Commit 254d96be authored by Mark Brown's avatar Mark Brown

Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/drm',...

Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/drm', 'asoc/topic/dwc' and 'asoc/topic/es8316' into asoc-next
...@@ -78,6 +78,7 @@ graph bindings specified in Documentation/devicetree/bindings/graph.txt. ...@@ -78,6 +78,7 @@ graph bindings specified in Documentation/devicetree/bindings/graph.txt.
remote endpoint phandle should be a reference to a valid mipi_dsi_host device remote endpoint phandle should be a reference to a valid mipi_dsi_host device
node. node.
- Video port 1 for the HDMI output - Video port 1 for the HDMI output
- Audio port 2 for the HDMI audio input
Example Example
...@@ -112,5 +113,12 @@ Example ...@@ -112,5 +113,12 @@ Example
remote-endpoint = <&hdmi_connector_in>; remote-endpoint = <&hdmi_connector_in>;
}; };
}; };
port@2 {
reg = <2>;
codec_endpoint: endpoint {
remote-endpoint = <&i2s0_cpu_endpoint>;
};
};
}; };
}; };
...@@ -25,7 +25,8 @@ Required properties: ...@@ -25,7 +25,8 @@ Required properties:
- clock-names: Shall contain "iahb" and "isfr" as defined in dw_hdmi.txt. - clock-names: Shall contain "iahb" and "isfr" as defined in dw_hdmi.txt.
- ports: See dw_hdmi.txt. The DWC HDMI shall have one port numbered 0 - ports: See dw_hdmi.txt. The DWC HDMI shall have one port numbered 0
corresponding to the video input of the controller and one port numbered 1 corresponding to the video input of the controller and one port numbered 1
corresponding to its HDMI output. Each port shall have a single endpoint. corresponding to its HDMI output, and one port numbered 2 corresponding to
sound input of the controller. Each port shall have a single endpoint.
Optional properties: Optional properties:
...@@ -59,6 +60,12 @@ Example: ...@@ -59,6 +60,12 @@ Example:
remote-endpoint = <&hdmi0_con>; remote-endpoint = <&hdmi0_con>;
}; };
}; };
port@2 {
reg = <2>;
rcar_dw_hdmi0_sound_in: endpoint {
remote-endpoint = <&hdmi_sound_out>;
};
};
}; };
}; };
......
...@@ -11,6 +11,7 @@ ...@@ -11,6 +11,7 @@
#include <sound/hdmi-codec.h> #include <sound/hdmi-codec.h>
#include <sound/pcm.h> #include <sound/pcm.h>
#include <sound/soc.h> #include <sound/soc.h>
#include <linux/of_graph.h>
#include "adv7511.h" #include "adv7511.h"
...@@ -182,10 +183,31 @@ static void audio_shutdown(struct device *dev, void *data) ...@@ -182,10 +183,31 @@ static void audio_shutdown(struct device *dev, void *data)
{ {
} }
static int adv7511_hdmi_i2s_get_dai_id(struct snd_soc_component *component,
struct device_node *endpoint)
{
struct of_endpoint of_ep;
int ret;
ret = of_graph_parse_endpoint(endpoint, &of_ep);
if (ret < 0)
return ret;
/*
* HDMI sound should be located as reg = <2>
* Then, it is sound port 0
*/
if (of_ep.port == 2)
return 0;
return -EINVAL;
}
static const struct hdmi_codec_ops adv7511_codec_ops = { static const struct hdmi_codec_ops adv7511_codec_ops = {
.hw_params = adv7511_hdmi_hw_params, .hw_params = adv7511_hdmi_hw_params,
.audio_shutdown = audio_shutdown, .audio_shutdown = audio_shutdown,
.audio_startup = audio_startup, .audio_startup = audio_startup,
.get_dai_id = adv7511_hdmi_i2s_get_dai_id,
}; };
static struct hdmi_codec_pdata codec_data = { static struct hdmi_codec_pdata codec_data = {
......
