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

[PATCH] ASoC: nau8821: Implement DRC controls
by Seven Lee 31 Jan '23

31 Jan '23

This patch is support dynamic range compression controls. Signed-off-by: Seven Lee <scott6986(a)gmail.com> --- sound/soc/codecs/nau8821.c | 96 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8821.h | 23 +++++++++ 2 files changed, 119 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 4a72b94e8410..fee970427a24 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -322,12 +322,92 @@ static const struct soc_enum nau8821_dac_oversampl_enum = SOC_ENUM_SINGLE(NAU8821_R2C_DAC_CTRL1, NAU8821_DAC_OVERSAMPLE_SFT, ARRAY_SIZE(nau8821_dac_oversampl), nau8821_dac_oversampl); +static const char * const nau8821_adc_drc_noise_gate[] = { + "1:1", "2:1", "4:1", "8:1" }; + +static const struct soc_enum nau8821_adc_drc_noise_gate_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, NAU8821_DRC_NG_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_adc_drc_noise_gate), + nau8821_adc_drc_noise_gate); + +static const char * const nau8821_adc_drc_expansion_slope[] = { + "1:1", "2:1", "4:1" }; + +static const struct soc_enum nau8821_adc_drc_expansion_slope_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, NAU8821_DRC_EXP_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_adc_drc_expansion_slope), + nau8821_adc_drc_expansion_slope); + +static const char * const nau8821_adc_drc_lower_region[] = { + "0", "1:2", "1:4", "1:8", "1:16", "", "", "1:1" }; + +static const struct soc_enum nau8821_adc_drc_lower_region_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, + NAU8821_DRC_CMP2_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_adc_drc_lower_region), + nau8821_adc_drc_lower_region); + +static const char * const nau8821_higher_region[] = { + "0", "1:2", "1:4", "1:8", "1:16", "", "", "1:1" }; + +static const struct soc_enum nau8821_higher_region_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, + NAU8821_DRC_CMP1_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_higher_region), + nau8821_higher_region); + +static const char * const nau8821_limiter_slope[] = { + "0", "1:2", "1:4", "1:8", "1:16", "1:32", "1:64", "1:1" }; + +static const struct soc_enum nau8821_limiter_slope_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, + NAU8821_DRC_LMT_SLP_ADC_SFT, ARRAY_SIZE(nau8821_limiter_slope), + nau8821_limiter_slope); + +static const char * const nau8821_detection_attack_time[] = { + "Ts", "3Ts", "7Ts", "15Ts", "31Ts", "63Ts", "127Ts", "255Ts", + "", "511Ts" }; + +static const struct soc_enum nau8821_detection_attack_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, + NAU8821_DRC_PK_COEF1_ADC_SFT, + ARRAY_SIZE(nau8821_detection_attack_time), + nau8821_detection_attack_time); + +static const char * const nau8821_detection_release_time[] = { + "63Ts", "127Ts", "255Ts", "511Ts", "1023Ts", "2047Ts", "4095Ts", + "8191Ts", "", "16383Ts" }; + +static const struct soc_enum nau8821_detection_release_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, + NAU8821_DRC_PK_COEF2_ADC_SFT, + ARRAY_SIZE(nau8821_detection_release_time), + nau8821_detection_release_time); + +static const char * const nau8821_attack_time[] = { + "Ts", "3Ts", "7Ts", "15Ts", "31Ts", "63Ts", "127Ts", "255Ts", + "511Ts", "1023Ts", "2047Ts", "4095Ts", "8191Ts" }; + +static const struct soc_enum nau8821_attack_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, NAU8821_DRC_ATK_ADC_SFT, + ARRAY_SIZE(nau8821_attack_time), nau8821_attack_time); + +static