Merge branch 'wip-2.6.34' into old-private-masterarchived-private-master

28 files changed, 797 insertions, 51 deletions

diff --git a/include/sound/Kbuild b/include/sound/Kbuild
index fd054a344324..e9dd9369ecb9 100644
--- a/include/sound/Kbuild
+++ b/include/sound/Kbuild
@@ -2,7 +2,6 @@ header-y += asound_fm.h
 header-y += hdsp.h
 header-y += hdspm.h
 header-y += sfnt_info.h
-header-y += sscape_ioctl.h
 
 unifdef-y += asequencer.h
 unifdef-y += asound.h
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 3dae3f799b9b..49400459b477 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -593,7 +593,7 @@ enum {
 
 struct ac97_quirk {
         unsigned short subvendor; /* PCI subsystem vendor id */
-        unsigned short subdevice; /* PCI sybsystem device id */
+        unsigned short subdevice; /* PCI subsystem device id */
         unsigned short mask;    /* device id bit mask, 0 = accept all */
         unsigned int codec_id;  /* codec id (if any), 0 = accept all */
         const char *name;       /* name shown as info */
diff --git a/include/sound/aci.h b/include/sound/aci.h
new file mode 100644
index 000000000000..ee639d355ef0
--- /dev/null
+++ b/include/sound/aci.h
@@ -0,0 +1,90 @@
+#ifndef _ACI_H_
+#define _ACI_H_
+
+#define ACI_REG_COMMAND         0       /* write register offset */
+#define ACI_REG_STATUS          1       /* read register offset */
+#define ACI_REG_BUSY            2       /* busy register offset */
+#define ACI_REG_RDS             2       /* PCM20: RDS register offset */
+#define ACI_MINTIME             500     /* ACI time out limit */
+
+#define ACI_SET_MUTE            0x0d
+#define ACI_SET_POWERAMP        0x0f
+#define ACI_SET_TUNERMUTE       0xa3
+#define ACI_SET_TUNERMONO       0xa4
+#define ACI_SET_IDE             0xd0
+#define ACI_SET_WSS             0xd1
+#define ACI_SET_SOLOMODE        0xd2
+#define ACI_SET_PREAMP          0x03
+#define ACI_GET_PREAMP          0x21
+#define ACI_WRITE_TUNE          0xa7
+#define ACI_READ_TUNERSTEREO    0xa8
+#define ACI_READ_TUNERSTATION   0xa9
+#define ACI_READ_VERSION        0xf1
+#define ACI_READ_IDCODE         0xf2
+#define ACI_INIT                0xff
+#define ACI_STATUS              0xf0
+#define ACI_S_GENERAL           0x00
+#define ACI_ERROR_OP            0xdf
+
+/* ACI Mixer */
+
+/* These are the values for the right channel GET registers.
+   Add an offset of 0x01 for the left channel register.
+   (left=right+0x01) */
+
+#define ACI_GET_MASTER          0x03
+#define ACI_GET_MIC             0x05
+#define ACI_GET_LINE            0x07
+#define ACI_GET_CD              0x09
+#define ACI_GET_SYNTH           0x0b
+#define ACI_GET_PCM             0x0d
+#define ACI_GET_LINE1           0x10    /* Radio on PCM20 */
+#define ACI_GET_LINE2           0x12
+
+#define ACI_GET_EQ1             0x22    /* from Bass ... */
+#define ACI_GET_EQ2             0x24
+#define ACI_GET_EQ3             0x26
+#define ACI_GET_EQ4             0x28
+#define ACI_GET_EQ5             0x2a
+#define ACI_GET_EQ6             0x2c
+#define ACI_GET_EQ7             0x2e    /* ... to Treble */
+
+/* And these are the values for the right channel SET registers.
+   For left channel access you have to add an offset of 0x08.
+   MASTER is an exception, which needs an offset of 0x01 */
+
+#define ACI_SET_MASTER          0x00
+#define ACI_SET_MIC             0x30
+#define ACI_SET_LINE            0x31
+#define ACI_SET_CD              0x34
+#define ACI_SET_SYNTH           0x33
+#define ACI_SET_PCM             0x32
+#define ACI_SET_LINE1           0x35    /* Radio on PCM20 */
+#define ACI_SET_LINE2           0x36
+
+#define ACI_SET_EQ1             0x40    /* from Bass ... */
+#define ACI_SET_EQ2             0x41
+#define ACI_SET_EQ3             0x42
+#define ACI_SET_EQ4             0x43
+#define ACI_SET_EQ5             0x44
+#define ACI_SET_EQ6             0x45
+#define ACI_SET_EQ7             0x46    /* ... to Treble */
+
+struct snd_miro_aci {
+        unsigned long aci_port;
+        int aci_vendor;
+        int aci_product;
+        int aci_version;
+        int aci_amp;
+        int aci_preamp;
+        int aci_solomode;
+
+        struct mutex aci_mutex;
+};
+
+int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3);
+
+struct snd_miro_aci *snd_aci_get_aci(void);
+
+#endif  /* _ACI_H_ */
+
diff --git a/include/sound/ak4113.h b/include/sound/ak4113.h
new file mode 100644
index 000000000000..2609048c1d44
--- /dev/null
+++ b/include/sound/ak4113.h
@@ -0,0 +1,321 @@
+#ifndef __SOUND_AK4113_H
+#define __SOUND_AK4113_H
+
+/*
+ *  Routines for Asahi Kasei AK4113
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *  Copyright (c) by Pavel Hofman <pavel.hofman@ivitera.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; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   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.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+/* AK4113 registers */
+/* power down */
+#define AK4113_REG_PWRDN        0x00
+/* format control */
+#define AK4113_REG_FORMAT       0x01
+/* input/output control */
+#define AK4113_REG_IO0          0x02
+/* input/output control */
+#define AK4113_REG_IO1          0x03
+/* interrupt0 mask */
+#define AK4113_REG_INT0_MASK    0x04
+/* interrupt1 mask */
+#define AK4113_REG_INT1_MASK    0x05
+/* DAT mask & DTS select */
+#define AK4113_REG_DATDTS       0x06
+/* receiver status 0 */
+#define AK4113_REG_RCS0         0x07
+/* receiver status 1 */
+#define AK4113_REG_RCS1         0x08
+/* receiver status 2 */
+#define AK4113_REG_RCS2         0x09
+/* RX channel status byte 0 */
+#define AK4113_REG_RXCSB0       0x0a
+/* RX channel status byte 1 */
+#define AK4113_REG_RXCSB1       0x0b
+/* RX channel status byte 2 */
+#define AK4113_REG_RXCSB2       0x0c
+/* RX channel status byte 3 */
+#define AK4113_REG_RXCSB3       0x0d
+/* RX channel status byte 4 */
+#define AK4113_REG_RXCSB4       0x0e
+/* burst preamble Pc byte 0 */
+#define AK4113_REG_Pc0          0x0f
+/* burst preamble Pc byte 1 */
+#define AK4113_REG_Pc1          0x10
+/* burst preamble Pd byte 0 */
+#define AK4113_REG_Pd0          0x11
+/* burst preamble Pd byte 1 */
+#define AK4113_REG_Pd1          0x12
+/* Q-subcode address + control */
+#define AK4113_REG_QSUB_ADDR    0x13
+/* Q-subcode track */
+#define AK4113_REG_QSUB_TRACK   0x14
+/* Q-subcode index */
+#define AK4113_REG_QSUB_INDEX   0x15
+/* Q-subcode minute */
+#define AK4113_REG_QSUB_MINUTE  0x16
+/* Q-subcode second */
+#define AK4113_REG_QSUB_SECOND  0x17
+/* Q-subcode frame */
+#define AK4113_REG_QSUB_FRAME   0x18
+/* Q-subcode zero */
+#define AK4113_REG_QSUB_ZERO    0x19
+/* Q-subcode absolute minute */
+#define AK4113_REG_QSUB_ABSMIN  0x1a
+/* Q-subcode absolute second */
+#define AK4113_REG_QSUB_ABSSEC  0x1b
+/* Q-subcode absolute frame */
+#define AK4113_REG_QSUB_ABSFRM  0x1c
+
+/* sizes */
+#define AK4113_REG_RXCSB_SIZE   ((AK4113_REG_RXCSB4-AK4113_REG_RXCSB0)+1)
+#define AK4113_REG_QSUB_SIZE    ((AK4113_REG_QSUB_ABSFRM-AK4113_REG_QSUB_ADDR)\
+                +1)
+
+#define AK4113_WRITABLE_REGS    (AK4113_REG_DATDTS + 1)
+
+/* AK4113_REG_PWRDN bits */
+/* Channel Status Select */
+#define AK4113_CS12             (1<<7)
+/* Block Start & C/U Output Mode */
+#define AK4113_BCU              (1<<6)
+/* Master Clock Operation Select */
+#define AK4113_CM1              (1<<5)
+/* Master Clock Operation Select */
+#define AK4113_CM0              (1<<4)
+/* Master Clock Frequency Select */
+#define AK4113_OCKS1            (1<<3)
+/* Master Clock Frequency Select */
+#define AK4113_OCKS0            (1<<2)
+/* 0 = power down, 1 = normal operation */
+#define AK4113_PWN              (1<<1)
+/* 0 = reset & initialize (except thisregister), 1 = normal operation */
+#define AK4113_RST              (1<<0)
+
+/* AK4113_REQ_FORMAT bits */
+/* V/TX Output select: 0 = Validity Flag Output, 1 = TX */
+#define AK4113_VTX              (1<<7)
+/* Audio Data Control */
+#define AK4113_DIF2             (1<<6)
+/* Audio Data Control */
+#define AK4113_DIF1             (1<<5)
+/* Audio Data Control */
+#define AK4113_DIF0             (1<<4)
+/* Deemphasis Autodetect Enable (1 = enable) */
+#define AK4113_DEAU             (1<<3)
+/* 32kHz-48kHz Deemphasis Control */
+#define AK4113_DEM1             (1<<2)