...@@ -82,9 +82,30 @@ static void dw_hdmi_i2s_audio_shutdown(struct device *dev, void *data) ...@@ -82,9 +82,30 @@ static void dw_hdmi_i2s_audio_shutdown(struct device *dev, void *data)
hdmi_write(audio, HDMI_AUD_CONF0_SW_RESET, HDMI_AUD_CONF0); hdmi_write(audio, HDMI_AUD_CONF0_SW_RESET, HDMI_AUD_CONF0);
} }
static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component,
struct device_node *endpoint)
{
struct of_endpoint of_ep;
int ret;
ret = of_graph_parse_endpoint(endpoint, &of_ep);
if (ret < 0)
return ret;
/*
* HDMI sound should be located as reg = <2>
* Then, it is sound port 0
*/
if (of_ep.port == 2)
return 0;
return -EINVAL;
}
static struct hdmi_codec_ops dw_hdmi_i2s_ops = { static struct hdmi_codec_ops dw_hdmi_i2s_ops = {
.hw_params = dw_hdmi_i2s_hw_params, .hw_params = dw_hdmi_i2s_hw_params,
.audio_shutdown = dw_hdmi_i2s_audio_shutdown, .audio_shutdown = dw_hdmi_i2s_audio_shutdown,
.get_dai_id = dw_hdmi_i2s_get_dai_id,
}; };
static int snd_dw_hdmi_probe(struct platform_device *pdev) static int snd_dw_hdmi_probe(struct platform_device *pdev)
......
...@@ -47,6 +47,7 @@ struct i2s_platform_data { ...@@ -47,6 +47,7 @@ struct i2s_platform_data {
#define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0)
#define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1)
#define DW_I2S_QUIRK_16BIT_IDX_OVERRIDE (1 << 2)
unsigned int quirks; unsigned int quirks;
unsigned int i2s_reg_comp1; unsigned int i2s_reg_comp1;
unsigned int i2s_reg_comp2; unsigned int i2s_reg_comp2;
......
...@@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS ...@@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_DA9055 if I2C select SND_SOC_DA9055 if I2C
select SND_SOC_DIO2125 select SND_SOC_DIO2125
select SND_SOC_DMIC select SND_SOC_DMIC
select SND_SOC_ES8316 if I2C
select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_SPI if SPI_MASTER
select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES8328_I2C if I2C
select SND_SOC_ES7134 select SND_SOC_ES7134
...@@ -543,6 +544,10 @@ config SND_SOC_HDMI_CODEC ...@@ -543,6 +544,10 @@ config SND_SOC_HDMI_CODEC
config SND_SOC_ES7134 config SND_SOC_ES7134
tristate "Everest Semi ES7134 CODEC" tristate "Everest Semi ES7134 CODEC"
config SND_SOC_ES8316
tristate "Everest Semi ES8316 CODEC"
depends on I2C
config SND_SOC_ES8328 config SND_SOC_ES8328
tristate tristate
......
...@@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o ...@@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o snd-soc-da9055-objs := da9055.o
snd-soc-dmic-objs := dmic.o snd-soc-dmic-objs := dmic.o
snd-soc-es7134-objs := es7134.o snd-soc-es7134-objs := es7134.o
snd-soc-es8316-objs := es8316.o
snd-soc-es8328-objs := es8328.o snd-soc-es8328-objs := es8328.o
snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-i2c-objs := es8328-i2c.o
snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-es8328-spi-objs := es8328-spi.o
...@@ -300,6 +301,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o ...@@ -300,6 +301,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o
obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o
obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o
obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
......
/*
* es8316.c -- es8316 ALSA SoC audio driver
* Copyright Everest Semiconductor Co.,Ltd
*
* Authors: David Yang <yangxiaohua@everest-semi.com>,
* Daniel Drake <drake@endlessm.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/acpi.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/mod_devicetable.h>
#include <linux/regmap.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include "es8316.h"
/* In slave mode at single speed, the codec is documented as accepting 5
* MCLK/LRCK ratios, but we also add ratio 400, which is commonly used on
* Intel Cherry Trail platforms (19.2MHz MCLK, 48kHz LRCK).