const char * const nau8821_decay_time[] = { + "63Ts", "127Ts", "255Ts", "511Ts", "1023Ts", "2047Ts", "4095Ts", + "8191Ts", "16383Ts", "32757Ts", "65535Ts" }; + +static const struct soc_enum nau8821_decay_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, NAU8821_DRC_DCY_ADC_SFT, + ARRAY_SIZE(nau8821_decay_time), nau8821_decay_time); + static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -6600, 2400); static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); static const DECLARE_TLV_DB_MINMAX(hp_vol_tlv, -900, 0); static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -6600, 50, 1); static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -7000, 2400); +static const DECLARE_TLV_DB_MINMAX(drc_knee4_tlv, -9800, -3500); +static const DECLARE_TLV_DB_MINMAX(drc_knee3_tlv, -8100, -1800); static const struct snd_kcontrol_new nau8821_controls[] = { SOC_DOUBLE_TLV("Mic Volume", NAU8821_R35_ADC_DGAIN_CTRL1, @@ -346,6 +426,22 @@ static const struct snd_kcontrol_new nau8821_controls[] = { SOC_DOUBLE_TLV("Headphone Crosstalk Volume", NAU8821_R2F_DAC_DGAIN_CTRL, 0, 8, 0xff, 0, crosstalk_vol_tlv), + SOC_SINGLE_TLV("ADC DRC KNEE4", NAU8821_R37_ADC_DRC_KNEE_IP34, + NAU8821_DRC_KNEE4_IP_ADC_SFT, 0x3f, 1, drc_knee4_tlv), + SOC_SINGLE_TLV("ADC DRC KNEE3", NAU8821_R37_ADC_DRC_KNEE_IP34, + NAU8821_DRC_KNEE3_IP_ADC_SFT, 0x3f, 1, drc_knee3_tlv), + + SOC_ENUM("ADC DRC Noise Gate", nau8821_adc_drc_noise_gate_enum), + SOC_ENUM("ADC DRC Expansion Slope", nau8821_adc_drc_expansion_slope_enum), + SOC_ENUM("ADC DRC Lower Region", nau8821_adc_drc_lower_region_enum), + SOC_ENUM("ADC DRC Higher Region", nau8821_higher_region_enum), + SOC_ENUM("ADC DRC Limiter Slope", nau8821_limiter_slope_enum), + SOC_ENUM("ADC DRC Peak Detection Attack Time", nau8821_detection_attack_time_enum), + SOC_ENUM("ADC DRC Peak Detection Release Time", nau8821_detection_release_time_enum), + SOC_ENUM("ADC DRC Attack Time", nau8821_attack_time_enum), + SOC_ENUM("ADC DRC Decay Time", nau8821_decay_time_enum), + SOC_SINGLE("DRC Enable Switch", NAU8821_R36_ADC_DRC_KNEE_IP12, + NAU8821_DRC_ENA_ADC_SFT, 1, 0), SOC_ENUM("ADC Decimation Rate", nau8821_adc_decimation_enum), SOC_ENUM("DAC Oversampling Rate", nau8821_dac_oversampl_enum), diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index c44251f54d48..d962293c218e 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -350,6 +350,29 @@ #define NAU8821_ADCL_CH_VOL_SFT 0 #define NAU8821_ADCL_CH_VOL_MASK 0xff +/* ADC_DRC_KNEE_IP12 (0x36) */ +#define NAU8821_DRC_ENA_ADC_SFT 15 +#define NAU8821_DRC_ENA_ADC_EN (0x1 << NAU8821_DRC_ENA_ADC_SFT) + +/* ADC_DRC_KNEE_IP34 (0x37) */ +#define NAU8821_DRC_KNEE4_IP_ADC_SFT 8 +#define NAU8821_DRC_KNEE4_IP_ADC_MASK (0xff << NAU8821_DRC_KNEE4_IP_ADC_SFT) +#define NAU8821_DRC_KNEE3_IP_ADC_SFT 0 +#define NAU8821_DRC_KNEE3_IP_ADC_MASK 0xff + +/* ADC_DRC_SLOPES (0x38) */ +#define NAU8821_DRC_NG_SLP_ADC_SFT 12 +#define NAU8821_DRC_EXP_SLP_ADC_SFT 9 +#define NAU8821_DRC_CMP2_SLP_ADC_SFT 6 +#define NAU8821_DRC_CMP1_SLP_ADC_SFT 3 +#define NAU8821_DRC_LMT_SLP_ADC_SFT 0 + +/* ADC_DRC_ATKDCY (0x39) */ +#define NAU8821_DRC_PK_COEF1_ADC_SFT 12 +#define NAU8821_DRC_PK_COEF2_ADC_SFT 8 +#define NAU8821_DRC_ATK_ADC_SFT 4 +#define NAU8821_DRC_DCY_ADC_SFT 0 + /* BIQ1_COF10 (0x4a) */ #define NAU8821_BIQ1_DAC_EN_SFT 3 #define NAU8821_BIQ1_DAC_EN_EN (0x1 << NAU8821_BIQ1_DAC_EN_SFT) -- 2.30.2

