Merge branch 'topic/misc' into for-next

Generic updates for sound 3.6

20 files changed, 360 insertions, 538 deletions

diff --git a/include/linux/ac97_codec.h b/include/linux/ac97_codec.h
deleted file mode 100644
index 0260c3e79fdd..000000000000
--- a/include/linux/ac97_codec.h
+++ /dev/null
@@ -1,362 +0,0 @@
-#ifndef _AC97_CODEC_H_
-#define _AC97_CODEC_H_
-
-#include <linux/types.h>
-#include <linux/soundcard.h>
-
-/* AC97 1.0 */
-#define  AC97_RESET               0x0000      //
-#define  AC97_MASTER_VOL_STEREO   0x0002      // Line Out
-#define  AC97_HEADPHONE_VOL       0x0004      // 
-#define  AC97_MASTER_VOL_MONO     0x0006      // TAD Output
-#define  AC97_MASTER_TONE         0x0008      //
-#define  AC97_PCBEEP_VOL          0x000a      // none
-#define  AC97_PHONE_VOL           0x000c      // TAD Input (mono)
-#define  AC97_MIC_VOL             0x000e      // MIC Input (mono)
-#define  AC97_LINEIN_VOL          0x0010      // Line Input (stereo)
-#define  AC97_CD_VOL              0x0012      // CD Input (stereo)
-#define  AC97_VIDEO_VOL           0x0014      // none
-#define  AC97_AUX_VOL             0x0016      // Aux Input (stereo)
-#define  AC97_PCMOUT_VOL          0x0018      // Wave Output (stereo)
-#define  AC97_RECORD_SELECT       0x001a      //
-#define  AC97_RECORD_GAIN         0x001c
-#define  AC97_RECORD_GAIN_MIC     0x001e
-#define  AC97_GENERAL_PURPOSE     0x0020
-#define  AC97_3D_CONTROL          0x0022
-#define  AC97_MODEM_RATE          0x0024
-#define  AC97_POWER_CONTROL       0x0026
-
-/* AC'97 2.0 */
-#define AC97_EXTENDED_ID          0x0028       /* Extended Audio ID */
-#define AC97_EXTENDED_STATUS      0x002A       /* Extended Audio Status */
-#define AC97_PCM_FRONT_DAC_RATE   0x002C       /* PCM Front DAC Rate */
-#define AC97_PCM_SURR_DAC_RATE    0x002E       /* PCM Surround DAC Rate */
-#define AC97_PCM_LFE_DAC_RATE     0x0030       /* PCM LFE DAC Rate */
-#define AC97_PCM_LR_ADC_RATE      0x0032       /* PCM LR ADC Rate */
-#define AC97_PCM_MIC_ADC_RATE     0x0034       /* PCM MIC ADC Rate */
-#define AC97_CENTER_LFE_MASTER    0x0036       /* Center + LFE Master Volume */
-#define AC97_SURROUND_MASTER      0x0038       /* Surround (Rear) Master Volume */
-#define AC97_RESERVED_3A          0x003A       /* Reserved in AC '97 < 2.2 */
-
-/* AC'97 2.2 */
-#define AC97_SPDIF_CONTROL        0x003A       /* S/PDIF Control */
-
-/* range 0x3c-0x58 - MODEM */
-#define AC97_EXTENDED_MODEM_ID    0x003C
-#define AC97_EXTEND_MODEM_STAT    0x003E
-#define AC97_LINE1_RATE           0x0040
-#define AC97_LINE2_RATE           0x0042
-#define AC97_HANDSET_RATE         0x0044
-#define AC97_LINE1_LEVEL          0x0046
-#define AC97_LINE2_LEVEL          0x0048
-#define AC97_HANDSET_LEVEL        0x004A
-#define AC97_GPIO_CONFIG          0x004C
-#define AC97_GPIO_POLARITY        0x004E
-#define AC97_GPIO_STICKY          0x0050
-#define AC97_GPIO_WAKE_UP         0x0052
-#define AC97_GPIO_STATUS          0x0054
-#define AC97_MISC_MODEM_STAT      0x0056
-#define AC97_RESERVED_58          0x0058
-
-/* registers 0x005a - 0x007a are vendor reserved */
-
-#define AC97_VENDOR_ID1           0x007c
-#define AC97_VENDOR_ID2           0x007e
-
-/* volume control bit defines */
-#define AC97_MUTE                 0x8000
-#define AC97_MICBOOST             0x0040
-#define AC97_LEFTVOL              0x3f00
-#define AC97_RIGHTVOL             0x003f
-
-/* record mux defines */
-#define AC97_RECMUX_MIC           0x0000
-#define AC97_RECMUX_CD            0x0101
-#define AC97_RECMUX_VIDEO         0x0202
-#define AC97_RECMUX_AUX           0x0303
-#define AC97_RECMUX_LINE          0x0404
-#define AC97_RECMUX_STEREO_MIX    0x0505
-#define AC97_RECMUX_MONO_MIX      0x0606
-#define AC97_RECMUX_PHONE         0x0707
-
-/* general purpose register bit defines */
-#define AC97_GP_LPBK              0x0080       /* Loopback mode */
-#define AC97_GP_MS                0x0100       /* Mic Select 0=Mic1, 1=Mic2 */
-#define AC97_GP_MIX               0x0200       /* Mono output select 0=Mix, 1=Mic */
-#define AC97_GP_RLBK              0x0400       /* Remote Loopback - Modem line codec */
-#define AC97_GP_LLBK              0x0800       /* Local Loopback - Modem Line codec */
-#define AC97_GP_LD                0x1000       /* Loudness 1=on */
-#define AC97_GP_3D                0x2000       /* 3D Enhancement 1=on */
-#define AC97_GP_ST                0x4000       /* Stereo Enhancement 1=on */
-#define AC97_GP_POP               0x8000       /* Pcm Out Path, 0=pre 3D, 1=post 3D */
-
-/* extended audio status and control bit defines */
-#define AC97_EA_VRA               0x0001       /* Variable bit rate enable bit */
-#define AC97_EA_DRA               0x0002       /* Double-rate audio enable bit */
-#define AC97_EA_SPDIF             0x0004       /* S/PDIF Enable bit */
-#define AC97_EA_VRM               0x0008       /* Variable bit rate for MIC enable bit */
-#define AC97_EA_CDAC              0x0040       /* PCM Center DAC is ready (Read only) */
-#define AC97_EA_SDAC              0x0040       /* PCM Surround DACs are ready (Read only) */
-#define AC97_EA_LDAC              0x0080       /* PCM LFE DAC is ready (Read only) */
-#define AC97_EA_MDAC              0x0100       /* MIC ADC is ready (Read only) */
-#define AC97_EA_SPCV              0x0400       /* S/PDIF configuration valid (Read only) */
-#define AC97_EA_PRI               0x0800       /* Turns the PCM Center DAC off */
