Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/kirkwood',...

Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/kirkwood', 'asoc/topic/max98090' and 'asoc/topic/mc13783' into asoc-next

Documentation/devicetree/bindings/sound/max98090.txt

@@ -4,7 +4,7 @@ This device supports I2C only.
 
 Required properties:
 
-- compatible : "maxim,max98090".
+- compatible : "maxim,max98090" or "maxim,max98091".
 
 - reg : The I2C address of the device.
 

arch/x86/include/asm/platform_sst_audio.h

+/*
+ * platform_sst_audio.h:  sst audio platform data header file
+ *
+ * Copyright (C) 2012-14 Intel Corporation
+ * Author: Jeeja KP <jeeja.kp@intel.com>
+ * 	Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
+ *	Vinod Koul ,vinod.koul@intel.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; version 2
+ * of the License.
+ */
+#ifndef _PLATFORM_SST_AUDIO_H_
+#define _PLATFORM_SST_AUDIO_H_
+
+#include <linux/sfi.h>
+
+enum sst_audio_task_id_mrfld {
+	SST_TASK_ID_NONE = 0,
+	SST_TASK_ID_SBA = 1,
+	SST_TASK_ID_MEDIA = 3,
+	SST_TASK_ID_MAX = SST_TASK_ID_MEDIA,
+};
+
+/* Device IDs for Merrifield are Pipe IDs,
+ * ref: DSP spec v0.75 */
+enum sst_audio_device_id_mrfld {
+	/* Output pipeline IDs */
+	PIPE_ID_OUT_START = 0x0,
+	PIPE_CODEC_OUT0 = 0x2,
+	PIPE_CODEC_OUT1 = 0x3,
+	PIPE_SPROT_LOOP_OUT = 0x4,
+	PIPE_MEDIA_LOOP1_OUT = 0x5,
+	PIPE_MEDIA_LOOP2_OUT = 0x6,
+	PIPE_VOIP_OUT = 0xC,
+	PIPE_PCM0_OUT = 0xD,
+	PIPE_PCM1_OUT = 0xE,
+	PIPE_PCM2_OUT = 0xF,
+	PIPE_MEDIA0_OUT = 0x12,
+	PIPE_MEDIA1_OUT = 0x13,
+/* Input Pipeline IDs */
+	PIPE_ID_IN_START = 0x80,
+	PIPE_CODEC_IN0 = 0x82,
+	PIPE_CODEC_IN1 = 0x83,
+	PIPE_SPROT_LOOP_IN = 0x84,
+	PIPE_MEDIA_LOOP1_IN = 0x85,
+	PIPE_MEDIA_LOOP2_IN = 0x86,
+	PIPE_VOIP_IN = 0x8C,
+	PIPE_PCM0_IN = 0x8D,
+	PIPE_PCM1_IN = 0x8E,
+	PIPE_MEDIA0_IN = 0x8F,
+	PIPE_MEDIA1_IN = 0x90,
+	PIPE_MEDIA2_IN = 0x91,
+	PIPE_RSVD = 0xFF,
+};
+
+/* The stream map for each platform consists of an array of the below
+ * stream map structure.
+ */
+struct sst_dev_stream_map {
+	u8 dev_num;		/* device id */
+	u8 subdev_num;		/* substream */
+	u8 direction;
+	u8 device_id;		/* fw id */
+	u8 task_id;		/* fw task */
+	u8 status;
+};
+
+struct sst_platform_data {
+	/* Intel software platform id*/
+	struct sst_dev_stream_map *pdev_strm_map;
+	unsigned int strm_map_size;
+};
+
+int add_sst_platform_device(void);
+#endif
+

include/sound/rt286.h

+/*
+ * linux/sound/rt286.h -- Platform data for RT286
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * 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 __LINUX_SND_RT286_H
+#define __LINUX_SND_RT286_H
+
+struct rt286_platform_data {
+	bool cbj_en; /*combo jack enable*/
+	bool gpio2_en; /*GPIO2 enable*/
+};
+
+#endif

sound/soc/codecs/Kconfig

@@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM3008
 	select SND_SOC_PCM512x_I2C if I2C
 	select SND_SOC_PCM512x_SPI if SPI_MASTER
+	select SND_SOC_RT286 if I2C
 	select SND_SOC_RT5631 if I2C
 	select SND_SOC_RT5640 if I2C
 	select SND_SOC_RT5645 if I2C
@@ -455,6 +456,9 @@ config SND_SOC_RL6231
 	default m if SND_SOC_RT5645=m
 	default m if SND_SOC_RT5651=m
 
+config SND_SOC_RT286
+	tristate
+
 config SND_SOC_RT5631
 	tristate
 

sound/soc/codecs/Makefile

@@ -69,6 +69,7 @@ snd-soc-pcm512x-objs := pcm512x.o
 snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
 snd-soc-pcm512x-spi-objs := pcm512x-spi.o
 snd-soc-rl6231-objs := rl6231.o
+snd-soc-rt286-objs := rt286.o
 snd-soc-rt5631-objs := rt5631.o
 snd-soc-rt5640-objs := rt5640.o
 snd-soc-rt5645-objs := rt5645.o
@@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_PCM512x)	+= snd-soc-pcm512x.o
 obj-$(CONFIG_SND_SOC_PCM512x_I2C)	+= snd-soc-pcm512x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM512x_SPI)	+= snd-soc-pcm512x-spi.o
 obj-$(CONFIG_SND_SOC_RL6231)	+= snd-soc-rl6231.o
+obj-$(CONFIG_SND_SOC_RT286)	+= snd-soc-rt286.o
 obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_RT5640)	+= snd-soc-rt5640.o
 obj-$(CONFIG_SND_SOC_RT5645)	+= snd-soc-rt5645.o

sound/soc/codecs/max98090.c

@@ -26,10 +26,6 @@
 #include <sound/max98090.h>
 #include "max98090.h"
 
-#define DEBUG
-#define EXTMIC_METHOD
-#define EXTMIC_METHOD_TEST
-
 /* Allows for sparsely populated register maps */
 static struct reg_default max98090_reg[] = {
 	{ 0x00, 0x00 }, /* 00 Software Reset */
@@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w,
 	else
 		val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT;
 