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[PATCH] [RFC] ALSA: hda: Fix the control element identification for multiple codecs
by Jaroslav Kysela 31 Jan '23

31 Jan '23

[This is a RFC for the discussion] Some motherboards have multiple HDA codecs connected to the serial bus. The current code may create multiple mixer controls with the almost identical identification. The current code use id.device field from the control element structure to store the codec address to avoid such clashes for multiple codecs. Unfortunately, the user space do not handle this correctly. For mixer controls, only name and index are used for the identifiers. This patch fixes this problem to compose the index using the codec address as an offset in case, when the control already exists. It is really unlikely that one codec will create 10 similar controls. This patch adds new kernel module parameter 'ctl_dev_id' to allow select the old behaviour, too. BugLink: https://github.com/alsa-project/alsa-lib/issues/294 BugLink: https://github.com/alsa-project/alsa-lib/issues/205 Fixes: 54d174031576 ("[ALSA] hda-codec - Fix connection list parsing") Fixes: 1afe206ab699 ("ALSA: hda - Try to find an empty control index when it's occupied") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> -- Discussion: There are several possibilities to handle the old behaviour - a kernel module parameter (proposed), a kernel configuration option or drop the old behaviour completely. --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_codec.c | 13 ++++++++++--- sound/pci/hda/hda_controller.c | 1 + sound/pci/hda/hda_controller.h | 1 + sound/pci/hda/hda_intel.c | 5 +++++ 5 files changed, 18 insertions(+), 3 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index eba23daf2c29..bbb7805e85d8 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -259,6 +259,7 @@ struct hda_codec { unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ unsigned int forced_resume:1; /* forced resume for jack */ unsigned int no_stream_clean_at_suspend:1; /* do not clean streams at suspend */ + unsigned int ctl_dev_id:1; /* old control element id build behaviour */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index edd653ece70d..8f20abc036bf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3390,7 +3390,12 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - if (addr > 0) + /* Do not use the id.device field for MIXER elements. + * This field is for real device numbers (like PCM) but codecs + * are hidden components from the user space view (unrelated + * to the mixer element identification). + */ + if (addr > 0 && codec->ctl_dev_id) kctl->id.device = addr; if (idx > 0) kctl->id.index = idx; @@ -3401,9 +3406,11 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, * the codec addr; if it still fails (or it's the * primary codec), then try another control index */ - if (!addr && codec->core.addr) + if (!addr && codec->core.addr) { addr = codec->core.addr; - else if (!idx && !knew->index) { + if (!codec->ctl_dev_id) + idx += 10 * addr; + } else if (!idx && !knew->index) { idx = find_empty_mixer_ctl_idx(codec, knew->name, 0); if (idx <= 0) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 0ff286b7b66b..083df287c1a4 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1231,6 +1231,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) continue; codec->jackpoll_interval = chip->jackpoll_interval; codec->beep_mode = chip->beep_mode; + codec->ctl_dev_id = chip->ctl_dev_id; codecs++; } } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index f5bf295eb830..8556031bcd68 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -124,6 +124,7 @@ struct azx { /* HD codec */ int codec_probe_mask; /* copied from probe_mask option */ unsigned int beep_mode; + bool ctl_dev_id; #ifdef CONFIG_SND_HDA_PATCH_LOADER const struct firmware *fw; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 87002670c0c9..2ef57dfa81f1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -119,6 +119,7 @@ static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif static bool dmic_detect = 1; +static bool ctl_dev_id[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -157,6 +158,8 @@ module_param(dmic_detect, bool, 0444); MODULE_PARM_DESC(dmic_detect, "Allow DSP driver selection (bypass this driver) " "(0=off, 1=on) (default=1); " "deprecated, use snd-intel-dspcfg.dsp_driver option instead"); +module_param_array(ctl_dev_id, bool, NULL, 0444); +MODULE_PARM_DESC(ctl_dev_id, "Use control device identifier (based on codec address)."); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2278,6 +2281,8 @@ static int azx_probe_continue(struct azx *chip) chip->beep_mode = beep_mode[dev]; #endif + chip->ctl_dev_id = ctl_dev_id[dev]; + /* create codec instances */ if (bus->codec_mask) { err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); -- 2.39.0