-#define AC97_EA_PRJ               0x1000       /* Turns the PCM Surround DACs off */
-#define AC97_EA_PRK               0x2000       /* Turns the PCM LFE DAC off */
-#define AC97_EA_PRL               0x4000       /* Turns the MIC ADC off */
-#define AC97_EA_SLOT_MASK         0xffcf       /* Mask for slot assignment bits */
-#define AC97_EA_SPSA_3_4          0x0000       /* Slot assigned to 3 & 4 */
-#define AC97_EA_SPSA_7_8          0x0010       /* Slot assigned to 7 & 8 */
-#define AC97_EA_SPSA_6_9          0x0020       /* Slot assigned to 6 & 9 */
-#define AC97_EA_SPSA_10_11        0x0030       /* Slot assigned to 10 & 11 */
-
-/* S/PDIF control bit defines */
-#define AC97_SC_PRO               0x0001       /* Professional status */
-#define AC97_SC_NAUDIO            0x0002       /* Non audio stream */
-#define AC97_SC_COPY              0x0004       /* Copyright status */
-#define AC97_SC_PRE               0x0008       /* Preemphasis status */
-#define AC97_SC_CC_MASK           0x07f0       /* Category Code mask */
-#define AC97_SC_L                 0x0800       /* Generation Level status */
-#define AC97_SC_SPSR_MASK         0xcfff       /* S/PDIF Sample Rate bits */
-#define AC97_SC_SPSR_44K          0x0000       /* Use 44.1kHz Sample rate */
-#define AC97_SC_SPSR_48K          0x2000       /* Use 48kHz Sample rate */
-#define AC97_SC_SPSR_32K          0x3000       /* Use 32kHz Sample rate */
-#define AC97_SC_DRS               0x4000       /* Double Rate S/PDIF */
-#define AC97_SC_V                 0x8000       /* Validity status */
-
-/* powerdown control and status bit defines */
-
-/* status */
-#define AC97_PWR_MDM              0x0010       /* Modem section ready */
-#define AC97_PWR_REF              0x0008       /* Vref nominal */
-#define AC97_PWR_ANL              0x0004       /* Analog section ready */
-#define AC97_PWR_DAC              0x0002       /* DAC section ready */
-#define AC97_PWR_ADC              0x0001       /* ADC section ready */
-
-/* control */
-#define AC97_PWR_PR0              0x0100       /* ADC and Mux powerdown */
-#define AC97_PWR_PR1              0x0200       /* DAC powerdown */
-#define AC97_PWR_PR2              0x0400       /* Output mixer powerdown (Vref on) */
-#define AC97_PWR_PR3              0x0800       /* Output mixer powerdown (Vref off) */
-#define AC97_PWR_PR4              0x1000       /* AC-link powerdown */
-#define AC97_PWR_PR5              0x2000       /* Internal Clk disable */
-#define AC97_PWR_PR6              0x4000       /* HP amp powerdown */
-#define AC97_PWR_PR7              0x8000       /* Modem off - if supported */
-
-/* extended audio ID register bit defines */
-#define AC97_EXTID_VRA            0x0001
-#define AC97_EXTID_DRA            0x0002
-#define AC97_EXTID_SPDIF          0x0004
-#define AC97_EXTID_VRM            0x0008
-#define AC97_EXTID_DSA0           0x0010
-#define AC97_EXTID_DSA1           0x0020
-#define AC97_EXTID_CDAC           0x0040
-#define AC97_EXTID_SDAC           0x0080
-#define AC97_EXTID_LDAC           0x0100
-#define AC97_EXTID_AMAP           0x0200
-#define AC97_EXTID_REV0           0x0400
-#define AC97_EXTID_REV1           0x0800
-#define AC97_EXTID_ID0            0x4000
-#define AC97_EXTID_ID1            0x8000
-
-/* extended status register bit defines */
-#define AC97_EXTSTAT_VRA          0x0001
-#define AC97_EXTSTAT_DRA          0x0002
-#define AC97_EXTSTAT_SPDIF        0x0004
-#define AC97_EXTSTAT_VRM          0x0008
-#define AC97_EXTSTAT_SPSA0        0x0010
-#define AC97_EXTSTAT_SPSA1        0x0020
-#define AC97_EXTSTAT_CDAC         0x0040
-#define AC97_EXTSTAT_SDAC         0x0080
-#define AC97_EXTSTAT_LDAC         0x0100
-#define AC97_EXTSTAT_MADC         0x0200
-#define AC97_EXTSTAT_SPCV         0x0400
-#define AC97_EXTSTAT_PRI          0x0800
-#define AC97_EXTSTAT_PRJ          0x1000
-#define AC97_EXTSTAT_PRK          0x2000
-#define AC97_EXTSTAT_PRL          0x4000
-
-/* extended audio ID register bit defines */
-#define AC97_EXTID_VRA            0x0001
-#define AC97_EXTID_DRA            0x0002
-#define AC97_EXTID_SPDIF          0x0004
-#define AC97_EXTID_VRM            0x0008
-#define AC97_EXTID_DSA0           0x0010
-#define AC97_EXTID_DSA1           0x0020
-#define AC97_EXTID_CDAC           0x0040
-#define AC97_EXTID_SDAC           0x0080
-#define AC97_EXTID_LDAC           0x0100
-#define AC97_EXTID_AMAP           0x0200
-#define AC97_EXTID_REV0           0x0400
-#define AC97_EXTID_REV1           0x0800
-#define AC97_EXTID_ID0            0x4000
-#define AC97_EXTID_ID1            0x8000
-
-/* extended status register bit defines */
-#define AC97_EXTSTAT_VRA          0x0001
-#define AC97_EXTSTAT_DRA          0x0002
-#define AC97_EXTSTAT_SPDIF        0x0004
-#define AC97_EXTSTAT_VRM          0x0008
-#define AC97_EXTSTAT_SPSA0        0x0010
-#define AC97_EXTSTAT_SPSA1        0x0020
-#define AC97_EXTSTAT_CDAC         0x0040
-#define AC97_EXTSTAT_SDAC         0x0080
-#define AC97_EXTSTAT_LDAC         0x0100
-#define AC97_EXTSTAT_MADC         0x0200
-#define AC97_EXTSTAT_SPCV         0x0400
-#define AC97_EXTSTAT_PRI          0x0800
-#define AC97_EXTSTAT_PRJ          0x1000
-#define AC97_EXTSTAT_PRK          0x2000
-#define AC97_EXTSTAT_PRL          0x4000
-
-/* useful power states */
-#define AC97_PWR_D0               0x0000      /* everything on */
-#define AC97_PWR_D1              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR4
-#define AC97_PWR_D2              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4
