdw_port_config struct ASoC: codecs: wsa88xx: remove unneeded includes ASoC: dt-bindings: samsung,odroid: correct codec DAI phandles ASoC: dt-bindings: samsung-i2s: drop unneeded assigned-clock* ASoC: dt-bindings: qcom,sm8250: use fallback for SDM845 sound cards ASoC: qcom: sdm845: add remark about unneeded compatibles ASoC: dt-bindings: renesas,rsnd: simplify list of compatibles ASoC: dt-bindings: maxim,max98090: Convert to dtschema ASoC: codecs: max98090: simplify snd_soc_dai_driver ASoC: dt-bindings: qcom,wsa881x: Allow sound-name-prefix ASoC: dt-bindings: qcom,wcd934x: Describe slim-ifc-dev ASoC: dt-bindings: qcom,wcd934x: Allow usage as IFD device ASoC: dt-bindings: qcom,q6apm-dai: adjust iommus for SM8550 ADSP ASoC: dt-bindings: maxim,max98095: Convert to dtschema Kuninori Morimoto (55): ASoC: soc-compress.c: fixup private_data on snd_soc_new_compress() 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: ak4613: enable Of-graph (Audio-Graph-Card) style ASoC: dt-bindings: renesas,rsnd: add missing playback/capture ASoC: dt-bindings: renesas,rsnd: tidyup rcar_sound,src ASoC: dt-bindings: renesas,rsnd: #sound-dai-cells is not mandatory ASoC: dt-bindings: simple-card: add missing #address-cells/#size-cells ASoC: soc-dai.h: add missing snd_soc_dai_set_widget() ASoC: soc-dai.h: add snd_soc_dai_dma_data_set/get() for low level ASoC: soc-dai.h: add snd_soc_dai_tdm_mask_set/get() helper ASoC: soc-dai.h: add snd_soc_dai_get/set_widget_playback/capture() helper ASoC: soc.h: add snd_soc_card_is_instantiated() helper ASoC: hdmi-codec: use helper function ASoC: hda: use helper function ASoC: max: use helper function ASoC: rt: use helper function ASoC: cirrus: use helper function ASoC: spear: use helper function ASoC: sdw-mockup: use helper function ASoC: intel: use helper function ASoC: mediatek: use helper function ASoC: meson: use helper function ASoC: rockchip: use helper function ASoC: sof: use helper function ASoC: tegra: use helper function ASoC: ti: use helper function ASoC: soc-core.c: use helper function ASoC: soc-dapm.c: use helper function ASoC: soc-dai.c: use helper function ASoC: soc-pcm.c: use helper function ASoC: soc-topology.c: use helper function ASoC: soc-dai.h: cleanup Playback/Capture data for snd_soc_dai ASoC: cs42l42: use helper function ASoC: rsnd: check whether playback/capture property exists ASoC: rsnd: fixup #endif position ASoC: rsnd: Remove unnecessary rsnd_dbg_dai_call() ASoC: rsnd: indicate necessary error when clock start failed ASoC: rsnd: indicate warning once if it can't handle requested rule ASoC: rsnd: use same debug message format on clkout ASoC: rsnd: remove unnecessary ADG flags ASoC: rsnd: rename clk to clkin ASoC: rsnd: moves clkout_name to top of the file ASoC: rsnd: use clkin/out_size ASoC: rsnd: use array for 44.1kHz/48kHz rate handling ASoC: rsnd: tidyup rsnd_dma_addr() ASoC: rsnd: dma.c: tidyup rsnd_dma_probe() ASoC: dt-bindings: renesas: add R8A779G0 V4H ASoC: rsnd: add R-Car Gen4 Sound support ASoC: dt-bindings: renesas,rsnd.yaml: tidyup reg/reg-name ASoC: rsnd: core.c: indicate warning if strange TDM width was set ASoC: dt-bindings: renesas,rsnd.yaml: drop "dmas/dma-names" from "rcar_sound,ssi" Luca Ceresoli (1): ASoC: fsl-asoc-card: constify fsl_asoc_card_dai Lucas Tanure (1): ASoC: soc-dapm.h: fixup warning struct snd_pcm_substream not declared Lukasz Majewski (3): ASoC: wm8940: Remove warning when no plat data present ASoC: wm8940: Rewrite code to set proper clocks ASoC: wm8940: Read chip ID when wm8940 codec probing Marek Vasut (1): ASoC: dt-bindings: fsl-sai: Simplify the VFxxx dmas binding Mario Limonciello (6): ASoC: amd: yc: Adjust the gain for PDM DMIC ASoC: amd: yc: Add a module parameter to influence pdm_gain ASoC: amd: renoir: Adjust the gain for PDM DMIC ASoC: amd: renoir: Add a module parameter to influence pdm_gain ASoC: amd: ps: Adjust the gain for PDM DMIC ASoC: amd: ps: Add a module parameter to influence pdm_gain Mark Brown (31): Add support for DSP volume controls ASoC: Drop empty platform remove functions ASoC: SOF: Extend the IPC ops optionality ASoC: microchip: power saving features and cleanups Add support for XCVR on i.MX93 platform ASoC: nau8822: add speaker Bridge Tied Output ASoC: fsl-asoc-card: Log error code when we fail to register Add support of MediaTek mt8188 to SOF Add tdm support in acp machine driver Fixes for avs driver ASoC: samsung: fsd: audio support for FSD SoC