*/
#define NR_SUPPORTED_MCLK_LRCK_RATIOS 6
static const unsigned int supported_mclk_lrck_ratios[] = {
256, 384, 400, 512, 768, 1024
};
struct es8316_priv {
unsigned int sysclk;
unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS];
struct snd_pcm_hw_constraint_list sysclk_constraints;
};
/*
* ES8316 controls
*/
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9600, 50, 1);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(0, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(250, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(450, 0, 0),
4, 4, TLV_DB_SCALE_ITEM(700, 0, 0),
5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0),
6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0),
7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0),
8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0),
9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0),
10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0),
);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv,
0, 0, TLV_DB_SCALE_ITEM(-4800, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(-2400, 1200, 0),
);
static const char * const ng_type_txt[] =
{ "Constant PGA Gain", "Mute ADC Output" };
static const struct soc_enum ng_type =
SOC_ENUM_SINGLE(ES8316_ADC_ALC_NG, 6, 2, ng_type_txt);
static const char * const adcpol_txt[] = { "Normal", "Invert" };
static const struct soc_enum adcpol =
SOC_ENUM_SINGLE(ES8316_ADC_MUTE, 1, 2, adcpol_txt);
static const char *const dacpol_txt[] =
{ "Normal", "R Invert", "L Invert", "L + R Invert" };
static const struct soc_enum dacpol =
SOC_ENUM_SINGLE(ES8316_DAC_SET1, 0, 4, dacpol_txt);
static const struct snd_kcontrol_new es8316_snd_controls[] = {
SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL,
4, 0, 3, 1, hpout_vol_tlv),
SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL,
0, 4, 7, 0, hpmixer_gain_tlv),
SOC_ENUM("Playback Polarity", dacpol),
SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
ES8316_DAC_VOLR, 0, 0xc0, 1, dac_vol_tlv),
SOC_SINGLE("DAC Soft Ramp Switch", ES8316_DAC_SET1, 4, 1, 1),
SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0),
SOC_SINGLE("DAC Notch Filter Switch", ES8316_DAC_SET2, 6, 1, 0),
SOC_SINGLE("DAC Double Fs Switch", ES8316_DAC_SET2, 7, 1, 0),
SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0),
SOC_ENUM("Capture Polarity", adcpol),
SOC_SINGLE("Mic Boost Switch", ES8316_ADC_D2SEPGA, 0, 1, 0),
SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME,
0, 0xc0, 1, adc_vol_tlv),
SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8316_ADC_PGAGAIN,
4, 10, 0, adc_pga_gain_tlv),
SOC_SINGLE("ADC Soft Ramp Switch", ES8316_ADC_MUTE, 4, 1, 0),
SOC_SINGLE("ADC Double Fs Switch", ES8316_ADC_DMIC, 4, 1, 0),
SOC_SINGLE("ALC Capture Switch", ES8316_ADC_ALC1, 6, 1, 0),
SOC_SINGLE_TLV("ALC Capture Max Volume", ES8316_ADC_ALC1, 0, 28, 0,
alc_max_gain_tlv),
SOC_SINGLE_TLV("ALC Capture Min Volume", ES8316_ADC_ALC2, 0, 28, 0,
alc_min_gain_tlv),
SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 10, 0,
alc_target_tlv),
SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0),
SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0),
SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4, 0, 10, 0),
SOC_SINGLE("ALC Capture Noise Gate Switch", ES8316_ADC_ALC_NG,
5, 1, 0),
SOC_SINGLE("ALC Capture Noise Gate Threshold", ES8316_ADC_ALC_NG,
0, 31, 0),
SOC_ENUM("ALC Capture Noise Gate Type", ng_type),
};
/* Analog Input Mux */
static const char * const es8316_analog_in_txt[] = {
"lin1-rin1",
"lin2-rin2",
"lin1-rin1 with 20db Boost",
"lin2-rin2 with 20db Boost"
};
static const unsigned int es8316_analog_in_values[] = { 0, 1, 2, 3 };
static const struct soc_enum es8316_analog_input_enum =
SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL, 4, 3,
ARRAY_SIZE(es8316_analog_in_txt),
es8316_analog_in_txt,
es8316_analog_in_values);
static const struct snd_kcontrol_new es8316_analog_in_mux_controls =
SOC_DAPM_ENUM("Route", es8316_analog_input_enum);
static const char * const es8316_dmic_txt[] = {
"dmic disable",
"dmic data at high level",
"dmic data at low level",
};
static const unsigned int es8316_dmic_values[] = { 0, 1, 2 };
static const struct soc_enum es8316_dmic_src_enum =
SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3,
ARRAY_SIZE(es8316_dmic_txt),
es8316_dmic_txt,
es8316_dmic_values);
static const struct snd_kcontrol_new es8316_dmic_src_controls =
SOC_DAPM_ENUM("Route", es8316_dmic_src_enum);
/* hp mixer mux */
static const char * const es8316_hpmux_texts[] = {
"lin1-rin1",
"lin2-rin2",
"lin-rin with Boost",
"lin-rin with Boost and PGA"
};
static const unsigned int es8316_hpmux_values[] = { 0, 1, 2, 3 };
static SOC_ENUM_SINGLE_DECL(es8316_left_hpmux_enum, ES8316_HPMIX_SEL,
4, es8316_hpmux_texts);
static const struct snd_kcontrol_new es8316_left_hpmux_controls =
SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum);