+/* 32kHz-48kHz Deemphasis Control */
+#define AK4113_DEM0             (1<<1)
+#define AK4113_DEM_OFF          (AK4113_DEM0)
+#define AK4113_DEM_44KHZ        (0)
+#define AK4113_DEM_48KHZ        (AK4113_DEM1)
+#define AK4113_DEM_32KHZ        (AK4113_DEM0|AK4113_DEM1)
+/* STDO: 16-bit, right justified */
+#define AK4113_DIF_16R          (0)
+/* STDO: 18-bit, right justified */
+#define AK4113_DIF_18R          (AK4113_DIF0)
+/* STDO: 20-bit, right justified */
+#define AK4113_DIF_20R          (AK4113_DIF1)
+/* STDO: 24-bit, right justified */
+#define AK4113_DIF_24R          (AK4113_DIF1|AK4113_DIF0)
+/* STDO: 24-bit, left justified */
+#define AK4113_DIF_24L          (AK4113_DIF2)
+/* STDO: I2S */
+#define AK4113_DIF_24I2S        (AK4113_DIF2|AK4113_DIF0)
+/* STDO: 24-bit, left justified; LRCLK, BICK = Input */
+#define AK4113_DIF_I24L         (AK4113_DIF2|AK4113_DIF1)
+/* STDO: I2S;  LRCLK, BICK = Input */
+#define AK4113_DIF_I24I2S       (AK4113_DIF2|AK4113_DIF1|AK4113_DIF0)
+
+/* AK4113_REG_IO0 */
+/* XTL1=0,XTL0=0 -> 11.2896Mhz; XTL1=0,XTL0=1 -> 12.288Mhz */
+#define AK4113_XTL1             (1<<6)
+/* XTL1=1,XTL0=0 -> 24.576Mhz; XTL1=1,XTL0=1 -> use channel status */
+#define AK4113_XTL0             (1<<5)
+/* Block Start Signal Output: 0 = U-bit, 1 = C-bit (req. BCU = 1) */
+#define AK4113_UCE              (1<<4)
+/* TX Output Enable (1 = enable) */
+#define AK4113_TXE              (1<<3)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS2             (1<<2)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS1             (1<<1)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS0             (1<<0)
+/* 11.2896 MHz ref. Xtal freq. */
+#define AK4113_XTL_11_2896M     (0)
+/* 12.288 MHz ref. Xtal freq. */
+#define AK4113_XTL_12_288M      (AK4113_XTL0)
+/* 24.576 MHz ref. Xtal freq. */
+#define AK4113_XTL_24_576M      (AK4113_XTL1)
+
+/* AK4113_REG_IO1 */
+/* Interrupt 0 pin Hold */
+#define AK4113_EFH1             (1<<7)
+/* Interrupt 0 pin Hold */
+#define AK4113_EFH0             (1<<6)
+#define AK4113_EFH_512LRCLK     (0)
+#define AK4113_EFH_1024LRCLK    (AK4113_EFH0)
+#define AK4113_EFH_2048LRCLK    (AK4113_EFH1)
+#define AK4113_EFH_4096LRCLK    (AK4113_EFH1|AK4113_EFH0)
+/* PLL Lock Time: 0 = 384/fs, 1 = 1/fs */
+#define AK4113_FAST             (1<<5)
+/* MCKO2 Output Select: 0 = CMx/OCKSx, 1 = Xtal */
+#define AK4113_XMCK             (1<<4)
+/* MCKO2 Output Freq. Select: 0 = x1, 1 = x0.5  (req. XMCK = 1) */
+#define AK4113_DIV              (1<<3)
+/* Input Recovery Data Select */
+#define AK4113_IPS2             (1<<2)
+/* Input Recovery Data Select */
+#define AK4113_IPS1             (1<<1)
+/* Input Recovery Data Select */
+#define AK4113_IPS0             (1<<0)
+#define AK4113_IPS(x)           ((x)&7)
+
+/* AK4113_REG_INT0_MASK && AK4113_REG_INT1_MASK*/
+/* mask enable for QINT bit */
+#define AK4113_MQI              (1<<7)
+/* mask enable for AUTO bit */
+#define AK4113_MAUT             (1<<6)
+/* mask enable for CINT bit */
+#define AK4113_MCIT             (1<<5)
+/* mask enable for UNLOCK bit */
+#define AK4113_MULK             (1<<4)
+/* mask enable for V bit */
+#define AK4113_V                (1<<3)
+/* mask enable for STC bit */
+#define AK4113_STC              (1<<2)
+/* mask enable for AUDN bit */
+#define AK4113_MAN              (1<<1)
+/* mask enable for PAR bit */
+#define AK4113_MPR              (1<<0)
+
+/* AK4113_REG_DATDTS */
+/* DAT Start ID Counter */
+#define AK4113_DCNT             (1<<4)
+/* DTS-CD 16-bit Sync Word Detect */
+#define AK4113_DTS16            (1<<3)
+/* DTS-CD 14-bit Sync Word Detect */
+#define AK4113_DTS14            (1<<2)
+/* mask enable for DAT bit (if 1, no INT1 effect */
+#define AK4113_MDAT1            (1<<1)
+/* mask enable for DAT bit (if 1, no INT0 effect */
+#define AK4113_MDAT0            (1<<0)
+
+/* AK4113_REG_RCS0 */
+/* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
+#define AK4113_QINT             (1<<7)
+/* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
+#define AK4113_AUTO             (1<<6)
+/* channel status buffer interrupt, 0 = no change, 1 = change */
+#define AK4113_CINT             (1<<5)
+/* PLL lock status, 0 = lock, 1 = unlock */
+#define AK4113_UNLCK            (1<<4)
+/* Validity bit, 0 = valid, 1 = invalid */
+#define AK4113_V                (1<<3)
+/* sampling frequency or Pre-emphasis change, 0 = no detect, 1 = detect */
+#define AK4113_STC              (1<<2)
+/* audio bit output, 0 = audio, 1 = non-audio */
+#define AK4113_AUDION           (1<<1)
+/* parity error or biphase error status, 0 = no error, 1 = error */
+#define AK4113_PAR              (1<<0)
+
+/* AK4113_REG_RCS1 */
+/* sampling frequency detection */
+#define AK4113_FS3              (1<<7)
+#define AK4113_FS2              (1<<6)
+#define AK4113_FS1              (1<<5)
+#define AK4113_FS0              (1<<4)
+/* Pre-emphasis detect, 0 = OFF, 1 = ON */
+#define AK4113_PEM              (1<<3)
+/* DAT Start ID Detect, 0 = no detect, 1 = detect */
+#define AK4113_DAT              (1<<2)
+/* DTS-CD bit audio stream detect, 0 = no detect, 1 = detect */
+#define AK4113_DTSCD            (1<<1)
+/* Non-PCM bit stream detection, 0 = no detect, 1 = detect */
+#define AK4113_NPCM             (1<<0)
+#define AK4113_FS_8000HZ        (AK4113_FS3|AK4113_FS0)
+#define AK4113_FS_11025HZ       (AK4113_FS2|AK4113_FS0)
+#define AK4113_FS_16000HZ       (AK4113_FS2|AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_22050HZ       (AK4113_FS2)
+#define AK4113_FS_24000HZ       (AK4113_FS2|AK4113_FS1)
+#define AK4113_FS_32000HZ       (AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_44100HZ       (0)
+#define AK4113_FS_48000HZ       (AK4113_FS1)
+#define AK4113_FS_64000HZ       (AK4113_FS3|AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_88200HZ       (AK4113_FS3)
+#define AK4113_FS_96000HZ       (AK4113_FS3|AK4113_FS1)
+#define AK4113_FS_176400HZ      (AK4113_FS3|AK4113_FS2)
+#define AK4113_FS_192000HZ      (AK4113_FS3|AK4113_FS2|AK4113_FS1)
+
+/* AK4113_REG_RCS2 */
+/* CRC for Q-subcode, 0 = no error, 1 = error */
+#define AK4113_QCRC             (1<<1)
+/* CRC for channel status, 0 = no error, 1 = error */
+#define AK4113_CCRC             (1<<0)
+
+/* flags for snd_ak4113_check_rate_and_errors() */
+#define AK4113_CHECK_NO_STAT    (1<<0)  /* no statistics */
+#define AK4113_CHECK_NO_RATE    (1<<1)  /* no rate check */
+
+#define AK4113_CONTROLS         13
+
+typedef void (ak4113_write_t)(void *private_data, unsigned char addr,
+                unsigned char data);
+typedef unsigned char (ak4113_read_t)(void *private_data, unsigned char addr);
+
+struct ak4113 {
+        struct snd_card *card;
+        ak4113_write_t *write;
+        ak4113_read_t *read;
+        void *private_data;
+        unsigned int init:1;
+        spinlock_t lock;
+        unsigned char regmap[AK4113_WRITABLE_REGS];
+        struct snd_kcontrol *kctls[AK4113_CONTROLS];
+        struct snd_pcm_substream *substream;
+        unsigned long parity_errors;
+        unsigned long v_bit_errors;
+        unsigned long qcrc_errors;
+        unsigned long ccrc_errors;
+        unsigned char rcs0;
+        unsigned char rcs1;
+        unsigned char rcs2;
+        struct delayed_work work;
+        unsigned int check_flags;
+        void *change_callback_private;
+        void (*change_callback)(struct ak4113 *ak4113, unsigned char c0,
+                        unsigned char c1);
+};
+
+int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
+                ak4113_write_t *write,
+                const unsigned char *pgm,
+                void *private_data, struct ak4113 **r_ak4113);
+void snd_ak4113_reg_write(struct ak4113 *ak4113, unsigned char reg,
+                unsigned char mask, unsigned char val);
+void snd_ak4113_reinit(struct ak4113 *ak4113);
+int snd_ak4113_build(struct ak4113 *ak4113,
+                struct snd_pcm_substream *capture_substream);
+int snd_ak4113_external_rate(struct ak4113 *ak4113);
+int snd_ak4113_check_rate_and_errors(struct ak4113 *ak4113, unsigned int flags);
+
+#endif /* __SOUND_AK4113_H */
+
diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h
index d293d36a66b8..3ce69fd92523 100644
--- a/include/sound/ak4114.h
+++ b/include/sound/ak4114.h
@@ -95,13 +95,13 @@
 
