25 files changed, 224 insertions, 117 deletions

diff --git a/sound/core/sound.c b/sound/core/sound.c
index 7872a02f6ca9..563d1967a0ad 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void)
         unregister_chrdev(major, "alsa");
 }
 
-module_init(alsa_sound_init)
-module_exit(alsa_sound_exit)
+subsys_initcall(alsa_sound_init);
+module_exit(alsa_sound_exit);
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 7fe12264ff80..0c164e5e4322 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
         default:
                 return -EINVAL;
         }
-        if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+        if (minor < 0 || minor >= SNDRV_OSS_MINORS)
                 return -EINVAL;
         return minor;
 }
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 751762f1c59a..0c40951b6523 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -377,12 +377,13 @@ init_arrays(struct snd_emu8000 *emu)
 static void __devinit
 size_dram(struct snd_emu8000 *emu)
 {
-        int i, size;
+        int i, size, detected_size;
 
         if (emu->dram_checked)
                 return;
 
         size = 0;
+        detected_size = 0;
 
         /* write out a magic number */
         snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE);
@@ -393,6 +394,8 @@ size_dram(struct snd_emu8000 *emu)
 
         while (size < EMU8000_MAX_DRAM) {
 
+                size += 512 * 1024;  /* increment 512kbytes */
+
                 /* Write a unique data on the test address.
                  * if the address is out of range, the data is written on
                  * 0x200000(=EMU8000_DRAM_OFFSET).  Then the id word is
@@ -414,7 +417,7 @@ size_dram(struct snd_emu8000 *emu)
                 if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2)
                         break; /* no memory at this address */
 
-                size += 512 * 1024;  /* increment 512kbytes */
+                detected_size = size;
 
                 snd_emu8000_read_wait(emu);
 
@@ -442,9 +445,9 @@ size_dram(struct snd_emu8000 *emu)
         snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE);
 
         snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n",
-                    emu->port1, size/1024);
+                    emu->port1, detected_size/1024);
 
-        emu->mem_size = size;
+        emu->mem_size = detected_size;
         emu->dram_checked = 1;
 }
 
diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c
index 08274c995d06..727bdb9ba2dc 100644
--- a/sound/oss/dev_table.c
+++ b/sound/oss/dev_table.c
@@ -67,14 +67,15 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
                 return -(EBUSY);
         }
         d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver)));
-
-        if (sound_nblocks < 1024)
-                sound_nblocks++;
+        sound_nblocks++;
+        if (sound_nblocks >= MAX_MEM_BLOCKS)
+                sound_nblocks = MAX_MEM_BLOCKS - 1;
 
         op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations)));
+        sound_nblocks++;
+        if (sound_nblocks >= MAX_MEM_BLOCKS)
+                sound_nblocks = MAX_MEM_BLOCKS - 1;
 
-        if (sound_nblocks < 1024)
-                sound_nblocks++;
         if (d == NULL || op == NULL) {
                 printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name);
                 sound_unload_audiodev(num);
@@ -128,9 +129,10 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
            until you unload sound! */
            
         op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations)));
+        sound_nblocks++;
+        if (sound_nblocks >= MAX_MEM_BLOCKS)
+                sound_nblocks = MAX_MEM_BLOCKS - 1;
 
-        if (sound_nblocks < 1024)
-                sound_nblocks++;
         if (op == NULL) {
                 printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
                 return -ENOMEM;
diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h
index 55271fbe7f49..9d35c4c65b9b 100644
--- a/sound/oss/sound_config.h
+++ b/sound/oss/sound_config.h
@@ -142,4 +142,6 @@ static inline int translate_mode(struct file *file)
 #define TIMER_ARMED     121234
 #define TIMER_NOT_ARMED 1
 
+#define MAX_MEM_BLOCKS 1024
+
 #endif
diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c
index 61aaedae6b7e..c62530943888 100644
--- a/sound/oss/soundcard.c
+++ b/sound/oss/soundcard.c
@@ -56,7 +56,7 @@
 /*
  * Table for permanently allocated memory (used when unloading the module)
  */
-void *          sound_mem_blocks[1024];
+void *          sound_mem_blocks[MAX_MEM_BLOCKS];
 int             sound_nblocks = 0;
 
 /* Persistent DMA buffers */
@@ -574,7 +574,7 @@ static int __init oss_init(void)
                                       NULL, "%s%d", dev_list[i].name, j);
         }
 
