13 files changed, 51 insertions, 53 deletions

diff --git a/sound/core/jack.c b/sound/core/jack.c
index dd4a12dc09aa..077a85262c1c 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -47,7 +47,7 @@ static int snd_jack_dev_register(struct snd_device *device)
         int err;
 
         snprintf(jack->name, sizeof(jack->name), "%s %s",
-                 card->longname, jack->id);
+                 card->shortname, jack->id);
         jack->input_dev->name = jack->name;
 
         /* Default to the sound card device. */
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index a466443c4a26..2fa9299a440d 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin,
                 while (dst_frames1 > 0) {
                         S1 = S2;
                         if (src_frames1-- > 0) {
-                                S1 = *src;
+                                S2 = *src;
                                 src += src_step;
                         }
                         if (pos & ~R_MASK) {
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 57d9f154c88b..38931f2f6967 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -847,23 +847,23 @@ static int __init AtaIrqInit(void)
            of events. So all we need to keep the music playing is
            to provide the sound hardware with new data upon
            an interrupt from timer A. */
-        mfp.tim_ct_a = 0;       /* ++roman: Stop timer before programming! */
-        mfp.tim_dt_a = 1;       /* Cause interrupt after first event. */
-        mfp.tim_ct_a = 8;       /* Turn on event counting. */
+        st_mfp.tim_ct_a = 0;    /* ++roman: Stop timer before programming! */
+        st_mfp.tim_dt_a = 1;    /* Cause interrupt after first event. */
+        st_mfp.tim_ct_a = 8;    /* Turn on event counting. */
         /* Register interrupt handler. */
         if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound",
                         AtaInterrupt))
                 return 0;
-        mfp.int_en_a |= 0x20;   /* Turn interrupt on. */
-        mfp.int_mk_a |= 0x20;
+        st_mfp.int_en_a |= 0x20;        /* Turn interrupt on. */
+        st_mfp.int_mk_a |= 0x20;
         return 1;
 }
 
 #ifdef MODULE
 static void AtaIrqCleanUp(void)
 {
-        mfp.tim_ct_a = 0;       /* stop timer */
-        mfp.int_en_a &= ~0x20;  /* turn interrupt off */
+        st_mfp.tim_ct_a = 0;            /* stop timer */
+        st_mfp.int_en_a &= ~0x20;       /* turn interrupt off */
         free_irq(IRQ_MFP_TIMA, AtaInterrupt);
 }
 #endif /* MODULE */
@@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void)
                 is_falcon = 0;
             } else
                 return -ENODEV;
-            if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0)
+            if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0)
                 return dmasound_init();
             else {
                 printk("DMA sound driver: Timer A interrupt already in use\n");
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3f00ddf450f8..c7c54e7748e9 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
 
 static struct pci_device_id snd_aw2_ids[] = {
-        {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID,
+        {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
          0, 0, 0},
         {0}
 };
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7958006a1d66..101a1c13a20d 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
          .ca0151_chip = 1,
          .spk71 = 1,
          .spdif_bug = 1,
+         .invert_shared_spdif = 1,      /* digital/analog switch swapped */
          .ac97_chip = 1} ,
         {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
          .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]",
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 482fb0304ca9..4ae51dcb81af 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -277,18 +277,19 @@ static ssize_t init_verbs_store(struct device *dev,
 {
         struct snd_hwdep *hwdep = dev_get_drvdata(dev);
         struct hda_codec *codec = hwdep->private_data;
-        char *p;
-        struct hda_verb verb, *v;
+        struct hda_verb *v;
+        int nid, verb, param;
 
-        verb.nid = simple_strtoul(buf, &p, 0);
-        verb.verb = simple_strtoul(p, &p, 0);
-        verb.param = simple_strtoul(p, &p, 0);
-        if (!verb.nid || !verb.verb || !verb.param)
+        if (sscanf(buf, "%i %i %i", &nid, &verb, &param) != 3)
+                return -EINVAL;
+        if (!nid || !verb)
                 return -EINVAL;
         v = snd_array_new(&codec->init_verbs);
         if (!v)
                 return -ENOMEM;
-        *v = verb;
+        v->nid = nid;
+        v->verb = verb;
+        v->param = param;
         return count;
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 11e791b965f6..5e909e0da04b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1947,16 +1947,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
         return 0;
 }
 
-static int azx_resume_early(struct pci_dev *pci)
-{
-        return pci_restore_state(pci);
-}
-
 static int azx_resume(struct pci_dev *pci)
 {
         struct snd_card *card = pci_get_drvdata(pci);
         struct azx *chip = card->private_data;
 
+        pci_set_power_state(pci, PCI_D0);
+        pci_restore_state(pci);
         if (pci_enable_device(pci) < 0) {
                 printk(KERN_ERR "hda-intel: pci_enable_device failed, "
                        "disabling device\n");
@@ -2098,6 +2095,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
         SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
         /* including bogus ALC268 in slot#2 that conflicts with ALC888 */
         SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
+        /* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */
+        SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03),
         {}
 };
 