 /* AK4114_REG_IO0 */
 #define AK4114_TX1E             (1<<7)  /* TX1 Output Enable (1 = enable) */
-#define AK4114_OPS12            (1<<2)  /* Output Though Data Selector for TX1 pin */
-#define AK4114_OPS11            (1<<1)  /* Output Though Data Selector for TX1 pin */
-#define AK4114_OPS10            (1<<0)  /* Output Though Data Selector for TX1 pin */
+#define AK4114_OPS12            (1<<6)  /* Output Data Selector for TX1 pin */
+#define AK4114_OPS11            (1<<5)  /* Output Data Selector for TX1 pin */
+#define AK4114_OPS10            (1<<4)  /* Output Data Selector for TX1 pin */
 #define AK4114_TX0E             (1<<3)  /* TX0 Output Enable (1 = enable) */
-#define AK4114_OPS02            (1<<2)  /* Output Though Data Selector for TX0 pin */
-#define AK4114_OPS01            (1<<1)  /* Output Though Data Selector for TX0 pin */
-#define AK4114_OPS00            (1<<0)  /* Output Though Data Selector for TX0 pin */
+#define AK4114_OPS02            (1<<2)  /* Output Data Selector for TX0 pin */
+#define AK4114_OPS01            (1<<1)  /* Output Data Selector for TX0 pin */
+#define AK4114_OPS00            (1<<0)  /* Output Data Selector for TX0 pin */
 