static SOC_ENUM_SINGLE_DECL(es8316_right_hpmux_enum, ES8316_HPMIX_SEL,
0, es8316_hpmux_texts);
static const struct snd_kcontrol_new es8316_right_hpmux_controls =
SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum);
/* headphone Output Mixer */
static const struct snd_kcontrol_new es8316_out_left_mix[] = {
SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH, 6, 1, 0),
SOC_DAPM_SINGLE("Left DAC Switch", ES8316_HPMIX_SWITCH, 7, 1, 0),
};
static const struct snd_kcontrol_new es8316_out_right_mix[] = {
SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH, 2, 1, 0),
SOC_DAPM_SINGLE("Right DAC Switch", ES8316_HPMIX_SWITCH, 3, 1, 0),
};
/* DAC data source mux */
static const char * const es8316_dacsrc_texts[] = {
"LDATA TO LDAC, RDATA TO RDAC",
"LDATA TO LDAC, LDATA TO RDAC",
"RDATA TO LDAC, RDATA TO RDAC",
"RDATA TO LDAC, LDATA TO RDAC",
};
static const unsigned int es8316_dacsrc_values[] = { 0, 1, 2, 3 };
static SOC_ENUM_SINGLE_DECL(es8316_dacsrc_mux_enum, ES8316_DAC_SET1,
6, es8316_dacsrc_texts);
static const struct snd_kcontrol_new es8316_dacsrc_mux_controls =
SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum);
static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("Bias", ES8316_SYS_PDN, 3, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Analog power", ES8316_SYS_PDN, 4, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Mic Bias", ES8316_SYS_PDN, 5, 1, NULL, 0),
SND_SOC_DAPM_INPUT("DMIC"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
/* Input Mux */
SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
&es8316_analog_in_mux_controls),
SND_SOC_DAPM_SUPPLY("ADC Vref", ES8316_SYS_PDN, 1, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("ADC bias", ES8316_SYS_PDN, 2, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("ADC Clock", ES8316_CLKMGR_CLKSW, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Line input PGA", ES8316_ADC_PDN_LINSEL,
7, 1, NULL, 0),
SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL, 6, 1),
SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0,
&es8316_dmic_src_controls),
/* Digital Interface */
SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 1,
ES8316_SERDATA_ADC, 6, 1),
SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("DAC Source Mux", SND_SOC_NOPM, 0, 0,
&es8316_dacsrc_mux_controls),
SND_SOC_DAPM_SUPPLY("DAC Vref", ES8316_SYS_PDN, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("DAC Clock", ES8316_CLKMGR_CLKSW, 2, 0, NULL, 0),
SND_SOC_DAPM_DAC("Right DAC", NULL, ES8316_DAC_PDN, 0, 1),
SND_SOC_DAPM_DAC("Left DAC", NULL, ES8316_DAC_PDN, 4, 1),
/* Headphone Output Side */
SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
&es8316_left_hpmux_controls),
SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
&es8316_right_hpmux_controls),
SND_SOC_DAPM_MIXER("Left Headphone Mixer", ES8316_HPMIX_PDN,
5, 1, &es8316_out_left_mix[0],
ARRAY_SIZE(es8316_out_left_mix)),
SND_SOC_DAPM_MIXER("Right Headphone Mixer", ES8316_HPMIX_PDN,
1, 1, &es8316_out_right_mix[0],
ARRAY_SIZE(es8316_out_right_mix)),
SND_SOC_DAPM_PGA("Left Headphone Mixer Out", ES8316_HPMIX_PDN,
4, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right Headphone Mixer Out", ES8316_HPMIX_PDN,
0, 1, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Left Headphone Charge Pump", ES8316_CPHP_OUTEN,
6, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Right Headphone Charge Pump", ES8316_CPHP_OUTEN,
2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8316_CPHP_PDN2,
5, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Headphone Charge Pump Clock", ES8316_CLKMGR_CLKSW,
4, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Left Headphone Driver", ES8316_CPHP_OUTEN,
5, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Right Headphone Driver", ES8316_CPHP_OUTEN,
1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Headphone Out", ES8316_CPHP_PDN1, 2, 1, NULL, 0),
/* pdn_Lical and pdn_Rical bits are documented as Reserved, but must
* be explicitly unset in order to enable HP output
*/
SND_SOC_DAPM_SUPPLY("Left Headphone ical", ES8316_CPHP_ICAL_VOL,
7, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Right Headphone ical", ES8316_CPHP_ICAL_VOL,
3, 1, NULL, 0),
SND_SOC_DAPM_OUTPUT("HPOL"),
SND_SOC_DAPM_OUTPUT("HPOR"),
};
static const struct snd_soc_dapm_route es8316_dapm_routes[] = {
/* Recording */
{"MIC1", NULL, "Mic Bias"},
{"MIC2", NULL, "Mic Bias"},
{"MIC1", NULL, "Bias"},
{"MIC2", NULL, "Bias"},
{"MIC1", NULL, "Analog power"},
{"MIC2", NULL, "Analog power"},
{"Differential Mux", "lin1-rin1", "MIC1"},
{"Differential Mux", "lin2-rin2", "MIC2"},
{"Line input PGA", NULL, "Differential Mux"},
{"Mono ADC", NULL, "ADC Clock"},
{"Mono ADC", NULL, "ADC Vref"},
{"Mono ADC", NULL, "ADC bias"},
{"Mono ADC", NULL, "Line input PGA"},
/* It's not clear why, but to avoid recording only silence,
* the DAC clock must be running for the ADC to work.
