Merge remote branch 'tiwai/topic/asoc' into for-2.6.38

include/sound/alc5623.h

+#ifndef _INCLUDE_SOUND_ALC5623_H
+#define _INCLUDE_SOUND_ALC5623_H
+struct alc5623_platform_data {
+	/* configure :                              */
+	/* Lineout/Speaker Amps Vmid ratio control  */
+	/* enable/disable adc/dac high pass filters */
+	unsigned int add_ctrl;
+	/* configure :                              */
+	/* output to enable when jack is low        */
+	/* output to enable when jack is high       */
+	/* jack detect (gpio/nc/jack detect [12]    */
+	unsigned int jack_det_ctrl;
+};
+#endif
+

sound/soc/codecs/Kconfig

@@ -22,11 +22,13 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
+	select SND_SOC_ALC5623 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS42L51 if I2C
 	select SND_SOC_CS4270 if I2C
+	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
-	select SND_SOC_JZ4740 if SOC_JZ4740
+	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
 	select SND_SOC_MAX98088 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
@@ -129,6 +131,9 @@ config SND_SOC_AK4642
 config SND_SOC_AK4671
 	tristate
 
+config SND_SOC_ALC5623
+       tristate
+
 config SND_SOC_CQ0093VC
 	tristate
 
@@ -317,3 +322,4 @@ config SND_SOC_WM2000
 
 config SND_SOC_WM9090
 	tristate
+

sound/soc/codecs/Makefile

@@ -17,6 +17,7 @@ snd-soc-da7210-objs := da7210.o
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-alc5623-objs := alc5623.o
 snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-stac9766-objs := stac9766.o
@@ -92,6 +93,7 @@ obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o

