Commit d001e978 authored by Takashi Iwai's avatar Takashi Iwai
Browse files

Merge tag 'asoc-fix-v6.10-merge-window' of... 

Merge tag 'asoc-fix-v6.10-merge-window' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.10

A bunch of fixes that came in during the merge window, all driver
specific and none of them especially remarkable.
parents b3b6f125 737ce4fb

Documentation/devicetree/bindings/sound/st,stm32-sai.yaml

+1 −1
Original line number Diff line number Diff line
@@ -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

include/sound/tas2781-dsp.h

+3 −4
Original line number Diff line number Diff line
@@ -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 <kevin-lu@ti.com>

//



#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);

sound/soc/codecs/es8326.c

+3 −3
Original line number Diff line number Diff line
@@ -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);

sound/soc/codecs/rt5645.c

+1 −0
Original line number Diff line number Diff line
@@ -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[] = {

sound/soc/codecs/tas2552.c

+13 −2
Original line number Diff line number Diff line
@@ -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,

	},

};