*/
{"Mono ADC", NULL, "DAC Clock"},
{"Digital Mic Mux", "dmic disable", "Mono ADC"},
{"I2S OUT", NULL, "Digital Mic Mux"},
/* Playback */
{"DAC Source Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"},
{"Left DAC", NULL, "DAC Clock"},
{"Right DAC", NULL, "DAC Clock"},
{"Left DAC", NULL, "DAC Vref"},
{"Right DAC", NULL, "DAC Vref"},
{"Left DAC", NULL, "DAC Source Mux"},
{"Right DAC", NULL, "DAC Source Mux"},
{"Left Headphone Mux", "lin-rin with Boost and PGA", "Line input PGA"},
{"Right Headphone Mux", "lin-rin with Boost and PGA", "Line input PGA"},
{"Left Headphone Mixer", "LLIN Switch", "Left Headphone Mux"},
{"Left Headphone Mixer", "Left DAC Switch", "Left DAC"},
{"Right Headphone Mixer", "RLIN Switch", "Right Headphone Mux"},
{"Right Headphone Mixer", "Right DAC Switch", "Right DAC"},
{"Left Headphone Mixer Out", NULL, "Left Headphone Mixer"},
{"Right Headphone Mixer Out", NULL, "Right Headphone Mixer"},
{"Left Headphone Charge Pump", NULL, "Left Headphone Mixer Out"},
{"Right Headphone Charge Pump", NULL, "Right Headphone Mixer Out"},
{"Left Headphone Charge Pump", NULL, "Headphone Charge Pump"},
{"Right Headphone Charge Pump", NULL, "Headphone Charge Pump"},
{"Left Headphone Charge Pump", NULL, "Headphone Charge Pump Clock"},
{"Right Headphone Charge Pump", NULL, "Headphone Charge Pump Clock"},
{"Left Headphone Driver", NULL, "Left Headphone Charge Pump"},
{"Right Headphone Driver", NULL, "Right Headphone Charge Pump"},
{"HPOL", NULL, "Left Headphone Driver"},
{"HPOR", NULL, "Right Headphone Driver"},
{"HPOL", NULL, "Left Headphone ical"},
{"HPOR", NULL, "Right Headphone ical"},
{"Headphone Out", NULL, "Bias"},
{"Headphone Out", NULL, "Analog power"},
{"HPOL", NULL, "Headphone Out"},
{"HPOR", NULL, "Headphone Out"},
};
static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec);
int i;
int count = 0;
es8316->sysclk = freq;
if (freq == 0)
return 0;
/* Limit supported sample rates to ones that can be autodetected
* by the codec running in slave mode.
*/
for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) {
const unsigned int ratio = supported_mclk_lrck_ratios[i];
if (freq % ratio == 0)
es8316->allowed_rates[count++] = freq / ratio;
}
es8316->sysclk_constraints.list = es8316->allowed_rates;
es8316->sysclk_constraints.count = count;
return 0;
}
static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 serdata1 = 0;
u8 serdata2 = 0;
u8 clksw;
u8 mask;
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
dev_err(codec->dev, "Codec driver only supports slave mode\n");
return -EINVAL;
}
if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) {
dev_err(codec->dev, "Codec driver only supports I2S format\n");
return -EINVAL;
}
/* Clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
serdata1 |= ES8316_SERDATA1_BCLK_INV;
serdata2 |= ES8316_SERDATA2_ADCLRP;
break;
case SND_SOC_DAIFMT_IB_NF:
serdata1 |= ES8316_SERDATA1_BCLK_INV;
break;
case SND_SOC_DAIFMT_NB_IF:
serdata2 |= ES8316_SERDATA2_ADCLRP;
break;
default:
return -EINVAL;
}
mask = ES8316_SERDATA1_MASTER | ES8316_SERDATA1_BCLK_INV;
snd_soc_update_bits(codec, ES8316_SERDATA1, mask, serdata1);
mask = ES8316_SERDATA2_FMT_MASK | ES8316_SERDATA2_ADCLRP;
snd_soc_update_bits(codec, ES8316_SERDATA_ADC, mask, serdata2);
snd_soc_update_bits(codec, ES8316_SERDATA_DAC, mask, serdata2);
/* Enable BCLK and MCLK inputs in slave mode */
clksw = ES8316_CLKMGR_CLKSW_MCLK_ON | ES8316_CLKMGR_CLKSW_BCLK_ON;
snd_soc_update_bits(codec, ES8316_CLKMGR_CLKSW, clksw, clksw);
return 0;
}
static int es8316_pcm_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec);
if (es8316->sysclk == 0) {
dev_err(codec->dev, "No sysclk provided\n");
return -EINVAL;
}
/* The set of sample rates that can be supported depends on the
* MCLK supplied to the CODEC.