-        if (sound_nblocks >= 1024)
+        if (sound_nblocks >= MAX_MEM_BLOCKS - 1)
                 printk(KERN_ERR "Sound warning: Deallocation table was too small.\n");
         
         return 0;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index c11920623009..a7630e9edf8a 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -83,6 +83,7 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = {
 { 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL },
 { 0x50534300, 0xffffff00, "Philips",            NULL,   NULL },
 { 0x53494c00, 0xffffff00, "Silicon Laboratory", NULL,   NULL },
+{ 0x53544d00, 0xffffff00, "STMicroelectronics", NULL,   NULL },
 { 0x54524100, 0xffffff00, "TriTech",            NULL,   NULL },
 { 0x54584e00, 0xffffff00, "Texas Instruments",  NULL,   NULL },
 { 0x56494100, 0xffffff00, "VIA Technologies",   NULL,   NULL },
@@ -161,6 +162,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x4e534350, 0xffffffff, "LM4550",             patch_lm4550,   NULL }, // volume wrap fix 
 { 0x50534304, 0xffffffff, "UCB1400",            patch_ucb1400,  NULL },
 { 0x53494c20, 0xffffffe0, "Si3036,8",           mpatch_si3036,  mpatch_si3036, AC97_MODEM_PATCH },
+{ 0x53544d02, 0xffffffff, "ST7597",             NULL,           NULL },
 { 0x54524102, 0xffffffff, "TR28022",            NULL,           NULL },
 { 0x54524103, 0xffffffff, "TR28023",            NULL,           NULL },
 { 0x54524106, 0xffffffff, "TR28026",            NULL,           NULL },
@@ -213,6 +215,14 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg)
 {
         /* filter some registers for buggy codecs */
         switch (ac97->id) {
+        case AC97_ID_ST_AC97_ID4:
+                if (reg == 0x08)
+                        return 0;
+                /* fall through */
+        case AC97_ID_ST7597:
+                if (reg == 0x22 || reg == 0x7a)
+                        return 1;
+                /* fall through */
         case AC97_ID_AK4540:
         case AC97_ID_AK4542:
                 if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c)
diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h
index c129492c82b3..d603147c4a96 100644
--- a/sound/pci/ac97/ac97_id.h
+++ b/sound/pci/ac97/ac97_id.h
@@ -62,3 +62,5 @@
 #define AC97_ID_CM9761_78       0x434d4978
 #define AC97_ID_CM9761_82       0x434d4982
 #define AC97_ID_CM9761_83       0x434d4983
+#define AC97_ID_ST7597          0x53544d02
+#define AC97_ID_ST_AC97_ID4     0x53544d04
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 139cf3b2b9d7..d9266bae2849 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1870,6 +1870,7 @@ static unsigned int ad1981_jacks_blacklist[] = {
         0x10140554, /* Thinkpad T42p/R50p */
         0x10140567, /* Thinkpad T43p 2668-G7U */
         0x10140581, /* Thinkpad X41-2527 */
+        0x10280160, /* Dell Dimension 2400 */
         0x104380b0, /* Asus A7V8X-MX */
         0x11790241, /* Toshiba Satellite A-15 S127 */
         0x144dc01a, /* Samsung NP-X20C004/SEG */
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index d6752dff2a44..42b4fbbd8e2b 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -297,6 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = {
 MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
 
 static struct snd_pci_quirk atiixp_quirks[] __devinitdata = {
+        SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0),
         SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0),
         { } /* terminator */
 };
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ec9c348336cc..565de38a3fc7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2332,6 +2332,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
  */
 static struct snd_pci_quirk msi_black_list[] __devinitdata = {
         SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
+        SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */
         {}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c7465053d6bb..da34095c707f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1093,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec)
         snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
 }
 
