From 04b5e2f9a75a3f33f29dec780c1363367642fd73 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 13 May 2024 17:07:30 +0300 Subject: [PATCH 1/6] ASoC: Intel: sof_sdw_rt_sdca_jack_common: Use name_prefix for `-sdca` detection Match against the correct string to decide to add the '-sdca' postfix: instead of codec_dai->name the correct one is component->name_prefix. The component->name_prefix is added previously to the card->components as hs. Fixes: 9a9d31b149f3 ("ASoC: Intel: sof_sdw_rt_sdca_jack_common: remove -sdca for new codecs") Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240513140730.27048-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) 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); From 1f900475314ef258af1a4c11bc9096fe2ffe263f Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Tue, 14 May 2024 02:31:22 +0000 Subject: [PATCH 2/6] ASoC: rt5645: mic-in detection threshold modification Modify mic-in detection threshold for better performance. Signed-off-by: Jack Yu Link: https://msgid.link/r/b7614d9e38054aa6ad8efa620edb4162@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) 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[] = { From 714f5df027b085c19c32af6f08a959bf35b9fb7c Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Wed, 15 May 2024 14:25:17 +0800 Subject: [PATCH 3/6] ASoC: codecs: ES8326: solve hp and button detect issue We got an error report about headphone type detection and button detection. We fixed the headphone type detection error by adjusting the condition of setting es8326->hp to 0.And we fixed the button detection error by adjusting micbias and vref. Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240515062517.23661-1-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 6 +++--- 1 file changed, 3 insertions(+), 3 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); From 7078ac4fd179a68d0bab448004fcd357e7a45f8d Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sat, 18 May 2024 11:35:15 +0800 Subject: [PATCH 4/6] ASoC: tas2552: Add TX path for capturing AUDIO-OUT data TAS2552 is a Smartamp with I/V sense data, add TX path to support capturing I/V data. Fixes: 38803ce7b53b ("ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()") Signed-off-by: Shenghao Ding Link: https://msgid.link/r/20240518033515.866-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) 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 */ @@ -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, }, }; From b195acf5266d2dee4067f89345c3e6b88d925311 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sat, 18 May 2024 22:15:46 +0800 Subject: [PATCH 5/6] ASoC: tas2781: Fix wrong loading calibrated data sequence Calibrated data will be set to default after loading DSP config params, which will cause speaker protection work abnormally. Reload calibrated data after loading DSP config params. Remove declaration of unused API which load calibrated data in wrong sequence, changed the copyright year and correct file name in license header. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding Link: https://msgid.link/r/20240518141546.1742-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781-dsp.h | 7 +- sound/soc/codecs/tas2781-fmwlib.c | 103 ++++++++---------------------- sound/soc/codecs/tas2781-i2c.c | 4 +- 3 files changed, 32 insertions(+), 82 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..7fba7ea26a4b 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -13,8 +13,8 @@ // Author: Kevin Lu // -#ifndef __TASDEVICE_DSP_H__ -#define __TASDEVICE_DSP_H__ +#ifndef __TAS2781_DSP_H__ +#define __TAS2781_DSP_H__ #define MAIN_ALL_DEVICES 0x0d #define MAIN_DEVICE_A 0x01 @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); 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 @@ int tasdevice_prmg_load(void *context, int prm_no) } 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) { From 737ce4fb96206f999ddea7530145fc0e8abd5d31 Mon Sep 17 00:00:00 2001 From: "Rob Herring (Arm)" Date: Mon, 20 May 2024 17:27:05 -0500 Subject: [PATCH 6/6] ASoC: dt-bindings: stm32: Ensure compatible pattern matches whole string The compatible pattern "st,stm32-sai-sub-[ab]" is missing starting and ending anchors, so any prefix and/or suffix would still be valid. This also fixes a warning on the example: Documentation/devicetree/bindings/sound/st,stm32-sai.example.dtb: /example-0/sai@4400b000/audio-controller@4400b004: failed to match any schema with compatible: ['st,stm32-sai-sub-a'] Signed-off-by: Rob Herring (Arm) Link: https://msgid.link/r/20240520222705.1742367-1-robh@kernel.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/st,stm32-sai.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index 59df8a832310..f555ccd6b00a 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -68,7 +68,7 @@ patternProperties: properties: compatible: description: Compatible for SAI sub-block A or B. - pattern: "st,stm32-sai-sub-[ab]" + pattern: "^st,stm32-sai-sub-[ab]$" "#sound-dai-cells": const: 0