-#define AC97_PWR_D3              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4
-#define AC97_PWR_ANLOFF          AC97_PWR_PR2|AC97_PWR_PR3  /* analog section off */
-
-/* Total number of defined registers.  */
-#define AC97_REG_CNT 64
-
-
-/* OSS interface to the ac97s.. */
-#define AC97_STEREO_MASK (SOUND_MASK_VOLUME|SOUND_MASK_PCM|\
-	SOUND_MASK_LINE|SOUND_MASK_CD|\
-	SOUND_MASK_ALTPCM|SOUND_MASK_IGAIN|\
-	SOUND_MASK_LINE1|SOUND_MASK_VIDEO)
-
-#define AC97_SUPPORTED_MASK (AC97_STEREO_MASK | \
-	SOUND_MASK_BASS|SOUND_MASK_TREBLE|\
-	SOUND_MASK_SPEAKER|SOUND_MASK_MIC|\
-	SOUND_MASK_PHONEIN|SOUND_MASK_PHONEOUT)
-
-#define AC97_RECORD_MASK (SOUND_MASK_MIC|\
-	SOUND_MASK_CD|SOUND_MASK_IGAIN|SOUND_MASK_VIDEO|\
-	SOUND_MASK_LINE1| SOUND_MASK_LINE|\
-	SOUND_MASK_PHONEIN)
-
-/* original check is not good enough in case FOO is greater than
- * SOUND_MIXER_NRDEVICES because the supported_mixers has exactly
- * SOUND_MIXER_NRDEVICES elements.
- * before matching the given mixer against the bitmask in supported_mixers we
- * check if mixer number exceeds maximum allowed size which is as mentioned
- * above SOUND_MIXER_NRDEVICES */
-#define supported_mixer(CODEC,FOO) ((FOO >= 0) && \
-                                    (FOO < SOUND_MIXER_NRDEVICES) && \
-                                    (CODEC)->supported_mixers & (1<<FOO) )
-
-struct ac97_codec {
-	/* Linked list of codecs */
-	struct list_head list;
-
-	/* AC97 controller connected with */
-	void *private_data;
-
-	char *name;
-	int id;
-	int dev_mixer; 
-	int type;
-	u32 model;
-
-	unsigned int modem:1;
-
-	struct ac97_ops *codec_ops;
-
-	/* controller specific lower leverl ac97 accessing routines.
-	   must be re-entrant safe */
-	u16  (*codec_read)  (struct ac97_codec *codec, u8 reg);
-	void (*codec_write) (struct ac97_codec *codec, u8 reg, u16 val);
-
-	/* Wait for codec-ready.  Ok to sleep here.  */
-	void  (*codec_wait)  (struct ac97_codec *codec);
-
-	/* callback used by helper drivers for interesting ac97 setups */
-	void  (*codec_unregister) (struct ac97_codec *codec);
-	
-	struct ac97_driver *driver;
-	void *driver_private;	/* Private data for the driver */
-	
-	spinlock_t lock;
-	
-	/* OSS mixer masks */
-	int modcnt;
-	int supported_mixers;
-	int stereo_mixers;
-	int record_sources;
-
-	/* Property flags */
-	int flags;
-
-	int bit_resolution;
-
-	/* OSS mixer interface */
-	int  (*read_mixer) (struct ac97_codec *codec, int oss_channel);
-	void (*write_mixer)(struct ac97_codec *codec, int oss_channel,
-			    unsigned int left, unsigned int right);
-	int  (*recmask_io) (struct ac97_codec *codec, int rw, int mask);
-	int  (*mixer_ioctl)(struct ac97_codec *codec, unsigned int cmd, unsigned long arg);
-
-	/* saved OSS mixer states */
-	unsigned int mixer_state[SOUND_MIXER_NRDEVICES];
-
-	/* Software Modem interface */
-	int  (*modem_ioctl)(struct ac97_codec *codec, unsigned int cmd, unsigned long arg);
-};
-
-/*
- *	Operation structures for each known AC97 chip
- */
- 
-struct ac97_ops
-{
-	/* Initialise */
-	int (*init)(struct ac97_codec *c);
-	/* Amplifier control */
-	int (*amplifier)(struct ac97_codec *codec, int on);
-	/* Digital mode control */
-	int (*digital)(struct ac97_codec *codec, int slots, int rate, int mode);
-#define AUDIO_DIGITAL		0x8000
-#define AUDIO_PRO		0x4000
-#define AUDIO_DRS		0x2000
-#define AUDIO_CCMASK		0x003F
-	
-#define AC97_DELUDED_MODEM	1	/* Audio codec reports its a modem */
-#define AC97_NO_PCM_VOLUME	2	/* Volume control is missing 	   */
-#define AC97_DEFAULT_POWER_OFF 4 /* Needs warm reset to power up */
-};
-
-extern int ac97_probe_codec(struct ac97_codec *);
-
-extern struct ac97_codec *ac97_alloc_codec(void);
-extern void ac97_release_codec(struct ac97_codec *codec);
-
-struct ac97_driver {
-	struct list_head list;
-	char *name;
-	u32 codec_id;
-	u32 codec_mask;
-	int (*probe) (struct ac97_codec *codec, struct ac97_driver *driver);
-	void (*remove) (struct ac97_codec *codec, struct ac97_driver *driver);
-};
-
-/* quirk types */
-enum {
-	AC97_TUNE_DEFAULT = -1, /* use default from quirk list (not valid in list) */
-	AC97_TUNE_NONE = 0,     /* nothing extra to do */
-	AC97_TUNE_HP_ONLY,      /* headphone (true line-out) control as master only */
-	AC97_TUNE_SWAP_HP,      /* swap headphone and master controls */
-	AC97_TUNE_SWAP_SURROUND, /* swap master and surround controls */
-	AC97_TUNE_AD_SHARING,   /* for AD1985, turn on OMS bit and use headphone */
-	AC97_TUNE_ALC_JACK,     /* for Realtek, enable JACK detection */
-};
-
-struct ac97_quirk {
-	unsigned short vendor;  /* PCI vendor id */
-	unsigned short device;  /* PCI device id */
-	unsigned short mask;    /* device id bit mask, 0 = accept all */
-	const char *name;       /* name shown as info */
-	int type;               /* quirk type above */
-};
-
-#endif /* _AC97_CODEC_H_ */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 0d1112815be3..e91e6047ca6f 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -810,7 +810,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa
 int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, 
 			       unsigned int cond,
 			       snd_pcm_hw_param_t var,
-			       struct snd_pcm_hw_constraint_list *l);
+			       const struct snd_pcm_hw_constraint_list *l);
 int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, 
 				  unsigned int cond,
 				  snd_pcm_hw_param_t var,
@@ -893,6 +893,7 @@ extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates;
 