sound/soc/codecs/alc5623.h

+/*
+ * alc5623.h  --  alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * Author: flove <flove@realtek.com>
+ * 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 version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _ALC5623_H
+#define _ALC5623_H
+
+#define ALC5623_RESET				0x00
+/*				5621 5622 5623  */
+/* speaker output vol		   2    2       */
+/* line output vol                      4    2  */
+/* HP output vol		   4    0    4  */
+#define ALC5623_SPK_OUT_VOL			0x02
+#define ALC5623_HP_OUT_VOL			0x04
+#define ALC5623_MONO_AUX_OUT_VOL		0x06
+#define ALC5623_AUXIN_VOL			0x08
+#define ALC5623_LINE_IN_VOL			0x0A
+#define ALC5623_STEREO_DAC_VOL			0x0C
+#define ALC5623_MIC_VOL				0x0E
+#define ALC5623_MIC_ROUTING_CTRL		0x10
+#define ALC5623_ADC_REC_GAIN			0x12
+#define ALC5623_ADC_REC_MIXER			0x14
+#define ALC5623_SOFT_VOL_CTRL_TIME		0x16
+/* ALC5623_OUTPUT_MIXER_CTRL :			*/
+/* same remark as for reg 2 line vs speaker	*/
+#define ALC5623_OUTPUT_MIXER_CTRL		0x1C
+#define ALC5623_MIC_CTRL			0x22
+
+#define	ALC5623_DAI_CONTROL			0x34
+#define ALC5623_DAI_SDP_MASTER_MODE		(0 << 15)
+#define ALC5623_DAI_SDP_SLAVE_MODE		(1 << 15)
+#define ALC5623_DAI_I2S_PCM_MODE		(1 << 14)
+#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL	(1 <<  7)
+#define ALC5623_DAI_ADC_DATA_L_R_SWAP		(1 <<  5)
+#define ALC5623_DAI_DAC_DATA_L_R_SWAP		(1 <<  4)
+#define ALC5623_DAI_I2S_DL_MASK			(3 <<  2)
+#define ALC5623_DAI_I2S_DL_32			(3 <<  2)
+#define	ALC5623_DAI_I2S_DL_24			(2 <<  2)
+#define ALC5623_DAI_I2S_DL_20			(1 <<  2)
+#define ALC5623_DAI_I2S_DL_16			(0 <<  2)
+#define ALC5623_DAI_I2S_DF_PCM			(3 <<  0)
+#define	ALC5623_DAI_I2S_DF_LEFT			(2 <<  0)
+#define ALC5623_DAI_I2S_DF_RIGHT		(1 <<  0)
+#define ALC5623_DAI_I2S_DF_I2S			(0 <<  0)
+
+#define ALC5623_STEREO_AD_DA_CLK_CTRL		0x36
+#define	ALC5623_COMPANDING_CTRL			0x38
+
+#define	ALC5623_PWR_MANAG_ADD1			0x3A
+#define ALC5623_PWR_ADD1_MAIN_I2S_EN		(1 << 15)
+#define ALC5623_PWR_ADD1_ZC_DET_PD_EN		(1 << 14)
+#define ALC5623_PWR_ADD1_MIC1_BIAS_EN		(1 << 11)
+#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN	(1 << 10)
+#define ALC5623_PWR_ADD1_SOFTGEN_EN		(1 <<  8) /* rsvd on 5622 */
+#define	ALC5623_PWR_ADD1_DEPOP_BUF_HP		(1 <<  6) /* rsvd on 5622 */
+#define	ALC5623_PWR_ADD1_HP_OUT_AMP		(1 <<  5)
+#define	ALC5623_PWR_ADD1_HP_OUT_ENH_AMP		(1 <<  4) /* rsvd on 5622 */
+#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX		(1 <<  2)
+#define ALC5623_PWR_ADD1_AUX_OUT_AMP		(1 <<  1)
+#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP	(1 <<  0) /* rsvd on 5622 */
+
+#define ALC5623_PWR_MANAG_ADD2			0x3C
+#define ALC5623_PWR_ADD2_LINEOUT		(1 << 15) /* rt5623 */
+#define ALC5623_PWR_ADD2_CLASS_AB		(1 << 15) /* rt5621 */
+#define ALC5623_PWR_ADD2_CLASS_D		(1 << 14) /* rt5621 */
+#define ALC5623_PWR_ADD2_VREF			(1 << 13)
+#define ALC5623_PWR_ADD2_PLL			(1 << 12)
+#define ALC5623_PWR_ADD2_DAC_REF_CIR		(1 << 10)
+#define ALC5623_PWR_ADD2_L_DAC_CLK		(1 <<  9)
+#define ALC5623_PWR_ADD2_R_DAC_CLK		(1 <<  8)
+#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN		(1 <<  7)
+#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN		(1 <<  6)
+#define ALC5623_PWR_ADD2_L_HP_MIXER		(1 <<  5)
+#define ALC5623_PWR_ADD2_R_HP_MIXER		(1 <<  4)
+#define ALC5623_PWR_ADD2_SPK_MIXER		(1 <<  3)
+#define ALC5623_PWR_ADD2_MONO_MIXER		(1 <<  2)
+#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER	(1 <<  1)
+#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER	(1 <<  0)
+
+#define ALC5623_PWR_MANAG_ADD3			0x3E
+#define ALC5623_PWR_ADD3_MAIN_BIAS		(1 << 15)
+#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP	(1 << 14)
+#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP	(1 << 13)
+#define ALC5623_PWR_ADD3_SPK_OUT		(1 << 12)
+#define ALC5623_PWR_ADD3_HP_L_OUT_VOL		(1 << 10)
+#define ALC5623_PWR_ADD3_HP_R_OUT_VOL		(1 <<  9)
+#define ALC5623_PWR_ADD3_LINEIN_L_VOL		(1 <<  7)
+#define ALC5623_PWR_ADD3_LINEIN_R_VOL		(1 <<  6)
+#define ALC5623_PWR_ADD3_AUXIN_L_VOL		(1 <<  5)
+#define ALC5623_PWR_ADD3_AUXIN_R_VOL		(1 <<  4)
+#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL		(1 <<  3)
+#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL		(1 <<  2)
+#define ALC5623_PWR_ADD3_MIC1_BOOST_AD		(1 <<  1)
+#define ALC5623_PWR_ADD3_MIC2_BOOST_AD		(1 <<  0)
+
+#define ALC5623_ADD_CTRL_REG			0x40
+
+#define	ALC5623_GLOBAL_CLK_CTRL_REG		0x42
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL	(1 << 15)
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK	(0 << 15)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK	(1 << 14)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK	(0 << 14)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8	(3 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4	(2 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2	(1 <<  1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1	(0 <<  1)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV2		(1 <<  0)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV1		(0 <<  0)
+
+#define ALC5623_PLL_CTRL			0x44
+#define ALC5623_PLL_CTRL_N_VAL(n)		(((n)&0xff) << 8)
+#define ALC5623_PLL_CTRL_K_VAL(k)		(((k)&0x7)  << 4)
+#define ALC5623_PLL_CTRL_M_VAL(m)		((m)&0xf)
+
+#define ALC5623_GPIO_OUTPUT_PIN_CTRL		0x4A
+#define ALC5623_GPIO_PIN_CONFIG			0x4C
+#define ALC5623_GPIO_PIN_POLARITY		0x4E
+#define ALC5623_GPIO_PIN_STICKY			0x50
+#define ALC5623_GPIO_PIN_WAKEUP			0x52
+#define ALC5623_GPIO_PIN_STATUS			0x54
+#define ALC5623_GPIO_PIN_SHARING		0x56
+#define	ALC5623_OVER_CURR_STATUS		0x58
+#define ALC5623_JACK_DET_CTRL			0x5A
+
+#define ALC5623_MISC_CTRL			0x5E
+#define ALC5623_MISC_DISABLE_FAST_VREG		(1 << 15)
+#define ALC5623_MISC_SPK_CLASS_AB_OC_PD		(1 << 13) /* 5621 */
+#define ALC5623_MISC_SPK_CLASS_AB_OC_DET	(1 << 12) /* 5621 */
+#define ALC5623_MISC_HP_DEPOP_MODE3_EN		(1 << 10)
+#define ALC5623_MISC_HP_DEPOP_MODE2_EN		(1 <<  9)
+#define ALC5623_MISC_HP_DEPOP_MODE1_EN		(1 <<  8)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN	(1 <<  6)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN	(1 <<  5)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN	(1 <<  4)
+#define ALC5623_MISC_M_DAC_L_INPUT		(1 <<  3)
+#define ALC5623_MISC_M_DAC_R_INPUT		(1 <<  2)
+#define ALC5623_MISC_IRQOUT_INV_CTRL		(1 <<  0)
+
+#define	ALC5623_PSEDUEO_SPATIAL_CTRL		0x60
+#define ALC5623_EQ_CTRL				0x62
+#define ALC5623_EQ_MODE_ENABLE			0x66
+#define ALC5623_AVC_CTRL			0x68
+#define ALC5623_HID_CTRL_INDEX			0x6A
+#define ALC5623_HID_CTRL_DATA			0x6C
+#define ALC5623_VENDOR_ID1			0x7C
+#define ALC5623_VENDOR_ID2			0x7E
+
+#define ALC5623_PLL_FR_MCLK			0
+#define ALC5623_PLL_FR_BCK			1
+#endif