-
 	if (val >= 1) {
 		if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) {
 			max98090->pa1en = val - 1; /* Update for volatile */
@@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux =
 	SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum);
 
 static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
-
 	SND_SOC_DAPM_INPUT("MIC1"),
 	SND_SOC_DAPM_INPUT("MIC2"),
 	SND_SOC_DAPM_INPUT("DMICL"),
@@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
-
 	SND_SOC_DAPM_INPUT("DMIC3"),
 	SND_SOC_DAPM_INPUT("DMIC4"),
 
@@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
-
 	{"MIC1 Input", NULL, "MIC1"},
 	{"MIC2 Input", NULL, "MIC2"},
 
@@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
 	{"SPKR", NULL, "SPK Right Out"},
 	{"RCVL", NULL, "RCV Left Out"},
 	{"RCVR", NULL, "RCV Right Out"},
-
 };
 
 static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
-
 	/* DMIC inputs */
 	{"DMIC3", NULL, "DMIC3_ENA"},
 	{"DMIC4", NULL, "DMIC4_ENA"},
 	{"DMIC3", NULL, "AHPF"},
 	{"DMIC4", NULL, "AHPF"},
-
 };
 
 static int max98090_add_widgets(struct snd_soc_codec *codec)
@@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec)
 
 		snd_soc_dapm_add_routes(dapm, max98091_dapm_routes,
 			ARRAY_SIZE(max98091_dapm_routes));
-
 	}
 
 	return 0;
@@ -2212,22 +2200,11 @@ static struct snd_soc_dai_driver max98090_dai[] = {
 }
 };
 
-static void max98090_handle_pdata(struct snd_soc_codec *codec)
-{
-	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
-	struct max98090_pdata *pdata = max98090->pdata;
-
-	if (!pdata) {
-		dev_err(codec->dev, "No platform data\n");
-		return;
-	}
-
-}
-
 static int max98090_probe(struct snd_soc_codec *codec)
 {
 	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
 	struct max98090_cdata *cdata;
+	enum max98090_type devtype;
 	int ret = 0;
 
 	dev_dbg(codec->dev, "max98090_probe\n");
@@ -2263,16 +2240,21 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	}
 
 	if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) {
-		max98090->devtype = MAX98090;
+		devtype = MAX98090;
 		dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret);
 	} else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) {
-		max98090->devtype = MAX98091;
+		devtype = MAX98091;
 		dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret);
 	} else {
-		max98090->devtype = MAX98090;
+		devtype = MAX98090;
 		dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret);
 	}
 
+	if (max98090->devtype != devtype) {
+		dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n");
+		max98090->devtype = devtype;
+	}
+
 	max98090->jack_state = M98090_JACK_STATE_NO_HEADSET;
 
 	INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work);
@@ -2317,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE,
 		M98090_MBVSEL_MASK, M98090_MBVSEL_2V8);
 
-	max98090_handle_pdata(codec);
-
 	max98090_add_widgets(codec);
 
 err_access:
@@ -2428,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev)
 }
 #endif
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int max98090_resume(struct device *dev)
 {
 	struct max98090_priv *max98090 = dev_get_drvdata(dev);
@@ -2460,12 +2440,14 @@ static const struct dev_pm_ops max98090_pm = {
 
 static const struct i2c_device_id max98090_i2c_id[] = {
 	{ "max98090", MAX98090 },
+	{ "max98091", MAX98091 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, max98090_i2c_id);
 
 static const struct of_device_id max98090_of_match[] = {
 	{ .compatible = "maxim,max98090", },
+	{ .compatible = "maxim,max98091", },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, max98090_of_match);

sound/soc/codecs/mc13783.c

@@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
 
 		ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port);
 		if (ret)
-			return ret;
+			goto out;
 
 		ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port);
 		if (ret)
-			return ret;
+			goto out;
 	}
 
 	dev_set_drvdata(&pdev->dev, priv);
@@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
 		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
 			mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
 
