Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs42l56',...

Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs42l56', 'asoc/topic/cs42xx8', 'asoc/topic/cx20442' and 'asoc/topic/davinci' into asoc-next

Documentation/devicetree/bindings/sound/cs4265.txt

+CS4265 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "cirrus,cs4265"
+
+  - reg : the I2C address of the device for I2C. The I2C address depends on
+          the state of the AD0 pin.  If AD0 is high, the i2c address is 0x4f.
+          If it is low, the i2c address is 0x4e.
+
+Optional properties:
+
+  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+Examples:
+
+codec_ad0_high: cs4265@4f { /* AD0 Pin is high */
+	compatible = "cirrus,cs4265";
+	reg = <0x4f>;
+};
+
+
+codec_ad0_low: cs4265@4e { /* AD0 Pin is low */
+	compatible = "cirrus,cs4265";
+	reg = <0x4e>;
+};

sound/soc/codecs/Kconfig

@@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS42L52 if I2C && INPUT
 	select SND_SOC_CS42L56 if I2C && INPUT
 	select SND_SOC_CS42L73 if I2C
+	select SND_SOC_CS4265 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CS42XX8_I2C if I2C
@@ -338,6 +339,11 @@ config SND_SOC_CS42L73
 	tristate "Cirrus Logic CS42L73 CODEC"
 	depends on I2C
 
+config SND_SOC_CS4265
+	tristate "Cirrus Logic CS4265 CODEC"
+	depends on I2C
+	select REGMAP_I2C
+
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
 	tristate "Cirrus Logic CS4270 CODEC"

sound/soc/codecs/Makefile

@@ -37,6 +37,7 @@ snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o
 snd-soc-cs42l52-objs := cs42l52.o
 snd-soc-cs42l56-objs := cs42l56.o
 snd-soc-cs42l73-objs := cs42l73.o
+snd-soc-cs4265-objs := cs4265.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cs4271-objs := cs4271.o
 snd-soc-cs42xx8-objs := cs42xx8.o
@@ -204,6 +205,7 @@ obj-$(CONFIG_SND_SOC_CS42L51_I2C)	+= snd-soc-cs42l51-i2c.o
 obj-$(CONFIG_SND_SOC_CS42L52)	+= snd-soc-cs42l52.o
 obj-$(CONFIG_SND_SOC_CS42L56)	+= snd-soc-cs42l56.o
 obj-$(CONFIG_SND_SOC_CS42L73)	+= snd-soc-cs42l73.o
+obj-$(CONFIG_SND_SOC_CS4265)	+= snd-soc-cs4265.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CS4271)	+= snd-soc-cs4271.o
 obj-$(CONFIG_SND_SOC_CS42XX8)	+= snd-soc-cs42xx8.o