sound/soc/codecs/wm8900.c

@@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg)
 {
 	switch (reg) {
 	case WM8900_REG_ID:
-	case WM8900_REG_POWER1:
 		return 1;
 	default:
 		return 0;
@@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec)
 		return -ENODEV;
 	}
 
-	/* Read back from the chip */
-	reg = snd_soc_read(codec, WM8900_REG_POWER1);
-	reg = (reg >> 12) & 0xf;
-	dev_info(codec->dev, "WM8900 revision %d\n", reg);
-
 	wm8900_reset(codec);
 
 	/* Turn the chip on */

sound/soc/codecs/wm8993.c

@@ -735,6 +735,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol,
 					    0);
 		}
 		wm8993->class_w_users++;
+		wm8993->hubs_data.class_w = true;
 	}
 
 	/* Implement the change */
@@ -751,6 +752,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol,
 					    WM8993_CP_DYN_V);
 		}
 		wm8993->class_w_users--;
+		wm8993->hubs_data.class_w = false;
 	}
 
 	dev_dbg(codec->dev, "Indirect DAC use count now %d\n",

sound/soc/codecs/wm8994.c

@@ -2228,6 +2228,7 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
 
 static void wm8994_update_class_w(struct snd_soc_codec *codec)
 {
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int enable = 1;
 	int source = 0;  /* GCC flow analysis can't track enable */
 	int reg, reg_r;
@@ -2278,11 +2279,13 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
 				    WM8994_CP_DYN_PWR |
 				    WM8994_CP_DYN_SRC_SEL_MASK,
 				    source | WM8994_CP_DYN_PWR);
+		wm8994->hubs.class_w = true;
 		
 	} else {
 		dev_dbg(codec->dev, "Class W disabled\n");
 		snd_soc_update_bits(codec, WM8994_CLASS_W_1,
 				    WM8994_CP_DYN_PWR, 0);
+		wm8994->hubs.class_w = false;
 	}
 }
 

sound/soc/codecs/wm_hubs.c

@@ -94,6 +94,18 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	u16 reg, reg_l, reg_r, dcs_cfg;
 
+	/* If we're using a digital only path and have a previously
+	 * callibrated DC servo offset stored then use that. */
+	if (hubs->class_w && hubs->class_w_dcs) {
+		dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
+			hubs->class_w_dcs);
+		snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+		wait_for_dc_servo(codec,
+				  WM8993_DCS_TRIG_DAC_WR_0 |
+				  WM8993_DCS_TRIG_DAC_WR_1);
+		return;
+	}
+
 	/* Set for 32 series updates */
 	snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
 			    WM8993_DCS_SERIES_NO_01_MASK,
@@ -101,34 +113,34 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 	wait_for_dc_servo(codec,
 			  WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1);
 