+out:
+	of_node_put(np);
 	return ret;
 }
 

sound/soc/codecs/rt286.h

+/*
+ * rt286.h  --  RT286 ALSA SoC audio driver
+ *
+ * Copyright 2011 Realtek Microelectronics
+ * Author: Johnny Hsu <johnnyhsu@realtek.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 __RT286_H__
+#define __RT286_H__
+
+#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
+
+#define RT286_AUDIO_FUNCTION_GROUP			0x01
+#define RT286_DAC_OUT1					0x02
+#define RT286_DAC_OUT2					0x03
+#define RT286_ADC_IN1					0x09
+#define RT286_ADC_IN2					0x08
+#define RT286_MIXER_IN					0x0b
+#define RT286_MIXER_OUT1				0x0c
+#define RT286_MIXER_OUT2				0x0d
+#define RT286_DMIC1					0x12
+#define RT286_DMIC2					0x13
+#define RT286_SPK_OUT					0x14
+#define RT286_MIC1					0x18
+#define RT286_LINE1					0x1a
+#define RT286_BEEP					0x1d
+#define RT286_SPDIF					0x1e
+#define RT286_VENDOR_REGISTERS				0x20
+#define RT286_HP_OUT					0x21
+#define RT286_MIXER_IN1					0x22
+#define RT286_MIXER_IN2					0x23
+
+#define RT286_SET_PIN_SFT				6
+#define RT286_SET_PIN_ENABLE				0x40
+#define RT286_SET_PIN_DISABLE				0
+#define RT286_SET_EAPD_HIGH				0x2
+#define RT286_SET_EAPD_LOW				0
+
+#define RT286_MUTE_SFT					7
+
+/* Verb commands */
+#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM)
+#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0)
+#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP)
+#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT)
+#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT)
+#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1)
+#define RT286_SPK_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0)
+#define RT286_HPO_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0)
+#define RT286_ADC0_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0)
+#define RT286_ADC1_MUX\
+	VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0)
+#define RT286_SET_MIC1\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0)
+#define RT286_SET_PIN_HPO\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0)
+#define RT286_SET_PIN_SPK\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0)
+#define RT286_SET_PIN_DMIC1\
+	VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0)
+#define RT286_SPK_EAPD\
+	VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0)
+#define RT286_SET_AMP_GAIN_HPO\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN1\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0)
+#define RT286_SET_AMP_GAIN_ADC_IN2\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0)
+#define RT286_GET_HP_SENSE\
+	VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0)
+#define RT286_GET_MIC1_SENSE\
+	VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0)
+#define RT286_SET_DMIC2_DEFAULT\
+	VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0)
+#define RT286_DACL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000)
+#define RT286_DACR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000)
+#define RT286_ADCL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000)
+#define RT286_ADCR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000)
+#define RT286_MIC_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000)
+#define RT286_SPOL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000)
+#define RT286_SPOR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000)
+#define RT286_HPOL_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000)
+#define RT286_HPOR_GAIN\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000)
+#define RT286_F_DAC_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000)
+#define RT286_F_RECMIX_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100)
+#define RT286_REC_MIC_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000)
+#define RT286_REC_I2S_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100)
+#define RT286_REC_LINE_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200)
+#define RT286_REC_BEEP_SWITCH\
+	VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300)
+#define RT286_DAC_FORMAT\
+	VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0)
+#define RT286_ADC_FORMAT\
+	VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0)
+#define RT286_COEF_INDEX\
+	VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0)
+#define RT286_PROC_COEF\
+	VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0)
+
+/* Index registers */
+#define RT286_A_BIAS_CTRL1	0x01
+#define RT286_A_BIAS_CTRL2	0x02
+#define RT286_POWER_CTRL1	0x03
+#define RT286_A_BIAS_CTRL3	0x04
+#define RT286_POWER_CTRL2	0x08
+#define RT286_I2S_CTRL1		0x09
+#define RT286_I2S_CTRL2		0x0a
+#define RT286_CLK_DIV		0x0b
+#define RT286_DC_GAIN		0x0d
+#define RT286_POWER_CTRL3	0x0f
+#define RT286_MIC1_DET_CTRL	0x19
+#define RT286_MISC_CTRL1	0x20
+#define RT286_IRQ_CTRL		0x33
+#define RT286_PLL_CTRL1		0x49
+#define RT286_CBJ_CTRL1		0x4f
+#define RT286_CBJ_CTRL2		0x50
+#define RT286_PLL_CTRL		0x63
+#define RT286_DEPOP_CTRL1	0x66
+#define RT286_DEPOP_CTRL2	0x67
+#define RT286_DEPOP_CTRL3	0x68
+#define RT286_DEPOP_CTRL4	0x69
+
+/* SPDIF (0x06) */
+#define RT286_SPDIF_SEL_SFT	0
+#define RT286_SPDIF_SEL_PCM0	0
+#define RT286_SPDIF_SEL_PCM1	1
+#define RT286_SPDIF_SEL_SPOUT	2
+#define RT286_SPDIF_SEL_PP	3
+
+/* RECMIX (0x0b) */
+#define RT286_M_REC_BEEP_SFT	0
+#define RT286_M_REC_LINE1_SFT	1
+#define RT286_M_REC_MIC1_SFT	2
+#define RT286_M_REC_I2S_SFT	3
+
+/* Front (0x0c) */
+#define RT286_M_FRONT_DAC_SFT	0
+#define RT286_M_FRONT_REC_SFT	1
+
+/* SPK-OUT (0x14) */
+#define RT286_M_SPK_MUX_SFT	14
+#define RT286_SPK_SEL_MASK	0x1
+#define RT286_SPK_SEL_SFT	0
+#define RT286_SPK_SEL_F		0
+#define RT286_SPK_SEL_S		1
+
+/* HP-OUT (0x21) */
+#define RT286_M_HP_MUX_SFT	14
+#define RT286_HP_SEL_MASK	0x1
+#define RT286_HP_SEL_SFT	0
+#define RT286_HP_SEL_F		0
+#define RT286_HP_SEL_S		1
+
+/* ADC (0x22) (0x23) */
+#define RT286_ADC_SEL_MASK	0x7
+#define RT286_ADC_SEL_SFT	0
+#define RT286_ADC_SEL_SURR	0
+#define RT286_ADC_SEL_FRONT	1
+#define RT286_ADC_SEL_DMIC	2
+#define RT286_ADC_SEL_BEEP	4
+#define RT286_ADC_SEL_LINE1	5
+#define RT286_ADC_SEL_I2S	6
+#define RT286_ADC_SEL_MIC1	7
+
+#define RT286_SCLK_S_MCLK	0
+#define RT286_SCLK_S_PLL	1
+
+enum {
+	RT286_AIF1,
+	RT286_AIF2,
+	RT286_AIFS,
+};
+
+int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
+
+#endif /* __RT286_H__ */
+

sound/soc/intel/Kconfig

@@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH
 	help
 	  This adds audio driver for Intel Baytrail platform based boards
 	  with the MAX98090 audio codec.
+
+config SND_SOC_INTEL_BROADWELL_MACH
+	tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint"
+	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC
+	select SND_SOC_INTEL_HASWELL
+	select SND_COMPRESS_OFFLOAD
+	select SND_SOC_RT286
+	help
+	  This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell
+	  Ultrabook platforms.
+	  Say Y if you have such a device
+	  If unsure select "N".

sound/soc/intel/Makefile

@@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o
 snd-soc-sst-haswell-objs := haswell.o
 snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
 snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
+snd-soc-sst-broadwell-objs := broadwell.o
 
