Merge branch 'fix/opl3sa2-suspend' into topic/isa-misc

38 files changed, 257 insertions, 160 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 89096e811a4b..772901e41ecb 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
          */
         do {
                 v = readl(aaci->base + AACI_SLFR);
-        } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--);
+        } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
 
         if (!timeout)
                 dev_err(&aaci->dev->dev,
@@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
          */
         do {
                 v = readl(aaci->base + AACI_SLFR);
-        } while ((v & SLFR_1TXB) && timeout--);
+        } while ((v & SLFR_1TXB) && --timeout);
 
         if (!timeout) {
                 dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
@@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
         do {
                 cond_resched();
                 v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
-        } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--);
+        } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
 
         if (!timeout) {
                 dev_err(&aaci->dev->dev, "timeout on RX valid\n");
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/pcm_oss.c b/sound/core/oss/pcm_oss.c
index e17836680f49..0a1798eafb0b 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1767,7 +1767,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
                        AFMT_S8 | AFMT_U16_LE |
                        AFMT_U16_BE |
                         AFMT_S32_LE | AFMT_S32_BE |
-                        AFMT_S24_LE | AFMT_S24_LE |
+                        AFMT_S24_LE | AFMT_S24_BE |
                         AFMT_S24_PACKED;
         params = kmalloc(sizeof(*params), GFP_KERNEL);
         if (!params)
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/drivers/mtpav.c b/sound/drivers/mtpav.c
index 5b89c0883d60..48b64e6b2670 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -706,7 +706,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
         mtp_card->card = card;
         mtp_card->irq = -1;
         mtp_card->share_irq = 0;
-        mtp_card->inmidiport = 0xffffffff;
         mtp_card->inmidistate = 0;
         mtp_card->outmidihwport = 0xffffffff;
         init_timer(&mtp_card->timer);
@@ -719,6 +718,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
         if (err < 0)
                 goto __error;
 
+        mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST;
+
         err = snd_mtpav_get_ISA(mtp_card);
         if (err < 0)
                 goto __error;
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 19b2d0420a26..3f0b877bc8b5 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -554,21 +554,27 @@ static int __devinit snd_opl3sa2_mixer(struct snd_card *card)
 #ifdef CONFIG_PM
 static int snd_opl3sa2_suspend(struct snd_card *card, pm_message_t state)
 {
-        struct snd_opl3sa2 *chip = card->private_data;
+        if (card) {
+                struct snd_opl3sa2 *chip = card->private_data;
 
-        snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
-        chip->wss->suspend(chip->wss);
-        /* power down */
-        snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3);
+                snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+                chip->wss->suspend(chip->wss);
+                /* power down */
+                snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3);
+        }
 
         return 0;
 }
 
 static int snd_opl3sa2_resume(struct snd_card *card)
 {
-        struct snd_opl3sa2 *chip = card->private_data;
+        struct snd_opl3sa2 *chip;
         int i;
 
+        if (!card)
+                return 0;
+
+        chip = card->private_data;
         /* power up */
         snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0);
 
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/Kconfig b/sound/pci/Kconfig
index 6e3a1848447c..82b9bddcdcd6 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -744,8 +744,8 @@ config SND_VIRTUOSO
         select SND_OXYGEN_LIB
         help
           Say Y here to include support for sound cards based on the
-          Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and
-          HDAV1.3 (Deluxe).
+          Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
+          Support for the HDAV1.3 (Deluxe) is very experimental.
 
           To compile this driver as a module, choose M here: the module
           will be called snd-virtuoso.
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_codec.c b/sound/pci/hda/hda_codec.c
index b7bba7dc7cf1..d03f99298be9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -487,7 +487,6 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
 {
         struct hda_bus *bus;
         int err;
-        char qname[8];
         static struct snd_device_ops dev_ops = {
                 .dev_register = snd_hda_bus_dev_register,
                 .dev_free = snd_hda_bus_dev_free,
@@ -517,10 +516,12 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
         mutex_init(&bus->cmd_mutex);
         INIT_LIST_HEAD(&bus->codec_list);
 