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+        if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+                return;
+        if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+                snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+                                    on ? 2 : 0);
+}
+
 static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
         unsigned int tmp;
@@ -1110,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
         case ALC_INIT_DEFAULT:
                 switch (codec->vendor_id) {
                 case 0x10ec0260:
-                        snd_hda_codec_write(codec, 0x0f, 0,
-                                            AC_VERB_SET_EAPD_BTLENABLE, 2);
-                        snd_hda_codec_write(codec, 0x10, 0,
-                                            AC_VERB_SET_EAPD_BTLENABLE, 2);
+                        set_eapd(codec, 0x0f, 1);
+                        set_eapd(codec, 0x10, 1);
                         break;
                 case 0x10ec0262:
                 case 0x10ec0267:
                 case 0x10ec0268:
                 case 0x10ec0269:
+                case 0x10ec0270:
                 case 0x10ec0272:
                 case 0x10ec0660:
                 case 0x10ec0662:
                 case 0x10ec0663:
                 case 0x10ec0862:
                 case 0x10ec0889:
-                        snd_hda_codec_write(codec, 0x14, 0,
-                                            AC_VERB_SET_EAPD_BTLENABLE, 2);
-                        snd_hda_codec_write(codec, 0x15, 0,
-                                            AC_VERB_SET_EAPD_BTLENABLE, 2);
+                        set_eapd(codec, 0x14, 1);
+                        set_eapd(codec, 0x15, 1);
                         break;
                 }
                 switch (codec->vendor_id) {
@@ -1230,6 +1237,8 @@ static void alc_init_auto_mic(struct hda_codec *codec)
                         return; /* invalid entry */
                 }
         }
+        if (!ext || !fixed)
+                return;
         if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
                 return; /* no unsol support */
         snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
@@ -1834,10 +1843,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void alc889_power_eapd(struct hda_codec *codec, int power)
 {
-        snd_hda_codec_write(codec, 0x14, 0,
-                            AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
-        snd_hda_codec_write(codec, 0x15, 0,
-                            AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+        set_eapd(codec, 0x14, power);
+        set_eapd(codec, 0x15, power);
 }
 #endif
 
@@ -4812,6 +4819,49 @@ static void fixup_automic_adc(struct hda_codec *codec)
         spec->auto_mic = 0; /* disable auto-mic to be sure */
 }
 
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+        hda_nid_t pin;
+        int i;
+
+        /* search for the input pin; there must be only one */
+        for (i = 0; i < AUTO_PIN_LAST; i++) {
+                if (spec->autocfg.input_pins[i]) {
+                        pin = spec->autocfg.input_pins[i];
+                        break;
+                }
+        }
+        if (!pin)
+                return;
+
+        /* set the default connection to that pin */
+        for (i = 0; i < spec->num_adc_nids; i++) {
+                hda_nid_t cap = spec->capsrc_nids ?
+                        spec->capsrc_nids[i] : spec->adc_nids[i];
+                int idx;
+
+                idx = get_connection_index(codec, cap, pin);
+                if (idx < 0)
+                        continue;
+                /* use only this ADC */
+                if (spec->capsrc_nids)
+                        spec->capsrc_nids += i;
+                spec->adc_nids += i;
+                spec->num_adc_nids = 1;
+                /* select or unmute this route */
+                if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+                        snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+                                                 HDA_AMP_MUTE, 0);
+                } else {
+                        snd_hda_codec_write_cache(codec, cap, 0,
+                                          AC_VERB_SET_CONNECT_SEL, idx);
+                }
+                return;
+        }
+}
+
 static void set_capture_mixer(struct hda_codec *codec)
 {
         struct alc_spec *spec = codec->spec;
@@ -4824,14 +4874,15 @@ static void set_capture_mixer(struct hda_codec *codec)
                   alc_capture_mixer3 },
         };
         if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
-                int mux;
-                if (spec->auto_mic) {
-                        mux = 0;
+                int mux = 0;
+                if (spec->auto_mic)
                         fixup_automic_adc(codec);
-                } else if (spec->input_mux && spec->input_mux->num_items > 1)
-                        mux = 1;
-                else
-                        mux = 0;
+                else if (spec->input_mux) {
+                        if (spec->input_mux->num_items > 1)
+                                mux = 1;
+                        else if (spec->input_mux->num_items == 1)
+                                fixup_single_adc(codec);
+                }
                 spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
         }
 }
@@ -7094,8 +7145,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
         HDA_BIND_MUTE   ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
         HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
         HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
-        HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
-        HDA_BIND_MUTE   ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT),
+        HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+        HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
         HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
         HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
         HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7496,6 +7547,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
         {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
         {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
         {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
         /* Front Mic pin: input vref at 80% */
         {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
         {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -7680,6 +7732,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
         spec->autocfg.speaker_pins[0] = 0x14;
 }
 