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o

sound/soc/intel/broadwell.c

+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt286.h"
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("DMIC1", NULL),
+	SND_SOC_DAPM_MIC("DMIC2", NULL),
+	SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+	/* speaker */
+	{"Speaker", NULL, "SPOR"},
+	{"Speaker", NULL, "SPOL"},
+
+	/* HP jack connectors - unknown if we have jack deteck */
+	{"Headphones", NULL, "HPO Pin"},
+
+	/* other jacks */
+	{"MIC1", NULL, "Mic Jack"},
+	{"LINE1", NULL, "Line Jack"},
+
+	/* digital mics */
+	{"DMIC1 Pin", NULL, "DMIC1"},
+	{"DMIC2 Pin", NULL, "DMIC2"},
+
+	/* CODEC BE connections */
+	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The ADSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
+static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+		SND_SOC_CLOCK_IN);
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+	.hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+	struct sst_hsw *broadwell = pdata->dsp;
+	int ret;
+
+	/* Set ADSP SSP port settings */
+	ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+		SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+		SST_HSW_DEVICE_CLOCK_MASTER, 9);
+	if (ret < 0) {
+		dev_err(rtd->dev, "error: failed to set device config\n");
+		return ret;
+	}
+
+	/* always connected - check HP for jack detect */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line Jack");
+	snd_soc_dapm_enable_pin(dapm, "DMIC1");
+	snd_soc_dapm_enable_pin(dapm, "DMIC2");
+
+	return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+	/* Front End DAI links */
+	{
+		.name = "System PCM",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.init = broadwell_rtd_init,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload0",
+		.stream_name = "Offload0 Playback",
+		.cpu_dai_name = "Offload0 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Offload1",
+		.stream_name = "Offload1 Playback",
+		.cpu_dai_name = "Offload1 Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	{
+		.name = "Loopback PCM",
+		.stream_name = "Loopback",
+		.cpu_dai_name = "Loopback Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 0,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "Capture PCM",
+		.stream_name = "Capture",
+		.cpu_dai_name = "Capture Pin",
+		.platform_name = "haswell-pcm-audio",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "Codec",
+		.be_id = 0,
+		.cpu_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "i2c-INT343A:00",
+		.codec_dai_name = "rt286-aif1",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = broadwell_ssp0_fixup,
+		.ops = &broadwell_rt286_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+	.name = "broadwell-rt286",
+	.owner = THIS_MODULE,
+	.dai_link = broadwell_rt286_dais,
+	.num_links = ARRAY_SIZE(broadwell_rt286_dais),
+	.dapm_widgets = broadwell_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+	.dapm_routes = broadwell_rt286_map,
+	.num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+	.fully_routed = true,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+	broadwell_rt286.dev = &pdev->dev;
+
+	return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&broadwell_rt286);
+	return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+	.probe = broadwell_audio_probe,
+	.remove = broadwell_audio_remove,
+	.driver = {
+		.name = "broadwell-audio",
+		.owner = THIS_MODULE,
+	},
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");

sound/soc/intel/byt-max98090.c

@@ -63,14 +63,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = {
 		.pin	= "Headset Mic",
 		.mask	= SND_JACK_MICROPHONE,
 	},
-	{
-		.pin	= "Ext Spk",
-		.mask	= SND_JACK_LINEOUT,
-	},
-	{
-		.pin	= "Int Mic",
-		.mask	= SND_JACK_LINEIN,
-	},
 };
 
 static struct snd_soc_jack_gpio hs_jack_gpios[] = {

sound/soc/intel/byt-rt5640.c

@@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+	{"Headset Mic", NULL, "MICBIAS1"},
 	{"IN2P", NULL, "Headset Mic"},
 	{"IN2N", NULL, "Headset Mic"},
 	{"DMIC1", NULL, "Internal Mic"},

sound/soc/intel/sst-atom-controls.h

+/*
+ *  Copyright (C) 2013-14 Intel Corp
+ *  Author: Ramesh Babu <ramesh.babu.koul@intel.com>
+ *  	Omair M Abdullah <omair.m.abdullah@intel.com>
+ *  	Samreen Nilofer <samreen.nilofer@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+
+#ifndef __SST_CONTROLS_V2_H__
+#define __SST_CONTROLS_V2_H__
+
+enum {
+	MERR_DPCM_AUDIO = 0,
+	MERR_DPCM_COMPR,
+};
+
+
+#endif

sound/soc/intel/sst-baytrail-ipc.c

@@ -122,6 +122,26 @@ struct sst_byt_tstamp {
 	u32 channel_peak[8];
 } __packed;
 
+struct sst_byt_fw_version {
+	u8 build;
+	u8 minor;
+	u8 major;
+	u8 type;
+} __packed;
+
+struct sst_byt_fw_build_info {
+	u8 date[16];
+	u8 time[16];
+} __packed;
+
+struct sst_byt_fw_init {
+	struct sst_byt_fw_version fw_version;
+	struct sst_byt_fw_build_info build_info;
+	u16 result;
+	u8 module_id;
+	u8 debug_info;
+} __packed;
+
 /* driver internal IPC message structure */
 struct ipc_message {
 	struct list_head list;
@@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 {
 	struct sst_byt *byt;
 	struct sst_fw *byt_sst_fw;
+	struct sst_byt_fw_init init;
 	int err;
 
 	dev_dbg(dev, "initialising Byt DSP IPC\n");
@@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
 		goto boot_err;
 	}
 
+	/* show firmware information */
+	sst_dsp_inbox_read(byt->dsp, &init, sizeof(init));
+	dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n",
+		 init.fw_version.major, init.fw_version.minor,
+		 init.fw_version.build, init.fw_version.type);
+	dev_info(byt->dev, "Build type: %x\n", init.fw_version.type);
+	dev_info(byt->dev, "Build date: %s %s\n",
+		 init.build_info.date, init.build_info.time);
+
 	pdata->dsp = byt;
 	byt->fw = byt_sst_fw;
 

sound/soc/intel/sst-dsp.c

@@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64);
 
 void sst_dsp_dump(struct sst_dsp *sst)
 {
-	sst->ops->dump(sst);
+	if (sst->ops->dump)
+		sst->ops->dump(sst);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_dump);
 
 void sst_dsp_reset(struct sst_dsp *sst)
 {
-	sst->ops->reset(sst);
+	if (sst->ops->reset)
+		sst->ops->reset(sst);
 }
 EXPORT_SYMBOL_GPL(sst_dsp_reset);
 
 int sst_dsp_boot(struct sst_dsp *sst)
 {
-	sst->ops->boot(sst);
+	if (sst->ops->boot)
+		sst->ops->boot(sst);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(sst_dsp_boot);

sound/soc/intel/sst-dsp.h

@@ -52,7 +52,11 @@
 #define SST_CLKCTL		0x78
 #define SST_CSR2		0x80
 #define SST_LTRC		0xE0
-#define SST_HDMC		0xE8
+#define SST_HMDC		0xE8
+
+#define SST_SHIM_BEGIN		SST_CSR
+#define SST_SHIM_END		SST_HDMC
+
 #define SST_DBGO		0xF0
 