-        snprintf(qname, sizeof(qname), "hda%d", card->number);
-        bus->workq = create_workqueue(qname);
+        snprintf(bus->workq_name, sizeof(bus->workq_name),
+                 "hd-audio%d", card->number);
+        bus->workq = create_singlethread_workqueue(bus->workq_name);
         if (!bus->workq) {
-                snd_printk(KERN_ERR "cannot create workqueue %s\n", qname);
+                snd_printk(KERN_ERR "cannot create workqueue %s\n",
+                           bus->workq_name);
                 kfree(bus);
                 return -ENOMEM;
         }
@@ -3087,6 +3088,16 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare);
 
+int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
+                                  struct hda_multi_out *mout)
+{
+        mutex_lock(&codec->spdif_mutex);
+        cleanup_dig_out_stream(codec, mout->dig_out_nid);
+        mutex_unlock(&codec->spdif_mutex);
+        return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup);
+
 /*
  * release the digital out
  */
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 5810ef588402..09a332ada0c6 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -614,6 +614,7 @@ struct hda_bus {
 
         /* unsolicited event queue */
         struct hda_bus_unsolicited *unsol;
+        char workq_name[16];
         struct workqueue_struct *workq; /* common workqueue for codecs */
 
         /* assigned PCMs */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 300ab407cf42..4ae51dcb81af 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -175,7 +175,7 @@ static int reconfig_codec(struct hda_codec *codec)
         err = snd_hda_codec_build_controls(codec);
         if (err < 0)
                 return err;
-        return 0;
+        return snd_card_register(codec->bus->card);
 }
 
 /*
@@ -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/hda_local.h b/sound/pci/hda/hda_local.h
index 1dd8716c387f..44f189cb97ae 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -251,6 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
                                   unsigned int stream_tag,
                                   unsigned int format,
                                   struct snd_pcm_substream *substream);
+int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
+                                  struct hda_multi_out *mout);
 int snd_hda_multi_out_analog_open(struct hda_codec *codec,
                                   struct hda_multi_out *mout,
                                   struct snd_pcm_substream *substream,
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 7ca66d654148..144b85276d5a 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -399,7 +399,8 @@ static void print_conn_list(struct snd_info_buffer *buffer,
 {
         int c, curr = -1;
 
-        if (conn_len > 1 && wid_type != AC_WID_AUD_MIX)
+        if (conn_len > 1 && wid_type != AC_WID_AUD_MIX &&
+            wid_type != AC_WID_VOL_KNB)
                 curr = snd_hda_codec_read(codec, nid, 0,
                                           AC_VERB_GET_CONNECT_SEL, 0);
         snd_iprintf(buffer, "  Connection: %d\n", conn_len);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2e7371ec2e23..e48612323aa0 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -275,6 +275,14 @@ static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
                                              format, substream);
 }
 
+static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+                                           struct hda_codec *codec,
+                                           struct snd_pcm_substream *substream)
+{
+        struct ad198x_spec *spec = codec->spec;
+        return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 /*
  * Analog capture
  */
@@ -333,7 +341,8 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = {
         .ops = {
                 .open = ad198x_dig_playback_pcm_open,
                 .close = ad198x_dig_playback_pcm_close,
-                .prepare = ad198x_dig_playback_pcm_prepare
+                .prepare = ad198x_dig_playback_pcm_prepare,
+                .cleanup = ad198x_dig_playback_pcm_cleanup
         },
 };
 
@@ -1885,8 +1894,8 @@ static hda_nid_t ad1988_capsrc_nids[3] = {
 #define AD1988_SPDIF_OUT_HDMI   0x0b
 #define AD1988_SPDIF_IN         0x07
 
-static hda_nid_t ad1989b_slave_dig_outs[2] = {
-        AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI
+static hda_nid_t ad1989b_slave_dig_outs[] = {
+        AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
 };
 
 static struct hda_input_mux ad1988_6stack_capture_source = {
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 75de40aaab0a..0177ef8f4c9e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -347,6 +347,7 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol,
                                      &spec->cur_mux[adc_idx]);
 }
 
+#ifdef CONFIG_SND_JACK
 static int conexant_add_jack(struct hda_codec *codec,
                 hda_nid_t nid, int type)
 {
@@ -394,7 +395,6 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid)
 
 static int conexant_init_jacks(struct hda_codec *codec)
 {
-#ifdef CONFIG_SND_JACK
         struct conexant_spec *spec = codec->spec;
         int i;
 