 /* AK4114_REG_IO1 */
 #define AK4114_EFH1             (1<<7)  /* Interrupt 0 pin Hold */
diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h
index 891cf1aea8b1..030b87c2f6d4 100644
--- a/include/sound/ak4xxx-adda.h
+++ b/include/sound/ak4xxx-adda.h
@@ -68,7 +68,7 @@ struct snd_akm4xxx {
         enum {
                 SND_AK4524, SND_AK4528, SND_AK4529,
                 SND_AK4355, SND_AK4358, SND_AK4381,
-                SND_AK5365
+                SND_AK5365, SND_AK4620,
         } type;
 
         /* (array) information of combined codecs */
@@ -76,6 +76,9 @@ struct snd_akm4xxx {
         const struct snd_akm4xxx_adc_channel *adc_info;
 
         struct snd_ak4xxx_ops ops;
+        unsigned int num_chips;
+        unsigned int total_regs;
+        const char *name;
 };
 
 void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 1f57bb92eb5a..098595500632 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -544,7 +544,7 @@ struct snd_rawmidi_status {
  *  Timer section - /dev/snd/timer
  */
 
-#define SNDRV_TIMER_VERSION             SNDRV_PROTOCOL_VERSION(2, 0, 5)
+#define SNDRV_TIMER_VERSION             SNDRV_PROTOCOL_VERSION(2, 0, 6)
 
 enum {
         SNDRV_TIMER_CLASS_NONE = -1,
diff --git a/include/sound/control.h b/include/sound/control.h
index ef96f07aa03b..112374dc0c58 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -56,7 +56,6 @@ struct snd_kcontrol_new {
 
 struct snd_kcontrol_volatile {
         struct snd_ctl_file *owner;     /* locked */
-        pid_t owner_pid;
         unsigned int access;    /* access rights */
 };
 
@@ -87,10 +86,12 @@ struct snd_kctl_event {
 
 #define snd_kctl_event(n) list_entry(n, struct snd_kctl_event, list)
 