ASoC: Merge up 6.2 fixes Add support to compress API to ipc_msg_data / ASoC: mediatek: Add support for MT8188 SoC Update lpass dt-bindings ASoC: simple-card-utils: create jack inputs for ASoC: SOF: ipc4: Multi-stream playback and capture support ASoC: dt-bindings: audio-graph-port related update ASoC: topology: Fixes and cleanups Add the Renesas IDT821034 codec support ASoC: codecs: Add Awinic AW88395 audio amplifier ASoC: mchp-spdifrx: add runtime PM support and fixes ASoC: use helper function and cleanup ASoC: cs42l42: Add SoundWire support Fix default DMIC gain on AMD PDM drivers ASoC: (SOF) topology: Regression fixes for next ASoC: SOF: core/ipc4/mtl: Add support for PCM delay Add the Infineon PEB2466 codec support ASoC: SOF: ipc4-topology: Configure copier sink format ASoC: qcom: q6dsp and lpass codec stablity fixes ASoC: SOF: Intel/ipc4: Support for low power playback Matthias Kaehlcke (1): SoC: rt5682s: Disable jack detection interrupt during suspend Nathan Chancellor (2): ASoC: amd: ps: Fix uninitialized ret in create_acp64_platform_devs() ASoC: mchp-spdifrx: Fix uninitialized use of mr in mchp_spdifrx_hw_params() Ondrej Jirman (2): ASoC: rt5640: Allow configuration of LOUT to mono differential mode ASoC: rt5640: Allow to describe how LOUT is wired Padmanabhan Rajanbabu (2): ASoC: dt-bindings: Add FSD I2S controller bindings ASoC: samsung: i2s: add support for FSD I2S Peter Ujfalusi (17): ASoC: SOF: sof-audio: Treat tplg_ops->route_setup() as optional ASoC: SOF: sof-audio: Update documentation for sof_ipc_tplg_ops ASoC: SOF: Add helper macro to be used to get an IPC ops ASoC: SOF: pcm: Extend the optionality of IPC ops to IPC as well ASoC: SOF: control: Extend the optionality of IPC ops to IPC as well ASoC: SOF: sof-audio: Extend the optionality of IPC ops to IPC as well ASoC: SOF: topology: Extend the optionality of IPC ops to IPC as well ASoC: SOF: pm: Extend the optionality of IPC ops to IPC as well ASoC: SOF: sof-priv: Mark fw_tracing ops optional in documentation ASoC: SOF: trace: Use sof_ipc_get_ops() in sof_fw_trace_init ASoC: SOF: trace: No need to check for op pointer in sof_fw_trace_free() ASoC: SOF: core: Print out the value of sof_debug if it is set ASoC: SOF: Avoid double decrementing use_count in sof_widget_setup on error ASoC: SOF: Protect swidget->use_count with mutex for kcontrol access race ASoC: SOF: ipc4-pcm: Do not run the trigger pipelines if no spipe is stored ASoC: SOF: topology: Add missed "else" in sof_connect_dai_widget ASoC: topology: Set correct unload callback for graph type Pierre-Louis Bossart (1): ASoC: amd: ps: use static function Rander Wang (14): ASoC: SOF: add ipc4_fw_reg header file ASoC: SOF: add fw_info_box support ASoC: SOF: add time info structure for ipc4 path ASoC: SOF: ipc4-pcm: allocate time info for pcm delay feature ASoC: SOF: ipc4-pcm: add hw_params ASoC: SOF: add delay function support in sof framework ASoC: SOF: add get_stream_position ops for pcm delay ASoC: SOF: Intel: mtl: add get_stream_position support ASoC: SOF: ipc4-pcm: add delay function support ASoC: SOF: Introduce a new set_pm_gate() IPC PM op ASoC: SOF: Intel: hda-dsp: use set_pm_gate according to ipc version ASoC: SOF: ipc4: Wake up dsp core before sending ipc msg ASoC: SOF: Intel: Enable d0i3 work for ipc4 ASoC: SOF: Intel: hda-dsp: Set streaming flag for d0i3 Randy Dunlap (1): ASoC: Kconfig: fix spelling of "up to" Ranjani Sridharan (16): ASoC: SOF: ipc4-topology: No need to unbind routes within a pipeline ASoC: soc-pcm: Export widget_in_list() ASoC: SOF: sof-audio: Set up/free DAI/AIF widgets only once ASoC: SOF: sof-audio: Only process widgets in the connected widget list ASoC: SOF: pcm: do not free widgets during suspend trigger ASoC: SOF: topology: Set IPC-specific trigger order for DAI links ASoC: SOF: Introduce PCM setup/free PCM IPC ops ASoC: SOF: ipc4-pcm: Define pcm_setup/free ops ASoC: SOF: ipc4: Add flag to skip triggering pipelines during FE DAI trigger ASoC: SOF: sof-audio: Populate the PCM stream pipeline_info ASoC: SOF: ipc4-pcm: Use the PCM stream's pipeline_info during trigger ASoC: SOF: Introduce struct snd_sof_pipeline ASoC: SOF: ipc4-pcm: Rename 'data' variable to trigger_list ASoC: SOF: ipc4-pcm: Implement pipeline trigger reference counting ASoC: SOF: ipc4-topology: Protect pipeline free with mutex ASoC: SOF: ipc4-topology: Print queue IDs in error Richard Fitzgerald (6): ASoC: cs42l42: Add SOFT_RESET_REBOOT