 int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
 unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
+unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
 
 static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream,
 					      struct snd_dma_buffer *bufp)
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index f494f1e3c900..37ae12e0ab06 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -22,6 +22,8 @@
  *
  */
 
+#include <sound/pcm.h>
+
 int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, 
 			   struct snd_pcm_hw_params *params,
 			   snd_pcm_hw_param_t var, int *dir);
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index 7067e2dfb0b9..a64d8fe3f855 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -38,21 +38,31 @@
 #define SNDRV_CTL_TLVT_DB_MINMAX 4	/* dB scale with min/max */
 #define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5	/* dB scale with min/max with mute */
 
+#define TLV_ITEM(type, ...) \
+	(type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__
+#define TLV_LENGTH(...) \
+	((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
+
+#define TLV_CONTAINER_ITEM(...) \
+	TLV_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
+#define DECLARE_TLV_CONTAINER(name, ...) \
+	unsigned int name[] = { TLV_CONTAINER_ITEM(__VA_ARGS__) }
+
 #define TLV_DB_SCALE_MASK	0xffff
 #define TLV_DB_SCALE_MUTE	0x10000
 #define TLV_DB_SCALE_ITEM(min, step, mute)			\
-	SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int),	\
-	(min), ((step) & TLV_DB_SCALE_MASK) | ((mute) ? TLV_DB_SCALE_MUTE : 0)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_SCALE,			\
+		 (min),					\
+		 ((step) & TLV_DB_SCALE_MASK) |		\
+			((mute) ? TLV_DB_SCALE_MUTE : 0))
 #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
 	unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
 
 /* dB scale specified with min/max values instead of step */
 #define TLV_DB_MINMAX_ITEM(min_dB, max_dB)			\
-	SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int),	\
-	(min_dB), (max_dB)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
 #define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB)			\
-	SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int),	\
-	(min_dB), (max_dB)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
 #define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
 	unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
 #define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
@@ -60,13 +70,16 @@
 
 /* linear volume between min_dB and max_dB (.01dB unit) */
 #define TLV_DB_LINEAR_ITEM(min_dB, max_dB)		    \
-	SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \
-	(min_dB), (max_dB)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
 #define DECLARE_TLV_DB_LINEAR(name, min_dB, max_dB)	\
 	unsigned int name[] = { TLV_DB_LINEAR_ITEM(min_dB, max_dB) }
 
 /* dB range container */
 /* Each item is: <min> <max> <TLV> */
+#define TLV_DB_RANGE_ITEM(...) \
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
+#define DECLARE_TLV_DB_RANGE(name, ...) \
+	unsigned int name[] = { TLV_DB_RANGE_ITEM(__VA_ARGS__) }
 /* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */
 #define TLV_DB_RANGE_HEAD(num)			\
 	SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int)
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 8f312fa6c282..7ae671923393 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1250,10 +1250,10 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params,
 int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime,
 			       unsigned int cond,
 			       snd_pcm_hw_param_t var,
-			       struct snd_pcm_hw_constraint_list *l)
+			       const struct snd_pcm_hw_constraint_list *l)
 {
 	return snd_pcm_hw_rule_add(runtime, cond, var,
-				   snd_pcm_hw_rule_list, l,
+				   snd_pcm_hw_rule_list, (void *)l,
 				   var, -1);
 }
 
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 9c9eff9afbac..d4fc1bfbe457 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -488,3 +488,21 @@ unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate)
 	return SNDRV_PCM_RATE_KNOT;
 }
 EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit);
+
+/**
+ * snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate
+ * @rate_bit: the rate bit to convert
+ *
+ * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag
+ * or 0 for an unknown rate bit
+ */
+unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit)
+{
+	unsigned int i;
+
+	for (i = 0; i < snd_pcm_known_rates.count; i++)
+		if ((1u << i) == rate_bit)
+			return snd_pcm_known_rates.list[i];
+	return 0;
+}
+EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate);
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index d7ccf28bd66a..f8fbe22515c9 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -135,10 +135,9 @@ struct snd_opti9xx {
 	unsigned long mc_base_size;
 #ifdef OPTi93X
 	unsigned long mc_indir_index;
-	unsigned long mc_indir_size;
 	struct resource *res_mc_indir;
-	struct snd_wss *codec;
 #endif	/* OPTi93X */
+	struct snd_wss *codec;
 	unsigned long pwd_reg;
 
 	spinlock_t lock;
@@ -245,10 +244,8 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip,
 	case OPTi9XX_HW_82C931:
 	case OPTi9XX_HW_82C933:
 		chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d;
-		if (!chip->mc_indir_index) {
+		if (!chip->mc_indir_index)
 			chip->mc_indir_index = 0xe0e;
-			chip->mc_indir_size = 2;
-		}
 		chip->password = 0xe4;
 		chip->pwd_reg = 0;
 		break;
@@ -351,7 +348,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg,
 		(snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
 
 
-static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip,
+static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 					   long port,
 					   int irq, int dma1, int dma2,
 					   long mpu_port, int mpu_irq)
@@ -403,7 +400,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip,
 