 #define SST_SHIM_SIZE		0x100
@@ -73,6 +77,8 @@
 #define SST_CSR_S0IOCS		(0x1 << 21)
 #define SST_CSR_S1IOCS		(0x1 << 23)
 #define SST_CSR_LPCS		(0x1 << 31)
+#define SST_CSR_24MHZ_LPCS	(SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS)
+#define SST_CSR_24MHZ_NO_LPCS	(SST_CSR_SBCS0 | SST_CSR_SBCS1)
 #define SST_BYT_CSR_RST		(0x1 << 0)
 #define SST_BYT_CSR_VECTOR_SEL	(0x1 << 1)
 #define SST_BYT_CSR_STALL	(0x1 << 2)
@@ -92,6 +98,14 @@
 #define SST_IMRX_DONE		(0x1 << 0)
 #define SST_BYT_IMRX_REQUEST	(0x1 << 1)
 
+/* IMRD / IMD */
+#define SST_IMRD_DONE		(0x1 << 0)
+#define SST_IMRD_BUSY		(0x1 << 1)
+#define SST_IMRD_SSP0		(0x1 << 16)
+#define SST_IMRD_DMAC0		(0x1 << 21)
+#define SST_IMRD_DMAC1		(0x1 << 22)
+#define SST_IMRD_DMAC		(SST_IMRD_DMAC0 | SST_IMRD_DMAC1)
+
 /*  IPCX / IPCC */
 #define	SST_IPCX_DONE		(0x1 << 30)
 #define	SST_IPCX_BUSY		(0x1 << 31)
@@ -118,9 +132,21 @@
 /* LTRC */
 #define SST_LTRC_VAL(x)		(x << 0)
 
-/* HDMC */
-#define SST_HDMC_HDDA0(x)	(x << 0)
-#define SST_HDMC_HDDA1(x)	(x << 7)
+/* HMDC */
+#define SST_HMDC_HDDA0(x)	(x << 0)
+#define SST_HMDC_HDDA1(x)	(x << 7)
+#define SST_HMDC_HDDA_E0_CH0	1
+#define SST_HMDC_HDDA_E0_CH1	2
+#define SST_HMDC_HDDA_E0_CH2	4
+#define SST_HMDC_HDDA_E0_CH3	8
+#define SST_HMDC_HDDA_E1_CH0	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0)
+#define SST_HMDC_HDDA_E1_CH1	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1)
+#define SST_HMDC_HDDA_E1_CH2	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2)
+#define SST_HMDC_HDDA_E1_CH3	SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E0_ALLCH	(SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \
+				 SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3)
+#define SST_HMDC_HDDA_E1_ALLCH	(SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \
+				 SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3)
 
 
 /* SST Vendor Defined Registers and bits */
@@ -130,11 +156,16 @@
 #define SST_VDRTCTL3		0xaC
 
 /* VDRTCTL0 */
+#define SST_VDRTCL0_APLLSE_MASK		1
 #define SST_VDRTCL0_DSRAMPGE_SHIFT	16
 #define SST_VDRTCL0_DSRAMPGE_MASK	(0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
 #define SST_VDRTCL0_ISRAMPGE_SHIFT	6
 #define SST_VDRTCL0_ISRAMPGE_MASK	(0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
 
+/* PMCS */
+#define SST_PMCS		0x84
+#define SST_PMCS_PS_MASK	0x3
+
 struct sst_dsp;
 
 /*

sound/soc/intel/sst-haswell-dsp.c

@@ -28,9 +28,6 @@
 #include <linux/firmware.h>
 #include <linux/pm_runtime.h>
 
-#include <linux/acpi.h>
-#include <acpi/acpi_bus.h>
-
 #include "sst-dsp.h"
 #include "sst-dsp-priv.h"
 #include "sst-haswell-ipc.h"
@@ -272,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst)
 		SST_CSR2_SDFD_SSP1);
 
 	/* enable DMA engine 0,1 all channels to access host memory */
-	sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC,
-		SST_HDMC_HDDA1(0xff)  | SST_HDMC_HDDA0(0xff),
-		SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff));
+	sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC,
+		SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff),
+		SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff));
 
 	/* disable all clock gating */
 	writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2);
@@ -313,9 +310,7 @@ static const struct sst_adsp_memregion lp_region[] = {
 
 /* wild cat point ADSP mem regions */
 static const struct sst_adsp_memregion wpt_region[] = {
-	{0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
-	{0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
-	{0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */
+	{0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */
 	{0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */
 };
 
@@ -339,21 +334,40 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
 	return 0;
 }
 
+struct sst_sram_shift {
+	u32 dev_id;	/* SST Device IDs  */
+	u32 iram_shift;
+	u32 dram_shift;
+};
+
+static const struct sst_sram_shift sram_shift[] = {
+	{SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */
+	{SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */
+};
 static u32 hsw_block_get_bit(struct sst_mem_block *block)
 {
-	u32 bit = 0, shift = 0;
+	u32 bit = 0, shift = 0, index;
+	struct sst_dsp *sst = block->dsp;
 
-	switch (block->type) {
-	case SST_MEM_DRAM:
-		shift = 16;
-		break;
-	case SST_MEM_IRAM:
-		shift = 6;
-		break;
-	default:
-		return 0;
+	for (index = 0; index < ARRAY_SIZE(sram_shift); index++) {
+		if (sram_shift[index].dev_id == sst->id)
+			break;
 	}
 
