20 files changed, 186 insertions, 132 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index b2a2db47aff5..4247406160e7 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -28,11 +28,6 @@ config SOUND
           and read <file:Documentation/sound/oss/README.modules>; the module
           will be called soundcore.
 
-          I'm told that even without a sound card, you can make your computer
-          say more than an occasional beep, by programming the PC speaker.
-          Kernel patches and supporting utilities to do that are in the pcsp
-          package, available at <ftp://ftp.infradead.org/pub/pcsp/>.
-
 source "sound/oss/dmasound/Kconfig"
 
 if !M68K
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 812f91b3de5b..6c8ab48c689a 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -259,8 +259,9 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev,
                 return minor;
         }
         snd_minors[minor] = preg;
-        preg->dev = device_create(sound_class, device, MKDEV(major, minor),
-                                  "%s", name);
+        preg->dev = device_create_drvdata(sound_class, device,
+                                          MKDEV(major, minor),
+                                          private_data, "%s", name);
         if (IS_ERR(preg->dev)) {
                 snd_minors[minor] = NULL;
                 mutex_unlock(&sound_mutex);
@@ -269,9 +270,6 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev,
                 return minor;
         }
 
-        if (preg->dev)
-                dev_set_drvdata(preg->dev, private_data);
-
         mutex_unlock(&sound_mutex);
         return 0;
 }
diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
index 379bcb074463..602b58e3b55d 100644
--- a/sound/drivers/Kconfig
+++ b/sound/drivers/Kconfig
@@ -5,9 +5,10 @@ menu "Generic devices"
 
 
 config SND_PCSP
-        tristate "PC-Speaker support"
+        tristate "PC-Speaker support (READ HELP!)"
         depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS
         depends on INPUT
+        depends on EXPERIMENTAL
         depends on SND
         select SND_PCM
         help
@@ -18,11 +19,21 @@ config SND_PCSP
 
           You can compile this as a module which will be called snd-pcsp.
 
+          WARNING: if you already have a soundcard, enabling this
+          driver may lead to a problem. Namely, it may get loaded
+          before the other sound driver of yours, making the
+          pc-speaker a default sound device. Which is likely not
+          what you want. To make this driver play nicely with other
+          sound driver, you can add this into your /etc/modprobe.conf:
+          options snd-pcsp index=2
+
           You don't need this driver if you only want your pc-speaker to beep.
           You don't need this driver if you have a tablet piezo beeper
           in your PC instead of the real speaker.
 
-          It should not hurt to say Y or M here in all other cases.
+          Say N if you have a sound card.
+          Say M if you don't.
+          Say Y only if you really know what you do.
 
 config SND_MPU401_UART
         tristate
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 54a1f9036c66..1899cf0685bc 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -96,7 +96,7 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev)
                 return -EINVAL;
 
         hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
-        pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE;
+        pcsp_chip.timer.cb_mode = HRTIMER_CB_SOFTIRQ;
         pcsp_chip.timer.function = pcsp_do_timer;
 
         card = snd_card_new(index, id, THIS_MODULE, 0);
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index f07cc1ee1fe7..1d661f795e8c 100644
--- a/sound/drivers/pcsp/pcsp.h
+++ b/sound/drivers/pcsp/pcsp.h
@@ -24,7 +24,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 /* default timer freq for PC-Speaker: 18643 Hz */
 #define DIV_18KHZ 64
 #define MAX_DIV DIV_18KHZ
-#define CUR_DIV() (MAX_DIV >> chip->treble)
+#define CALC_DIV(d) (MAX_DIV >> (d))
+#define CUR_DIV() CALC_DIV(chip->treble)
 #define PCSP_MAX_TREBLE 1
 