+struct pid;
+
 struct snd_ctl_file {
         struct list_head list;          /* list of all control files */
         struct snd_card *card;
-        pid_t pid;
+        struct pid *pid;
         int prefer_pcm_subdevice;
         int prefer_rawmidi_subdevice;
         wait_queue_head_t change_sleep;
diff --git a/include/sound/core.h b/include/sound/core.h
index a61499c22b0b..89e0ac17f44a 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -458,5 +458,8 @@ struct snd_pci_quirk {
 const struct snd_pci_quirk *
 snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list);
 
+const struct snd_pci_quirk *
+snd_pci_quirk_lookup_id(u16 vendor, u16 device,
+                        const struct snd_pci_quirk *list);
 
 #endif /* __SOUND_CORE_H */
diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h
index 92647532c454..66d28c2cb53d 100644
--- a/include/sound/cs4231-regs.h
+++ b/include/sound/cs4231-regs.h
@@ -70,7 +70,6 @@
 #define AD1845_PWR_DOWN         0x1b    /* power down control */
 #define CS4235_LEFT_MASTER      0x1b    /* left master output control */
 #define CS4231_REC_FORMAT       0x1c    /* clock and data format - record - bits 7-0 MCE */
-#define CS4231_PLY_VAR_FREQ     0x1d    /* playback variable frequency */
 #define AD1845_CLOCK            0x1d    /* crystal clock select and total power down */
 #define CS4235_RIGHT_MASTER     0x1d    /* right master output control */
 #define CS4231_REC_UPR_CNT      0x1e    /* record upper count */
diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h
index 7c44667e79a6..49b03c9e5e55 100644
--- a/include/sound/cs46xx_dsp_spos.h
+++ b/include/sound/cs46xx_dsp_spos.h
@@ -118,9 +118,11 @@ struct dsp_scb_descriptor {
 
         struct snd_info_entry *proc_info;
         int ref_count;
-        spinlock_t lock;
 
-        int deleted;
+        u16 volume[2];
+        unsigned int deleted :1;
+        unsigned int updated :1;
+        unsigned int volume_set :1;
 };
 
 struct dsp_task_descriptor {
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index de6d981de5d6..8b611a561985 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -262,6 +262,8 @@ struct snd_pcm_hw_constraint_list {
         unsigned int mask;
 };
 
+struct snd_pcm_hwptr_log;
+
 struct snd_pcm_runtime {
         /* -- Status -- */
         struct snd_pcm_substream *trigger_master;
@@ -310,7 +312,9 @@ struct snd_pcm_runtime {
         struct snd_pcm_mmap_control *control;
 
         /* -- locking / scheduling -- */
-        wait_queue_head_t sleep;
+        unsigned int twake: 1;          /* do transfer (!poll) wakeup */
+        wait_queue_head_t sleep;        /* poll sleep */
+        wait_queue_head_t tsleep;       /* transfer sleep */
         struct fasync_struct *fasync;
 
         /* -- private section -- */
@@ -340,6 +344,10 @@ struct snd_pcm_runtime {
         /* -- OSS things -- */
         struct snd_pcm_oss_runtime oss;
 #endif
+
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+        struct snd_pcm_hwptr_log *hwptr_log;
+#endif
 };
 
 struct snd_pcm_group {          /* keep linked substreams */
@@ -348,6 +356,8 @@ struct snd_pcm_group {		/* keep linked substreams */
         int count;
 };
 
+struct pid;
+
 struct snd_pcm_substream {
         struct snd_pcm *pcm;
         struct snd_pcm_str *pstr;
@@ -379,6 +389,7 @@ struct snd_pcm_substream {
         atomic_t mmap_count;
         unsigned int f_flags;
         void (*pcm_release)(struct snd_pcm_substream *);
+        struct pid *pid;
 #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
         /* -- OSS things -- */
         struct snd_pcm_oss_substream oss;
@@ -831,6 +842,8 @@ void snd_pcm_set_sync(struct snd_pcm_substream *substream);
 int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream);
 int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream,
                       unsigned int cmd, void *arg);                      
+int snd_pcm_update_state(struct snd_pcm_substream *substream,
+                         struct snd_pcm_runtime *runtime);
 int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream);
 int snd_pcm_playback_xrun_check(struct snd_pcm_substream *substream);
 int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream);
@@ -902,6 +915,44 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
 int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size);
 int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream);
 
+int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream,
+                                      size_t size, gfp_t gfp_flags);
+int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream);
+struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream,
+                                          unsigned long offset);
+#if 0 /* for kernel-doc */
+/**
+ * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer
+ * @substream: the substream to allocate the buffer to
+ * @size: the requested buffer size, in bytes
+ *
+ * Allocates the PCM substream buffer using vmalloc(), i.e., the memory is
+ * contiguous in kernel virtual space, but not in physical memory.  Use this
+ * if the buffer is accessed by kernel code but not by device DMA.
+ *
+ * Returns 1 if the buffer was changed, 0 if not changed, or a negative error
+ * code.
+ */
+static int snd_pcm_lib_alloc_vmalloc_buffer
+                        (struct snd_pcm_substream *substream, size_t size);
+/**
+ * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer
+ * @substream: the substream to allocate the buffer to
+ * @size: the requested buffer size, in bytes
+ *
+ * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses
+ * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory.
+ */
+static int snd_pcm_lib_alloc_vmalloc_32_buffer
+                        (struct snd_pcm_substream *substream, size_t size);
+#endif
+#define snd_pcm_lib_alloc_vmalloc_buffer(subs, size) \
+        _snd_pcm_lib_alloc_vmalloc_buffer \
+                        (subs, size, GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO)
+#define snd_pcm_lib_alloc_vmalloc_32_buffer(subs, size) \
+        _snd_pcm_lib_alloc_vmalloc_buffer \
+                        (subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO)
+
 #ifdef CONFIG_SND_DMA_SGBUF
 /*
  * SG-buffer handling
@@ -972,6 +1023,10 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s
 #define snd_pcm_lib_mmap_iomem  NULL
 #endif
 