+	if (index < ARRAY_SIZE(sram_shift)) {
+		switch (block->type) {
+		case SST_MEM_DRAM:
+			shift = sram_shift[index].dram_shift;
+			break;
+		case SST_MEM_IRAM:
+			shift = sram_shift[index].iram_shift;
+			break;
+		default:
+			shift = 0;
+		}
+	} else
+		shift = 0;
+
 	bit = 1 << (block->index + shift);
 
 	return bit;
@@ -501,8 +515,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
 		}
 	}
 
-	/* set default power gating mask */
-	writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0);
+	/* set default power gating control, enable power gating control for all blocks. that is,
+	can't be accessed, please enable each block before accessing. */
+	writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0);
 
 	return 0;
 }

sound/soc/intel/sst-haswell-ipc.c

@@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready {
 	u32 inbox_size;
 	u32 outbox_size;
 	u32 fw_info_size;
-	u8 fw_info[1];
+	u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
 } __attribute__((packed));
 
 struct ipc_message {
@@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work)
 		return;
 	}
 
-	/* if the DSP is busy we will TX messages after IRQ */
+	/* if the DSP is busy, we will TX messages after IRQ.
+	 * also postpone if we are in the middle of procesing completion irq*/
 	ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX);
-	if (ipcx & SST_IPCX_BUSY) {
+	if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) {
 		spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
 		return;
 	}
@@ -502,6 +503,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg,
 		ipc_shim_dbg(hsw, "message timeout");
 
 		trace_ipc_error("error message timeout for", msg->header);
+		list_del(&msg->list);
 		ret = -ETIMEDOUT;
 	} else {
 
@@ -569,6 +571,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
 {
 	struct sst_hsw_ipc_fw_ready fw_ready;
 	u32 offset;
+	u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)];
+	char *tmp[5], *pinfo;
+	int i = 0;
 
 	offset = (header & 0x1FFFFFFF) << 3;
 
@@ -589,6 +594,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
 		fw_ready.inbox_offset, fw_ready.inbox_size);
 	dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n",
 		fw_ready.outbox_offset, fw_ready.outbox_size);
+	if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) {
+		fw_ready.fw_info[fw_ready.fw_info_size] = 0;
+		dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info);
+
+		/* log the FW version info got from the mailbox here. */
+		memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size);
+		pinfo = &fw_info[0];
+		for (i = 0; i < sizeof(tmp) / sizeof(char *); i++)
+			tmp[i] = strsep(&pinfo, " ");
+		dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - "
+			"version: %s.%s, build %s, source commit id: %s\n",
+			tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]);
+	}
 }
 
 static void hsw_notification_work(struct work_struct *work)
@@ -671,7 +689,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg)
 	switch (stream_msg) {
 	case IPC_STR_STAGE_MESSAGE:
 	case IPC_STR_NOTIFICATION:
+		break;
 	case IPC_STR_RESET:
+		trace_ipc_notification("stream reset", stream->reply.stream_hw_id);
 		break;
 	case IPC_STR_PAUSE:
 		stream->running = false;
@@ -762,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
 	}
 
 	/* update any stream states */
-	hsw_stream_update(hsw, msg);
+	if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE)
+		hsw_stream_update(hsw, msg);
 
 	/* wake up and return the error if we have waiters on this message ? */
 	list_del(&msg->list);
@@ -1628,7 +1649,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 	enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
 {
 	u32 header, state_;
-	int ret;
+	int ret, item;
 
 	header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
 	state_ = state;
@@ -1642,6 +1663,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw,
 		return ret;
 	}
 
+	for (item = 0; item < dx->entries_no; item++) {
+		dev_dbg(hsw->dev,
+			"Item[%d] offset[%x] - size[%x] - source[%x]\n",
+			item, dx->mem_info[item].offset,
+			dx->mem_info[item].size,
+			dx->mem_info[item].source);
+	}
 	dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n",
 		dx->entries_no, state);
 
@@ -1775,8 +1803,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
 
 	/* get the FW version */
 	sst_hsw_fw_get_version(hsw, &version);
-	dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n",
-		version.type, version.major, version.minor, version.build);
 
 	/* get the globalmixer */
 	ret = sst_hsw_mixer_get_info(hsw);

sound/soc/intel/sst-mfld-platform-compress.c

@@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
 	int retval;
 	struct snd_sst_params str_params;
 	struct sst_compress_cb cb;
+	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct sst_data *ctx = snd_soc_platform_get_drvdata(platform);
 
 	stream = cstream->runtime->private_data;
 	/* construct fw structure for this*/
 	memset(&str_params, 0, sizeof(str_params));
 
-	str_params.ops = STREAM_OPS_PLAYBACK;
-	str_params.stream_type = SST_STREAM_TYPE_MUSIC;
-	str_params.device_type = SND_SST_DEVICE_COMPRESS;
+	/* fill the device type and stream id to pass to SST driver */
+	retval = sst_fill_stream_params(cstream, ctx, &str_params, true);
+	pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval);
+	if (retval < 0)
+		return retval;
 
 	switch (params->codec.id) {
 	case SND_AUDIOCODEC_MP3: {

sound/soc/intel/sst-mfld-platform.h

@@ -39,9 +39,10 @@ extern struct sst_device *sst;
 
 struct pcm_stream_info {
 	int str_id;
-	void *mad_substream;
-	void (*period_elapsed) (void *mad_substream);
+	void *arg;
+	void (*period_elapsed) (void *arg);
 	unsigned long long buffer_ptr;
+	unsigned long long pcm_delay;
 	int sfreq;
 };
 
@@ -62,7 +63,9 @@ enum sst_controls {
 	SST_SND_BUFFER_POINTER =	0x05,
 	SST_SND_STREAM_INIT =		0x06,
 	SST_SND_START	 =		0x07,
-	SST_MAX_CONTROLS =		0x07,
+	SST_SET_BYTE_STREAM =           0x100A,
+	SST_GET_BYTE_STREAM =           0x100B,
+	SST_MAX_CONTROLS = SST_GET_BYTE_STREAM,
 };
 
 enum sst_stream_ops {
@@ -124,8 +127,9 @@ struct compress_sst_ops {
 };
 
 struct sst_ops {
-	int (*open) (struct sst_stream_params *str_param);
+	int (*open) (struct snd_sst_params *str_param);
 	int (*device_control) (int cmd, void *arg);
+	int (*set_generic_params)(enum sst_controls cmd, void *arg);
 	int (*close) (unsigned int str_id);
 };
 