+	/* Different chips in the family support different readback
+	 * methods.
+	 */
+	switch (hubs->dcs_readback_mode) {
+	case 0:
+		reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+			& WM8993_DCS_INTEG_CHAN_0_MASK;;
+		reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
+			& WM8993_DCS_INTEG_CHAN_1_MASK;
+		break;
+	case 1:
+		reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+		reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+			>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+		break;
+	default:
+		WARN(1, "Unknown DCS readback method\n");
+		break;
+	}
+
+	dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
+
 	/* Apply correction to DC servo result */
 	if (hubs->dcs_codes) {
 		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
 			hubs->dcs_codes);
 
-		/* Different chips in the family support different
-		 * readback methods.
-		 */
-		switch (hubs->dcs_readback_mode) {
-		case 0:
-			reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
-				& WM8993_DCS_INTEG_CHAN_0_MASK;;
-			reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
-				& WM8993_DCS_INTEG_CHAN_1_MASK;
-			break;
-		case 1:
-			reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
-			reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
-				>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
-			reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
-			break;
-		default:
-			WARN(1, "Unknown DCS readback method");
-			break;
-		}
-
-		dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
-
 		/* HPOUT1L */
 		if (reg_l + hubs->dcs_codes > 0 &&
 		    reg_l + hubs->dcs_codes < 0xff)
@@ -148,7 +160,15 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 		wait_for_dc_servo(codec,
 				  WM8993_DCS_TRIG_DAC_WR_0 |
 				  WM8993_DCS_TRIG_DAC_WR_1);
+	} else {
+		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		dcs_cfg |= reg_r;
 	}
+
+	/* Save the callibrated offset if we're in class W mode and
+	 * therefore don't have any analogue signal mixed in. */
+	if (hubs->class_w)
+		hubs->class_w_dcs = dcs_cfg;
 }
 
 /*
@@ -163,6 +183,9 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 
 	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
 
+	/* Updating the analogue gains invalidates the DC servo cache */
+	hubs->class_w_dcs = 0;
+
 	/* If we're applying an offset correction then updating the
 	 * callibration would be likely to introduce further offsets. */
 	if (hubs->dcs_codes)

sound/soc/codecs/wm_hubs.h

@@ -23,6 +23,9 @@ struct wm_hubs_data {
 	int dcs_codes;
 	int dcs_readback_mode;
 	int hp_startup_mode;
+
+	bool class_w;
+	u16 class_w_dcs;
 };
 
 extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);

sound/soc/kirkwood/Kconfig

@@ -18,3 +18,12 @@ config SND_KIRKWOOD_SOC_OPENRD
 	  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
+	select SND_KIRKWOOD_SOC_I2S
+	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

@@ -5,5 +5,7 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
 obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
 
 snd-soc-openrd-objs := kirkwood-openrd.o
+snd-soc-t5325-objs := kirkwood-t5325.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-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 <sound/soc-dapm.h>
+#include <mach/kirkwood.h>
+#include <plat/audio.h>
+#include <asm/mach-types.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;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+	unsigned int freq, fmt;
+
+	fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	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 int t5325_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+
+	snd_soc_dapm_new_controls(codec, t5325_dapm_widgets,
+				ARRAY_SIZE(t5325_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, t5325_route, ARRAY_SIZE(t5325_route));
+
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Speaker");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link t5325_dai[] = {
+{
+	.name = "ALC5621",
+	.stream_name = "ALC5621 HiFi",
+	.cpu_dai_name = "kirkwood-i2s",
+	.platform_name = "kirkwood-pcm-audio",
+	.codec_dai_name = "alc5621-hifi",
+	.codec_name = "alc562x-codec.0-001a",
+	.ops = &t5325_ops,
+	.init = t5325_dai_init,
+},
+};
+
+
+static struct snd_soc_card t5325 = {
+	.name = "t5325",
+	.dai_link = t5325_dai,
+	.num_links = ARRAY_SIZE(t5325_dai),
+};
+
+static struct platform_device *t5325_snd_device;
+
+static int __init t5325_init(void)
+{
+	int ret;
+
+	if (!machine_is_t5325())
+		return 0;
+
+	t5325_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!t5325_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(t5325_snd_device,
+			&t5325);
+
+	ret = platform_device_add(t5325_snd_device);
+	if (ret) {
+		printk(KERN_ERR "%s: platform_device_add failed\n", __func__);
+		platform_device_put(t5325_snd_device);
+	}
+
+	return ret;
+}
+module_init(t5325_init);
+
+static void __exit t5325_exit(void)
+{
+	platform_device_unregister(t5325_snd_device);
+}
+module_exit(t5325_exit);
+
+MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
+MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
+MODULE_LICENSE("GPL");

sound/soc/pxa/tosa.c

@@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);

sound/soc/soc-core.c

@@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev,
 {
 	struct snd_soc_pcm_runtime *rtd =
 			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	int ret;
 
-	strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	if (ret)
+		return ret;
 
 	return count;
 }