Merge branch 'topic/misc' into for-next
Generic updates for sound 3.6
diff --git a/include/linux/ac97_codec.h b/include/linux/ac97_codec.h
deleted file mode 100644
index 0260c3e..0000000
--- 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 0d11128..e91e604 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -810,7 +810,7 @@
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 @@
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 f494f1e..37ae12e 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 7067e2d..a64d8fe 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 8f312fa..7ae6719 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1250,10 +1250,10 @@
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 9c9eff9..d4fc1bf 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -488,3 +488,21 @@
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 d7ccf28..f8fbe22 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -135,10 +135,9 @@
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 @@
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 @@
(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 @@
#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 @@
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 @@
#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 @@
&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 @@
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 @@
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 49c8a0c..360b08b 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1456,7 +1456,6 @@
{
.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 @@
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 09d4648..7d8803a 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 @@
} \
})
+/* 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 557c782..fa13efb 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 @@
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 82c8d8c..a41106d 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1321,35 +1321,30 @@
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 deef213..adb3b4c 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -361,74 +361,6 @@
#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 0435f45..e3ac1f7 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1368,6 +1368,67 @@
}
}
+/* 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 @@
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 bda776c..a4c602c 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -103,6 +103,7 @@
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 304411c..b33db1e 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -504,6 +504,8 @@
[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 @@
[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 @@
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 @@
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 be01737..a81ab6b 100644
--- a/sound/pci/pcxhr/pcxhr_core.h
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -79,6 +79,8 @@
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 @@
#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 1cb82c0..84fe576 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 @@
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 5a37a00..5971b99 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.h
+++ b/sound/pci/pcxhr/pcxhr_mix22.h
@@ -34,6 +34,7 @@
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 64aed43..7da0d0a 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -485,7 +485,7 @@
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 41f4b69..690000d 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 @@
}
/*
+ * 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 @@
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 @@
}
}
+/*
+ * 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 @@
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;
}
