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-rw-r--r--Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt8
-rw-r--r--Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt9
-rw-r--r--drivers/gpu/drm/bridge/adv7511/adv7511_audio.c22
-rw-r--r--drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c21
-rw-r--r--include/sound/designware_i2s.h1
-rw-r--r--sound/soc/codecs/Kconfig5
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/es8316.c637
-rw-r--r--sound/soc/codecs/es8316.h129
-rw-r--r--sound/soc/davinci/davinci-mcasp.c12
-rw-r--r--sound/soc/dwc/dwc-i2s.c6
11 files changed, 845 insertions, 7 deletions
diff --git a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt
index 00ea670b8c4d..06668bca7ffc 100644
--- a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt
+++ b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.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
node.
- Video port 1 for the HDMI output
+- Audio port 2 for the HDMI audio input
Example
@@ -112,5 +113,12 @@ Example
remote-endpoint = <&hdmi_connector_in>;
};
};
+
+ port@2 {
+ reg = <2>;
+ codec_endpoint: endpoint {
+ remote-endpoint = <&i2s0_cpu_endpoint>;
+ };
+ };
};
};
diff --git a/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt b/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt
index f6b3f36d422b..81b68580e199 100644
--- a/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt
+++ b/Documentation/devicetree/bindings/display/bridge/renesas,dw-hdmi.txt
@@ -25,7 +25,8 @@ Required properties:
- 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
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:
@@ -59,6 +60,12 @@ Example:
remote-endpoint = <&hdmi0_con>;
};
};
+ port@2 {
+ reg = <2>;
+ rcar_dw_hdmi0_sound_in: endpoint {
+ remote-endpoint = <&hdmi_sound_out>;
+ };
+ };
};
};
diff --git a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c
index cf92ebfe6ab7..67469c26bae8 100644
--- a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c
+++ b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c
@@ -11,6 +11,7 @@
#include <sound/hdmi-codec.h>
#include <sound/pcm.h>
#include <sound/soc.h>
+#include <linux/of_graph.h>
#include "adv7511.h"
@@ -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 = {
.hw_params = adv7511_hdmi_hw_params,
.audio_shutdown = audio_shutdown,
.audio_startup = audio_startup,
+ .get_dai_id = adv7511_hdmi_i2s_get_dai_id,
};
static struct hdmi_codec_pdata codec_data = {
diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c
index aaf287d2e91d..b2cf59f54c88 100644
--- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c
+++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c
@@ -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);
}
+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 = {
.hw_params = dw_hdmi_i2s_hw_params,
.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)
diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h
index 5681855396c4..830f5caa915c 100644
--- a/include/sound/designware_i2s.h
+++ b/include/sound/designware_i2s.h
@@ -47,6 +47,7 @@ struct i2s_platform_data {
#define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0)
#define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1)
+ #define DW_I2S_QUIRK_16BIT_IDX_OVERRIDE (1 << 2)
unsigned int quirks;
unsigned int i2s_reg_comp1;
unsigned int i2s_reg_comp2;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 883ed4c8a551..f0f794186186 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_DA9055 if I2C
select SND_SOC_DIO2125
select SND_SOC_DMIC
+ select SND_SOC_ES8316 if I2C
select SND_SOC_ES8328_SPI if SPI_MASTER
select SND_SOC_ES8328_I2C if I2C
select SND_SOC_ES7134
@@ -543,6 +544,10 @@ config SND_SOC_HDMI_CODEC
config SND_SOC_ES7134
tristate "Everest Semi ES7134 CODEC"
+config SND_SOC_ES8316
+ tristate "Everest Semi ES8316 CODEC"
+ depends on I2C
+
config SND_SOC_ES8328
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 28a63fdaf982..e878306ce46e 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o
snd-soc-dmic-objs := dmic.o
snd-soc-es7134-objs := es7134.o
+snd-soc-es8316-objs := es8316.o
snd-soc-es8328-objs := es8328.o
snd-soc-es8328-i2c-objs := es8328-i2c.o
snd-soc-es8328-spi-objs := es8328-spi.o
@@ -300,6 +301,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
obj-$(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-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
new file mode 100644
index 000000000000..ecc02449c569
--- /dev/null
+++ b/sound/soc/codecs/es8316.c
@@ -0,0 +1,637 @@
+/*
+ * 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");
diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h
new file mode 100644
index 000000000000..6bcdd63ea459
--- /dev/null
+++ b/sound/soc/codecs/es8316.h
@@ -0,0 +1,129 @@
+/*
+ * 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
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 3c5a9804d3f5..56ec1d301ac2 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -629,7 +629,7 @@ static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
if (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;
for (i = 2; i <= serializers; i++)
@@ -1297,7 +1297,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_CHANNELS,
- 2, max_channels);
+ 0, max_channels);
snd_pcm_hw_constraint_list(substream->runtime,
0, SNDRV_PCM_HW_PARAM_CHANNELS,
@@ -1459,13 +1459,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.suspend = davinci_mcasp_suspend,
.resume = davinci_mcasp_resume,
.playback = {
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
},
.capture = {
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 32 * 16,
.rates = DAVINCI_MCASP_RATES,
.formats = DAVINCI_MCASP_PCM_FMTS,
@@ -1971,12 +1971,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
*/
mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
- (32 + mcasp->num_serializer - 2),
+ (32 + mcasp->num_serializer - 1),
GFP_KERNEL);
mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
devm_kzalloc(mcasp->dev, sizeof(unsigned int) *
- (32 + mcasp->num_serializer - 2),
+ (32 + mcasp->num_serializer - 1),
GFP_KERNEL);
if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index 9c46e4112026..916067638180 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -496,6 +496,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
idx = COMP1_TX_WORDSIZE_0(comp1);
if (WARN_ON(idx >= ARRAY_SIZE(formats)))
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_max =
1 << (COMP1_TX_CHANNELS(comp1) + 1);
@@ -508,6 +510,8 @@ static int dw_configure_dai(struct dw_i2s_dev *dev,
idx = COMP2_RX_WORDSIZE_0(comp2);
if (WARN_ON(idx >= ARRAY_SIZE(formats)))
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_max =
1 << (COMP1_RX_CHANNELS(comp1) + 1);
@@ -543,6 +547,8 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev,
if (ret < 0)
return ret;
+ if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE)
+ idx = 1;
/* Set DMA slaves info */
dev->play_dma_data.pd.data = pdata->play_dma_data;
dev->capture_dma_data.pd.data = pdata->capture_dma_data;
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-03 20:56:41 +0000