@@ -2468,7 +2467,6 @@ static struct pci_driver driver = {
         .remove = __devexit_p(azx_remove),
 #ifdef CONFIG_PM
         .suspend = azx_suspend,
-        .resume_early = azx_resume_early,
         .resume = azx_resume,
 #endif
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ed8fcbd60003..6c26afcb8262 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7017,6 +7017,7 @@ static int patch_alc882(struct hda_codec *codec)
                 case 0x106b3e00: /* iMac 24 Aluminium */
                         board_config = ALC885_IMAC24;
                         break;
+                case 0x106b00a0: /* MacBookPro3,1 - Another revision */
                 case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
                 case 0x106b00a4: /* MacbookPro4,1 */
                 case 0x106b2c00: /* Macbook Pro rev3 */
@@ -8469,6 +8470,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
                 ALC888_ACER_ASPIRE_4930G),
         SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
                 ALC888_ACER_ASPIRE_4930G),
+        SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
+                ALC888_ACER_ASPIRE_4930G),
         SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
         SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
         SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
@@ -10554,6 +10557,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
         SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC),
         SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC),
         SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC),
+        SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC),
         SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
         SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
         SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8027edf3c8f2..3bc427645da8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4989,7 +4989,7 @@ again:
         case STAC_DELL_M4_3:
                 spec->num_dmics = 1;
                 spec->num_smuxes = 0;
-                spec->num_dmuxes = 0;
+                spec->num_dmuxes = 1;
                 break;
         default:
                 spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 18c7c91786bc..6c870c12a177 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -26,7 +26,7 @@
  * SPI 0 -> 1st PCM1796 (front)
  * SPI 1 -> 2nd PCM1796 (surround)
  * SPI 2 -> 3rd PCM1796 (center/LFE)
- * SPI 4 -> 4th PCM1796 (back) and EEPROM self-destruct (do not use!)
+ * SPI 4 -> 4th PCM1796 (back)
  *
  * GPIO 2 -> M0 of CS5381
  * GPIO 3 -> M1 of CS5381
@@ -207,12 +207,6 @@ static void xonar_gpio_changed(struct oxygen *chip);
 static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
                                      u8 reg, u8 value)
 {
-        /*
-         * We don't want to do writes on SPI 4 because the EEPROM, which shares
-         * the same pin, might get confused and broken.  We'd better take care
-         * that the driver works with the default register values ...
-         */
-#if 0
         /* maps ALSA channel pair number to SPI output */
         static const u8 codec_map[4] = {
                 0, 1, 2, 4
@@ -223,7 +217,6 @@ static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
                          (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) |
                          OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
                          (reg << 8) | value);
-#endif
 }
 
 static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec,
@@ -757,9 +750,6 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
 
 static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
 {
-        if (!strncmp(template->name, "Master Playback ", 16))
-                /* disable volume/mute because they would require SPI writes */
-                return 1;
         if (!strncmp(template->name, "CD Capture ", 11))
                 /* CD in is actually connected to the video in pin */
                 template->private_value ^= AC97_CD ^ AC97_VIDEO;
@@ -850,8 +840,9 @@ static const struct oxygen_model model_xonar_d2 = {
         .dac_volume_min = 0x0f,
         .dac_volume_max = 0xff,
         .misc_flags = OXYGEN_MISC_MIDI,
-        .function_flags = OXYGEN_FUNCTION_SPI,
-        .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S,
+        .function_flags = OXYGEN_FUNCTION_SPI |
+                          OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+        .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
         .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 84131a916c92..69d87dee6995 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -97,12 +97,12 @@ struct pcxhr_mgr {
         int capture_chips;
         int fw_file_set;
         int firmware_num;
-        int is_hr_stereo:1;
-        int board_has_aes1:1;   /* if 1 board has AES1 plug and SRC */
-        int board_has_analog:1; /* if 0 the board is digital only */
-        int board_has_mic:1;    /* if 1 the board has microphone input */
-        int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
-        int mono_capture:1;     /* if 1 the board does mono capture */
+        unsigned int is_hr_stereo:1;
+        unsigned int board_has_aes1:1;  /* if 1 board has AES1 plug and SRC */
+        unsigned int board_has_analog:1; /* if 0 the board is digital only */
+        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 */
 
         struct snd_dma_buffer hostport;
 
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 2ab83129d9b0..19e37451c216 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
                  * build the rate table and bitmap flags
                  */
                 int r, idx;
-                unsigned int nonzero_rates = 0;
 
                 fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
                 if (fp->rate_table == NULL) {
@@ -2532,24 +2531,27 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
                         return -1;
                 }
 
-                fp->nr_rates = nr_rates;
-                fp->rate_min = fp->rate_max = combine_triple(&fmt[8]);
+                fp->nr_rates = 0;
+                fp->rate_min = fp->rate_max = 0;
                 for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) {
                         unsigned int rate = combine_triple(&fmt[idx]);
+                        if (!rate)
+                                continue;
                         /* C-Media CM6501 mislabels its 96 kHz altsetting */
                         if (rate == 48000 && nr_rates == 1 &&
-                            chip->usb_id == USB_ID(0x0d8c, 0x0201) &&
+                            (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
+                             chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
                             fp->altsetting == 5 && fp->maxpacksize == 392)
                                 rate = 96000;
-                        fp->rate_table[r] = rate;
-                        nonzero_rates |= rate;
-                        if (rate < fp->rate_min)
+                        fp->rate_table[fp->nr_rates] = rate;
+                        if (!fp->rate_min || rate < fp->rate_min)
                                 fp->rate_min = rate;
-                        else if (rate > fp->rate_max)
+                        if (!fp->rate_max || rate > fp->rate_max)
                                 fp->rate_max = rate;
                         fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+                        fp->nr_rates++;
                 }
-                if (!nonzero_rates) {
+                if (!fp->nr_rates) {
                         hwc_debug("All rates were zero. Skipping format!\n");
                         return -1;
                 }
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 320641ab5be7..26bad373fe65 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
         }
 
         ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+        ep_info.out_interval = 0;
         ep_info.out_cables = endpoint->out_cables & 0x5555;
         err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
         if (err < 0)