*/
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&es8316->sysclk_constraints);
return 0;
}
static int es8316_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec);
u8 wordlen = 0;
if (!es8316->sysclk) {
dev_err(codec->dev, "No MCLK configured\n");
return -EINVAL;
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
wordlen = ES8316_SERDATA2_LEN_16;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
wordlen = ES8316_SERDATA2_LEN_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
wordlen = ES8316_SERDATA2_LEN_24;
break;
case SNDRV_PCM_FORMAT_S32_LE:
wordlen = ES8316_SERDATA2_LEN_32;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, ES8316_SERDATA_DAC,
ES8316_SERDATA2_LEN_MASK, wordlen);
snd_soc_update_bits(codec, ES8316_SERDATA_ADC,
ES8316_SERDATA2_LEN_MASK, wordlen);
return 0;
}
static int es8316_mute(struct snd_soc_dai *dai, int mute)
{
snd_soc_update_bits(dai->codec, ES8316_DAC_SET1, 0x20,
mute ? 0x20 : 0);
return 0;
}
#define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops es8316_ops = {
.startup = es8316_pcm_startup,
.hw_params = es8316_pcm_hw_params,
.set_fmt = es8316_set_dai_fmt,
.set_sysclk = es8316_set_dai_sysclk,
.digital_mute = es8316_mute,
};
static struct snd_soc_dai_driver es8316_dai = {
.name = "ES8316 HiFi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = ES8316_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = ES8316_FORMATS,
},
.ops = &es8316_ops,
.symmetric_rates = 1,
};
static int es8316_probe(struct snd_soc_codec *codec)
{
/* Reset codec and enable current state machine */
snd_soc_write(codec, ES8316_RESET, 0x3f);
usleep_range(5000, 5500);
snd_soc_write(codec, ES8316_RESET, ES8316_RESET_CSM_ON);
msleep(30);
/*
* Documentation is unclear, but this value from the vendor driver is
* needed otherwise audio output is silent.
*/
snd_soc_write(codec, ES8316_SYS_VMIDSEL, 0xff);
/*
* Documentation for this register is unclear and incomplete,
* but here is a vendor-provided value that improves volume
* and quality for Intel CHT platforms.
*/
snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x32);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_es8316 = {
.probe = es8316_probe,
.idle_bias_off = true,
.component_driver = {
.controls = es8316_snd_controls,
.num_controls = ARRAY_SIZE(es8316_snd_controls),
.dapm_widgets = es8316_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets),
.dapm_routes = es8316_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes),
},
};
static const struct regmap_config es8316_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = 0x53,
.cache_type = REGCACHE_RBTREE,
};
static int es8316_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
struct es8316_priv *es8316;
struct regmap *regmap;
es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv),
GFP_KERNEL);
if (es8316 == NULL)
return -ENOMEM;
i2c_set_clientdata(i2c_client, es8316);
regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap);
if (IS_ERR(regmap))
return PTR_ERR(regmap);
return snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316,
&es8316_dai, 1);
}
static int es8316_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
return 0;
}
static const struct i2c_device_id es8316_i2c_id[] = {
{"es8316", 0 },
{}
};
MODULE_DEVICE_TABLE(i2c, es8316_i2c_id);
static const struct of_device_id es8316_of_match[] = {
{ .compatible = "everest,es8316", },
{},
};
MODULE_DEVICE_TABLE(of, es8316_of_match);
static const struct acpi_device_id es8316_acpi_match[] = {
{"ESSX8316", 0},
{},
};
MODULE_DEVICE_TABLE(acpi, es8316_acpi_match);
static struct i2c_driver es8316_i2c_driver = {
.driver = {
.name = "es8316",
.acpi_match_table = ACPI_PTR(es8316_acpi_match),
.of_match_table = of_match_ptr(es8316_of_match),
},
.probe = es8316_i2c_probe,
.remove = es8316_i2c_remove,
.id_table = es8316_i2c_id,
};