+static void alc885_mb5_automute(struct hda_codec *codec)
+{
+        unsigned int present;
+
+        present = snd_hda_codec_read(codec, 0x14, 0,
+                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+        snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+                                 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+        snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+                                 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+
+}
+
+static void alc885_mb5_unsol_event(struct hda_codec *codec,
+                                    unsigned int res)
+{
+        /* Headphone insertion or removal. */
+        if ((res >> 26) == ALC880_HP_EVENT)
+                alc885_mb5_automute(codec);
+}
+
 static void alc885_imac91_automute(struct hda_codec *codec)
 {
         unsigned int present;
@@ -9126,6 +9199,8 @@ static struct alc_config_preset alc882_presets[] = {
                 .input_mux = &mb5_capture_source,
                 .dig_out_nid = ALC882_DIGOUT_NID,
                 .dig_in_nid = ALC882_DIGIN_NID,
+                .unsol_event = alc885_mb5_unsol_event,
+                .init_hook = alc885_mb5_automute,
         },
         [ALC885_MACPRO] = {
                 .mixers = { alc882_macpro_mixer },
@@ -9403,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = {
                 .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
                 .channel_mode = alc883_3ST_6ch_modes,
                 .need_dac_fix = 1,
+                .const_channel_count = 6,
                 .num_mux_defs =
                         ARRAY_SIZE(alc888_2_capture_sources),
                 .input_mux = alc888_2_capture_sources,
@@ -10307,7 +10383,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
         struct alc_spec *spec = codec->spec;
 
         spec->autocfg.hp_pins[0] = 0x15;
-        spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+        spec->autocfg.speaker_pins[0] = 0x14;
 }
 
 static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -11179,7 +11255,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 #define alc262_auto_create_input_ctls \
-        alc880_auto_create_input_ctls
+        alc882_auto_create_input_ctls
 
 /*
  * generic initialization of ADC, input mixers and output mixers
@@ -11718,9 +11794,9 @@ static struct alc_config_preset alc262_presets[] = {
                 .num_channel_mode = ARRAY_SIZE(alc262_modes),
                 .channel_mode = alc262_modes,
                 .input_mux = &alc262_capture_source,
-                .unsol_event = alc_automute_amp_unsol_event,
+                .unsol_event = alc_sku_unsol_event,
                 .setup = alc262_hp_t5735_setup,
-                .init_hook = alc_automute_amp,
+                .init_hook = alc_inithook,
         },
         [ALC262_HP_RP5700] = {
                 .mixers = { alc262_hp_rp5700_mixer },
@@ -12471,6 +12547,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
                 dac = 0x02;
                 break;
         case 0x15:
+        case 0x21:
                 dac = 0x03;
                 break;
         default:
@@ -14855,6 +14932,8 @@ static int patch_alc861(struct hda_codec *codec)
         spec->stream_digital_playback = &alc861_pcm_digital_playback;
         spec->stream_digital_capture = &alc861_pcm_digital_capture;
 
+        if (!spec->cap_mixer)
+                set_capture_mixer(codec);
         set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
 
         spec->vmaster_nid = 0x03;
@@ -15493,7 +15572,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
                                                 const struct auto_pin_cfg *cfg)
 {
-        return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+        return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0);
 }
 
 
@@ -17251,7 +17330,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
         SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
         SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
                       ALC662_3ST_6ch_DIG),
-        SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
+        SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
         SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
         SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
                       ALC662_3ST_6ch_DIG),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2291a8396817..799ba2570902 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4730,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
         }
 }
 