@@ -422,10 +422,19 @@ static int conexant_init_jacks(struct hda_codec *codec)
                         ++hv;
                 }
         }
-#endif
         return 0;
 
 }
+#else
+static inline void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid)
+{
+}
+
+static inline int conexant_init_jacks(struct hda_codec *codec)
+{
+        return 0;
+}
+#endif
 
 static int conexant_init(struct hda_codec *codec)
 {
@@ -1566,6 +1575,7 @@ static struct snd_pci_quirk cxt5047_cfg_tbl[] = {
         SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP),
         SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP),
         SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP),
+        SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP),
         SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD),
         {}
 };
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 3564f4e4b74c..fcc77fec4487 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -49,11 +49,6 @@ static struct hda_verb pinout_enable_verb[] = {
         {} /* terminator */
 };
 
-static struct hda_verb pinout_disable_verb[] = {
-        {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00},
-        {}
-};
-
 static struct hda_verb unsolicited_response_verb[] = {
         {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN |
                                                   INTEL_HDMI_EVENT_TAG},
@@ -248,10 +243,6 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
 
 static void hdmi_enable_output(struct hda_codec *codec)
 {
-        /* Enable Audio InfoFrame Transmission */
-        hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
-        snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
-                                                AC_DIPXMIT_BEST);
         /* Unmute */
         if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
                 snd_hda_codec_write(codec, PIN_NID, 0,
@@ -260,17 +251,24 @@ static void hdmi_enable_output(struct hda_codec *codec)
         snd_hda_sequence_write(codec, pinout_enable_verb);
 }
 
-static void hdmi_disable_output(struct hda_codec *codec)
+/*
+ * Enable Audio InfoFrame Transmission
+ */
+static void hdmi_start_infoframe_trans(struct hda_codec *codec)
 {
-        snd_hda_sequence_write(codec, pinout_disable_verb);
-        if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
-                snd_hda_codec_write(codec, PIN_NID, 0,
-                                AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+        hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+        snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+                                                AC_DIPXMIT_BEST);
+}
 
-        /*
-         * FIXME: noises may arise when playing music after reloading the
-         * kernel module, until the next X restart or monitor repower.
-         */
+/*
+ * Disable Audio InfoFrame Transmission
+ */
+static void hdmi_stop_infoframe_trans(struct hda_codec *codec)
+{
+        hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+        snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+                                                AC_DIPXMIT_DISABLE);
 }
 
 static int hdmi_get_channel_count(struct hda_codec *codec)
@@ -368,11 +366,16 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
                                         struct hdmi_audio_infoframe *ai)
 {
         u8 *params = (u8 *)ai;
+        u8 sum = 0;
         int i;
 
         hdmi_debug_dip_size(codec);
         hdmi_clear_dip_buffers(codec); /* be paranoid */
 
+        for (i = 0; i < sizeof(ai); i++)
+                sum += params[i];
+        ai->checksum = - sum;
+
         hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
         for (i = 0; i < sizeof(ai); i++)
                 hdmi_write_dip_byte(codec, PIN_NID, params[i]);
@@ -419,14 +422,18 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec,
         /*
          * CA defaults to 0 for basic stereo audio
          */
-        if (!eld->eld_ver)
-                return 0;
-        if (!eld->spk_alloc)
-                return 0;
         if (channels <= 2)
                 return 0;
 
         /*
+         * HDMI sink's ELD info cannot always be retrieved for now, e.g.
+         * in console or for audio devices. Assume the highest speakers
+         * configuration, to _not_ prohibit multi-channel audio playback.
+         */
+        if (!eld->spk_alloc)
+                eld->spk_alloc = 0xffff;
+
+        /*
          * expand ELD's speaker allocation mask
          *
          * ELD tells the speaker mask in a compact(paired) form,
@@ -485,6 +492,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
         hdmi_setup_channel_mapping(codec, &ai);
 
         hdmi_fill_audio_infoframe(codec, &ai);
+        hdmi_start_infoframe_trans(codec);
 }
 
 
@@ -562,7 +570,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo,
 {
         struct intel_hdmi_spec *spec = codec->spec;
 