register ASoC: cs42l42: Ensure MCLKint is a multiple of the sample rate ASoC: cs42l42: Separate ASP config from PLL config ASoC: cs42l42: Export some functions for SoundWire ASoC: cs42l42: Add SoundWire support ASoC: cs42l42: Don't set idle_bias_on Sameer Pujar (1): ASoC: rt5640: Update MCLK rate in set_sysclk() Seppo Ingalsuo (1): ASoC: SOF: IPC3 topology: Print the conflicting bytes sizes Shengjiu Wang (1): ASoC: fsl_sai: initialize is_dsp_mode flag Shuming Fan (3): ASoC: rt711-sdca: add jack detection mode for JD2 100K ASoC: rt712-sdca: Add RT712 SDCA driver for Jack and Amp topology ASoC: rt712-sdca: fix coding style and unconditionally return issues Srinivas Kandagatla (8): ASoC: qcom: q6apm-lpass-dai: unprepare stream if its already prepared ASoC: qcom: q6apm-dai: fix race condition while updating the position pointer ASoC: qcom: q6apm-dai: Add SNDRV_PCM_INFO_BATCH flag ASoC: qcom: audioreach: fix ADSP ready check ASoC: codecs: lpass: register mclk after runtime pm ASoC: codecs: lpass: fix incorrect mclk rate ASoC: codecs: lpass: do not reset soundwire block on clk enable ASoC: codecs: lpass: remove not so useful verbose log Srinivasa Rao Mandadapu (3): ASoC: qdsp6: audioreach: Add ADSP ready check ASoC: qcom: dt-bindings: lpass-va-macro: Update clock name ASoC: dt-bindings: google,sc7280-herobrine: Add platform property Stefan Binding (2): soundwire: stream: Add specific prep/deprep commands to port_prep callback ASoC: cs42l42: Wait for debounce interval after resume Steffen Aschbacher (5): ASoC: codecs: tas5720: add support for TAS5720A-Q1 (automotive) variant ASoC: tas5720: set bit 7 in ANALOG_CTRL_REG for TAS5720A-Q1 during probe ASoC: dt-bindings: add entry for TAS5720A-Q1 driver ASoC: tlv320adcx140: fix 'ti,gpio-config' DT property init ASoC: tlv320adcx140: extend list of supported samplerates Tinghan Shen (3): dt-bindings: dsp: mediatek: Add mt8188 dsp compatible ASoC: SOF: mediatek: Support mt8188 platform ASoC: SOF: mediatek: Provide debugfs_add_region_item ops for core Tom Rix (1): ASoC: codecs: aw88395: initialize cur_scene_id to 0 Trevor Wu (14): 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 ASoC: dt-bindings: mediatek,mt8188-afe: add audio afe document ASoC: mediatek: common: add soundcard driver common code ASoC: mediatek: mt8188: add machine driver with mt6359 ASoC: dt-bindings: mediatek,mt8188-mt6359: add mt8188-mt6359 document ASoC: mediatek: mt8188: remove etdm dead code Uwe Kleine-König (17): ASoC: pxa: Drop empty platform remove function ASoC: amd: Drop empty platform remove function ASoC: atmel-classd: Drop empty platform remove function ASoC: atmel-pdmic: Drop empty platform remove function ASoC: 88pm860x: Drop empty platform remove function ASoC: ac97: Drop empty platform remove function ASoC: adau7002: Drop empty platform remove function ASoC: bt-sco: Drop empty platform remove function ASoC: cq93vc: Drop empty platform remove function ASoC: mc13783: Drop empty platform remove function ASoC: wl1273: Drop empty platform remove function ASoC: pxa: e740_wm9705: Drop empty platform remove function ASoC: pxa: e750_wm9705: Drop empty platform remove function ASoC: pxa: e800_wm9712: Drop empty platform remove function ASoC: SOF: nocodec: Drop empty platform remove function ASoC: pxa: Drop empty platform remove function ASoC: sma1303: Convert to i2c's .probe_new() V sujith kumar Reddy (1): ASoC: SOF: amd: Enable cache for AMD Rembrandt platform Venkata Prasad Potturu (6): ASoC: amd: acp: Add new cpu dai's in machine driver ASoC: amd: acp: Refactor dai format implementation ASoC: amd: acp: Refactor i2s clocks programming sequence ASoC: amd: acp: Add i2s tdm support in machine driver ASoC: amd: acp: Enable i2s tdm support for skyrim platforms ASoC: amd: acp: Refactor bit width calculation Vijendar Mukunda (8): ASoC: amd: ps: implement api to retrieve acp device config ASoC: amd: ps: refactor platform device creation logic ASoC: amd: ps: update dev index value in irq handler ASoC: amd: ps: move irq handler registration ASoC: amd: ps: add mutex lock for accessing common registers ASoC: amd: ps: use acp_lock to protect common registers in pdm driver ASoC: amd: ps: remove unused variable ASoC: amd: update ps platform acp header file Weidong Wang (8): ASoC: codecs: Add i2c and codec registration for aw88395 and their associated operation functions ASoC: codecs: ACF bin parsing and check library file for aw88395 ASoC: codecs: Aw88395 function for ALSA Audio Driver ASoC: codecs: Aw88395 chip register file, data type file and Kconfig Makefile ASoC: dt-bindings: Add schema for "awinic,aw88395" ASoC: codecs: Fixed a spelling error in the function name ASoC: codecs: Modify the log print statement ASoC: codecs: Modify error implicit declaration of function 'gpiod_set_value_cansleep' YC Hung (1): ASoC: SOF: mediatek: mt8195: remove a redundant comparison of sram Yang Li (2): ASoC: codecs: fix platform_no_drv_owner.cocci warning ASoC: codecs: Remove unneeded semicolon Yassine Oudjana (1): ASoC: qcom: apq8096: set driver name correctly 강신형 (1): ASoC: soc-compress: Reposition and add pcm_mutex .../bindings/dsp/mediatek,mt8186-dsp.yaml | 4 +- .../devicetree/bindings/sound/adi,adau7002.txt | 19 - .../devicetree/bindings/sound/adi,adau7002.yaml | 40 + .../devicetree/bindings/sound/ak4613.yaml | 7 + .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 34 - .../bindings/sound/amlogic,axg-fifo.yaml | 112 + .../devicetree/bindings/sound/amlogic,axg-pdm.txt | 29 - .../devicetree/bindings/sound/amlogic,axg-pdm.yaml | 82 + .../bindings/sound/amlogic,axg-sound-card.txt | 124 - .../bindings/sound/amlogic,axg-sound-card.yaml | 183 ++ .../bindings/sound/amlogic,axg-spdifin.txt | 27 - .../bindings/sound/amlogic,axg-spdifin.yaml | 86 + .../bindings/sound/amlogic,axg-spdifout.txt | 25 - .../bindings/sound/amlogic,axg-spdifout.yaml | 79 + .../bindings/sound/amlogic,axg-tdm-formatters.txt | 36 - .../bindings/sound/amlogic,axg-tdm-formatters.yaml | 88 + .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 - .../bindings/sound/amlogic,axg-tdm-iface.yaml | 55 + .../bindings/sound/amlogic,gx-sound-card.yaml | 2 +- .../bindings/sound/audio-graph-port.yaml | 65 +- .../devicetree/bindings/sound/awinic,aw88395.yaml | 53 + .../devicetree/bindings/sound/cirrus,cs35l41.yaml | 43 +- .../bindings/sound/component-common.yaml | 21 + .../devicetree/bindings/sound/dai-common.yaml | 11 +- .../devicetree/bindings/sound/fsl,sai.yaml | 38 +- .../devicetree/bindings/sound/fsl,xcvr.yaml | 1 + .../bindings/sound/google,sc7280-herobrine.yaml | 12 + .../bindings/sound/infineon,peb2466.yaml | 91 + .../bindings/sound/irondevice,sma1303.yaml | 48 + .../devicetree/bindings/sound/max98090.txt | 59 - .../devicetree/bindings/sound/max98095.txt | 22 - .../devicetree/bindings/sound/maxim,max98090.yaml | 84 + .../devicetree/bindings/sound/maxim,max98095.yaml | 54 + .../bindings/sound/mediatek,mt8188-afe.yaml | 208 ++ .../bindings/sound/mediatek,mt8188-mt6359.yaml | 97 + ...,i2s-mcc.yaml => microchip,sama7g5-i2smcc.yaml} | 2 +- ...ochip,pdmc.yaml => microchip,sama7g5-pdmc.yaml} | 2 +- ...spdifrx.yaml => microchip,sama7g5-spdifrx.yaml} | 2 +- ...spdiftx.yaml => microchip,sama7g5-spdiftx.yaml} | 2 +- .../devicetree/bindings/sound/nau8822.txt | 16 - .../devicetree/bindings/sound/nuvoton,nau8822.yaml | 46 + .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 136 +- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +- .../devicetree/bindings/sound/qcom,q6apm-dai.yaml | 3 +- .../devicetree/bindings/sound/qcom,sm8250.yaml | 24 +- .../devicetree/bindings/sound/qcom,wcd934x.yaml | 62 +- .../devicetree/bindings/sound/qcom,wsa881x.yaml | 5 +- .../bindings/sound/renesas,idt821034.yaml | 75 + .../devicetree/bindings/sound/renesas,rsnd.yaml | 59 +- Documentation/devicetree/bindings/sound/rt5640.txt | 3 + .../devicetree/bindings/sound/samsung,odroid.yaml | 3 +- .../devicetree/bindings/sound/samsung-i2s.yaml | 11 +- .../devicetree/bindings/sound/simple-card.yaml | 18 +- .../devicetree/bindings/sound/tas5720.txt | 2 + .../devicetree/bindings/sound/ti,pcm3168a.txt | 56 - .../devicetree/bindings/sound/ti,pcm3168a.yaml | 107 + .../devicetree/bindings/sound/ti,tlv320aic3x.yaml | 165 + .../devicetree/bindings/sound/tlv320aic3x.txt | 97 - .../devicetree/bindings/vendor-prefixes.yaml | 2 + MAINTAINERS | 21 + drivers/soundwire/stream.c | 4 +- include/linux/platform_data/asoc-ux500-msp.h | 20 - include/linux/soundwire/sdw.h | 8 +- include/sound/acp63_chip_offset_byte.h | 751 ++--- include/sound/cs42l42.h | 5 + include/sound/simple_card_utils.h | 3 + include/sound/soc-component.h | 2 + include/sound/soc-dai.h | 90 +- include/sound/soc-dapm.h | 1 + include/sound/soc-dpcm.h | 2 + include/sound/soc-topology.h | 2 +- include/sound/soc.h | 6 + include/sound/sof/ipc4/header.h | 3 + include/uapi/sound/intel/avs/tokens.h | 4 + sound/soc/amd/acp/acp-legacy-mach.c | 13 + sound/soc/amd/acp/acp-mach-common.c | 465 ++- sound/soc/amd/acp/acp-mach.h | 4 + sound/soc/amd/acp/acp-sof-mach.c | 14 + sound/soc/amd/ps/acp63.h | 14 +- sound/soc/amd/ps/pci-ps.c | 211 +- sound/soc/amd/ps/ps-pdm-dma.c | 40 +- sound/soc/amd/raven/acp3x-i2s.c | 8 - sound/soc/amd/renoir/acp3x-pdm-dma.c | 8 +- sound/soc/amd/renoir/rn_acp3x.h | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 8 +- sound/soc/amd/yc/acp6x.h | 2 +- sound/soc/atmel/atmel-classd.c | 6 - sound/soc/atmel/atmel-pdmic.c | 6 - sound/soc/atmel/mchp-pdmc.c | 145 +- sound/soc/atmel/mchp-spdifrx.c | 554 +++- sound/soc/atmel/mchp-spdiftx.c | 8 +- sound/soc/cirrus/ep93xx-ac97.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 6 +- sound/soc/codecs/88pm860x-codec.c | 6 - sound/soc/codecs/Kconfig | 69 +- sound/soc/codecs/Makefile | 19 +- sound/soc/codecs/ac97.c | 6 - sound/soc/codecs/adau7002.c | 6 - sound/soc/codecs/aw88395/aw88395.c | 579 ++++ sound/soc/codecs/aw88395/aw88395.h | 58 + sound/soc/codecs/aw88395/aw88395_data_type.h | 142 + sound/soc/codecs/aw88395/aw88395_device.c | 1748 ++++++++++ sound/soc/codecs/aw88395/aw88395_device.h | 194 ++ sound/soc/codecs/aw88395/aw88395_lib.c | 1066 +++++++ sound/soc/codecs/aw88395/aw88395_lib.h | 92 + sound/soc/codecs/aw88395/aw88395_reg.h | 383 +++ sound/soc/codecs/bt-sco.c | 6 - sound/soc/codecs/cq93vc.c | 6 - sound/soc/codecs/cs35l45-i2c.c | 1 - sound/soc/codecs/cs35l45-spi.c | 1 - sound/soc/codecs/cs35l45-tables.c | 12 +- sound/soc/codecs/cs35l45.c | 1 - sound/soc/codecs/cs42l42-sdw.c | 607 ++++ sound/soc/codecs/cs42l42.c | 133 +- sound/soc/codecs/cs42l42.h | 9 +- sound/soc/codecs/da7213.c | 30 +- sound/soc/codecs/da7213.h | 3 + sound/soc/codecs/hda.c | 7 +- sound/soc/codecs/hdmi-codec.c | 11 +- sound/soc/codecs/idt821034.c | 1178 +++++++ sound/soc/codecs/jz4760.c | 9 + sound/soc/codecs/lpass-rx-macro.c | 31 +- sound/soc/codecs/lpass-tx-macro.c | 25 +- sound/soc/codecs/lpass-va-macro.c | 43 +- sound/soc/codecs/lpass-wsa-macro.c | 27 +- sound/soc/codecs/max98090.c | 8 +- sound/soc/codecs/max98373-sdw.c | 5 +- sound/soc/codecs/mc13783.c | 6 - sound/soc/codecs/nau8822.c | 9 + sound/soc/codecs/nau8822.h | 9 + sound/soc/codecs/peb2466.c | 2071 ++++++++++++ sound/soc/codecs/rt1308-sdw.c | 5 +- sound/soc/codecs/rt1316-sdw.c | 7 +- sound/soc/codecs/rt1318-sdw.c | 7 +- sound/soc/codecs/rt5640.c | 9 + sound/soc/codecs/rt5640.h | 2 + sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5665.c | 4 +- sound/soc/codecs/rt5668.c | 4 +- sound/soc/codecs/rt5682-sdw.c | 5 +- sound/soc/codecs/rt5682.c | 4 +- sound/soc/codecs/rt5682s.c | 14 +- sound/soc/codecs/rt5682s.h | 1 + sound/soc/codecs/rt700.c | 7 +- sound/soc/codecs/rt711-sdca-sdw.c | 2 +- sound/soc/codecs/rt711-sdca.c | 22 +- sound/soc/codecs/rt711-sdca.h | 8 +- sound/soc/codecs/rt711.c | 7 +- sound/soc/codecs/rt712-sdca-sdw.c | 485 +++ sound/soc/codecs/rt712-sdca-sdw.h | 108 + sound/soc/codecs/rt712-sdca.c | 1340 ++++++++ sound/soc/codecs/rt712-sdca.h | 220 ++ sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 5 +- sound/soc/codecs/rt715.c | 5 +- sound/soc/codecs/sdw-mockup.c | 5 +- sound/soc/codecs/sma1303.c | 1820 +++++++++++ sound/soc/codecs/sma1303.h | 609 ++++ sound/soc/codecs/tas5720.c | 128 +- sound/soc/codecs/tas5720.h | 16 +- sound/soc/codecs/tlv320adcx140.c | 2 +- sound/soc/codecs/tlv320adcx140.h | 4 +- sound/soc/codecs/ts3a227e.c | 20 +- sound/soc/codecs/wcd934x.c | 10 +- sound/soc/codecs/wcd938x-sdw.c | 2 +- sound/soc/codecs/wl1273.c | 6 - sound/soc/codecs/wm8940.c | 116 +- sound/soc/codecs/wm8940.h | 3 + sound/soc/codecs/wsa881x.c | 63 +- sound/soc/codecs/wsa883x.c | 22 +- sound/soc/fsl/fsl-asoc-card.c | 4 +- sound/soc/fsl/fsl_sai.c | 6 +- sound/soc/fsl/fsl_xcvr.c | 155 +- sound/soc/fsl/fsl_xcvr.h | 7 + sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 6 +- sound/soc/generic/simple-card-utils.c | 49 + sound/soc/generic/simple-card.c | 4 + sound/soc/intel/atom/sst-atom-controls.c | 9 +- sound/soc/intel/avs/Makefile | 2 +- sound/soc/intel/avs/boards/nau8825.c | 5 +- sound/soc/intel/avs/boards/rt286.c | 2 +- sound/soc/intel/avs/boards/rt298.c | 2 +- sound/soc/intel/avs/control.c | 105 + sound/soc/intel/avs/control.h | 23 + sound/soc/intel/avs/messages.c | 29 + sound/soc/intel/avs/messages.h | 33 + sound/soc/intel/avs/path.c | 62 + sound/soc/intel/avs/pcm.c | 45 +- sound/soc/intel/avs/probes.c | 23 - sound/soc/intel/avs/topology.c | 76 + sound/soc/intel/avs/topology.h | 3 + sound/soc/intel/avs/trace.