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

diff --git a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt
index 00ea670b8c4dc49ea35324a1b606c009a5b87df0..06668bca7ffcd9385a08b9d1ed80a4321bf83ab5 100644 (file)
--- 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 f6b3f36d422b2a85d35f5af4ac64deec1a51ec60..81b68580e19995e5c292ce70d9f5395d5e30d8c3 100644 (file)
--- 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 cf92ebfe6ab70fcba79fcd53af1fd7b15dfefa29..67469c26bae88f9ada64e0866d73fbdcfe793886 100644 (file)
--- 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 aaf287d2e91d057b3bf5e149ba4a21033755cf06..b2cf59f54c889e528b1a7c7bc6371c899594bd20 100644 (file)
--- 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 5681855396c411210b51fffe1b368d6ccf5a1027..830f5caa915cc6a94085b838aec5a92eecbb9b71 100644 (file)
--- 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 883ed4c8a5510fb9090ad1889976003960781d89..f0f79418618650979accffeebf1835782ff09cea 100644 (file)
--- 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 28a63fdaf982fdf6b15afcdc56fd11f89698c250..e878306ce46e972ded5cbf6a8db4866f57dc6e26 100644 (file)
--- 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 (file)
index 0000000..ecc0244
--- /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 (file)
index 0000000..6bcdd63
--- /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 3c5a9804d3f5edc259b25bd129b40918fe80197b..56ec1d301ac2927b5ebabb123a28c2a56dbc532f 100644 (file)
--- 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 9c46e411202649a31c5a597cebd4af83e2826ce7..91606763818089ffe2c4d058eb214f1e5714765a 100644 (file)
--- 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;