-        hdmi_disable_output(codec);
+        hdmi_stop_infoframe_trans(codec);
 
         return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
@@ -582,8 +590,6 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 
         hdmi_setup_audio_infoframe(codec, substream);
 
-        hdmi_enable_output(codec);
-
         return 0;
 }
 
@@ -628,8 +634,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec)
 
 static int intel_hdmi_init(struct hda_codec *codec)
 {
-        /* disable audio output as early as possible */
-        hdmi_disable_output(codec);
+        hdmi_enable_output(codec);
 
         snd_hda_sequence_write(codec, unsolicited_response_verb);
 
@@ -679,6 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
         { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
         { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
         { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
+        { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
         { .id = 0x10951392, .name = "SiI1392 HDMI",     .patch = patch_intel_hdmi },
         {} /* terminator */
 };
@@ -687,6 +693,7 @@ MODULE_ALIAS("snd-hda-codec-id:808629fb");
 MODULE_ALIAS("snd-hda-codec-id:80862801");
 MODULE_ALIAS("snd-hda-codec-id:80862802");
 MODULE_ALIAS("snd-hda-codec-id:80862803");
+MODULE_ALIAS("snd-hda-codec-id:80862804");
 MODULE_ALIAS("snd-hda-codec-id:10951392");
 
 MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 82dd08431970..6c26afcb8262 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1037,6 +1037,7 @@ do_sku:
                 case 0x10ec0267:
                 case 0x10ec0268:
                 case 0x10ec0269:
+                case 0x10ec0272:
                 case 0x10ec0660:
                 case 0x10ec0662:
                 case 0x10ec0663:
@@ -1065,6 +1066,7 @@ do_sku:
                 case 0x10ec0882:
                 case 0x10ec0883:
                 case 0x10ec0885:
+                case 0x10ec0887:
                 case 0x10ec0889:
                         snd_hda_codec_write(codec, 0x20, 0,
                                             AC_VERB_SET_COEF_INDEX, 7);
@@ -7012,12 +7014,15 @@ static int patch_alc882(struct hda_codec *codec)
                         break;
                 case 0x106b1000: /* iMac 24 */
                 case 0x106b2800: /* AppleTV */
+                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 */
                 case 0x106b3600: /* Macbook 3.1 */
+                case 0x106b3800: /* MacbookPro4,1 - latter revision */
                         board_config = ALC885_MBP3;
                         break;
                 default:
@@ -8465,6 +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),
@@ -8474,10 +8481,12 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
         SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
         SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
         SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+        SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
         SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
         SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
+        SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
         SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
         SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
         SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
@@ -8512,6 +8521,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
         SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
         SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
+        SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550",
+                      ALC883_FUJITSU_PI2515),
         SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515),
         SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530",
                 ALC888_FUJITSU_XA3530),
@@ -8526,6 +8537,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
         SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
         SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
+        SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
         SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL),
         SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
         {}
@@ -10545,6 +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 c39deebb588f..6094344fb223 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -81,6 +81,7 @@ enum {
 
 enum {
         STAC_92HD83XXX_REF,
+        STAC_92HD83XXX_PWR_REF,
         STAC_92HD83XXX_MODELS
 };
 
@@ -334,7 +335,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
 };
 
 static unsigned int stac92hd83xxx_pwr_mapping[4] = {
-        0x03, 0x0c, 0x10, 0x40,
+        0x03, 0x0c, 0x20, 0x40,
 };
 
 static hda_nid_t stac92hd83xxx_amp_nids[1] = {
@@ -841,10 +842,6 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = {
 };
 
 static struct hda_verb stac92hd83xxx_core_init[] = {
-        /* start of config #1 */
-        { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3},
-
-        /* start of config #2 */
         { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
         { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
         { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
@@ -885,8 +882,8 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 static struct hda_verb stac925x_core_init[] = {
         /* set dac0mux for dac converter */
         { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00},
-        /* unmute and set max the selector */
-        { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f },
+        /* mute the master volume */
+        { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
         {}
 };
 