+static int hp_blike_system(u32 subsystem_id);
+
+static void set_hp_led_gpio(struct hda_codec *codec)
+{
+        struct sigmatel_spec *spec = codec->spec;
+        switch (codec->vendor_id) {
+        case 0x111d7608:
+                /* GPIO 0 */
+                spec->gpio_led = 0x01;
+                break;
+        case 0x111d7600:
+        case 0x111d7601:
+        case 0x111d7602:
+        case 0x111d7603:
+                /* GPIO 3 */
+                spec->gpio_led = 0x08;
+                break;
+        }
+}
+
 /*
  * This method searches for the mute LED GPIO configuration
  * provided as OEM string in SMBIOS. The format of that string
@@ -4741,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
  *
  * So, HP B-series like systems may have HP_Mute_LED_0 (current models)
  * or  HP_Mute_LED_0_3 (future models) OEM SMBIOS strings
+ *
+ *
+ * The dv-series laptops don't seem to have the HP_Mute_LED* strings in
+ * SMBIOS - at least the ones I have seen do not have them - which include
+ * my own system (HP Pavilion dv6-1110ax) and my cousin's
+ * HP Pavilion dv9500t CTO.
+ * Need more information on whether it is true across the entire series.
+ * -- kunal
  */
 static int find_mute_led_gpio(struct hda_codec *codec)
 {
@@ -4751,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec)
                 while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
                                                                 NULL, dev))) {
                         if (sscanf(dev->name, "HP_Mute_LED_%d_%d",
-                              &spec->gpio_led_polarity,
-                              &spec->gpio_led) == 2) {
+                                  &spec->gpio_led_polarity,
+                                  &spec->gpio_led) == 2) {
                                 spec->gpio_led = 1 << spec->gpio_led;
                                 return 1;
                         }
                         if (sscanf(dev->name, "HP_Mute_LED_%d",
-                              &spec->gpio_led_polarity) == 1) {
-                                switch (codec->vendor_id) {
-                                case 0x111d7608:
-                                        /* GPIO 0 */
-                                        spec->gpio_led = 0x01;
-                                        return 1;
-                                case 0x111d7600:
-                                case 0x111d7601:
-                                case 0x111d7602:
-                                case 0x111d7603:
-                                        /* GPIO 3 */
-                                        spec->gpio_led = 0x08;
-                                        return 1;
-                                }
+                                  &spec->gpio_led_polarity) == 1) {
+                                set_hp_led_gpio(codec);
+                                return 1;
                         }
                 }
+
+                /*
+                 * Fallback case - if we don't find the DMI strings,
+                 * we statically set the GPIO - if not a B-series system.
+                 */
+                if (!hp_blike_system(codec->subsystem_id)) {
+                        set_hp_led_gpio(codec);
+                        spec->gpio_led_polarity = 1;
+                        return 1;
+                }
         }
         return 0;
 }
@@ -5548,6 +5575,8 @@ again:
         spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
         spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
+        snd_printdd("Found board config: %d\n", spec->board_config);
+
         switch (spec->board_config) {
         case STAC_HP_M4:
                 /* enable internal microphone */
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index b5ca02e2038c..e66ef2b69b5d 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate)
                                  rptr.retwords[2] != M &&
                                  rptr.retwords[3] != N &&
                                  i++ < MAX_WRITE_RETRY);
-                        if (i == MAX_WRITE_RETRY) {
+                        if (i > MAX_WRITE_RETRY) {
                                 snd_printdd("sent samplerate %d: %d failed\n",
                                             *intdec, rate);
                                 return -EIO;
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 69bd0acc81c8..a1bbe16b7f96 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev)
         INIT_LIST_HEAD(&codec->dapm_widgets);
         INIT_LIST_HEAD(&codec->dapm_paths);
 
+        ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+        if (ret < 0) {
+                printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n");
+                goto err;
+        }
+
         /* register pcms */
         ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
         if (ret < 0)
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a9dc5fb54774..da589d8664d0 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
         u16 reg;
 
         /* Sync reg_cache with the hardware */
-        for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
+        for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
                 u16 val = tlv320aic23_read_reg_cache(codec, reg);
                 tlv320aic23_write(codec, reg, val);
         }
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ebbf11b653a4..718ef912e758 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
                 iface |= 0x3 << 8;
                 break;
         case SND_SOC_DAIFMT_DSP_B:
-                iface |= 0x3 << 8;      /* lg not sure which mode */
+                iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
                 break;
         default:
                 return -EINVAL;
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 265e68c75df8..af8cb6995a1f 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -424,23 +424,23 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
 
         /* filter coefficient */
         switch (params_rate(params)) {
-        case SNDRV_PCM_RATE_8000:
+        case 8000:
                 adn |= 0x5 << 1;
                 break;
-        case SNDRV_PCM_RATE_11025:
+        case 11025:
                 adn |= 0x4 << 1;
                 break;
-        case SNDRV_PCM_RATE_16000:
+        case 16000:
                 adn |= 0x3 << 1;
                 break;
-        case SNDRV_PCM_RATE_22050:
+        case 22050:
                 adn |= 0x2 << 1;
                 break;
-        case SNDRV_PCM_RATE_32000:
+        case 32000:
                 adn |= 0x1 << 1;
                 break;
-        case SNDRV_PCM_RATE_44100:
-        case SNDRV_PCM_RATE_48000:
+        case 44100:
+        case 48000:
                 break;
         }
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce5515e3f2b0..3595bd57c4eb 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
         struct i2c_client *i2c = codec->control_data;
         int i;
         u16 *reg_cache = codec->reg_cache;
-        u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+        u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
                                  GFP_KERNEL);
 