sound/soc/codecs/cs4265.c

+/*
+ * cs4265.c -- CS4265 ALSA SoC audio driver
+ *
+ * Copyright 2014 Cirrus Logic, Inc.
+ *
+ * Author: Paul Handrigan <paul.handrigan@cirrus.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/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/gpio/consumer.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include "cs4265.h"
+
+struct cs4265_private {
+	struct device *dev;
+	struct regmap *regmap;
+	struct gpio_desc *reset_gpio;
+	u8 format;
+	u32 sysclk;
+};
+
+static const struct reg_default cs4265_reg_defaults[] = {
+	{ CS4265_PWRCTL, 0x0F },
+	{ CS4265_DAC_CTL, 0x08 },
+	{ CS4265_ADC_CTL, 0x00 },
+	{ CS4265_MCLK_FREQ, 0x00 },
+	{ CS4265_SIG_SEL, 0x40 },
+	{ CS4265_CHB_PGA_CTL, 0x00 },
+	{ CS4265_CHA_PGA_CTL, 0x00 },
+	{ CS4265_ADC_CTL2, 0x19 },
+	{ CS4265_DAC_CHA_VOL, 0x00 },
+	{ CS4265_DAC_CHB_VOL, 0x00 },
+	{ CS4265_DAC_CTL2, 0xC0 },
+	{ CS4265_SPDIF_CTL1, 0x00 },
+	{ CS4265_SPDIF_CTL2, 0x00 },
+	{ CS4265_INT_MASK, 0x00 },
+	{ CS4265_STATUS_MODE_MSB, 0x00 },
+	{ CS4265_STATUS_MODE_LSB, 0x00 },
+};
+
+static bool cs4265_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS4265_PWRCTL:
+	case CS4265_DAC_CTL:
+	case CS4265_ADC_CTL:
+	case CS4265_MCLK_FREQ:
+	case CS4265_SIG_SEL:
+	case CS4265_CHB_PGA_CTL:
+	case CS4265_CHA_PGA_CTL:
+	case CS4265_ADC_CTL2:
+	case CS4265_DAC_CHA_VOL:
+	case CS4265_DAC_CHB_VOL:
+	case CS4265_DAC_CTL2:
+	case CS4265_SPDIF_CTL1:
+	case CS4265_SPDIF_CTL2:
+	case CS4265_INT_MASK:
+	case CS4265_STATUS_MODE_MSB:
+	case CS4265_STATUS_MODE_LSB:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool cs4265_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case CS4265_INT_STATUS:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static DECLARE_TLV_DB_SCALE(pga_tlv, -1200, 50, 0);
+
+static DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 0);
+
+static const char * const digital_input_mux_text[] = {
+	"SDIN1", "SDIN2"
+};
+
+static SOC_ENUM_SINGLE_DECL(digital_input_mux_enum, CS4265_SIG_SEL, 7,
+		digital_input_mux_text);
+
+static const struct snd_kcontrol_new digital_input_mux =
+	SOC_DAPM_ENUM("Digital Input Mux", digital_input_mux_enum);
+
+static const char * const mic_linein_text[] = {
+	"MIC", "LINEIN"
+};
+
+static SOC_ENUM_SINGLE_DECL(mic_linein_enum, CS4265_ADC_CTL2, 0,
+		mic_linein_text);
+
+static const char * const cam_mode_text[] = {
+	"One Byte", "Two Byte"
+};
+
+static SOC_ENUM_SINGLE_DECL(cam_mode_enum, CS4265_SPDIF_CTL1, 5,
+		cam_mode_text);
+
+static const char * const cam_mono_stereo_text[] = {
+	"Stereo", "Mono"
+};
+
+static SOC_ENUM_SINGLE_DECL(spdif_mono_stereo_enum, CS4265_SPDIF_CTL2, 2,
+		cam_mono_stereo_text);
+
+static const char * const mono_select_text[] = {
+	"Channel A", "Channel B"
+};
+
+static SOC_ENUM_SINGLE_DECL(spdif_mono_select_enum, CS4265_SPDIF_CTL2, 0,
+		mono_select_text);
+
+static const struct snd_kcontrol_new mic_linein_mux =
+	SOC_DAPM_ENUM("ADC Input Capture Mux", mic_linein_enum);
+
+static const struct snd_kcontrol_new loopback_ctl =
+	SOC_DAPM_SINGLE("Switch", CS4265_SIG_SEL, 1, 1, 0);
+
+static const struct snd_kcontrol_new spdif_switch =
+	SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 0, 0);
+
+static const struct snd_kcontrol_new dac_switch =
+	SOC_DAPM_SINGLE("Switch", CS4265_PWRCTL, 1, 1, 0);
+
+static const struct snd_kcontrol_new cs4265_snd_controls[] = {
+
+	SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL,
+			      CS4265_CHB_PGA_CTL, 0, 0x28, 0x30, pga_tlv),
+	SOC_DOUBLE_R_TLV("DAC Volume", CS4265_DAC_CHA_VOL,
+		      CS4265_DAC_CHB_VOL, 0, 0xFF, 1, dac_tlv),
+	SOC_SINGLE("De-emp 44.1kHz Switch", CS4265_DAC_CTL, 1,
+				1, 0),
+	SOC_SINGLE("DAC INV Switch", CS4265_DAC_CTL2, 5,
+				1, 0),
+	SOC_SINGLE("DAC Zero Cross Switch", CS4265_DAC_CTL2, 6,
+				1, 0),
+	SOC_SINGLE("DAC Soft Ramp Switch", CS4265_DAC_CTL2, 7,
+				1, 0),
+	SOC_SINGLE("ADC HPF Switch", CS4265_ADC_CTL, 1,
+				1, 0),
+	SOC_SINGLE("ADC Zero Cross Switch", CS4265_ADC_CTL2, 3,
+				1, 1),
+	SOC_SINGLE("ADC Soft Ramp Switch", CS4265_ADC_CTL2, 7,
+				1, 0),
+	SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
+				6, 1, 0),
+	SOC_ENUM("C Data Access", cam_mode_enum),
+	SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
+				3, 1, 0),
+	SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
+	SOC_SINGLE("MMTLR Data Switch", 0,
+				1, 1, 0),
+	SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
+	SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
+};
+
+static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = {
+
+	SND_SOC_DAPM_INPUT("LINEINL"),
+	SND_SOC_DAPM_INPUT("LINEINR"),
+	SND_SOC_DAPM_INPUT("MICL"),
+	SND_SOC_DAPM_INPUT("MICR"),
+
+	SND_SOC_DAPM_AIF_OUT("DOUT", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SPDIFOUT", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_MUX("ADC Mux", SND_SOC_NOPM, 0, 0, &mic_linein_mux),
+
+	SND_SOC_DAPM_ADC("ADC", NULL, CS4265_PWRCTL, 2, 1),
+	SND_SOC_DAPM_PGA("Pre-amp MIC", CS4265_PWRCTL, 3,
+			1, NULL, 0),
+
+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM,
+			 0, 0, &digital_input_mux),
+
+	SND_SOC_DAPM_MIXER("SDIN1 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("SDIN2 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("SPDIF Transmitter", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0,
+			&loopback_ctl),
+	SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0,
+			&spdif_switch),
+	SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1,
+			&dac_switch),
+
+	SND_SOC_DAPM_AIF_IN("DIN1", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DIN2", NULL,  0,
+			SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("TXIN", NULL,  0,
+			CS4265_SPDIF_CTL2, 5, 1),
+
+	SND_SOC_DAPM_OUTPUT("LINEOUTL"),
+	SND_SOC_DAPM_OUTPUT("LINEOUTR"),
+
+};
+
+static const struct snd_soc_dapm_route cs4265_audio_map[] = {
+
+	{"DIN1", NULL, "DAI1 Playback"},
+	{"DIN2", NULL, "DAI2 Playback"},
+	{"SDIN1 Input Mixer", NULL, "DIN1"},
+	{"SDIN2 Input Mixer", NULL, "DIN2"},
+	{"Input Mux", "SDIN1", "SDIN1 Input Mixer"},
+	{"Input Mux", "SDIN2", "SDIN2 Input Mixer"},
+	{"DAC", "Switch", "Input Mux"},
+	{"SPDIF", "Switch", "Input Mux"},
+	{"LINEOUTL", NULL, "DAC"},
+	{"LINEOUTR", NULL, "DAC"},
+	{"SPDIFOUT", NULL, "SPDIF"},
+
+	{"ADC Mux", "LINEIN", "LINEINL"},
+	{"ADC Mux", "LINEIN", "LINEINR"},
+	{"ADC Mux", "MIC", "MICL"},
+	{"ADC Mux", "MIC", "MICR"},
+	{"ADC", NULL, "ADC Mux"},
+	{"DOUT", NULL, "ADC"},
+	{"DAI1 Capture", NULL, "DOUT"},
+	{"DAI2 Capture", NULL, "DOUT"},
+
+	/* Loopback */
+	{"Loopback", "Switch", "ADC"},
+	{"DAC", NULL, "Loopback"},
+};
+
+struct cs4265_clk_para {
+	u32 mclk;
+	u32 rate;
+	u8 fm_mode; /* values 1, 2, or 4 */
+	u8 mclkdiv;
+};
+
+static const struct cs4265_clk_para clk_map_table[] = {
+	/*32k*/
+	{8192000, 32000, 0, 0},
+	{12288000, 32000, 0, 1},
+	{16384000, 32000, 0, 2},
+	{24576000, 32000, 0, 3},
+	{32768000, 32000, 0, 4},
+
+	/*44.1k*/
+	{11289600, 44100, 0, 0},
+	{16934400, 44100, 0, 1},
+	{22579200, 44100, 0, 2},
+	{33868000, 44100, 0, 3},
+	{45158400, 44100, 0, 4},
+
+	/*48k*/
+	{12288000, 48000, 0, 0},
+	{18432000, 48000, 0, 1},
+	{24576000, 48000, 0, 2},
+	{36864000, 48000, 0, 3},
+	{49152000, 48000, 0, 4},
+
+	/*64k*/
+	{8192000, 64000, 1, 0},
+	{1228800, 64000, 1, 1},
+	{1693440, 64000, 1, 2},
+	{2457600, 64000, 1, 3},
+	{3276800, 64000, 1, 4},
+
+	/* 88.2k */
+	{11289600, 88200, 1, 0},
+	{16934400, 88200, 1, 1},
+	{22579200, 88200, 1, 2},
+	{33868000, 88200, 1, 3},
+	{45158400, 88200, 1, 4},
+
+	/* 96k */
+	{12288000, 96000, 1, 0},
+	{18432000, 96000, 1, 1},
+	{24576000, 96000, 1, 2},
+	{36864000, 96000, 1, 3},
+	{49152000, 96000, 1, 4},
+
+	/* 128k */
+	{8192000, 128000, 2, 0},
+	{12288000, 128000, 2, 1},
+	{16934400, 128000, 2, 2},
+	{24576000, 128000, 2, 3},
+	{32768000, 128000, 2, 4},
+
+	/* 176.4k */
+	{11289600, 176400, 2, 0},
+	{16934400, 176400, 2, 1},
+	{22579200, 176400, 2, 2},
+	{33868000, 176400, 2, 3},
+	{49152000, 176400, 2, 4},
+
+	/* 192k */
+	{12288000, 192000, 2, 0},
+	{18432000, 192000, 2, 1},
+	{24576000, 192000, 2, 2},
+	{36864000, 192000, 2, 3},
+	{49152000, 192000, 2, 4},
+};
+
+static int cs4265_get_clk_index(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
+		if (clk_map_table[i].rate == rate &&
+				clk_map_table[i].mclk == mclk)
+			return i;
+	}
+	return -EINVAL;
+}
+