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/sof_ssp_amp.c | 4 - sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 19 +- sound/soc/intel/skylake/skl-topology.h | 5 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/Makefile | 2 +- sound/soc/mediatek/common/mtk-base-afe.h | 19 + sound/soc/mediatek/common/mtk-dsp-sof-common.c | 18 +- sound/soc/mediatek/common/mtk-soundcard-driver.c | 79 + sound/soc/mediatek/common/mtk-soundcard-driver.h | 14 + sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 8 +- sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 7 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 1 + sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 7 +- .../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- sound/soc/mediatek/mt8188/Makefile | 15 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 658 ++++ sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 115 + sound/soc/mediatek/mt8188/mt8188-afe-common.h | 151 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 3358 ++++++++++++++++++++ sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 205 ++ sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 15 + sound/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 | 2573 +++++++++++++++ sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 368 +++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 785 +++++ sound/soc/mediatek/mt8188/mt8188-reg.h | 3180 ++++++++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 8 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 7 +- sound/soc/meson/aiu-fifo-i2s.c | 4 +- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.c | 21 +- sound/soc/meson/axg-tdm-interface.c | 34 +- sound/soc/meson/axg-tdmin.c | 2 +- sound/soc/meson/axg-tdmout.c | 2 +- sound/soc/meson/meson-codec-glue.c | 8 +- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/e740_wm9705.c | 6 - sound/soc/pxa/e750_wm9705.c | 6 - sound/soc/pxa/e800_wm9712.c | 6 - sound/soc/pxa/spitz.c | 6 - sound/soc/qcom/apq8096.c | 1 + sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 22 +- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 5 + sound/soc/qcom/qdsp6/q6apm.c | 14 + sound/soc/qcom/qdsp6/q6apm.h | 2 + sound/soc/qcom/qdsp6/q6prm.c | 4 + sound/soc/qcom/sdm845.c | 1 + sound/soc/rockchip/Kconfig | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s-regs.h | 1 + sound/soc/samsung/i2s.c | 53 + sound/soc/sh/rcar/adg.c | 169 +- sound/soc/sh/rcar/core.c | 73 +- sound/soc/sh/rcar/dma.c | 57 +- sound/soc/sh/rcar/gen.c | 70 + sound/soc/sh/rcar/rsnd.h | 23 +- sound/soc/sh/rcar/ssi.c | 13 +- sound/soc/sh/rcar/ssiu.c | 15 +- sound/soc/soc-ac97.c | 68 +- sound/soc/soc-component.c | 20 + sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dai.c | 13 +- sound/soc/soc-dapm.c | 26 +- sound/soc/soc-pcm.c | 15 +- sound/soc/soc-topology.c | 187 +- sound/soc/sof/amd/acp-dsp-offset.h | 4 + sound/soc/sof/amd/acp-ipc.c | 8 +- sound/soc/sof/amd/acp-loader.c | 7 + sound/soc/sof/amd/acp.h | 7 +- sound/soc/sof/compress.c | 9 + sound/soc/sof/control.c | 44 +- sound/soc/sof/core.c | 4 + sound/soc/sof/intel/Kconfig | 4 +- sound/soc/sof/intel/cnl.c | 2 + sound/soc/sof/intel/hda-dai.c | 92 +- sound/soc/sof/intel/hda-dsp.c | 46 +- sound/soc/sof/intel/hda-ipc.c | 36 +- sound/soc/sof/intel/hda.h | 6 +- sound/soc/sof/intel/mtl.c | 16 + sound/soc/sof/intel/mtl.h | 6 + sound/soc/sof/ipc3-control.c | 46 +- sound/soc/sof/ipc3-pcm.c | 3 +- sound/soc/sof/ipc3-topology.c | 39 +- sound/soc/sof/ipc3.c | 22 +- sound/soc/sof/ipc4-control.c | 33 +- sound/soc/sof/ipc4-fw-reg.h | 155 + sound/soc/sof/ipc4-pcm.c | 573 +++- sound/soc/sof/ipc4-priv.h | 17 + sound/soc/sof/ipc4-topology.c | 212 +- sound/soc/sof/ipc4-topology.h | 60 + sound/soc/sof/ipc4.c | 29 + sound/soc/sof/mediatek/mt8186/mt8186.c | 22 +- sound/soc/sof/mediatek/mt8186/mt8186.h | 10 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 8 +- sound/soc/sof/mediatek/mt8195/mt8195.h | 2 - sound/soc/sof/nocodec.c | 6 - sound/soc/sof/ops.h | 18 +- sound/soc/sof/pcm.c | 36 +- sound/soc/sof/pm.c | 12 +- sound/soc/sof/sof-audio.c | 281 +- sound/soc/sof/sof-audio.h | 67 +- sound/soc/sof/sof-priv.h | 30 +- sound/soc/sof/stream-ipc.c | 53 +- sound/soc/sof/topology.c | 309 +- sound/soc/sof/trace.c | 8 +- sound/soc/spear/spdif_out.c | 3 +- sound/soc/tegra/Kconfig | 4 +- sound/soc/tegra/tegra20_ac97.c | 4 +- sound/soc/tegra/tegra20_i2s.c | 4 +- sound/soc/tegra/tegra20_spdif.c | 3 +- sound/soc/tegra/tegra210_admaif.