 #else	/* OPTi93X */
 	case OPTi9XX_HW_82C931:
-	case OPTi9XX_HW_82C933:
+		/* disable 3D sound (set GPIO1 as output, low) */
+		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
+	case OPTi9XX_HW_82C933: /* FALL THROUGH */
 		/*
 		 * The BTC 1817DW has QS1000 wavetable which is connected
 		 * to the serial digital input of the OPTI931.
@@ -696,8 +695,7 @@ static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip)
 		if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)))
 			return 0;
 #else	/* OPTi93X */
-	chip->res_mc_indir = request_region(chip->mc_indir_index,
-					    chip->mc_indir_size,
+	chip->res_mc_indir = request_region(chip->mc_indir_index, 2,
 					    "OPTi93x MC");
 	if (chip->res_mc_indir == NULL)
 		return -EBUSY;
@@ -770,8 +768,9 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip,
 #ifdef OPTi93X
 	port = pnp_port_start(pdev, 0) - 4;
 	fm_port = pnp_port_start(pdev, 1) + 8;
-	chip->mc_indir_index = pnp_port_start(pdev, 3) + 2;
-	chip->mc_indir_size = pnp_port_len(pdev, 3) - 2;
+	/* adjust mc_indir_index - some cards report it at 0xe?d,
+	   other at 0xe?c but it really is always at 0xe?e */
+	chip->mc_indir_index = (pnp_port_start(pdev, 3) & ~0xf) | 0xe;
 #else
 	devmc = pnp_request_card_device(card, pid->devs[2].id, NULL);
 	if (devmc == NULL)
@@ -871,9 +870,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
 			       &codec);
 	if (error < 0)
 		return error;
-#ifdef OPTi93X
 	chip->codec = codec;
-#endif
 	error = snd_wss_pcm(codec, 0, &pcm);
 	if (error < 0)
 		return error;
@@ -1054,11 +1051,55 @@ static int __devexit snd_opti9xx_isa_remove(struct device *devptr,
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int snd_opti9xx_suspend(struct snd_card *card)
+{
+	struct snd_opti9xx *chip = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	chip->codec->suspend(chip->codec);
+	return 0;
+}
+
+static int snd_opti9xx_resume(struct snd_card *card)
+{
+	struct snd_opti9xx *chip = card->private_data;
+	int error, xdma2;
+#if defined(CS4231) || defined(OPTi93X)
+	xdma2 = dma2;
+#else
+	xdma2 = -1;
+#endif
+
+	error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2,
+				      mpu_port, mpu_irq);
+	if (error)
+		return error;
+	chip->codec->resume(chip->codec);
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+
+static int snd_opti9xx_isa_suspend(struct device *dev, unsigned int n,
+				   pm_message_t state)
+{
+	return snd_opti9xx_suspend(dev_get_drvdata(dev));
+}
+
+static int snd_opti9xx_isa_resume(struct device *dev, unsigned int n)
+{
+	return snd_opti9xx_resume(dev_get_drvdata(dev));
+}
+#endif
+
 static struct isa_driver snd_opti9xx_driver = {
 	.match		= snd_opti9xx_isa_match,
 	.probe		= snd_opti9xx_isa_probe,
 	.remove		= __devexit_p(snd_opti9xx_isa_remove),
-	/* FIXME: suspend/resume */
+#ifdef CONFIG_PM
+	.suspend	= snd_opti9xx_isa_suspend,
+	.resume		= snd_opti9xx_isa_resume,
+#endif
 	.driver		= {
 		.name	= DEV_NAME
 	},
@@ -1124,12 +1165,29 @@ static void __devexit snd_opti9xx_pnp_remove(struct pnp_card_link * pcard)
 	snd_opti9xx_pnp_is_probed = 0;
 }
 
+#ifdef CONFIG_PM
+static int snd_opti9xx_pnp_suspend(struct pnp_card_link *pcard,
+				   pm_message_t state)
+{
+	return snd_opti9xx_suspend(pnp_get_card_drvdata(pcard));
+}
+
+static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard)
+{
+	return snd_opti9xx_resume(pnp_get_card_drvdata(pcard));
+}
+#endif
+
 static struct pnp_card_driver opti9xx_pnpc_driver = {
 	.flags		= PNP_DRIVER_RES_DISABLE,
 	.name		= "opti9xx",
 	.id_table	= snd_opti9xx_pnpids,
 	.probe		= snd_opti9xx_pnp_probe,
 	.remove		= __devexit_p(snd_opti9xx_pnp_remove),
+#ifdef CONFIG_PM
+	.suspend	= snd_opti9xx_pnp_suspend,
+	.resume		= snd_opti9xx_pnp_resume,
+#endif
 };
 #endif
 
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 49c8a0c2442c..360b08b03e1d 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1456,7 +1456,6 @@ static struct snd_pcm_hardware snd_wss_playback =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_MMAP_VALID |
-				 SNDRV_PCM_INFO_RESUME |
 				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		(SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM |
 				 SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE),
@@ -1657,6 +1656,10 @@ static void snd_wss_resume(struct snd_wss *chip)
 			break;
 		}
 	}
+	/* Yamaha needs this to resume properly */
+	if (chip->hardware == WSS_HW_OPL3SA2)
+		snd_wss_out(chip, CS4231_PLAYBK_FORMAT,
+			    chip->image[CS4231_PLAYBK_FORMAT]);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 #if 1
 	snd_wss_mce_down(chip);
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 09d46484bc1a..7d8803a00b79 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -69,7 +69,6 @@
 #include <linux/sound.h>
 #include <linux/slab.h>
 #include <linux/soundcard.h>
-#include <linux/ac97_codec.h>
 #include <linux/pci.h>
 #include <linux/bitops.h>
 #include <linux/interrupt.h>
@@ -199,6 +198,22 @@ static const char invalid_magic[] =
         }                                          \
 })
 
+/* AC97 registers */
+#define AC97_MASTER_VOL_STEREO   0x0002      /* Line Out		*/
+#define AC97_PCBEEP_VOL          0x000a      /* none			*/
+#define AC97_PHONE_VOL           0x000c      /* TAD Input (mono)	*/
+#define AC97_MIC_VOL             0x000e      /* MIC Input (mono)	*/
+#define AC97_LINEIN_VOL          0x0010      /* Line Input (stereo)	*/
+#define AC97_CD_VOL              0x0012      /* CD Input (stereo)	*/
+#define AC97_AUX_VOL             0x0016      /* Aux Input (stereo)	*/
+#define AC97_PCMOUT_VOL          0x0018      /* Wave Output (stereo)	*/
+#define AC97_RECORD_SELECT       0x001a      /*			*/
+#define AC97_RECORD_GAIN         0x001c
+#define AC97_GENERAL_PURPOSE     0x0020
+#define AC97_3D_CONTROL          0x0022
+#define AC97_POWER_CONTROL       0x0026
+#define AC97_VENDOR_ID1           0x007c
+
 struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs };
 
 typedef struct serdma_descr_s {
diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c
index 557c782ae4fc..fa13efbebdaf 100644
--- a/sound/pci/au88x0/au88x0_mixer.c
+++ b/sound/pci/au88x0/au88x0_mixer.c
@@ -10,6 +10,15 @@
 #include <sound/core.h>
 #include "au88x0.h"
 