+static int cs4265_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
+			unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	if (clk_id != 0) {
+		dev_err(codec->dev, "Invalid clk_id %d\n", clk_id);
+		return -EINVAL;
+	}
+	for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
+		if (clk_map_table[i].mclk == freq) {
+			cs4265->sysclk = freq;
+			return 0;
+		}
+	}
+	cs4265->sysclk = 0;
+	dev_err(codec->dev, "Invalid freq parameter %d\n", freq);
+	return -EINVAL;
+}
+
+static int cs4265_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
+	u8 iface = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		snd_soc_update_bits(codec, CS4265_ADC_CTL,
+				CS4265_ADC_MASTER,
+				CS4265_ADC_MASTER);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		snd_soc_update_bits(codec, CS4265_ADC_CTL,
+				CS4265_ADC_MASTER,
+				0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	 /* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= SND_SOC_DAIFMT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface |= SND_SOC_DAIFMT_RIGHT_J;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= SND_SOC_DAIFMT_LEFT_J;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	cs4265->format = iface;
+	return 0;
+}
+
+static int cs4265_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	if (mute) {
+		snd_soc_update_bits(codec, CS4265_DAC_CTL,
+			CS4265_DAC_CTL_MUTE,
+			CS4265_DAC_CTL_MUTE);
+		snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
+			CS4265_SPDIF_CTL2_MUTE,
+			CS4265_SPDIF_CTL2_MUTE);
+	} else {
+		snd_soc_update_bits(codec, CS4265_DAC_CTL,
+			CS4265_DAC_CTL_MUTE,
+			0);
+		snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
+			CS4265_SPDIF_CTL2_MUTE,
+			0);
+	}
+	return 0;
+}
+
+static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
+	int index;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
+		((cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK)
+		== SND_SOC_DAIFMT_RIGHT_J))
+		return -EINVAL;
+
+	index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
+	if (index >= 0) {
+		snd_soc_update_bits(codec, CS4265_ADC_CTL,
+			CS4265_ADC_FM, clk_map_table[index].fm_mode);
+		snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
+			CS4265_MCLK_FREQ_MASK,
+			clk_map_table[index].mclkdiv);
+
+	} else {
+		dev_err(codec->dev, "can't get correct mclk\n");
+		return -EINVAL;
+	}
+
+	switch (cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		snd_soc_update_bits(codec, CS4265_DAC_CTL,
+			CS4265_DAC_CTL_DIF, (1 << 4));
+		snd_soc_update_bits(codec, CS4265_ADC_CTL,
+			CS4265_ADC_DIF, (1 << 4));
+		snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
+			CS4265_SPDIF_CTL2_DIF, (1 << 6));
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		if (params_width(params) == 16) {
+			snd_soc_update_bits(codec, CS4265_DAC_CTL,
+				CS4265_DAC_CTL_DIF, (1 << 5));
+			snd_soc_update_bits(codec, CS4265_ADC_CTL,
+				CS4265_SPDIF_CTL2_DIF, (1 << 7));
+		} else {
+			snd_soc_update_bits(codec, CS4265_DAC_CTL,
+				CS4265_DAC_CTL_DIF, (3 << 5));
+			snd_soc_update_bits(codec, CS4265_ADC_CTL,
+				CS4265_SPDIF_CTL2_DIF, (1 << 7));
+		}
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		snd_soc_update_bits(codec, CS4265_DAC_CTL,
+			CS4265_DAC_CTL_DIF, 0);
+		snd_soc_update_bits(codec, CS4265_ADC_CTL,
+			CS4265_ADC_DIF, 0);
+		snd_soc_update_bits(codec, CS4265_ADC_CTL,
+			CS4265_SPDIF_CTL2_DIF, (1 << 6));
+
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int cs4265_set_bias_level(struct snd_soc_codec *codec,
+					enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_update_bits(codec, CS4265_PWRCTL,
+			CS4265_PWRCTL_PDN, 0);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, CS4265_PWRCTL,
+			CS4265_PWRCTL_PDN,
+			CS4265_PWRCTL_PDN);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, CS4265_PWRCTL,
+			CS4265_PWRCTL_PDN,
+			CS4265_PWRCTL_PDN);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define CS4265_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+			SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+			SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+#define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+
+static const struct snd_soc_dai_ops cs4265_ops = {
+	.hw_params	= cs4265_pcm_hw_params,
+	.digital_mute	= cs4265_digital_mute,
+	.set_fmt	= cs4265_set_fmt,
+	.set_sysclk	= cs4265_set_sysclk,
+};
+
+static struct snd_soc_dai_driver cs4265_dai[] = {
+	{
+		.name = "cs4265-dai1",
+		.playback = {
+			.stream_name = "DAI1 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS4265_RATES,
+			.formats = CS4265_FORMATS,
+		},
+		.capture = {
+			.stream_name = "DAI1 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS4265_RATES,
+			.formats = CS4265_FORMATS,
+		},
+		.ops = &cs4265_ops,
+	},
+	{
+		.name = "cs4265-dai2",
+		.playback = {
+			.stream_name = "DAI2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS4265_RATES,
+			.formats = CS4265_FORMATS,
+		},
+		.capture = {
+			.stream_name = "DAI2 Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = CS4265_RATES,
+			.formats = CS4265_FORMATS,
+		},
+		.ops = &cs4265_ops,
+	},
+};
+
+static const struct snd_soc_codec_driver soc_codec_cs4265 = {
+	.set_bias_level = cs4265_set_bias_level,
+
+	.dapm_widgets = cs4265_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(cs4265_dapm_widgets),
+	.dapm_routes = cs4265_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(cs4265_audio_map),
+
+	.controls = cs4265_snd_controls,
+	.num_controls = ARRAY_SIZE(cs4265_snd_controls),
+};
+
+static const struct regmap_config cs4265_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = CS4265_MAX_REGISTER,
+	.reg_defaults = cs4265_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(cs4265_reg_defaults),
+	.readable_reg = cs4265_readable_register,
+	.volatile_reg = cs4265_volatile_register,
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int cs4265_i2c_probe(struct i2c_client *i2c_client,
+			     const struct i2c_device_id *id)
+{
+	struct cs4265_private *cs4265;
+	int ret = 0;
+	unsigned int devid = 0;
+	unsigned int reg;
+
+	cs4265 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4265_private),
+			       GFP_KERNEL);
+	if (cs4265 == NULL)
+		return -ENOMEM;
+	cs4265->dev = &i2c_client->dev;
+
+	cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap);
+	if (IS_ERR(cs4265->regmap)) {
+		ret = PTR_ERR(cs4265->regmap);
+		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
+		return ret;
+	}
+
+	cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev,
+		"reset-gpios");
+	if (IS_ERR(cs4265->reset_gpio)) {
+		ret = PTR_ERR(cs4265->reset_gpio);
+		if (ret != -ENOENT && ret != -ENOSYS)
+			return ret;
+
+		cs4265->reset_gpio = NULL;
+	} else {
+		ret = gpiod_direction_output(cs4265->reset_gpio, 0);
+		if (ret)
+			return ret;
+		mdelay(1);
+		gpiod_set_value_cansleep(cs4265->reset_gpio, 1);
+
+	}
+
+	i2c_set_clientdata(i2c_client, cs4265);
+
+	ret = regmap_read(cs4265->regmap, CS4265_CHIP_ID, &reg);
+	devid = reg & CS4265_CHIP_ID_MASK;
+	if (devid != CS4265_CHIP_ID_VAL) {
+		ret = -ENODEV;
+		dev_err(&i2c_client->dev,
+			"CS4265 Device ID (%X). Expected %X\n",
+			devid, CS4265_CHIP_ID);
+		return ret;
+	}
+	dev_info(&i2c_client->dev,
+		"CS4265 Version %x\n",
+			reg & CS4265_REV_ID_MASK);
+
+	regmap_write(cs4265->regmap, CS4265_PWRCTL, 0x0F);
+
+	ret =  snd_soc_register_codec(&i2c_client->dev,
+			&soc_codec_cs4265, cs4265_dai,
+			ARRAY_SIZE(cs4265_dai));
+	return ret;
+}
+
+static int cs4265_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static const struct of_device_id cs4265_of_match[] = {
+	{ .compatible = "cirrus,cs4265", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, cs4265_of_match);
+
+static const struct i2c_device_id cs4265_id[] = {
+	{ "cs4265", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, cs4265_id);
+
+static struct i2c_driver cs4265_i2c_driver = {
+	.driver = {
+		.name = "cs4265",
+		.owner = THIS_MODULE,
+		.of_match_table = cs4265_of_match,
+	},
+	.id_table = cs4265_id,
+	.probe =    cs4265_i2c_probe,
+	.remove =   cs4265_i2c_remove,
+};
+
+module_i2c_driver(cs4265_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS4265 driver");
+MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan@cirrus.com>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/cs4265.h