+int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream,
+                               struct vm_area_struct *area);
+#define snd_pcm_lib_mmap_vmalloc        snd_pcm_lib_mmap_noncached
+
 static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
 {
         *max = dma < 4 ? 64 * 1024 : 128 * 1024;
diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h
index cc4e226f35fd..760c969d885d 100644
--- a/include/sound/pcm_oss.h
+++ b/include/sound/pcm_oss.h
@@ -61,7 +61,7 @@ struct snd_pcm_oss_runtime {
         struct snd_pcm_plugin *plugin_first;
         struct snd_pcm_plugin *plugin_last;
 #endif
-        unsigned int prev_hw_ptr_interrupt;
+        unsigned int prev_hw_ptr_period;
 };
 
 struct snd_pcm_oss_file {
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index c23c26585700..2480e7d10dcf 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -46,6 +46,7 @@
 struct snd_rawmidi;
 struct snd_rawmidi_substream;
 struct snd_seq_port_info;
+struct pid;
 
 struct snd_rawmidi_ops {
         int (*open) (struct snd_rawmidi_substream * substream);
@@ -97,6 +98,7 @@ struct snd_rawmidi_substream {
         struct snd_rawmidi_str *pstr;
         char name[32];
         struct snd_rawmidi_runtime *runtime;
+        struct pid *pid;
         /* hardware layer */
         struct snd_rawmidi_ops *ops;
 };
diff --git a/include/sound/sb.h b/include/sound/sb.h
index 4e62ee1e4115..95353542256a 100644
--- a/include/sound/sb.h
+++ b/include/sound/sb.h
@@ -33,6 +33,7 @@ enum sb_hw_type {
         SB_HW_20,
         SB_HW_201,
         SB_HW_PRO,
+        SB_HW_JAZZ16,           /* Media Vision Jazz16 */
         SB_HW_16,
         SB_HW_16CSP,            /* SB16 with CSP chip */
         SB_HW_ALS100,           /* Avance Logic ALS100 chip */
diff --git a/include/sound/sh_dac_audio.h b/include/sound/sh_dac_audio.h
new file mode 100644
index 000000000000..f5deaf1ddb9f
--- /dev/null
+++ b/include/sound/sh_dac_audio.h
@@ -0,0 +1,21 @@
+/*
+ * SH_DAC specific configuration, for the dac_audio platform_device
+ *
+ * Copyright (C) 2009 Rafael Ignacio Zurita <rizurita@yahoo.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 __INCLUDE_SH_DAC_AUDIO_H
+#define __INCLUDE_SH_DAC_AUDIO_H
+
+struct dac_audio_pdata {
+        int buffer_size;
+        int channel;
+        void (*start)(struct dac_audio_pdata *pd);
+        void (*stop)(struct dac_audio_pdata *pd);
+};
+
+#endif /* __INCLUDE_SH_DAC_AUDIO_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 97ca9af414dc..0a0b019d41ad 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -16,6 +16,8 @@
 
 #include <linux/list.h>
 
+#include <sound/soc.h>
+
 struct snd_pcm_substream;
 
 /*
@@ -30,6 +32,7 @@ struct snd_pcm_substream;
 #define SND_SOC_DAIFMT_DSP_A            3 /* L data MSB after FRM LRC */
 #define SND_SOC_DAIFMT_DSP_B            4 /* L data MSB during FRM LRC */
 #define SND_SOC_DAIFMT_AC97             5 /* AC97 */
+#define SND_SOC_DAIFMT_PDM              6 /* Pulse density modulation */
 
 /* left and right justified also known as MSB and LSB respectively */
 #define SND_SOC_DAIFMT_MSB              SND_SOC_DAIFMT_LEFT_J
@@ -106,7 +109,7 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
         int div_id, int div);
 
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
-        int pll_id, unsigned int freq_in, unsigned int freq_out);
+        int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
 
 /* Digital Audio interface formatting */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
@@ -114,6 +117,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
         unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
 
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+        unsigned int tx_num, unsigned int *tx_slot,
+        unsigned int rx_num, unsigned int *rx_slot);
+
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 
 /* Digital Audio Interface mute */
@@ -136,8 +143,8 @@ struct snd_soc_dai_ops {
          */
         int (*set_sysclk)(struct snd_soc_dai *dai,
                 int clk_id, unsigned int freq, int dir);
-        int (*set_pll)(struct snd_soc_dai *dai,
-                int pll_id, unsigned int freq_in, unsigned int freq_out);
+        int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
+                unsigned int freq_in, unsigned int freq_out);
         int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
 
         /*
@@ -148,6 +155,9 @@ struct snd_soc_dai_ops {
         int (*set_tdm_slot)(struct snd_soc_dai *dai,
                 unsigned int tx_mask, unsigned int rx_mask,
                 int slots, int slot_width);
+        int (*set_channel_map)(struct snd_soc_dai *dai,
+                unsigned int tx_num, unsigned int *tx_slot,
+                unsigned int rx_num, unsigned int *rx_slot);
         int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 
         /*
@@ -209,7 +219,6 @@ struct snd_soc_dai {
         struct snd_soc_codec *codec;
         unsigned int active;
         unsigned char pop_wait:1;
-        void *dma_data;
 
         /* DAI private data */
         void *private_data;
@@ -220,4 +229,21 @@ struct snd_soc_dai {
         struct list_head list;
 };
 
+static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
+                                             const struct snd_pcm_substream *ss)
+{
+        return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+                dai->playback.dma_data : dai->capture.dma_data;
+}
+
+static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
+                                            const struct snd_pcm_substream *ss,
+                                            void *data)
+{
+        if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                dai->playback.dma_data = data;
+        else
+                dai->capture.dma_data = data;
+}
+
 #endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c1410e3191e3..c0922a034223 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -95,6 +95,21 @@
         .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \
         .num_kcontrols = 1}
 