+static int remove_ctl(struct snd_card *card, const char *name)
+{
+	struct snd_ctl_elem_id id;
+	memset(&id, 0, sizeof(id));
+	strcpy(id.name, name);
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	return snd_ctl_remove_id(card, &id);
+}
+
 static int __devinit snd_vortex_mixer(vortex_t * vortex)
 {
 	struct snd_ac97_bus *pbus;
@@ -28,5 +37,7 @@ static int __devinit snd_vortex_mixer(vortex_t * vortex)
 	ac97.scaps = AC97_SCAP_NO_SPDIF;
 	err = snd_ac97_mixer(pbus, &ac97, &vortex->codec);
 	vortex->isquad = ((vortex->codec == NULL) ?  0 : (vortex->codec->ext_id&0x80));
+	remove_ctl(vortex->card, "Master Mono Playback Volume");
+	remove_ctl(vortex->card, "Master Mono Playback Switch");
 	return err;
 }
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 82c8d8c5c52a..a41106d745ca 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1321,35 +1321,30 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static unsigned int db_scale_master[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_master,
 	0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1),
 	54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0),
-};
+);
 
-static unsigned int db_scale_audio1[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_audio1,
 	0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0),
-};
+);
 
-static unsigned int db_scale_audio2[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_audio2,
 	0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0),
-};
+);
 
-static unsigned int db_scale_mic[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_mic,
 	0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(0, 150, 0),
-};
+);
 
-static unsigned int db_scale_line[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_line,
 	0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0),
-};
+);
 
 static const DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0);
 
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index deef21399586..adb3b4c7917e 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -361,74 +361,6 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)");
 #define DSP2HOST_REQ_I2SRATE    0x02
 #define DSP2HOST_REQ_TIMER      0x04
 
-/* AC97 registers */
-/* XXX fix this crap up */
-/*#define AC97_RESET              0x00*/
-
-#define AC97_VOL_MUTE_B         0x8000
-#define AC97_VOL_M              0x1F
-#define AC97_LEFT_VOL_S         8
-
-#define AC97_MASTER_VOL         0x02
-#define AC97_LINE_LEVEL_VOL     0x04
-#define AC97_MASTER_MONO_VOL    0x06
-#define AC97_PC_BEEP_VOL        0x0A
-#define AC97_PC_BEEP_VOL_M      0x0F
-#define AC97_SROUND_MASTER_VOL  0x38
-#define AC97_PC_BEEP_VOL_S      1
-
-/*#define AC97_PHONE_VOL          0x0C
-#define AC97_MIC_VOL            0x0E*/
-#define AC97_MIC_20DB_ENABLE    0x40
-
-/*#define AC97_LINEIN_VOL         0x10
-#define AC97_CD_VOL             0x12
-#define AC97_VIDEO_VOL          0x14
-#define AC97_AUX_VOL            0x16*/
-#define AC97_PCM_OUT_VOL        0x18
-/*#define AC97_RECORD_SELECT      0x1A*/
-#define AC97_RECORD_MIC         0x00
-#define AC97_RECORD_CD          0x01
-#define AC97_RECORD_VIDEO       0x02
-#define AC97_RECORD_AUX         0x03
-#define AC97_RECORD_MONO_MUX    0x02
-#define AC97_RECORD_DIGITAL     0x03
-#define AC97_RECORD_LINE        0x04
-#define AC97_RECORD_STEREO      0x05
-#define AC97_RECORD_MONO        0x06
-#define AC97_RECORD_PHONE       0x07
-
-/*#define AC97_RECORD_GAIN        0x1C*/
-#define AC97_RECORD_VOL_M       0x0F
-
-/*#define AC97_GENERAL_PURPOSE    0x20*/
-#define AC97_POWER_DOWN_CTRL    0x26
-#define AC97_ADC_READY          0x0001
-#define AC97_DAC_READY          0x0002
-#define AC97_ANALOG_READY       0x0004
-#define AC97_VREF_ON            0x0008
-#define AC97_PR0                0x0100
-#define AC97_PR1                0x0200
-#define AC97_PR2                0x0400
-#define AC97_PR3                0x0800
-#define AC97_PR4                0x1000
-
-#define AC97_RESERVED1          0x28
-
-#define AC97_VENDOR_TEST        0x5A
-
-#define AC97_CLOCK_DELAY        0x5C
-#define AC97_LINEOUT_MUX_SEL    0x0001
-#define AC97_MONO_MUX_SEL       0x0002
-#define AC97_CLOCK_DELAY_SEL    0x1F
-#define AC97_DAC_CDS_SHIFT      6
-#define AC97_ADC_CDS_SHIFT      11
-
-#define AC97_MULTI_CHANNEL_SEL  0x74
-
-/*#define AC97_VENDOR_ID1         0x7C
-#define AC97_VENDOR_ID2         0x7E*/
-
 /*
  * ASSP control regs
  */
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 0435f45e9513..e3ac1f768ff6 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1368,6 +1368,67 @@ static void pcxhr_proc_gpo_write(struct snd_info_entry *entry,
 	}
 }
 