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[PATCH RESEND] ALSA: doc: Fix PCM interface section typos
by Miquel Raynal 31 Jan '23

31 Jan '23

Fix two mistakes in the PCM interface section: 1/ Members of the snd_pcm_hardware structure are channels_{min,max} and not channel_{min,max} (mind the 's'). 2/ Another sentence is incomplete as the reference to one structure member (period_bytes_max) is missing. There is no relevant 'Fixes:' tag to apply as both typos predate the Git era. Signed-off-by: Miquel Raynal <miquel.raynal(a)bootlin.com> --- Hello, I wrote and sent this patch in 2019 but I likely only Cc'd Jon and the doc ML, which might have not been enough, so just in case, here is a resend. Link: https://lore.kernel.org/linux-doc/20190829145512.3752-1-miquel.raynal@bootl… .../sound/kernel-api/writing-an-alsa-driver.rst | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 07a620c5ca74..5c9523b7d55c 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -1720,16 +1720,16 @@ Typically, you'll have a hardware descriptor as below: - ``rate_min`` and ``rate_max`` define the minimum and maximum sample rate. This should correspond somehow to ``rates`` bits. -- ``channel_min`` and ``channel_max`` define, as you might already +- ``channels_min`` and ``channels_max`` define, as you might already expected, the minimum and maximum number of channels. - ``buffer_bytes_max`` defines the maximum buffer size in bytes. There is no ``buffer_bytes_min`` field, since it can be calculated from the minimum period size and the minimum number of - periods. Meanwhile, ``period_bytes_min`` and define the minimum and - maximum size of the period in bytes. ``periods_max`` and - ``periods_min`` define the maximum and minimum number of periods in - the buffer. + periods. Meanwhile, ``period_bytes_min`` and ``period_bytes_max`` + define the minimum and maximum size of the period in bytes. + ``periods_max`` and ``periods_min`` define the maximum and minimum + number of periods in the buffer. The “period” is a term that corresponds to a fragment in the OSS world. The period defines the size at which a PCM interrupt is -- 2.34.1

3 2

[PATCH] ALSA: firewire-motu: fix unreleased lock warning in hwdep device
by Takashi Sakamoto 31 Jan '23

31 Jan '23

Smatch static analysis tool detects that acquired lock is not released in hwdep device when condition branch is passed due to no event. It is unlikely to occur, while fulfilling is preferable for better coding. Reported-by: Dan Carpenter <error27(a)gmail.com> Fixes: 634ec0b2906e ("ALSA: firewire-motu: notify event for parameter change in register DSP model") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-hwdep.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/firewire/motu/motu-hwdep.c b/sound/firewire/motu/motu-hwdep.c index a900fc0e7644..88d1f4b56e4b 100644 --- a/sound/firewire/motu/motu-hwdep.c +++ b/sound/firewire/motu/motu-hwdep.c @@ -87,6 +87,10 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, return -EFAULT; count = consumed; + } else { + spin_unlock_irq(&motu->lock); + + count = 0; } return count; -- 2.37.2

2 1

[PATCH] ALSA: firewire-lib: fix uninitialized local variable
by Takashi Sakamoto 31 Jan '23