+/*
+ * cs4265.h -- CS4265 ALSA SoC audio driver
+ *
+ * Copyright 2014 Cirrus Logic, Inc.
+ *
+ * Author: Paul Handrigan <paul.handrigan@cirrus.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 __CS4265_H__
+#define __CS4265_H__
+
+#define CS4265_CHIP_ID				0x1
+#define CS4265_CHIP_ID_VAL			0xD0
+#define CS4265_CHIP_ID_MASK			0xF0
+#define CS4265_REV_ID_MASK			0x0F
+
+#define CS4265_PWRCTL				0x02
+#define CS4265_PWRCTL_PDN			1
+
+#define CS4265_DAC_CTL				0x3
+#define CS4265_DAC_CTL_MUTE			(1 << 2)
+#define CS4265_DAC_CTL_DIF			(3 << 4)
+
+#define CS4265_ADC_CTL				0x4
+#define CS4265_ADC_MASTER			1
+#define CS4265_ADC_DIF				(1 << 4)
+#define CS4265_ADC_FM				(3 << 6)
+
+#define CS4265_MCLK_FREQ			0x5
+#define CS4265_MCLK_FREQ_MASK			(7 << 4)
+
+#define CS4265_SIG_SEL				0x6
+#define CS4265_SIG_SEL_LOOP			(1 << 1)
+
+#define CS4265_CHB_PGA_CTL			0x7
+#define CS4265_CHA_PGA_CTL			0x8
+
+#define CS4265_ADC_CTL2				0x9
+
+#define CS4265_DAC_CHA_VOL			0xA
+#define CS4265_DAC_CHB_VOL			0xB
+
+#define CS4265_DAC_CTL2				0xC
+
+#define CS4265_INT_STATUS			0xD
+#define CS4265_INT_MASK				0xE
+#define CS4265_STATUS_MODE_MSB			0xF
+#define CS4265_STATUS_MODE_LSB			0x10
+
+#define CS4265_SPDIF_CTL1			0x11
+
+#define CS4265_SPDIF_CTL2			0x12
+#define CS4265_SPDIF_CTL2_MUTE			(1 << 4)
+#define CS4265_SPDIF_CTL2_DIF			(3 << 6)
+
+#define CS4265_C_DATA_BUFF			0x13
+#define CS4265_MAX_REGISTER			0x2A
+
+#endif