         /* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
                 for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
                         if (tmp_cache[i] != reg_cache[i])
                                 snd_soc_write(codec, i, tmp_cache[i]);
+                kfree(tmp_cache);
         } else {
                 dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
         }
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 3d850b97037a..31e39ffd1d8e 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -378,23 +378,23 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
                 iface |= (1 << 9);
 
         switch (params_rate(params)) {
-        case SNDRV_PCM_RATE_8000:
+        case 8000:
                 addcntrl |= (0x5 << 1);
                 break;
-        case SNDRV_PCM_RATE_11025:
+        case 11025:
                 addcntrl |= (0x4 << 1);
                 break;
-        case SNDRV_PCM_RATE_16000:
+        case 16000:
                 addcntrl |= (0x3 << 1);
                 break;
-        case SNDRV_PCM_RATE_22050:
+        case 22050:
                 addcntrl |= (0x2 << 1);
                 break;
-        case SNDRV_PCM_RATE_32000:
+        case 32000:
                 addcntrl |= (0x1 << 1);
                 break;
-        case SNDRV_PCM_RATE_44100:
-        case SNDRV_PCM_RATE_48000:
+        case 44100:
+        case 48000:
                 break;
         }
         ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index a808675388fc..8812751da8c9 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -482,23 +482,23 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
 
         /* filter coefficient */
         switch (params_rate(params)) {
-        case SNDRV_PCM_RATE_8000:
+        case 8000:
                 adn |= 0x5 << 1;
                 break;
-        case SNDRV_PCM_RATE_11025:
+        case 11025:
                 adn |= 0x4 << 1;
                 break;
-        case SNDRV_PCM_RATE_16000:
+        case 16000:
                 adn |= 0x3 << 1;
                 break;
-        case SNDRV_PCM_RATE_22050:
+        case 22050:
                 adn |= 0x2 << 1;
                 break;
-        case SNDRV_PCM_RATE_32000:
+        case 32000:
                 adn |= 0x1 << 1;
                 break;
-        case SNDRV_PCM_RATE_44100:
-        case SNDRV_PCM_RATE_48000:
+        case 44100:
+        case 48000:
                 break;
         }
 
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3db8a6c523f4..19283e5edfbf 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index c7af09729c6e..5263ab18f827 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -42,42 +42,12 @@ static struct snd_soc_device fsi_snd_devdata = {
         .codec_dev      = &soc_codec_dev_ak4642,
 };
 
-#define AK4642_BUS 0
-#define AK4642_ADR 0x12
-static int ak4642_add_i2c_device(void)
-{
-        struct i2c_board_info info;
-        struct i2c_adapter *adapter;
-        struct i2c_client *client;
-
-        memset(&info, 0, sizeof(struct i2c_board_info));
-        info.addr = AK4642_ADR;
-        strlcpy(info.type, "ak4642", I2C_NAME_SIZE);
-
-        adapter = i2c_get_adapter(AK4642_BUS);
-        if (!adapter) {
-                printk(KERN_DEBUG "can't get i2c adapter\n");
-                return -ENODEV;
-        }
-
-        client = i2c_new_device(adapter, &info);
-        i2c_put_adapter(adapter);
-        if (!client) {
-                printk(KERN_DEBUG "can't add i2c device\n");
-                return -ENODEV;
-        }
-
-        return 0;
-}
-
 static struct platform_device *fsi_snd_device;
 
 static int __init fsi_ak4642_init(void)
 {
         int ret = -ENOMEM;
 
-        ak4642_add_i2c_device();
-
         fsi_snd_device = platform_device_alloc("soc-audio", -1);
         if (!fsi_snd_device)
                 goto out;
diff --git a/sound/sound_core.c b/sound/sound_core.c
index dbca7c909a31..7c2d677a2df5 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void)
         class_destroy(sound_class);
 }
 
-module_init(init_soundcore);
+subsys_initcall(init_soundcore);
 module_exit(cleanup_soundcore);
 
 
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 4963defee18a..9edef4684978 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -1936,7 +1936,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
         struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
         struct snd_usb_substream *subs = &as->substream[direction];
 
-        if (subs->interface >= 0) {
+        if (!as->chip->shutdown && subs->interface >= 0) {
                 usb_set_interface(subs->dev, subs->interface, 0);
                 subs->interface = -1;
         }