module_i2c_driver(es8316_i2c_driver);
MODULE_DESCRIPTION("Everest Semi ES8316 ALSA SoC Codec Driver");
MODULE_AUTHOR("David Yang <yangxiaohua@everest-semi.com>");
MODULE_LICENSE("GPL v2");
/*
* Copyright Everest Semiconductor Co.,Ltd
*
* Author: David Yang <yangxiaohua@everest-semi.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
*/
#ifndef _ES8316_H
#define _ES8316_H
/*
* ES8316 register space
*/
/* Reset Control */
#define ES8316_RESET 0x00
/* Clock Management */
#define ES8316_CLKMGR_CLKSW 0x01
#define ES8316_CLKMGR_CLKSEL 0x02
#define ES8316_CLKMGR_ADCOSR 0x03
#define ES8316_CLKMGR_ADCDIV1 0x04
#define ES8316_CLKMGR_ADCDIV2 0x05
#define ES8316_CLKMGR_DACDIV1 0x06
#define ES8316_CLKMGR_DACDIV2 0x07
#define ES8316_CLKMGR_CPDIV 0x08
/* Serial Data Port Control */
#define ES8316_SERDATA1 0x09
#define ES8316_SERDATA_ADC 0x0a
#define ES8316_SERDATA_DAC 0x0b
/* System Control */
#define ES8316_SYS_VMIDSEL 0x0c
#define ES8316_SYS_PDN 0x0d
#define ES8316_SYS_LP1 0x0e
#define ES8316_SYS_LP2 0x0f
#define ES8316_SYS_VMIDLOW 0x10
#define ES8316_SYS_VSEL 0x11
#define ES8316_SYS_REF 0x12
/* Headphone Mixer */
#define ES8316_HPMIX_SEL 0x13
#define ES8316_HPMIX_SWITCH 0x14
#define ES8316_HPMIX_PDN 0x15
#define ES8316_HPMIX_VOL 0x16
/* Charge Pump Headphone driver */
#define ES8316_CPHP_OUTEN 0x17
#define ES8316_CPHP_ICAL_VOL 0x18
#define ES8316_CPHP_PDN1 0x19
#define ES8316_CPHP_PDN2 0x1a
#define ES8316_CPHP_LDOCTL 0x1b
/* Calibration */
#define ES8316_CAL_TYPE 0x1c
#define ES8316_CAL_SET 0x1d
#define ES8316_CAL_HPLIV 0x1e
#define ES8316_CAL_HPRIV 0x1f
#define ES8316_CAL_HPLMV 0x20
#define ES8316_CAL_HPRMV 0x21
/* ADC Control */
#define ES8316_ADC_PDN_LINSEL 0x22
#define ES8316_ADC_PGAGAIN 0x23
#define ES8316_ADC_D2SEPGA 0x24
#define ES8316_ADC_DMIC 0x25
#define ES8316_ADC_MUTE 0x26
#define ES8316_ADC_VOLUME 0x27
#define ES8316_ADC_ALC1 0x29
#define ES8316_ADC_ALC2 0x2a
#define ES8316_ADC_ALC3 0x2b
#define ES8316_ADC_ALC4 0x2c
#define ES8316_ADC_ALC5 0x2d
#define ES8316_ADC_ALC_NG 0x2e
/* DAC Control */
#define ES8316_DAC_PDN 0x2f
#define ES8316_DAC_SET1 0x30
#define ES8316_DAC_SET2 0x31
#define ES8316_DAC_SET3 0x32
#define ES8316_DAC_VOLL 0x33
#define ES8316_DAC_VOLR 0x34
/* GPIO */
#define ES8316_GPIO_SEL 0x4d
#define ES8316_GPIO_DEBOUNCE 0x4e
#define ES8316_GPIO_FLAG 0x4f
/* Test mode */
#define ES8316_TESTMODE 0x50
#define ES8316_TEST1 0x51
#define ES8316_TEST2 0x52
#define ES8316_TEST3 0x53
/*
* Field definitions
*/
/* ES8316_RESET */
#define ES8316_RESET_CSM_ON 0x80
/* ES8316_CLKMGR_CLKSW */
#define ES8316_CLKMGR_CLKSW_MCLK_ON 0x40
#define ES8316_CLKMGR_CLKSW_BCLK_ON 0x20
/* ES8316_SERDATA1 */
#define ES8316_SERDATA1_MASTER 0x80
#define ES8316_SERDATA1_BCLK_INV 0x20
/* ES8316_SERDATA_ADC and _DAC */
#define ES8316_SERDATA2_FMT_MASK 0x3
#define ES8316_SERDATA2_FMT_I2S 0x00
#define ES8316_SERDATA2_FMT_LEFTJ 0x01
#define ES8316_SERDATA2_FMT_RIGHTJ 0x02
#define ES8316_SERDATA2_FMT_PCM 0x03
#define ES8316_SERDATA2_ADCLRP 0x20
#define ES8316_SERDATA2_LEN_MASK 0x1c
#define ES8316_SERDATA2_LEN_24 0x00
#define ES8316_SERDATA2_LEN_20 0x04
#define ES8316_SERDATA2_LEN_18 0x08
#define ES8316_SERDATA2_LEN_16 0x0c
#define ES8316_SERDATA2_LEN_32 0x10
#endif
...@@ -629,7 +629,7 @@ static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream, ...@@ -629,7 +629,7 @@ static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
if (mcasp->tdm_mask[stream]) if (mcasp->tdm_mask[stream])
slots = hweight32(mcasp->tdm_mask[stream]); slots = hweight32(mcasp->tdm_mask[stream]);
for (i = 2; i <= slots; i++) for (i = 1; i <= slots; i++)
list[count++] = i; list[count++] = i;
for (i = 2; i <= serializers; i++) for (i = 2; i <= serializers; i++)
...@@ -1297,7 +1297,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, ...@@ -1297,7 +1297,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