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
+         wcontrols) \
+{       .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
+        .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \
+         wcontrols)\
+{       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+        .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \
+         wcontrols)\
+{       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
+        .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \
+        .num_kcontrols = ARRAY_SIZE(wcontrols)}
+
 /* path domain with event - event handler must return 0 for success */
 #define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \
         wncontrols, wevent, wflags) \
@@ -126,6 +141,23 @@
         .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \
         .event = wevent, .event_flags = wflags}
 
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+        wevent, wflags) \
+{       .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
+        .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+        .event = wevent, .event_flags = wflags}
+#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+        wevent, wflags) \
+{       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+        .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+        .event = wevent, .event_flags = wflags}
+#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \
+        wcontrols, wevent, wflags) \
+{       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+        .invert = winvert, .kcontrols = wcontrols, \
+        .num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags}
+
 /* events that are pre and post DAPM */
 #define SND_SOC_DAPM_PRE(wname, wevent) \
 {       .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \
@@ -206,6 +238,12 @@
         .get = snd_soc_dapm_get_enum_double, \
         .put = snd_soc_dapm_put_enum_double, \
         .private_value = (unsigned long)&xenum }
+#define SOC_DAPM_ENUM_VIRT(xname, xenum)                    \
+{       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+        .info = snd_soc_info_enum_double, \
+        .get = snd_soc_dapm_get_enum_virt, \
+        .put = snd_soc_dapm_put_enum_virt, \
+        .private_value = (unsigned long)&xenum }
 #define SOC_DAPM_VALUE_ENUM(xname, xenum) \
 {       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
         .info = snd_soc_info_enum_double, \
@@ -260,6 +298,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol);
 int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol);
 int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol);
 int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
@@ -333,6 +375,10 @@ struct snd_soc_dapm_route {
         const char *sink;
         const char *control;
         const char *source;
+
+        /* Note: currently only supported for links where source is a supply */
+        int (*connected)(struct snd_soc_dapm_widget *source,
+                         struct snd_soc_dapm_widget *sink);
 };
 
 /* dapm audio path between two widgets */
@@ -349,6 +395,9 @@ struct snd_soc_dapm_path {
         u32 connect:1;  /* source and sink widgets are connected */
         u32 walked:1;   /* path has been walked */
 
+        int (*connected)(struct snd_soc_dapm_widget *source,
+                         struct snd_soc_dapm_widget *sink);
+
         struct list_head list_source;
         struct list_head list_sink;
         struct list_head list;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 475cb7ed6bec..a57fbfcd4c8f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -169,6 +169,23 @@
         .private_value = (unsigned long)&xenum }
 
 /*
+ * Simplified versions of above macros, declaring a struct and calculating
+ * ARRAY_SIZE internally
+ */
+#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \
+        struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
+                                                ARRAY_SIZE(xtexts), xtexts)
+#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \
+        SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
+#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \
+        struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
+#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \
+        struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \
+                                                        ARRAY_SIZE(xtexts), xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
+        SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+
+/*
  * Bias levels
  *
  * @ON:      Bias is fully on for audio playback and capture operations.
@@ -223,15 +240,15 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
                                int addr_bits, int data_bits,
                                enum snd_soc_control_type control);
 
-#ifdef CONFIG_PM
-int snd_soc_suspend_device(struct device *dev);
-int snd_soc_resume_device(struct device *dev);
-#endif
-
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
-int snd_soc_init_card(struct snd_soc_device *socdev);
+
+/* Utility functions to get clock rates from various things */
+int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
+int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
 
 /* set runtime hw params */
 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
@@ -253,6 +270,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
 /* codec register bit access */
 int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
                                 unsigned int mask, unsigned int value);
+int snd_soc_update_bits_locked(struct snd_soc_codec *codec,
+                               unsigned short reg, unsigned int mask,
+                               unsigned int value);
 int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
                                 unsigned int mask, unsigned int value);
 
@@ -333,6 +353,8 @@ struct snd_soc_jack_gpio {
         int debounce_time;
         struct snd_soc_jack *jack;
         struct work_struct work;
+
+        int (*jack_status_check)(void);
 };
 #endif
 
@@ -353,6 +375,7 @@ struct snd_soc_pcm_stream {
         unsigned int channels_min;      /* min channels */
         unsigned int channels_max;      /* max channels */
         unsigned int active:1;          /* stream is in use */
+        void *dma_data;                 /* used by platform code */
 };
 
 /* SoC audio ops */
@@ -400,6 +423,10 @@ struct snd_soc_codec {
         short reg_cache_size;
         short reg_cache_step;
 
+        unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
+        unsigned int cache_only:1;  /* Suppress writes to hardware */
+        unsigned int cache_sync:1; /* Cache needs to be synced to hardware */
+
         /* dapm */
         u32 pop_time;
         struct list_head dapm_widgets;
@@ -413,6 +440,7 @@ struct snd_soc_codec {
         unsigned int num_dai;
 
 #ifdef CONFIG_DEBUG_FS
+        struct dentry *debugfs_codec_root;
         struct dentry *debugfs_reg;
         struct dentry *debugfs_pop_time;
         struct dentry *debugfs_dapm;
@@ -494,6 +522,8 @@ struct snd_soc_card {
         int (*set_bias_level)(struct snd_soc_card *,
                               enum snd_soc_bias_level level);
 