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/ti/davinci-i2s.c | 5 +- sound/soc/ti/davinci-mcasp.c | 5 +- sound/soc/ti/davinci-vcif.c | 5 +- sound/soc/ux500/mop500.c | 8 +- sound/soc/ux500/ux500_msp_dai.c | 37 +- sound/soc/ux500/ux500_msp_i2s.c | 66 +- sound/soc/ux500/ux500_msp_i2s.h | 14 +- sound/soc/ux500/ux500_pcm.c | 83 +- 326 files changed, 32241 insertions(+), 3385 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw88395.yaml create mode 100644 Documentation/devicetree/bindings/sound/component-common.yaml create mode 100644 Documentation/devicetree/bindings/sound/infineon,peb2466.yaml create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml delete mode 100644 Documentation/devicetree/bindings/sound/max98090.txt delete mode 100644 Documentation/devicetree/bindings/sound/max98095.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98090.yaml create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98095.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml rename Documentation/devicetree/bindings/sound/{mchp,i2s-mcc.yaml => microchip,sama7g5-i2smcc.yaml} (97%) rename Documentation/devicetree/bindings/sound/{microchip,pdmc.yaml => microchip,sama7g5-pdmc.yaml} (97%) rename Documentation/devicetree/bindings/sound/{mchp,spdifrx.yaml => microchip,sama7g5-spdifrx.yaml} (95%) rename Documentation/devicetree/bindings/sound/{mchp,spdiftx.yaml => microchip,sama7g5-spdiftx.yaml} (95%) delete mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,idt821034.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt delete mode 100644 include/linux/platform_data/asoc-ux500-msp.h create mode 100644 sound/soc/codecs/aw88395/aw88395.c create mode 100644 sound/soc/codecs/aw88395/aw88395.h create mode 100644 sound/soc/codecs/aw88395/aw88395_data_type.h create mode 100644 sound/soc/codecs/aw88395/aw88395_device.c create mode 100644 sound/soc/codecs/aw88395/aw88395_device.h create mode 100644 sound/soc/codecs/aw88395/aw88395_lib.c create mode 100644 sound/soc/codecs/aw88395/aw88395_lib.h create mode 100644 sound/soc/codecs/aw88395/aw88395_reg.h create mode 100644 sound/soc/codecs/cs42l42-sdw.c create mode 100644 sound/soc/codecs/idt821034.c create mode 100644 sound/soc/codecs/peb2466.c create mode 100644 sound/soc/codecs/rt712-sdca-sdw.c create mode 100644 sound/soc/codecs/rt712-sdca-sdw.h create mode 100644 sound/soc/codecs/rt712-sdca.c create mode 100644 sound/soc/codecs/rt712-sdca.h create mode 100644 sound/soc/codecs/sma1303.c create mode 100644 sound/soc/codecs/sma1303.h create mode 100644 sound/soc/intel/avs/control.c create mode 100644 sound/soc/intel/avs/control.h create mode 100644 sound/soc/mediatek/common/mtk-soundcard-driver.c create mode 100644 sound/soc/mediatek/common/mtk-soundcard-driver.h 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 create mode 100644 sound/soc/sof/ipc4-fw-reg.h

2 1

[GIT PULL] ASoC fixes for v6.2-rc8
by Mark Brown 16 Feb '23

16 Feb '23

The following changes since commit c173ee5b2fa6195066674d66d1d7e191010fb1ff: ASoC: topology: Return -ENOMEM on memory allocation failure (2023-02-07 14:06:26 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.2-rc8 for you to fetch changes up to 5afc7eefe41645259da84898fc55f6f46cb4de47: ASoC: SOF: ops: refine parameters order in function snd_sof_dsp_update8 (2023-02-08 11:51:45 +0000) ---------------------------------------------------------------- ASoC: Fix for v6.2 One non-urgent fix for v6.2, this could possibly wait till the merge window. ---------------------------------------------------------------- Rander Wang (1): ASoC: SOF: ops: refine parameters order in function snd_sof_dsp_update8 sound/soc/sof/ops.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

2 1

[PATCH] ASoC: da7219: Improve the IRQ process to increase the stability
by David Rau 16 Feb '23

16 Feb '23