sound/soc/codecs/cs42l56.c

@@ -318,24 +318,32 @@ static const struct soc_enum adca_swap_enum =
 			      ARRAY_SIZE(left_swap_text),
 			      left_swap_text,
 			      swap_values);
+static const struct snd_kcontrol_new adca_swap_mux =
+	SOC_DAPM_ENUM("Route", adca_swap_enum);
 
 static const struct soc_enum pcma_swap_enum =
 	SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 4, 3,
 			      ARRAY_SIZE(left_swap_text),
 			      left_swap_text,
 			      swap_values);
+static const struct snd_kcontrol_new pcma_swap_mux =
+	SOC_DAPM_ENUM("Route", pcma_swap_enum);
 
 static const struct soc_enum adcb_swap_enum =
 	SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 2, 3,
 			      ARRAY_SIZE(right_swap_text),
 			      right_swap_text,
 			      swap_values);
+static const struct snd_kcontrol_new adcb_swap_mux =
+	SOC_DAPM_ENUM("Route", adcb_swap_enum);
 
 static const struct soc_enum pcmb_swap_enum =
 	SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 6, 3,
 			      ARRAY_SIZE(right_swap_text),
 			      right_swap_text,
 			      swap_values);
+static const struct snd_kcontrol_new pcmb_swap_mux =
+	SOC_DAPM_ENUM("Route", pcmb_swap_enum);
 
 static const struct snd_kcontrol_new hpa_switch =
 	SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1);
@@ -467,11 +475,6 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
 	SOC_SINGLE("ADCA Invert", CS42L56_MISC_ADC_CTL, 2, 1, 1),
 	SOC_SINGLE("ADCB Invert", CS42L56_MISC_ADC_CTL, 3, 1, 1),
 