@@ -143,10 +147,27 @@ struct sst_device {
 	char *name;
 	struct device *dev;
 	struct sst_ops *ops;
+	struct platform_device *pdev;
 	struct compress_sst_ops *compr_ops;
 };
 
+struct sst_data;
 void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
+int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
+			   struct snd_sst_params *str_params, bool is_compress);
+
+struct sst_algo_int_control_v2 {
+	struct soc_mixer_control mc;
+	u16 module_id; /* module identifieer */
+	u16 pipe_id; /* location info: pipe_id + instance_id */
+	u16 instance_id;
+	unsigned int value; /* Value received is stored here */
+};
+struct sst_data {
+	struct platform_device *pdev;
+	struct sst_platform_data *pdata;
+	struct mutex lock;
+};
 int sst_register_dsp(struct sst_device *sst);
 int sst_unregister_dsp(struct sst_device *sst);
 #endif

sound/soc/kirkwood/Kconfig

 config SND_KIRKWOOD_SOC
 	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
-	depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST
+	depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
@@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB
 	  Say Y if you want to add support for SoC audio on
 	  the Armada 370 Development Board.
 
-config SND_KIRKWOOD_SOC_OPENRD
-	tristate "SoC Audio support for Kirkwood Openrd Client"
-	depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
-	depends on I2C
-	select SND_SOC_CS42L51
-	help
-	  Say Y if you want to add support for SoC audio on
-	  Openrd Client.
-
-config SND_KIRKWOOD_SOC_T5325
-	tristate "SoC Audio support for HP t5325"
-	depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C
-	select SND_SOC_ALC5623
-	help
-	  Say Y if you want to add support for SoC audio on
-	  the HP t5325 thin client.
-

sound/soc/kirkwood/Makefile

@@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o
 
 obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
 
-snd-soc-openrd-objs := kirkwood-openrd.o
-snd-soc-t5325-objs := kirkwood-t5325.o
 snd-soc-armada-370-db-objs := armada-370-db.o
 
 obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
-obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o

sound/soc/kirkwood/kirkwood-dma.c

@@ -28,11 +28,12 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
 }
 
 static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
-	.info = (SNDRV_PCM_INFO_INTERLEAVED |
-		 SNDRV_PCM_INFO_MMAP |
-		 SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_PAUSE),
+	.info = SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_PAUSE |
+		SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
 	.buffer_bytes_max	= KIRKWOOD_SND_MAX_BUFFER_BYTES,
 	.period_bytes_min	= KIRKWOOD_SND_MIN_PERIOD_BYTES,
 	.period_bytes_max	= KIRKWOOD_SND_MAX_PERIOD_BYTES,

sound/soc/kirkwood/kirkwood-i2s.c

@@ -212,7 +212,8 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
 				    KIRKWOOD_PLAYCTL_SIZE_MASK);
 		priv->ctl_play |= ctl_play;
 	} else {
-		priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK;
+		priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK |
+				   KIRKWOOD_RECCTL_SIZE_MASK);
 		priv->ctl_rec |= ctl_rec;
 	}
 
@@ -221,14 +222,24 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static unsigned kirkwood_i2s_play_mute(unsigned ctl)
+{
+	if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN))
+		ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE;
+	if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN))
+		ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE;
+	return ctl;
+}
+
 static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 				int cmd, struct snd_soc_dai *dai)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai);
 	uint32_t ctl, value;
 
 	ctl = readl(priv->io + KIRKWOOD_PLAYCTL);
-	if (ctl & KIRKWOOD_PLAYCTL_PAUSE) {
+	if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) {
 		unsigned timeout = 5000;
 		/*
 		 * The Armada510 spec says that if we enter pause mode, the
@@ -256,14 +267,16 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 			ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;	/* i2s */
 		else
 			ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;	/* spdif */
-
+		ctl = kirkwood_i2s_play_mute(ctl);
 		value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
 		/* enable interrupts */
-		value = readl(priv->io + KIRKWOOD_INT_MASK);
-		value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
-		writel(value, priv->io + KIRKWOOD_INT_MASK);
+		if (!runtime->no_period_wakeup) {
+			value = readl(priv->io + KIRKWOOD_INT_MASK);
+			value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
+			writel(value, priv->io + KIRKWOOD_INT_MASK);
+		}
 
 		/* enable playback */
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
@@ -295,6 +308,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE |
 				KIRKWOOD_PLAYCTL_SPDIF_MUTE);
+		ctl = kirkwood_i2s_play_mute(ctl);
 		writel(ctl, priv->io + KIRKWOOD_PLAYCTL);
 		break;
 
@@ -322,8 +336,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
 		else
 			ctl &= ~KIRKWOOD_RECCTL_I2S_EN;		/* spdif */
 
-		value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN |
-				KIRKWOOD_RECCTL_SPDIF_EN);
+		value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_RECCTL);
 
 		/* enable interrupts */
@@ -347,7 +360,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
 
 		/* disable all records */
 		value = readl(priv->io + KIRKWOOD_RECCTL);
-		value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+		value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_RECCTL);
 		break;
 