Remove the sleep control in IRQ thread and create an individual task to handel it for Jack plug in event. This commit improves the control of ground switches in the AAD IRQ. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- sound/soc/codecs/da7219-aad.c | 41 +++++++++++++++++++++++++++++------ sound/soc/codecs/da7219-aad.h | 4 ++++ 2 files changed, 38 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index c55b033d89da..4a4f09f924bc 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -339,11 +339,39 @@ static void da7219_aad_hptest_work(struct work_struct *work) SND_JACK_HEADSET | SND_JACK_LINEOUT); } +static void da7219_aad_jack_det_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, jack_det_work); + struct snd_soc_component *component = da7219_aad->component; + u8 srm_st; + + mutex_lock(&da7219_aad->jack_det_mutex); + + srm_st = snd_soc_component_read(component, DA7219_PLL_SRM_STS) & DA7219_PLL_SRM_STS_MCLK; + msleep(da7219_aad->gnd_switch_delay * ((srm_st == 0x0) ? 2 : 1) - 4); + /* Enable ground switch */ + snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); + + mutex_unlock(&da7219_aad->jack_det_mutex); +} + /* * IRQ */ +static irqreturn_t da7219_aad_pre_irq_thread(int irq, void *data) +{ + + struct da7219_aad_priv *da7219_aad = data; + + if (!da7219_aad->jack_inserted) + schedule_work(&da7219_aad->jack_det_work); + + return IRQ_WAKE_THREAD; +} + static irqreturn_t da7219_aad_irq_thread(int irq, void *data) { struct da7219_aad_priv *da7219_aad = data; @@ -351,14 +379,9 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; - u8 statusa, srm_st; + u8 statusa; int i, report = 0, mask = 0; - srm_st = snd_soc_component_read(component, DA7219_PLL_SRM_STS) & DA7219_PLL_SRM_STS_MCLK; - msleep(da7219_aad->gnd_switch_delay * ((srm_st == 0x0) ? 2 : 1) - 4); - /* Enable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); - /* Read current IRQ events */ regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, events, DA7219_AAD_IRQ_REG_MAX); @@ -377,6 +400,9 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B], statusa); + if (!da7219_aad->jack_inserted) + cancel_work_sync(&da7219_aad->jack_det_work); + if (statusa & DA7219_JACK_INSERTION_STS_MASK) { /* Jack Insertion */ if (events[DA7219_AAD_IRQ_REG_A] & @@ -940,8 +966,9 @@ int da7219_aad_init(struct snd_soc_component *component) INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); + INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work); - ret = request_threaded_irq(da7219_aad->irq, NULL, + ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread, da7219_aad_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "da7219-aad", da7219_aad); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h index 21fdf53095cc..be87ee47edde 100644 --- a/sound/soc/codecs/da7219-aad.h +++ b/sound/soc/codecs/da7219-aad.h @@ -11,6 +11,7 @@ #define __DA7219_AAD_H #include <linux/timer.h> +#include <linux/mutex.h> #include <sound/soc.h> #include <sound/jack.h> #include <sound/da7219-aad.h> @@ -196,6 +197,9 @@ struct da7219_aad_priv { struct work_struct btn_det_work; struct work_struct hptest_work; + struct work_struct jack_det_work; + + struct mutex jack_det_mutex; struct snd_soc_jack *jack; bool micbias_resume_enable; -- 2.17.1

4 5

[PATCHv2] hda/hdmi: Register with vga_switcheroo on Dual GPU Macbooks
by Orlando Chamberlain 16 Feb '23

16 Feb '23

Commit 586bc4aab878 ("ALSA: hda/hdmi - fix vgaswitcheroo detection for AMD") caused only AMD gpu's with PX to have their audio component register with vga_switcheroo. This meant that Apple Macbooks with apple-gmux as the gpu switcher no longer had the audio client registering, so when the gpu is powered off by vga_switcheroo snd_hda_intel is unaware that it should have suspended the device: amdgpu: switched off snd_hda_intel 0000:03:00.1: Unable to change power state from D3hot to D0, device inaccessible snd_hda_intel 0000:03:00.1: CORB reset timeout#2, CORBRP = 65535 To resolve this, we use apple_gmux_detect() and register a vga_switcheroo audio client when apple-gmux is detected. Fixes: 586bc4aab878 ("ALSA: hda/hdmi - fix vgaswitcheroo detection for AMD") Link: https://lore.kernel.org/all/20230210044826.9834-9-orlandoch.dev@gmail.com/ Signed-off-by: Orlando Chamberlain <orlandoch.dev(a)gmail.com> --- v1->v2: Use apple_gmux_detect(), split this patch out of the patch series sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 87002670c0c9..cfd2ddfde112 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -50,6 +50,7 @@ #include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> +#include <linux/apple-gmux.h> #include <linux/firmware.h> #include <sound/hda_codec.h> #include "hda_controller.h" @@ -1463,7 +1464,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) * vgaswitcheroo. */ if (((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) && - atpx_present()) + (atpx_present() || apple_gmux_detect(NULL, NULL))) return p; pci_dev_put(p); } -- 2.39.1

2 1