-	SOC_ENUM("PCMA Swap", pcma_swap_enum),
-	SOC_ENUM("PCMB Swap", pcmb_swap_enum),
-	SOC_ENUM("ADCA Swap", adca_swap_enum),
-	SOC_ENUM("ADCB Swap", adcb_swap_enum),
-
 	SOC_DOUBLE("HPF Switch", CS42L56_HPF_CTL, 5, 7, 1, 1),
 	SOC_DOUBLE("HPF Freeze Switch", CS42L56_HPF_CTL, 4, 6, 1, 1),
 	SOC_ENUM("HPFA Corner Freq", hpfa_freq_enum),
@@ -570,6 +573,16 @@ static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = {
 	SND_SOC_DAPM_ADC("ADCA", NULL, CS42L56_PWRCTL_1, 1, 1),
 	SND_SOC_DAPM_ADC("ADCB", NULL, CS42L56_PWRCTL_1, 2, 1),
 
+	SND_SOC_DAPM_MUX("ADCA Swap Mux", SND_SOC_NOPM, 0, 0,
+		&adca_swap_mux),
+	SND_SOC_DAPM_MUX("ADCB Swap Mux", SND_SOC_NOPM, 0, 0,
+		&adcb_swap_mux),
+
+	SND_SOC_DAPM_MUX("PCMA Swap Mux", SND_SOC_NOPM, 0, 0,
+		&pcma_swap_mux),
+	SND_SOC_DAPM_MUX("PCMB Swap Mux", SND_SOC_NOPM, 0, 0,
+		&pcmb_swap_mux),
+
 	SND_SOC_DAPM_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0),
 
@@ -607,8 +620,19 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = {
 	{"Digital Output Mux", NULL, "ADCA"},
 	{"Digital Output Mux", NULL, "ADCB"},
 
-	{"ADCB", NULL, "ADCB Mux"},
-	{"ADCA", NULL, "ADCA Mux"},
+	{"ADCB", NULL, "ADCB Swap Mux"},
+	{"ADCA", NULL, "ADCA Swap Mux"},
+
+	{"ADCA Swap Mux", NULL, "ADCA"},
+	{"ADCB Swap Mux", NULL, "ADCB"},
+
+	{"DACA", "Left", "ADCA Swap Mux"},
+	{"DACA", "LR 2", "ADCA Swap Mux"},
+	{"DACA", "Right", "ADCA Swap Mux"},
+
+	{"DACB", "Left", "ADCB Swap Mux"},
+	{"DACB", "LR 2", "ADCB Swap Mux"},
+	{"DACB", "Right", "ADCB Swap Mux"},
 
 	{"ADCA Mux", NULL, "AIN3A"},
 	{"ADCA Mux", NULL, "AIN2A"},
@@ -633,30 +657,32 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = {
 	{"PGAB Input Mux", NULL, "AIN2B"},
 	{"PGAB Input Mux", NULL, "AIN3B"},
 
-	{"LOB", NULL, "Lineout Right"},
-	{"LOA", NULL, "Lineout Left"},
-
-	{"Lineout Right", "Switch", "LINEOUTB Input Mux"},
-	{"Lineout Left", "Switch", "LINEOUTA Input Mux"},
+	{"LOB", "Switch", "LINEOUTB Input Mux"},
+	{"LOA", "Switch", "LINEOUTA Input Mux"},
 
 	{"LINEOUTA Input Mux", "PGAA", "PGAA"},
 	{"LINEOUTB Input Mux", "PGAB", "PGAB"},
 	{"LINEOUTA Input Mux", "DACA", "DACA"},
 	{"LINEOUTB Input Mux", "DACB", "DACB"},
 
-	{"HPA", NULL, "Headphone Left"},
-	{"HPB", NULL, "Headphone Right"},
-
-	{"Headphone Right", "Switch", "HPB Input Mux"},
-	{"Headphone Left", "Switch", "HPA Input Mux"},
+	{"HPA", "Switch", "HPB Input Mux"},
+	{"HPB", "Switch", "HPA Input Mux"},
 
 	{"HPA Input Mux", "PGAA", "PGAA"},
 	{"HPB Input Mux", "PGAB", "PGAB"},
 	{"HPA Input Mux", "DACA", "DACA"},
 	{"HPB Input Mux", "DACB", "DACB"},
 
-	{"DACB", NULL, "HiFi Playback"},
-	{"DACA", NULL, "HiFi Playback"},
+	{"DACA", NULL, "PCMA Swap Mux"},
+	{"DACB", NULL, "PCMB Swap Mux"},
+
+	{"PCMB Swap Mux", "Left", "HiFi Playback"},
+	{"PCMB Swap Mux", "LR 2", "HiFi Playback"},
+	{"PCMB Swap Mux", "Right", "HiFi Playback"},
+
+	{"PCMA Swap Mux", "Left", "HiFi Playback"},
+	{"PCMA Swap Mux", "LR 2", "HiFi Playback"},
+	{"PCMA Swap Mux", "Right", "HiFi Playback"},
 
 };
 

sound/soc/codecs/cs42xx8.c

@@ -219,6 +219,9 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	case SND_SOC_DAIFMT_RIGHT_J:
 		val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ;
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM;
+		break;
 	default:
 		dev_err(codec->dev, "unsupported dai format\n");
 		return -EINVAL;
@@ -422,7 +425,7 @@ const struct cs42xx8_driver_data cs42888_data = {
 };
 EXPORT_SYMBOL_GPL(cs42888_data);
 