31 Jan '23

The function local variable, curr_cycle_time is declared without initialization. When tracepoints event is not probed, it looks to be used as is. This commit fixes it. Fortunately, the value of local variable is not used unless the event is probed, thus this commit is for better coding. Reported-by: Dan Carpenter <error27(a)gmail.com> Fixes: fef4e61b0b7 ("ALSA: firewire-lib: extend tracepoints event including CYCLE_TIME of 1394 OHCI") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f7e1329efa0b..a13c0b408aad 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -910,7 +910,7 @@ static int generate_tx_packet_descs(struct amdtp_stream *s, struct pkt_desc *des unsigned int dbc = s->data_block_counter; unsigned int packet_index = s->packet_index; unsigned int queue_size = s->queue_size; - u32 curr_cycle_time; + u32 curr_cycle_time = 0; int i; int err; -- 2.37.2

2 1

[PATCH v2 0/6] Fix default DMIC gain on AMD PDM drivers
by Mario Limonciello 30 Jan '23

30 Jan '23

It's been reported that a number of laptops have a low volume level from the digital microphone compared to Windows. AMD offers a register that can adjust the gain for PDM which is not configured at maximum gain by default. To fix this change the default for all 3 drivers to raise the gain but also offer a module parameter. The module parameter can be used for debugging if the gain is too high on a given laptop. This is intentionally split into multiple patches for default and parameter so that if the default really does behave better universally we can bring it back to stable too later. v1->v2: * Add guards for parameter usage * More comments in commit messages 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 sound/soc/amd/ps/acp63.h | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 10 +++++++++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 10 +++++++++- sound/soc/amd/renoir/rn_acp3x.h | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 10 +++++++++- sound/soc/amd/yc/acp6x.h | 2 +- 6 files changed, 30 insertions(+), 6 deletions(-) -- 2.34.1

2 6

[PATCH 00/11] ASoC: topology: Fixes and cleanups
by Amadeusz Sławiński 30 Jan '23

30 Jan '23

Following is series of fixes and cleanups for core topology code. Few patches fixing various problems all around and few fixing function names. v2: - fix commit messages - change order of last two patches - as there is no code changes, add Reviewed-by from Ranjani Amadeusz Sławiński (11): 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 include/sound/soc-topology.h | 2 +- sound/soc/soc-topology.c | 183 ++++++++++++++--------------------- 2 files changed, 74 insertions(+), 111 deletions(-) -- 2.25.1

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[PATCH v2 1/4] ASoC: codecs: tas5720: split a tas5720_mute_soc_component() function
by Alexandru Ardelean 30 Jan '23

30 Jan '23

This is to be re-used in tas5720_mute() (which is part of the dai_ops) and also in the tas5720_fault_check_work() hook. The benefit here isn't too great (now). It's only when we add support for a new device with a slightly different regmap that this becomes more useful. Signed-off-by: Alexandru Ardelean <alex(a)shruggie.ro> --- Changelog v1 -> v2: - https://lore.kernel.org/alsa-devel/20230115193347.24190-1-alex@shruggie.ro/ - No change sound/soc/codecs/tas5720.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 3885c0bf0b01..0587c9bb3c4e 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -199,9 +199,8 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, return ret; } -static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) +static int tas5720_mute_soc_component(struct snd_soc_component *component, int mute) { - struct snd_soc_component *component = dai->component; int ret; ret = snd_soc_component_update_bits(component, TAS5720_DIGITAL_CTRL2_REG, @@ -214,6 +213,11 @@ static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) return 0; } +static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + return tas5720_mute_soc_component(dai->component, mute); +} + static void tas5720_fault_check_work(struct work_struct *work) { struct tas5720_data *tas5720 = container_of(work, struct tas5720_data, @@ -318,8 +322,7 @@ static int tas5720_codec_probe(struct snd_soc_component *component) expected_device_id, device_id); /* Set device to mute */ - ret = snd_soc_component_update_bits(component, TAS5720_DIGITAL_CTRL2_REG, - TAS5720_MUTE, TAS5720_MUTE); + ret = tas5720_mute_soc_component(component, 1); if (ret < 0) goto error_snd_soc_component_update_bits; -- 2.34.1

