Merge branch 'topic/misc' into for-next

Generic updates for sound 3.6

16 files changed, 335 insertions, 167 deletions

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;
         }