@@ -1138,6 +1135,8 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac925x_mixer[] = {
+        HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
         STAC_INPUT_SOURCE(1),
         HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
         HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
@@ -1208,7 +1207,7 @@ static const char *slave_vols[] = {
         "LFE Playback Volume",
         "Side Playback Volume",
         "Headphone Playback Volume",
-        "Headphone Playback Volume",
+        "Headphone2 Playback Volume",
         "Speaker Playback Volume",
         "External Speaker Playback Volume",
         "Speaker2 Playback Volume",
@@ -1222,7 +1221,7 @@ static const char *slave_sws[] = {
         "LFE Playback Switch",
         "Side Playback Switch",
         "Headphone Playback Switch",
-        "Headphone Playback Switch",
+        "Headphone2 Playback Switch",
         "Speaker Playback Switch",
         "External Speaker Playback Switch",
         "Speaker2 Playback Switch",
@@ -1736,10 +1735,12 @@ static unsigned int ref92hd83xxx_pin_configs[14] = {
 
 static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
         [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
+        [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
 };
 
 static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
         [STAC_92HD83XXX_REF] = "ref",
+        [STAC_92HD83XXX_PWR_REF] = "mic-ref",
 };
 
 static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1798,9 +1799,13 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
         SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
                       "HP dv5", STAC_HP_M4),
         SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
-                      "HP dv7", STAC_HP_M4),
+                      "HP dv7", STAC_HP_DV5),
+        SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
+                      "HP dv4", STAC_HP_DV5),
         SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
                       "HP dv7", STAC_HP_M4),
+        SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600,
+                      "HP dv5", STAC_HP_DV5),
         SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603,
                       "HP dv5", STAC_HP_DV5),
         SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
@@ -2437,6 +2442,14 @@ static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
                                              stream_tag, format, substream);
 }
 
+static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+                                        struct hda_codec *codec,
+                                        struct snd_pcm_substream *substream)
+{
+        struct sigmatel_spec *spec = codec->spec;
+        return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 
 /*
  * Analog capture callbacks
@@ -2481,7 +2494,8 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = {
         .ops = {
                 .open = stac92xx_dig_playback_pcm_open,
                 .close = stac92xx_dig_playback_pcm_close,
-                .prepare = stac92xx_dig_playback_pcm_prepare
+                .prepare = stac92xx_dig_playback_pcm_prepare,
+                .cleanup = stac92xx_dig_playback_pcm_cleanup
         },
 };
 
@@ -2536,6 +2550,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 
         info->name = "STAC92xx Analog";
         info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback;
+        info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+                spec->multiout.dac_nids[0];
         info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture;
         info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
         info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs;
@@ -3500,6 +3516,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
         if (! spec->autocfg.line_outs)
                 return 0; /* can't find valid pin config */
 
+#if 0 /* FIXME: temporarily disabled */
         /* If we have no real line-out pin and multiple hp-outs, HPs should
          * be set up as multi-channel outputs.
          */
@@ -3519,6 +3536,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
                 spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
                 spec->autocfg.hp_outs = 0;
         }
+#endif /* FIXME: temporarily disabled */
         if (spec->autocfg.mono_out_pin) {
                 int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
                         (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
@@ -3573,13 +3591,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
                 err = stac92xx_auto_fill_dac_nids(codec);
                 if (err < 0)
                         return err;
+                err = stac92xx_auto_create_multi_out_ctls(codec,
+                                                          &spec->autocfg);
+                if (err < 0)
+                        return err;
         }
 
-        err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg);
-
-        if (err < 0)
-                return err;
-
         /* setup analog beep controls */
         if (spec->anabeep_nid > 0) {
                 err = stac92xx_auto_create_beep_ctls(codec,
@@ -4753,7 +4770,9 @@ static struct hda_input_mux stac92hd83xxx_dmux = {
 static int patch_stac92hd83xxx(struct hda_codec *codec)
 {
         struct sigmatel_spec *spec;
+        hda_nid_t conn[STAC92HD83_DAC_COUNT + 1];
         int err;
+        int num_dacs;
 
         spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
         if (spec == NULL)
@@ -4772,15 +4791,16 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
         spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
         spec->multiout.dac_nids = spec->dac_nids;
 
-        spec->init = stac92hd83xxx_core_init;
-        switch (codec->vendor_id) {
-        case 0x111d7605:
-                break;
-        default:
-                spec->num_pwrs--;
-                spec->init++; /* switch to config #2 */
-        }
 