-const struct of_device_id cs42xx8_of_match[] = {
+static const struct of_device_id cs42xx8_of_match[] = {
 	{ .compatible = "cirrus,cs42448", .data = &cs42448_data, },
 	{ .compatible = "cirrus,cs42888", .data = &cs42888_data, },
 	{ /* sentinel */ }

sound/soc/codecs/cs42xx8.h

@@ -128,8 +128,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap);
 #define CS42XX8_INTF_DAC_DIF_RIGHTJ		(2 << CS42XX8_INTF_DAC_DIF_SHIFT)
 #define CS42XX8_INTF_DAC_DIF_RIGHTJ_16		(3 << CS42XX8_INTF_DAC_DIF_SHIFT)
 #define CS42XX8_INTF_DAC_DIF_ONELINE_20		(4 << CS42XX8_INTF_DAC_DIF_SHIFT)
-#define CS42XX8_INTF_DAC_DIF_ONELINE_24		(6 << CS42XX8_INTF_DAC_DIF_SHIFT)
-#define CS42XX8_INTF_DAC_DIF_TDM		(7 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_ONELINE_24		(5 << CS42XX8_INTF_DAC_DIF_SHIFT)
+#define CS42XX8_INTF_DAC_DIF_TDM		(6 << CS42XX8_INTF_DAC_DIF_SHIFT)
 #define CS42XX8_INTF_ADC_DIF_SHIFT		0
 #define CS42XX8_INTF_ADC_DIF_WIDTH		3
 #define CS42XX8_INTF_ADC_DIF_MASK		(((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT)
@@ -138,8 +138,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap);
 #define CS42XX8_INTF_ADC_DIF_RIGHTJ		(2 << CS42XX8_INTF_ADC_DIF_SHIFT)
 #define CS42XX8_INTF_ADC_DIF_RIGHTJ_16		(3 << CS42XX8_INTF_ADC_DIF_SHIFT)
 #define CS42XX8_INTF_ADC_DIF_ONELINE_20		(4 << CS42XX8_INTF_ADC_DIF_SHIFT)
-#define CS42XX8_INTF_ADC_DIF_ONELINE_24		(6 << CS42XX8_INTF_ADC_DIF_SHIFT)
-#define CS42XX8_INTF_ADC_DIF_TDM		(7 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_ONELINE_24		(5 << CS42XX8_INTF_ADC_DIF_SHIFT)
+#define CS42XX8_INTF_ADC_DIF_TDM		(6 << CS42XX8_INTF_ADC_DIF_SHIFT)
 
 /* ADC Control & DAC De-Emphasis (Address 05h) */
 #define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT	7

sound/soc/codecs/cx20442.c

@@ -383,8 +383,8 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
 	struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec);
 
 	if (cx20442->control_data) {
-			struct tty_struct *tty = cx20442->control_data;
-			tty_hangup(tty);
+		struct tty_struct *tty = cx20442->control_data;
+		tty_hangup(tty);
 	}
 
 	if (!IS_ERR(cx20442->por)) {

sound/soc/davinci/Kconfig

 config SND_DAVINCI_SOC
-	tristate "SoC Audio for TI DAVINCI or AM33XX/AM43XX chips"
-	depends on ARCH_DAVINCI || SOC_AM33XX || SOC_AM43XX
+	tristate "SoC Audio for TI DAVINCI"
+	depends on ARCH_DAVINCI
+
+config SND_EDMA_SOC
+	tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)"
+	depends on SOC_AM33XX || SOC_AM43XX
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	help
+	  Say Y or M here if you want audio support for TI SoC which uses eDMA.
+	  The following line of SoCs are supported by this platform driver:
+	  - AM335x
+	  - AM437x/AM438x
 
 config SND_DAVINCI_SOC_I2S
 	tristate
 
 config SND_DAVINCI_SOC_MCASP
-	depends on SND_DAVINCI_SOC || SND_OMAP_SOC
-	tristate
+	tristate "Multichannel Audio Serial Port (McASP) support"
+	depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC
+	help
+	  Say Y or M here if you want to have support for McASP IP found in
+	  various Texas Instruments SoCs like:
+	  - daVinci devices
+	  - Sitara line of SoCs (AM335x, AM438x, etc)
+	  - DRA7x devices
 
 config SND_DAVINCI_SOC_VCIF
 	tristate
@@ -19,7 +35,7 @@ config SND_DAVINCI_SOC_GENERIC_EVM
 
 config SND_AM33XX_SOC_EVM
 	tristate "SoC Audio for the AM33XX chip based boards"
-	depends on SND_DAVINCI_SOC && SOC_AM33XX && I2C
+	depends on SND_EDMA_SOC && SOC_AM33XX && I2C
 	select SND_DAVINCI_SOC_GENERIC_EVM
 	help
 	  Say Y or M if you want to add support for SoC audio on AM33XX

sound/soc/davinci/Makefile

 # DAVINCI Platform Support
 snd-soc-davinci-objs := davinci-pcm.o
+snd-soc-edma-objs := edma-pcm.o
 snd-soc-davinci-i2s-objs := davinci-i2s.o
 snd-soc-davinci-mcasp-objs:= davinci-mcasp.o
 snd-soc-davinci-vcif-objs:= davinci-vcif.o
 
 obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o
+obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o
 obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
 obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o
 obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o

sound/soc/davinci/davinci-mcasp.c

@@ -27,6 +27,7 @@
 #include <linux/of_platform.h>
 #include <linux/of_device.h>
 
+#include <sound/asoundef.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -36,6 +37,7 @@
 #include <sound/omap-pcm.h>
 