snd_pcm_hw_constraint_minmax(substream->runtime, snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS, SNDRV_PCM_HW_PARAM_CHANNELS,
2, max_channels); 0, max_channels);
snd_pcm_hw_constraint_list(substream->runtime, snd_pcm_hw_constraint_list(substream->runtime,
0, SNDRV_PCM_HW_PARAM_CHANNELS, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
...@@ -1459,13 +1459,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { ...@@ -1459,13 +1459,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.suspend = davinci_mcasp_suspend, .suspend = davinci_mcasp_suspend,
.resume = davinci_mcasp_resume, .resume = davinci_mcasp_resume,
.playback = { .playback = {
.channels_min = 2, .channels_min = 1,
.channels_max = 32 * 16, .channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES, .rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS, .formats = DAVINCI_MCASP_PCM_FMTS,
}, },
.capture = { .capture = {
.channels_min = 2, .channels_min = 1,
.channels_max = 32 * 16, .channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES, .rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS, .formats = DAVINCI_MCASP_PCM_FMTS,
...@@ -1971,12 +1971,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ...@@ -1971,12 +1971,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
*/ */
mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list = mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
devm_kzalloc(mcasp->dev, sizeof(unsigned int) * devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
(32 + mcasp->num_serializer - 2), (32 + mcasp->num_serializer - 1),
GFP_KERNEL); GFP_KERNEL);
mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list = mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
devm_kzalloc(mcasp->dev, sizeof(unsigned int) * devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
(32 + mcasp->num_serializer - 2), (32 + mcasp->num_serializer - 1),
GFP_KERNEL); GFP_KERNEL);
if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
......
...@@ -496,6 +496,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, ...@@ -496,6 +496,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
idx = COMP1_TX_WORDSIZE_0(comp1); idx = COMP1_TX_WORDSIZE_0(comp1);
if (WARN_ON(idx >= ARRAY_SIZE(formats))) if (WARN_ON(idx >= ARRAY_SIZE(formats)))
return -EINVAL; return -EINVAL;
if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE)
idx = 1;
dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM;
dw_i2s_dai->playback.channels_max = dw_i2s_dai->playback.channels_max =
1 << (COMP1_TX_CHANNELS(comp1) + 1); 1 << (COMP1_TX_CHANNELS(comp1) + 1);
...@@ -508,6 +510,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, ...@@ -508,6 +510,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
idx = COMP2_RX_WORDSIZE_0(comp2); idx = COMP2_RX_WORDSIZE_0(comp2);
if (WARN_ON(idx >= ARRAY_SIZE(formats))) if (WARN_ON(idx >= ARRAY_SIZE(formats)))
return -EINVAL; return -EINVAL;
if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE)
idx = 1;
dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM;
dw_i2s_dai->capture.channels_max = dw_i2s_dai->capture.channels_max =
1 << (COMP1_RX_CHANNELS(comp1) + 1); 1 << (COMP1_RX_CHANNELS(comp1) + 1);
...@@ -543,6 +547,8 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, ...@@ -543,6 +547,8 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev,
if (ret < 0) if (ret < 0)
return ret; return ret;
if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE)
idx = 1;
/* Set DMA slaves info */ /* Set DMA slaves info */
dev->play_dma_data.pd.data = pdata->play_dma_data; dev->play_dma_data.pd.data = pdata->play_dma_data;
dev->capture_dma_data.pd.data = pdata->capture_dma_data; dev->capture_dma_data.pd.data = pdata->capture_dma_data;
......
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