+        /* set port 0xe to select the last DAC
+         */
+        num_dacs = snd_hda_get_connections(codec, 0x0e,
+                conn, STAC92HD83_DAC_COUNT + 1) - 1;
+
+        snd_hda_codec_write_cache(codec, 0xe, 0,
+                AC_VERB_SET_CONNECT_SEL, num_dacs);
+
+        spec->init = stac92hd83xxx_core_init;
         spec->mixer = stac92hd83xxx_mixer;
         spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
         spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids);
@@ -4806,6 +4826,15 @@ again:
                 return err;
         }
 
+        switch (codec->vendor_id) {
+        case 0x111d7604:
+        case 0x111d7605:
+                if (spec->board_config == STAC_92HD83XXX_PWR_REF)
+                        break;
+                spec->num_pwrs = 0;
+                break;
+        }
+
         err = stac92xx_parse_auto_config(codec, 0x1d, 0);
         if (!err) {
                 if (spec->board_config < 0) {
@@ -4962,7 +4991,7 @@ 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/intel8x0.c b/sound/pci/intel8x0.c
index 19d3391e229f..e900cdc84849 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -617,7 +617,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip)
         int time = 100;
         if (chip->buggy_semaphore)
                 return 0; /* just ignore ... */
-        while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY))
+        while (--time && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY))
                 udelay(1);
         if (! time && ! chip->in_ac97_init)
                 snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n");
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index e9e829e83d7a..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,
@@ -683,7 +676,7 @@ static void xonar_hdav_uart_input(struct oxygen *chip)
         if (chip->uart_input_count >= 2 &&
             chip->uart_input[chip->uart_input_count - 2] == 'O' &&
             chip->uart_input[chip->uart_input_count - 1] == 'K') {
-                printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:");
+                printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n");
                 print_hex_dump_bytes("", DUMP_PREFIX_OFFSET,
                                      chip->uart_input, chip->uart_input_count);
                 chip->uart_input_count = 0;
@@ -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,
 };
 
@@ -908,6 +899,7 @@ static const struct oxygen_model model_xonar_hdav = {
         .dac_channels = 8,
         .dac_volume_min = 0x0f,
         .dac_volume_max = 0xff,
+        .misc_flags = OXYGEN_MISC_MIDI,
         .function_flags = OXYGEN_FUNCTION_2WIRE,
         .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/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 1fac5efd285b..3dcdc4e3cfa0 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -44,8 +44,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <mach/hardware.h>
-
 #include "atmel-pcm.h"
 
 
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index c5d67900d666..ff0054b76502 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -10,7 +10,7 @@
  * Based on at91-ssc.c by
  * Frank Mandarino <fmandarino@endrelia.com>
  * Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index a828746e8a2f..391135f9c6c1 100644
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -10,7 +10,7 @@
  * Based on at91-ssc.c by
  * Frank Mandarino <fmandarino@endrelia.com>
  * Based on pxa2xx Platform drivers by
- * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Liam Girdwood <lrg@slimlogic.co.uk>
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b47a749c5ea2..aea0cb72d80a 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
                                         struct snd_ctl_elem_value *ucontrol)
 {
         struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
-        int reg = kcontrol->private_value & 0xff;
-        int shift = (kcontrol->private_value >> 8) & 0x0f;
-        int mask = (kcontrol->private_value >> 16) & 0xff;
-        int invert = (kcontrol->private_value >> 24) & 0x01;
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
+        unsigned int reg = mc->reg;
+        unsigned int shift = mc->shift;
+        int max = mc->max;
+        unsigned int mask = (1 << fls(max)) - 1;
+        unsigned int invert = mc->invert;
         unsigned short val, val_mask;
         int ret;
         struct snd_soc_dapm_path *path;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index e3989d406f54..35d99750c383 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
  *
- * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com>
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 6c21b50c9375..77620ab98756 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1451,7 +1451,14 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 },
 };
 
-struct snd_soc_dai wm8753_dai[2];
+struct snd_soc_dai wm8753_dai[] = {
+        {
+                .name = "WM8753 DAI 0",
+        },
+        {
+                .name = "WM8753 DAI 1",
+        },
+};
 EXPORT_SYMBOL_GPL(wm8753_dai);
 
 static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5b5afc144478..a5731faa150c 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -2,8 +2,7 @@
  * wm8990.c  --  WM8990 ALSA Soc Audio driver
  *
  * Copyright 2008 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -177,7 +176,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol)
 {
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-        int reg = kcontrol->private_value & 0xff;
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
+        int reg = mc->reg;
         int ret;
         u16 val;
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index bcec3f60bad9..acf39a646b2f 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -183,16 +183,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 };
 