+/* Access to the results of the CMD_GET_TIME_CODE RMH */
+#define TIME_CODE_VALID_MASK	0x00800000
+#define TIME_CODE_NEW_MASK	0x00400000
+#define TIME_CODE_BACK_MASK	0x00200000
+#define TIME_CODE_WAIT_MASK	0x00100000
+
+/* Values for the CMD_MANAGE_SIGNAL RMH */
+#define MANAGE_SIGNAL_TIME_CODE	0x01
+#define MANAGE_SIGNAL_MIDI	0x02
+
+/* linear time code read proc*/
+static void pcxhr_proc_ltc(struct snd_info_entry *entry,
+			   struct snd_info_buffer *buffer)
+{
+	struct snd_pcxhr *chip = entry->private_data;
+	struct pcxhr_mgr *mgr = chip->mgr;
+	struct pcxhr_rmh rmh;
+	unsigned int ltcHrs, ltcMin, ltcSec, ltcFrm;
+	int err;
+	/* commands available when embedded DSP is running */
+	if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))) {
+		snd_iprintf(buffer, "no firmware loaded\n");
+		return;
+	}
+	if (!mgr->capture_ltc) {
+		pcxhr_init_rmh(&rmh, CMD_MANAGE_SIGNAL);
+		rmh.cmd[0] |= MANAGE_SIGNAL_TIME_CODE;
+		err = pcxhr_send_msg(mgr, &rmh);
+		if (err) {
+			snd_iprintf(buffer, "ltc not activated (%d)\n", err);
+			return;
+		}
+		if (mgr->is_hr_stereo)
+			hr222_manage_timecode(mgr, 1);
+		else
+			pcxhr_write_io_num_reg_cont(mgr, REG_CONT_VALSMPTE,
+						    REG_CONT_VALSMPTE, NULL);
+		mgr->capture_ltc = 1;
+	}
+	pcxhr_init_rmh(&rmh, CMD_GET_TIME_CODE);
+	err = pcxhr_send_msg(mgr, &rmh);
+	if (err) {
+		snd_iprintf(buffer, "ltc read error (err=%d)\n", err);
+		return ;
+	}
+	ltcHrs = 10*((rmh.stat[0] >> 8) & 0x3) + (rmh.stat[0] & 0xf);
+	ltcMin = 10*((rmh.stat[1] >> 16) & 0x7) + ((rmh.stat[1] >> 8) & 0xf);
+	ltcSec = 10*(rmh.stat[1] & 0x7) + ((rmh.stat[2] >> 16) & 0xf);
+	ltcFrm = 10*((rmh.stat[2] >> 8) & 0x3) + (rmh.stat[2] & 0xf);
+
+	snd_iprintf(buffer, "timecode: %02u:%02u:%02u-%02u\n",
+			    ltcHrs, ltcMin, ltcSec, ltcFrm);
+	snd_iprintf(buffer, "raw: 0x%04x%06x%06x\n", rmh.stat[0] & 0x00ffff,
+			    rmh.stat[1] & 0xffffff, rmh.stat[2] & 0xffffff);
+	/*snd_iprintf(buffer, "dsp ref time: 0x%06x%06x\n",
+			    rmh.stat[3] & 0xffffff, rmh.stat[4] & 0xffffff);*/
+	if (!(rmh.stat[0] & TIME_CODE_VALID_MASK)) {
+		snd_iprintf(buffer, "warning: linear timecode not valid\n");
+	}
+}
+
 static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 {
 	struct snd_info_entry *entry;
@@ -1383,6 +1444,8 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 		entry->c.text.write = pcxhr_proc_gpo_write;
 		entry->mode |= S_IWUSR;
 	}
+	if (!snd_card_proc_new(chip->card, "ltc", &entry))
+		snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc);
 }
 /* end of proc interface */
 
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index bda776c49884..a4c602c45173 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -103,6 +103,7 @@ struct pcxhr_mgr {
 	unsigned int board_has_mic:1; /* if 1 the board has microphone input */
 	unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
 	unsigned int mono_capture:1; /* if 1 the board does mono capture */
+	unsigned int capture_ltc:1; /* if 1 the board captures LTC input */
 
 	struct snd_dma_buffer hostport;
 
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 304411c1fe4b..b33db1e006e7 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -504,6 +504,8 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = {
 [CMD_FORMAT_STREAM_IN] =		{ 0x870000, 0, RMH_SSIZE_FIXED },
 [CMD_STREAM_SAMPLE_COUNT] =		{ 0x902000, 2, RMH_SSIZE_FIXED },
 [CMD_AUDIO_LEVEL_ADJUST] =		{ 0xc22000, 0, RMH_SSIZE_FIXED },
+[CMD_GET_TIME_CODE] =			{ 0x060000, 5, RMH_SSIZE_FIXED },
+[CMD_MANAGE_SIGNAL] =			{ 0x0f0000, 0, RMH_SSIZE_FIXED },
 };
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE
@@ -533,6 +535,8 @@ static char* cmd_names[] = {
 [CMD_FORMAT_STREAM_IN] =		"CMD_FORMAT_STREAM_IN",
 [CMD_STREAM_SAMPLE_COUNT] =		"CMD_STREAM_SAMPLE_COUNT",
 [CMD_AUDIO_LEVEL_ADJUST] =		"CMD_AUDIO_LEVEL_ADJUST",
+[CMD_GET_TIME_CODE] =			"CMD_GET_TIME_CODE",
+[CMD_MANAGE_SIGNAL] =			"CMD_MANAGE_SIGNAL",
 };
 #endif
 
@@ -1133,13 +1137,12 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr,
 	hw_sample_count = ((u_int64_t)rmh.stat[0]) << 24;
 	hw_sample_count += (u_int64_t)rmh.stat[1];
 
-	snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n",
+	snd_printdd("stream %c%d : abs samples real(%llu) timer(%llu)\n",
 		    stream->pipe->is_capture ? 'C' : 'P',
 		    stream->substream->number,
-		    (long unsigned int)hw_sample_count,
-		    (long unsigned int)(stream->timer_abs_periods +
-					stream->timer_period_frag +
-					mgr->granularity));
+		    hw_sample_count,
+		    stream->timer_abs_periods + stream->timer_period_frag +
+						mgr->granularity);
 	return hw_sample_count;
 }
 
@@ -1243,10 +1246,18 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 
 		if ((dsp_time_diff < 0) &&
 		    (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) {
-			snd_printdd("ERROR DSP TIME old(%d) new(%d) -> "
-				    "resynchronize all streams\n",
+			/* handle dsp counter wraparound without resync */
+			int tmp_diff = dsp_time_diff + PCXHR_DSP_TIME_MASK + 1;
+			snd_printdd("WARNING DSP timestamp old(%d) new(%d)",
 				    mgr->dsp_time_last, dsp_time_new);
-			mgr->dsp_time_err++;
+			if (tmp_diff > 0 && tmp_diff <= (2*mgr->granularity)) {
+				snd_printdd("-> timestamp wraparound OK: "
+					    "diff=%d\n", tmp_diff);
+				dsp_time_diff = tmp_diff;
+			} else {
+				snd_printdd("-> resynchronize all streams\n");
+				mgr->dsp_time_err++;
+			}
 		}
 #ifdef CONFIG_SND_DEBUG_VERBOSE
 		if (dsp_time_diff == 0)
diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h
index be0173796cdb..a81ab6b811e7 100644
--- a/sound/pci/pcxhr/pcxhr_core.h
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -79,6 +79,8 @@ enum {
 	CMD_FORMAT_STREAM_IN,		/* cmd_len >= 4	stat_len = 0 */
 	CMD_STREAM_SAMPLE_COUNT,	/* cmd_len = 2	stat_len = (2 * nb_stream) */
 	CMD_AUDIO_LEVEL_ADJUST,		/* cmd_len = 3	stat_len = 0 */
+	CMD_GET_TIME_CODE,		/* cmd_len = 1  stat_len = 5 */
+	CMD_MANAGE_SIGNAL,		/* cmd_len = 1  stat_len = 0 */
 	CMD_LAST_INDEX
 };
 