+        long pmdown_time;
+
         /* CPU <--> Codec DAI links  */
         struct snd_soc_dai_link *dai_link;
         int num_links;
diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h
deleted file mode 100644
index 0d8885969c64..000000000000
--- a/include/sound/sscape_ioctl.h
+++ /dev/null
@@ -1,21 +0,0 @@
-#ifndef SSCAPE_IOCTL_H
-#define SSCAPE_IOCTL_H
-
-
-struct sscape_bootblock
-{
-  unsigned char code[256];
-  unsigned version;
-};
-
-#define SSCAPE_MICROCODE_SIZE  65536
-
-struct sscape_microcode
-{
-  unsigned char __user *code;
-};
-
-#define SND_SSCAPE_LOAD_BOOTB  _IOWR('P', 100, struct sscape_bootblock)
-#define SND_SSCAPE_LOAD_MCODE  _IOW ('P', 101, struct sscape_microcode)
-
-#endif
diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h
new file mode 100644
index 000000000000..ac0665264bdf
--- /dev/null
+++ b/include/sound/tlv320dac33-plat.h
@@ -0,0 +1,21 @@
+/*
+ * Platform header for Texas Instruments TLV320DAC33 codec driver
+ *
+ * Author:      Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *
+ * Copyright:   (C) 2009 Nokia Corporation
+ *
+ * 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 __TLV320DAC33_PLAT_H
+#define __TLV320DAC33_PLAT_H
+
+struct tlv320dac33_platform_data {
+        int power_gpio;
+        u8 burst_bclkdiv;
+};
+
+#endif /* __TLV320DAC33_PLAT_H */
diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
new file mode 100644
index 000000000000..e29fde6b5cbe
--- /dev/null
+++ b/include/sound/tpa6130a2-plat.h
@@ -0,0 +1,36 @@
+/*
+ * TPA6130A2 driver platform header
+ *
+ * Copyright (C) Nokia Corporation
+ *
+ * Written by Peter Ujfalusi <peter.ujfalusi@nokia.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.
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef TPA6130A2_PLAT_H
+#define TPA6130A2_PLAT_H
+
+enum tpa_model {
+        TPA6130A2,
+        TPA6140A2,
+};
+
+struct tpa6130a2_platform_data {
+        enum tpa_model id;
+        int power_gpio;
+};
+
+#endif
diff --git a/include/sound/version.h b/include/sound/version.h
index 22939142dd23..7fed23442db8 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
 /* include/version.h */
-#define CONFIG_SND_VERSION "1.0.21"
+#define CONFIG_SND_VERSION "1.0.22.1"
 #define CONFIG_SND_DATE ""
diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h
new file mode 100644
index 000000000000..aa388ca9ec64
--- /dev/null
+++ b/include/sound/wm2000.h
@@ -0,0 +1,26 @@
+/*
+ * linux/sound/wm2000.h -- Platform data for WM2000
+ *
+ * Copyright 2010 Wolfson Microelectronics. PLC.
+ *
+ * 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_WM2000_H
+#define __LINUX_SND_WM2000_H
+
+struct wm2000_platform_data {
+        /** Filename for system-specific image to download to device. */
+        const char *download_file;
+
+        /** Divide MCLK by 2 for system clock? */
+        unsigned int mclkdiv2:1;
+
+        /** Disable speech clarity enhancement, for use when an
+         * external algorithm is used. */
+        unsigned int speech_enh_disable:1;
+};
+
+#endif
diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h
new file mode 100644
index 000000000000..d66575a601be
--- /dev/null
+++ b/include/sound/wm8904.h
@@ -0,0 +1,57 @@
+/*
+ * Platform data for WM8904
+ *
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef __MFD_WM8994_PDATA_H__
+#define __MFD_WM8994_PDATA_H__
+
+#define WM8904_DRC_REGS 4
+#define WM8904_EQ_REGS  25
+
+/**
+ * DRC configurations are specified with a label and a set of register
+ * values to write (the enable bits will be ignored).  At runtime an
+ * enumerated control will be presented for each DRC block allowing
+ * the user to choose the configration to use.
+ *
+ * Configurations may be generated by hand or by using the DRC control
+ * panel provided by the WISCE - see  http://www.wolfsonmicro.com/wisce/
+ * for details.
+ */
+struct wm8904_drc_cfg {
+        const char *name;
+        u16 regs[WM8904_DRC_REGS];
+};
+
+/**
+ * ReTune Mobile configurations are specified with a label, sample
+ * rate and set of values to write (the enable bits will be ignored).
+ *
+ * Configurations are expected to be generated using the ReTune Mobile
+ * control panel in WISCE - see http://www.wolfsonmicro.com/wisce/
+ */
+struct wm8904_retune_mobile_cfg {
+        const char *name;
+        unsigned int rate;
+        u16 regs[WM8904_EQ_REGS];
+};
+
+struct wm8904_pdata {
+        int num_drc_cfgs;
+        struct wm8904_drc_cfg *drc_cfgs;
+
+        int num_retune_mobile_cfgs;
+        struct wm8904_retune_mobile_cfg *retune_mobile_cfgs;
+};
+
+#endif
diff --git a/include/sound/wm8955.h b/include/sound/wm8955.h
new file mode 100644
index 000000000000..5074ef499f40
--- /dev/null
+++ b/include/sound/wm8955.h
@@ -0,0 +1,26 @@
+/*
+ * Platform data for WM8955
+ *
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef __WM8955_PDATA_H__
+#define __WM8955_PDATA_H__
+
+struct wm8955_pdata {
+        /* Configure LOUT2/ROUT2 to drive a speaker */
+        unsigned int out2_speaker:1;
+
+        /* Configure MONOIN+/- in differential mode */
+        unsigned int monoin_diff:1;
+};
+
+#endif
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
index 9c661f2f8cda..eee19f63c0d8 100644
--- a/include/sound/wm8993.h
+++ b/include/sound/wm8993.h
@@ -36,7 +36,7 @@ struct wm8993_platform_data {
         unsigned int micbias1_lvl:1;
         unsigned int micbias2_lvl:1;
 
-        /* Jack detect threashold levels, see datasheet for values */
+        /* Jack detect threshold levels, see datasheet for values */
         unsigned int jd_scthr:2;
         unsigned int jd_thr:2;
 };
diff --git a/include/sound/wss.h b/include/sound/wss.h
index 6d65f322f1d5..fd01f22825cd 100644
--- a/include/sound/wss.h
+++ b/include/sound/wss.h
@@ -154,7 +154,6 @@ int snd_wss_create(struct snd_card *card,
                       unsigned short hardware,
                       unsigned short hwshare,
                       struct snd_wss **rchip);
-int snd_wss_free(struct snd_wss *chip);
 int snd_wss_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm);
 int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer);
 int snd_wss_mixer(struct snd_wss *chip);