 /**
- * mpc8610_hpcd_machine: ASoC machine data
- */
-static struct snd_soc_card mpc8610_hpcd_machine = {
-        .probe = mpc8610_hpcd_machine_probe,
-        .remove = mpc8610_hpcd_machine_remove,
-        .name = "MPC8610 HPCD",
-        .num_links = 1,
-};
-
-/**
  * mpc8610_hpcd_probe: OF probe function for the fabric driver
  *
  * This function gets called when an SSI node is found in the device tree.
@@ -455,7 +445,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
         machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */
         machine_data->dai.ops = &mpc8610_hpcd_ops;
 
-        mpc8610_hpcd_machine.dai_link = &machine_data->dai;
+        machine_data->machine.probe = mpc8610_hpcd_machine_probe;
+        machine_data->machine.remove = mpc8610_hpcd_machine_remove;
+        machine_data->machine.name = "MPC8610 HPCD";
+        machine_data->machine.num_links = 1;
+        machine_data->machine.dai_link = &machine_data->dai;
 
         /* Allocate a new audio platform device structure */
         sound_device = platform_device_alloc("soc-audio", -1);
@@ -465,7 +459,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
                 goto error;
         }
 
-        machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
+        machine_data->sound_devdata.card = &machine_data->machine;
         machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
         machine_data->machine.platform = &fsl_soc_platform;
 
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index ec5e18a78758..05dd5abcddf4 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -302,6 +302,10 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
         regs->spcr1     |= RINTM(3);
         regs->rcr2      |= RFIG;
         regs->xcr2      |= XFIG;
+        if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+                regs->xccr = DXENDLY(1) | XDMAEN;
+                regs->rccr = RFULL_CYCLE | RDMAEN;
+        }
 
         switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
         case SND_SOC_DAIFMT_I2S:
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index b0362dfd5b71..dd3bb2933762 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -175,9 +175,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct omap_runtime_data *prtd = runtime->private_data;
+        unsigned long flags;
         int ret = 0;
 
-        spin_lock_irq(&prtd->lock);
+        spin_lock_irqsave(&prtd->lock, flags);
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_RESUME:
@@ -195,7 +196,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
         default:
                 ret = -EINVAL;
         }
-        spin_unlock_irq(&prtd->lock);
+        spin_unlock_irqrestore(&prtd->lock, flags);
 
         return ret;
 }
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index ad97836818b1..e226fa75669c 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -91,7 +91,7 @@ static struct snd_soc_dai_link sdp3430_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+static struct snd_soc_card snd_soc_sdp3430 = {
         .name = "SDP3430",
         .platform = &omap_soc_platform,
         .dai_link = &sdp3430_dai,
@@ -100,7 +100,7 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = {
 
 /* Audio subsystem */
 static struct snd_soc_device sdp3430_snd_devdata = {
-        .machine = &snd_soc_machine_sdp3430,
+        .card = &snd_soc_sdp3430,
         .codec_dev = &soc_codec_dev_twl4030,
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 55fdb4abb179..ec3f8bb4b51d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1385,7 +1385,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
 
         mutex_lock(&codec->mutex);
 #ifdef CONFIG_SND_SOC_AC97_BUS
-        if (ac97) {
+        /* Only instantiate AC97 if not already done by the adaptor
+         * for the generic AC97 subsystem.
+         */
+        if (ac97 && strcmp(codec->name, "AC97") != 0) {
                 ret = soc_ac97_dev_register(codec);
                 if (ret < 0) {
                         printk(KERN_ERR "asoc: AC97 device register failed\n");
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c709b9563226..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;
                 }
@@ -2966,6 +2968,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
                 return -EINVAL;
         }
         alts = &iface->altsetting[fp->altset_idx];
+        fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
         usb_set_interface(chip->dev, fp->iface, 0);
         init_usb_pitch(chip->dev, fp->iface, alts, fp);
         init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max);
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)