@@ -116,7 +118,7 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh);
 #define IO_NUM_REG_OUT_ANA_LEVEL	20
 #define IO_NUM_REG_IN_ANA_LEVEL		21
 
-
+#define REG_CONT_VALSMPTE		0x000800
 #define REG_CONT_UNMUTE_INPUTS		0x020000
 
 /* parameters used with register IO_NUM_REG_STATUS */
diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c
index 1cb82c0a9cb3..84fe57626eba 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.c
+++ b/sound/pci/pcxhr/pcxhr_mix22.c
@@ -53,6 +53,7 @@
 #define PCXHR_DSP_RESET_DSP	0x01
 #define PCXHR_DSP_RESET_MUTE	0x02
 #define PCXHR_DSP_RESET_CODEC	0x08
+#define PCXHR_DSP_RESET_SMPTE	0x10
 #define PCXHR_DSP_RESET_GPO_OFFSET	5
 #define PCXHR_DSP_RESET_GPO_MASK	0x60
 
@@ -527,6 +528,16 @@ int hr222_write_gpo(struct pcxhr_mgr *mgr, int value)
 	return 0;
 }
 
+int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable)
+{
+	if (enable)
+		mgr->dsp_reset |= PCXHR_DSP_RESET_SMPTE;
+	else
+		mgr->dsp_reset &= ~PCXHR_DSP_RESET_SMPTE;
+
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset);
+	return 0;
+}
 
 int hr222_update_analog_audio_level(struct snd_pcxhr *chip,
 				    int is_capture, int channel)
diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h
index 5a37a0007e8f..5971b9933f41 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.h
+++ b/sound/pci/pcxhr/pcxhr_mix22.h
@@ -34,6 +34,7 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr,
 
 int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value);
 int hr222_write_gpo(struct pcxhr_mgr *mgr, int value);
+int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable);
 
 #define HR222_LINE_PLAYBACK_LEVEL_MIN		0	/* -25.5 dB */
 #define HR222_LINE_PLAYBACK_ZERO_LEVEL		51	/* 0.0 dB */
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 64aed432ae22..7da0d0aa72cb 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf,
 		     const struct usb_device_id *id)
 {
 	int ret;
-	struct snd_card *card;
+	struct snd_card *card = NULL;
 	struct usb_device *device = interface_to_usbdev(intf);
 
 	ret = create_card(device, intf, &card);
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 41f4b6911920..690000db0ec0 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -42,6 +42,13 @@
 
 extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
 
+struct std_mono_table {
+	unsigned int unitid, control, cmask;
+	int val_type;
+	const char *name;
+	snd_kcontrol_tlv_rw_t *tlv_callback;
+};
+
 /* private_free callback */
 static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
 {
@@ -114,6 +121,25 @@ static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer,
 }
 
 /*
+ * Create a set of standard UAC controls from a table
+ */
+static int snd_create_std_mono_table(struct usb_mixer_interface *mixer,
+				struct std_mono_table *t)
+{
+	int err;
+
+	while (t->name != NULL) {
+		err = snd_create_std_mono_ctl(mixer, t->unitid, t->control,
+				t->cmask, t->val_type, t->name, t->tlv_callback);
+		if (err < 0)
+			return err;
+		t++;
+	}
+
+	return 0;
+}
+
+/*
  * Sound Blaster remote control configuration
  *
  * format of remote control data:
@@ -916,61 +942,6 @@ static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
-
-/*
- * Create mixer for Electrix Ebox-44
- *
- * The mixer units from this device are corrupt, and even where they
- * are valid they presents mono controls as L and R channels of
- * stereo. So we create a good mixer in code.
- */
-
-static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer)
-{
-	int err;
-
-	err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Headphone Playback Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16,
-				"Headphone A Mix Playback Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16,
-				"Headphone B Mix Playback Volume", NULL);
-	if (err < 0)
-		return err;
-
-	err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Output Playback Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16,
-				"Output A Playback Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16,
-				"Output B Playback Volume", NULL);
-	if (err < 0)
-		return err;
-
-	err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Input Capture Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16,
-				"Input A Capture Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16,
-				"Input B Capture Volume", NULL);
-	if (err < 0)
-		return err;
-
-	return 0;
-}
-
 void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 			       unsigned char samplerate_id)
 {
@@ -990,6 +961,81 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 	}
 }
 
+/*
+ * The mixer units for Ebox-44 are corrupt, and even where they
+ * are valid they presents mono controls as L and R channels of
+ * stereo. So we provide a good mixer here.
+ */
+struct std_mono_table ebox44_table[] = {
+	{
+		.unitid = 4,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Headphone Playback Switch"
+	},
+	{
+		.unitid = 4,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Headphone A Mix Playback Volume"
+	},
+	{
+		.unitid = 4,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Headphone B Mix Playback Volume"
+	},
+
+	{
+		.unitid = 7,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Output Playback Switch"
+	},
+	{
+		.unitid = 7,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Output A Playback Volume"
+	},
+	{
+		.unitid = 7,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Output B Playback Volume"
+	},
+
+	{
+		.unitid = 10,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Input Capture Switch"
+	},
+	{
+		.unitid = 10,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Input A Capture Volume"
+	},
+	{
+		.unitid = 10,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Input B Capture Volume"
+	},
+
+	{}
+};
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -1035,7 +1081,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 		break;
 
 	case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */
-		err = snd_ebox44_create_mixer(mixer);
+		/* detection is disabled in mixer_maps.c */
+		err = snd_create_std_mono_table(mixer, ebox44_table);
 		break;
 	}