@@ -411,7 +424,7 @@ static int kirkwood_i2s_init(struct kirkwood_dma_data *priv)
 	writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
 	value = readl(priv->io + KIRKWOOD_RECCTL);
-	value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+	value &= ~KIRKWOOD_RECCTL_ENABLE_MASK;
 	writel(value, priv->io + KIRKWOOD_RECCTL);
 
 	return 0;

sound/soc/kirkwood/kirkwood-openrd.c

-/*
- * kirkwood-openrd.c
- *
- * (c) 2010 Arnaud Patard <apatard@mandriva.com>
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <sound/soc.h>
-#include <linux/platform_data/asoc-kirkwood.h>
-#include "../codecs/cs42l51.h"
-
-static int openrd_client_hw_params(struct snd_pcm_substream *substream,
-		struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	unsigned int freq;
-
-	switch (params_rate(params)) {
-	default:
-	case 44100:
-		freq = 11289600;
-		break;
-	case 48000:
-		freq = 12288000;
-		break;
-	case 96000:
-		freq = 24576000;
-		break;
-	}
-
-	return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
-
-}
-
-static struct snd_soc_ops openrd_client_ops = {
-	.hw_params = openrd_client_hw_params,
-};
-
-
-static struct snd_soc_dai_link openrd_client_dai[] = {
-{
-	.name = "CS42L51",
-	.stream_name = "CS42L51 HiFi",
-	.cpu_dai_name = "i2s",
-	.platform_name = "mvebu-audio",
-	.codec_dai_name = "cs42l51-hifi",
-	.codec_name = "cs42l51-codec.0-004a",
-	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
-	.ops = &openrd_client_ops,
-},
-};
-
-
-static struct snd_soc_card openrd_client = {
-	.name = "OpenRD Client",
-	.owner = THIS_MODULE,
-	.dai_link = openrd_client_dai,
-	.num_links = ARRAY_SIZE(openrd_client_dai),
-};
-
-static int openrd_probe(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = &openrd_client;
-	int ret;
-
-	card->dev = &pdev->dev;
-
-	ret = snd_soc_register_card(card);
-	if (ret)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
-	return ret;
-}
-
-static int openrd_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
-static struct platform_driver openrd_driver = {
-	.driver		= {
-		.name	= "openrd-client-audio",
-		.owner	= THIS_MODULE,
-	},
-	.probe		= openrd_probe,
-	.remove		= openrd_remove,
-};
-
-module_platform_driver(openrd_driver);
-
-/* Module information */
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC OpenRD Client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:openrd-client-audio");

sound/soc/kirkwood/kirkwood-t5325.c

-/*
- * kirkwood-t5325.c
- *
- * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
- *
- *  This program is free software; you can redistribute  it and/or modify it
- *  under  the terms of  the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the  License, or (at your
- *  option) any later version.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <sound/soc.h>
-#include <linux/platform_data/asoc-kirkwood.h>
-#include "../codecs/alc5623.h"
-
-static int t5325_hw_params(struct snd_pcm_substream *substream,
-		struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	unsigned int freq;
-
-	freq = params_rate(params) * 256;
-
-	return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
-
-}
-
-static struct snd_soc_ops t5325_ops = {
-	.hw_params = t5325_hw_params,
-};
-
-static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone Jack", NULL),
-	SND_SOC_DAPM_SPK("Speaker", NULL),
-	SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route t5325_route[] = {
-	{ "Headphone Jack",	NULL,	"HPL" },
-	{ "Headphone Jack",	NULL,	"HPR" },
-
-	{"Speaker",		NULL,	"SPKOUT"},
-	{"Speaker",		NULL,	"SPKOUTN"},
-
-	{ "MIC1",		NULL,	"Mic Jack" },
-	{ "MIC2",		NULL,	"Mic Jack" },
-};
-
-static struct snd_soc_dai_link t5325_dai[] = {
-{
-	.name = "ALC5621",
-	.stream_name = "ALC5621 HiFi",
-	.cpu_dai_name = "i2s",
-	.platform_name = "mvebu-audio",
-	.codec_dai_name = "alc5621-hifi",
-	.codec_name = "alc562x-codec.0-001a",
-	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
-	.ops = &t5325_ops,
-},
-};
-
-static struct snd_soc_card t5325 = {
-	.name = "t5325",
-	.owner = THIS_MODULE,
-	.dai_link = t5325_dai,
-	.num_links = ARRAY_SIZE(t5325_dai),
-
-	.dapm_widgets = t5325_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets),
-	.dapm_routes = t5325_route,
-	.num_dapm_routes = ARRAY_SIZE(t5325_route),
-};
-
-static int t5325_probe(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = &t5325;
-	int ret;
-
-	card->dev = &pdev->dev;
-
-	ret = snd_soc_register_card(card);
-	if (ret)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
-			ret);
-	return ret;
-}
-
-static int t5325_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_card(card);
-	return 0;
-}
-
-static struct platform_driver t5325_driver = {
-	.driver		= {
-		.name	= "t5325-audio",
-		.owner	= THIS_MODULE,
-	},
-	.probe		= t5325_probe,
-	.remove		= t5325_remove,
-};
-
-module_platform_driver(t5325_driver);
-
-MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
-MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:t5325-audio");

sound/soc/kirkwood/kirkwood.h

@@ -38,6 +38,9 @@
 #define KIRKWOOD_RECCTL_SIZE_24		(1<<0)
 #define KIRKWOOD_RECCTL_SIZE_32		(0<<0)
 
+#define KIRKWOOD_RECCTL_ENABLE_MASK		(KIRKWOOD_RECCTL_SPDIF_EN | \
+						 KIRKWOOD_RECCTL_I2S_EN)
+
 #define KIRKWOOD_REC_BUF_ADDR			0x1004
 #define KIRKWOOD_REC_BUF_SIZE			0x1008
 #define KIRKWOOD_REC_BYTE_COUNT			0x100C
@@ -121,9 +124,9 @@
 
 /* Theses values come from the marvell alsa driver */
 /* need to find where they come from               */
-#define KIRKWOOD_SND_MIN_PERIODS		8
+#define KIRKWOOD_SND_MIN_PERIODS		2
 #define KIRKWOOD_SND_MAX_PERIODS		16
-#define KIRKWOOD_SND_MIN_PERIOD_BYTES		0x800
+#define KIRKWOOD_SND_MIN_PERIOD_BYTES		256
 #define KIRKWOOD_SND_MAX_PERIOD_BYTES		0x8000
 #define KIRKWOOD_SND_MAX_BUFFER_BYTES		(KIRKWOOD_SND_MAX_PERIOD_BYTES \
 						 * KIRKWOOD_SND_MAX_PERIODS)