diff options
author | Linus Torvalds | 2024-05-24 08:48:51 -0700 |
---|---|---|
committer | Linus Torvalds | 2024-05-24 08:48:51 -0700 |
commit | 041c9f71a47b8b98f6bdbbf4c0312f9782ca9a70 (patch) | |
tree | 93fd6c207323aa6d2e11becaa703ee8ad60d5d90 /sound/soc | |
parent | e292ead0c9dad3580cfd45693a59902c8d31a0a7 (diff) | |
parent | d001e978c1c45b25d823489171151d13fd28ef4e (diff) |
Merge tag 'sound-fix-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes for 6.10-rc1. Most of changes are various
device-specific fixes and quirks, while there are a few small changes
in ALSA core timer and module / built-in fixes"
* tag 'sound-fix-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek: fix mute/micmute LEDs don't work for ProBook 440/460 G11.
ALSA: core: Enable proc module when CONFIG_MODULES=y
ALSA: core: Fix NULL module pointer assignment at card init
ALSA: hda/realtek: Enable headset mic of JP-IK LEAP W502 with ALC897
ASoC: dt-bindings: stm32: Ensure compatible pattern matches whole string
ASoC: tas2781: Fix wrong loading calibrated data sequence
ASoC: tas2552: Add TX path for capturing AUDIO-OUT data
ALSA: usb-audio: Fix for sampling rates support for Mbox3
Documentation: sound: Fix trailing whitespaces
ALSA: timer: Set lower bound of start tick time
ASoC: codecs: ES8326: solve hp and button detect issue
ASoC: rt5645: mic-in detection threshold modification
ASoC: Intel: sof_sdw_rt_sdca_jack_common: Use name_prefix for `-sdca` detection
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/codecs/es8326.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/rt5645.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/tas2552.c | 15 | ||||
-rw-r--r-- | sound/soc/codecs/tas2781-fmwlib.c | 103 | ||||
-rw-r--r-- | sound/soc/codecs/tas2781-i2c.c | 4 | ||||
-rw-r--r-- | sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 2 |
6 files changed, 47 insertions, 84 deletions
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 833ea52638ab..03b539ba540f 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -829,8 +829,8 @@ static void es8326_jack_detect_handler(struct work_struct *work) /* mute adc when mic path switch */ regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); - es8326->hp = 0; } + es8326->hp = 0; regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a); regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03); @@ -981,7 +981,7 @@ static int es8326_resume(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); usleep_range(10000, 15000); regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9); - regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb); + regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xd8); /* set headphone default type and detect pin */ regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83); regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05); @@ -1018,7 +1018,7 @@ static int es8326_resume(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F); /* select vdda as micbias source */ - regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x23); + regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x03); /* set dac dsmclip = 1 */ regmap_write(es8326->regmap, ES8326_DAC_DSM, 0x08); regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15); diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 05f574bf8b8f..cdb7ff7020e9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -81,6 +81,7 @@ static const struct reg_sequence init_list[] = { static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, {RT5645_PWR_ANLG1, 0x02}, + {RT5645_IL_CMD3, 0x0018}, }; static const struct reg_default rt5645_reg[] = { diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 40f5f27e74c0..a7ed59ec49a6 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -2,7 +2,8 @@ /* * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier * - * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com + * Copyright (C) 2014 - 2024 Texas Instruments Incorporated - + * https://www.ti.com * * Author: Dan Murphy <dmurphy@ti.com> */ @@ -119,12 +120,14 @@ static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] = &tas2552_input_mux_control), SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "DAC Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0), SND_SOC_DAPM_POST("Post Event", tas2552_post_event), - SND_SOC_DAPM_OUTPUT("OUT") + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") }; static const struct snd_soc_dapm_route tas2552_audio_map[] = { @@ -134,6 +137,7 @@ static const struct snd_soc_dapm_route tas2552_audio_map[] = { {"ClassD", NULL, "Input selection"}, {"OUT", NULL, "ClassD"}, {"ClassD", NULL, "PLL"}, + {"ASI OUT", NULL, "DMIC"} }; #ifdef CONFIG_PM @@ -538,6 +542,13 @@ static struct snd_soc_dai_driver tas2552_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = TAS2552_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, .ops = &tas2552_speaker_dai_ops, }, }; diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index a6be81adcb83..265a8ca25cbb 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2151,6 +2151,24 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, return ret; } +static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) +{ + struct tasdevice_calibration *cal; + struct tasdevice_fw *cal_fmw; + + cal_fmw = priv->tasdevice[i].cali_data_fmw; + + /* No calibrated data for current devices, playback will go ahead. */ + if (!cal_fmw) + return; + + cal = cal_fmw->calibrations; + if (cal) + return; + + load_calib_data(priv, &cal->dev_data); +} + int tasdevice_select_tuningprm_cfg(void *context, int prm_no, int cfg_no, int rca_conf_no) { @@ -2210,21 +2228,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, for (i = 0; i < tas_priv->ndev; i++) { if (tas_priv->tasdevice[i].is_loaderr == true) continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration - *cal = cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } + if (tas_priv->tasdevice[i].is_loaderr == false && + tas_priv->tasdevice[i].is_loading == true) tas_priv->tasdevice[i].cur_prog = prm_no; - } } } @@ -2245,11 +2251,15 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tasdevice_load_data(tas_priv, &(conf->dev_data)); for (i = 0; i < tas_priv->ndev; i++) { if (tas_priv->tasdevice[i].is_loaderr == true) { - status |= 1 << (i + 4); + status |= BIT(i + 4); continue; - } else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) + } + + if (tas_priv->tasdevice[i].is_loaderr == false && + tas_priv->tasdevice[i].is_loading == true) { + tasdev_load_calibrated_data(tas_priv, i); tas_priv->tasdevice[i].cur_conf = cfg_no; + } } } else dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", @@ -2308,65 +2318,6 @@ out: } EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); -int tasdevice_prmg_calibdata_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - int prog_status = 0; - int i; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - goto out; - } - - if (prm_no >= tas_fmw->nr_programs) { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - goto out; - } - - for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) { - tas_priv->tasdevice[i].cur_conf = -1; - tas_priv->tasdevice[i].is_loading = true; - prog_status++; - } - tas_priv->tasdevice[i].is_loaderr = false; - } - - if (prog_status) { - program = &(tas_fmw->programs[prm_no]); - tasdevice_load_data(tas_priv, &(program->dev_data)); - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].is_loaderr == true) - continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration *cal = - cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } - tas_priv->tasdevice[i].cur_prog = prm_no; - } - } - } - -out: - return prog_status; -} -EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, - SND_SOC_TAS2781_FMWLIB); - void tasdevice_tuning_switch(void *context, int state) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b5abff230e43..9350972dfefe 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, __func__, tas_priv->cal_binaryname[i]); } - tasdevice_prmg_calibdata_load(tas_priv, 0); + tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 701b0372f59e..012195c50519 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -109,7 +109,7 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *d return -ENOMEM; for (i = 0; i < ARRAY_SIZE(need_sdca_suffix); i++) { - if (strstr(codec_dai->name, need_sdca_suffix[i])) { + if (strstr(component->name_prefix, need_sdca_suffix[i])) { /* Add -sdca suffix for existing UCMs */ card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s-sdca", card->components); |