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[PATCH v3 00/11] ASoC: dt-bindings: audio-graph-port related update
by Kuninori Morimoto 30 Jan '23

30 Jan '23

Hi ASoC ML, DT ML Cc Geert These v3 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. This patch-set fixup some Renesas's "make dtbs_check". v2 -> v3 - add Acked-by / Reviewed-by - include missing v1 -> v2 - remove patch for Nvidia - tidyup ti,pcm3168a schema - maintainers - ports - explain more on git-log Link: https://lore.kernel.org/r/87358hj2ub.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87v8mepyoy.wl-kuninori.morimoto.gx@renesas.com 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: ak4613: enable Of-graph (Audio-Graph-Card) style ASoC: dt-bindings: renesas,rsnd: add missing playback/capture ASoC: dt-bindings: brenesas,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 .../devicetree/bindings/sound/ak4613.yaml | 7 ++ .../bindings/sound/audio-graph-port.yaml | 65 +++++++---- .../bindings/sound/renesas,rsnd.yaml | 31 +++-- .../bindings/sound/simple-card.yaml | 4 + .../devicetree/bindings/sound/ti,pcm3168a.txt | 56 --------- .../bindings/sound/ti,pcm3168a.yaml | 107 ++++++++++++++++++ 6 files changed, 184 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -- 2.25.1

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[PATCH 1/2] ASoC: dt-bindings: fsl-sai: Simplify the VFxxx dmas binding
by Marek Vasut 30 Jan '23

30 Jan '23

Get rid of the vf610 sai special case, instead update the vfxxx.dtsi DT to use the same DMA channel ordering as all the other devices. The sai DMA channel ordering has not been aligned with other IP DMA channel ordering in the vfxxx.dtsi anyway. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Adam Ford <aford173(a)gmail.com> Cc: Alexander Stein <alexander.stein(a)ew.tq-group.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Joakim Zhang <qiangqing.zhang(a)nxp.com> Cc: Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> Cc: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Lucas Stach <l.stach(a)pengutronix.de> Cc: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Markus Niebel <Markus.Niebel(a)ew.tq-group.com> Cc: NXP Linux Team <linux-imx(a)nxp.com> Cc: Paul Elder <paul.elder(a)ideasonboard.com> Cc: Peng Fan <peng.fan(a)nxp.com> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: Richard Cochran <richardcochran(a)gmail.com> Cc: Richard Zhu <hongxing.zhu(a)nxp.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Shawn Guo <shawnguo(a)kernel.org> Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> Cc: Stefan Agner <stefan(a)agner.ch> Cc: Tim Harvey <tharvey(a)gateworks.com> Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org To: linux-arm-kernel(a)lists.infradead.org --- .../devicetree/bindings/sound/fsl,sai.yaml | 38 ++++--------------- 1 file changed, 8 insertions(+), 30 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml index 7e56337d8edc1..088c26b001cc0 100644 --- a/Documentation/devicetree/bindings/sound/fsl,sai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -76,10 +76,14 @@ properties: minItems: 4 dmas: - maxItems: 2 + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX dma-names: - maxItems: 2 + items: + - const: rx + - const: tx interrupts: items: @@ -142,31 +146,6 @@ properties: allOf: - $ref: dai-common.yaml# - - if: - properties: - compatible: - contains: - const: fsl,vf610-sai - then: - properties: - dmas: - items: - - description: DMA controller phandle and request line for TX - - description: DMA controller phandle and request line for RX - dma-names: - items: - - const: tx - - const: rx - else: - properties: - dmas: - items: - - description: DMA controller phandle and request line for RX - - description: DMA controller phandle and request line for TX - dma-names: - items: - - const: rx - - const: tx - if: required: - fsl,sai-asynchronous @@ -199,9 +178,8 @@ examples: <&clks VF610_CLK_SAI2>, <&clks 0>, <&clks 0>; clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 21>, - <&edma0 0 20>; + dma-names = "rx", "tx"; + dmas = <&edma0 0 20>, <&edma0 0 21>; big-endian; lsb-first; }; -- 2.39.0

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