32 files changed, 105 insertions, 73 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/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 4690b8b5681f..e570649184e2 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -692,6 +692,9 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer,
                 snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
                 if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME)
                         snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
+        } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) {
+                snd_mixer_oss_put_volume1_vol(fmixer, pslot,
+                        slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
         } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
                 snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
         } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) {
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index e17836680f49..699d2890535c 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)
@@ -2872,7 +2872,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
                         setup = kmalloc(sizeof(*setup), GFP_KERNEL);
                         if (! setup) {
                                 buffer->error = -ENOMEM;
-                                mutex_lock(&pstr->oss.setup_mutex);
+                                mutex_unlock(&pstr->oss.setup_mutex);
                                 return;
                         }
                         if (pstr->oss.setup_list == NULL)
@@ -2886,7 +2886,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
                         if (! template.task_name) {
                                 kfree(setup);
                                 buffer->error = -ENOMEM;
-                                mutex_lock(&pstr->oss.setup_mutex);
+                                mutex_unlock(&pstr->oss.setup_mutex);
                                 return;
                         }
                 }
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/core/sgbuf.c b/sound/core/sgbuf.c
index d4564edd61d7..4e7ec2b49873 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -38,6 +38,10 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
         if (! sgbuf)
                 return -EINVAL;
 
+        if (dmab->area)
+                vunmap(dmab->area);
+        dmab->area = NULL;
+
         tmpb.dev.type = SNDRV_DMA_TYPE_DEV;
         tmpb.dev.dev = sgbuf->dev;
         for (i = 0; i < sgbuf->pages; i++) {
@@ -48,9 +52,6 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
                 tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT;
                 snd_dma_free_pages(&tmpb);
         }
-        if (dmab->area)
-                vunmap(dmab->area);
-        dmab->area = NULL;
 
         kfree(sgbuf->table);
         kfree(sgbuf->page_table);
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/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/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/mixart/mixart.c b/sound/pci/mixart/mixart.c
index f23a73577c22..bb162507fe6c 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -607,6 +607,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs,
         /* set the format to the board */
         err = mixart_set_format(stream, format);
         if(err < 0) {
+                mutex_unlock(&mgr->setup_mutex);
                 return err;
         }
 
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 98c6a8c65d81..6c870c12a177 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -676,7 +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;
@@ -899,6 +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/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 8f9e3859c37c..ff321110ec02 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -477,7 +477,7 @@ static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream)
                 card->dma_start_bus_addr[SND_PS3_CH_R] =
                         runtime->dma_addr + (runtime->dma_bytes / 2);
 
-                pr_debug("%s: vaddr=%p bus=%#lx\n", __func__,
+                pr_debug("%s: vaddr=%p bus=%#llx\n", __func__,
                          card->dma_start_vaddr[SND_PS3_CH_L],
                          card->dma_start_bus_addr[SND_PS3_CH_L]);
 
@@ -1030,7 +1030,7 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
                 pr_info("%s: nullbuffer alloc failed\n", __func__);
                 goto clean_preallocate;
         }
-        pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__,
+        pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__,
                  the_card.null_buffer_start_vaddr,
                  the_card.null_buffer_start_dma_addr);
         /* set default sample rate/word width */
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/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 493a4e8aa273..a2f1da8b4646 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -720,7 +720,8 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
         struct snd_soc_dapm_path *path;
         int found = 0;
 
-        if (widget->id != snd_soc_dapm_mux)
+        if (widget->id != snd_soc_dapm_mux &&
+            widget->id != snd_soc_dapm_value_mux)
                 return -ENODEV;
 
         if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
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)
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 92115755d98e..5d8ef09b9dcc 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -128,6 +128,14 @@
         .bInterfaceClass = USB_CLASS_AUDIO,
         .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
 },
+{
+        USB_DEVICE(0x046d, 0x0990),
+        .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+                .vendor_name = "Logitech, Inc.",
+                .product_name = "QuickCam Pro 9000",
+                .ifnum = QUIRK_NO_INTERFACE
+        }
+},
 
 /*
  * Yamaha devices