 #include "davinci-pcm.h"
+#include "edma-pcm.h"
 #include "davinci-mcasp.h"
 
 #define MCASP_MAX_AFIFO_DEPTH	64
@@ -63,6 +65,7 @@ struct davinci_mcasp {
 	u8	num_serializer;
 	u8	*serial_dir;
 	u8	version;
+	u8	bclk_div;
 	u16	bclk_lrclk_ratio;
 	int	streams;
 
@@ -417,6 +420,7 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
 			       ACLKXDIV(div - 1), ACLKXDIV_MASK);
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
 			       ACLKRDIV(div - 1), ACLKRDIV_MASK);
+		mcasp->bclk_div = div;
 		break;
 
 	case 2:		/* BCLK/LRCLK ratio */
@@ -637,8 +641,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream)
 }
 
 /* S/PDIF */
-static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp)
+static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp,
+			      unsigned int rate)
 {
+	u32 cs_value = 0;
+	u8 *cs_bytes = (u8*) &cs_value;
+
 	/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
 	   and LSB first */
 	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15));
@@ -660,6 +668,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp)
 	/* Enable the DIT */
 	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN);
 
+	/* Set S/PDIF channel status bits */
+	cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT;
+	cs_bytes[1] = IEC958_AES1_CON_PCM_CODER;
+
+	switch (rate) {
+	case 22050:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_22050;
+		break;
+	case 24000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_24000;
+		break;
+	case 32000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_32000;
+		break;
+	case 44100:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_44100;
+		break;
+	case 48000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_48000;
+		break;
+	case 88200:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_88200;
+		break;
+	case 96000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_96000;
+		break;
+	case 176400:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_176400;
+		break;
+	case 192000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_192000;
+		break;
+	default:
+		printk(KERN_WARNING "unsupported sampling rate: %d\n", rate);
+		return -EINVAL;
+	}
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value);
+
 	return 0;
 }
 
@@ -675,15 +723,22 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	int period_size = params_period_size(params);
 	int ret;
 
-	/* If mcasp is BCLK master we need to set BCLK divider */
-	if (mcasp->bclk_master) {
+	/*
+	 * If mcasp is BCLK master, and a BCLK divider was not provided by
+	 * the machine driver, we need to calculate the ratio.
+	 */
+	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
 		unsigned int bclk_freq = snd_soc_params_to_bclk(params);
+		unsigned int div = mcasp->sysclk_freq / bclk_freq;
 		if (mcasp->sysclk_freq % bclk_freq != 0) {
-			dev_err(mcasp->dev, "Can't produce required BCLK\n");
-			return -EINVAL;
+			if (((mcasp->sysclk_freq / div) - bclk_freq) >
+			    (bclk_freq - (mcasp->sysclk_freq / (div+1))))
+				div++;
+			dev_warn(mcasp->dev,
+				 "Inaccurate BCLK: %u Hz / %u != %u Hz\n",
+				 mcasp->sysclk_freq, div, bclk_freq);
 		}
-		davinci_mcasp_set_clkdiv(
-			cpu_dai, 1, mcasp->sysclk_freq / bclk_freq);
+		davinci_mcasp_set_clkdiv(cpu_dai, 1, div);
 	}
 
 	ret = mcasp_common_hw_param(mcasp, substream->stream,
@@ -692,7 +747,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
-		ret = mcasp_dit_hw_param(mcasp);
+		ret = mcasp_dit_hw_param(mcasp, params_rate(params));
 	else
 		ret = mcasp_i2s_hw_param(mcasp, substream->stream);
 
@@ -782,7 +837,7 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
-	if (mcasp->version == MCASP_VERSION_4) {
+	if (mcasp->version >= MCASP_VERSION_3) {
 		/* Using dmaengine PCM */
 		dai->playback_dma_data =
 				&mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
@@ -1232,10 +1287,16 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	 IS_MODULE(CONFIG_SND_DAVINCI_SOC))
 	case MCASP_VERSION_1:
 	case MCASP_VERSION_2:
-	case MCASP_VERSION_3:
 		ret = davinci_soc_platform_register(&pdev->dev);
 		break;
 #endif
+#if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \
+	(IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+	 IS_MODULE(CONFIG_SND_EDMA_SOC))
+	case MCASP_VERSION_3:
+		ret = edma_pcm_platform_register(&pdev->dev);
+		break;
+#endif
 #if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \
 	(IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
 	 IS_MODULE(CONFIG_SND_OMAP_SOC))

sound/soc/davinci/edma-pcm.c

@@ -28,8 +28,8 @@
 static const struct snd_pcm_hardware edma_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
-				  SNDRV_PCM_INFO_BATCH |
 				  SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
 				  SNDRV_PCM_INFO_INTERLEAVED,
 	.buffer_bytes_max	= 128 * 1024,
 	.period_bytes_min	= 32,

sound/soc/davinci/edma-pcm.h

@@ -20,6 +20,13 @@
 #ifndef __EDMA_PCM_H__
 #define __EDMA_PCM_H__
 
+#if IS_ENABLED(CONFIG_SND_EDMA_SOC)
 int edma_pcm_platform_register(struct device *dev);
+#else
+static inline int edma_pcm_platform_register(struct device *dev)
+{
+	return 0;
+}
+#endif /* CONFIG_SND_EDMA_SOC */
 
 #endif /* __EDMA_PCM_H__ */