 /* unfortunately, with hrtimers 37KHz does not work very well :( */
@@ -36,7 +37,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 #define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1)
 #define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV)
 #define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble))
-#define PCSP_RATE() (PIT_TICK_RATE / CUR_DIV())
+#define PCSP_CALC_RATE(i) (PIT_TICK_RATE / CALC_DIV(i))
+#define PCSP_RATE() PCSP_CALC_RATE(chip->treble)
 #define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE
 #define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE
 #define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1)
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index ac6238e93513..e341f3f83b6a 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -9,7 +9,6 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <sound/pcm.h>
-#include <linux/interrupt.h>
 #include <asm/io.h>
 #include "pcsp.h"
 
@@ -18,36 +17,12 @@ module_param(nforce_wa, bool, 0444);
 MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround "
                 "(expect bad sound)");
 
-static void pcsp_start_timer(unsigned long dummy)
-{
-        hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
-}
-
-/*
- * We need the hrtimer_start as a tasklet to avoid
- * the nasty locking problem. :(
- * The problem:
- * - The timer handler is called with the cpu_base->lock
- *   already held by hrtimer code.
- * - snd_pcm_period_elapsed() takes the
- *   substream->self_group.lock.
- * So far so good.
- * But the snd_pcsp_trigger() is called with the
- * substream->self_group.lock held, and it calls
- * hrtimer_start(), which takes the cpu_base->lock.
- * You see the problem. We have the code pathes
- * which take two locks in a reverse order. This
- * can deadlock and the lock validator complains.
- * The only solution I could find was to move the
- * hrtimer_start() into a tasklet. -stsp
- */
-static DECLARE_TASKLET(pcsp_start_timer_tasklet, pcsp_start_timer, 0);
+#define DMIX_WANTS_S16  1
 
 enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 {
-        unsigned long flags;
         unsigned char timer_cnt, val;
-        int periods_elapsed;
+        int fmt_size, periods_elapsed;
         u64 ns;
         size_t period_bytes, buffer_bytes;
         struct snd_pcm_substream *substream;
@@ -64,9 +39,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
                 return HRTIMER_RESTART;
         }
 
-        /* hrtimer calls us from both hardirq and softirq contexts,
-         * so irqsave :( */
-        spin_lock_irqsave(&chip->substream_lock, flags);
+        spin_lock_irq(&chip->substream_lock);
         /* Takashi Iwai says regarding this extra lock:
 
         If the irq handler handles some data on the DMA buffer, it should
@@ -92,8 +65,11 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
                 goto exit_nr_unlock2;
 
         runtime = substream->runtime;
-        /* assume it is u8 mono */
-        val = runtime->dma_area[chip->playback_ptr];
+        fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3;
+        /* assume it is mono! */
+        val = runtime->dma_area[chip->playback_ptr + fmt_size - 1];
+        if (snd_pcm_format_signed(runtime->format))
+                val ^= 0x80;
         timer_cnt = val * CUR_DIV() / 256;
 
         if (timer_cnt && chip->enable) {
@@ -111,12 +87,14 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 
         period_bytes = snd_pcm_lib_period_bytes(substream);
         buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
-        chip->playback_ptr += PCSP_INDEX_INC();
+        chip->playback_ptr += PCSP_INDEX_INC() * fmt_size;
         periods_elapsed = chip->playback_ptr - chip->period_ptr;
         if (periods_elapsed < 0) {
-                printk(KERN_WARNING "PCSP: playback_ptr inconsistent "
+#if PCSP_DEBUG
+                printk(KERN_INFO "PCSP: buffer_bytes mod period_bytes != 0 ? "
                         "(%zi %zi %zi)\n",
                         chip->playback_ptr, period_bytes, buffer_bytes);
+#endif
                 periods_elapsed += buffer_bytes;
         }
         periods_elapsed /= period_bytes;
@@ -132,7 +110,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
                 chip->period_ptr %= buffer_bytes;
         }
 
-        spin_unlock_irqrestore(&chip->substream_lock, flags);
+        spin_unlock_irq(&chip->substream_lock);
 
         if (!atomic_read(&chip->timer_active))
                 return HRTIMER_NORESTART;
@@ -146,7 +124,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 exit_nr_unlock2:
         snd_pcm_stream_unlock(substream);
 exit_nr_unlock1:
-        spin_unlock_irqrestore(&chip->substream_lock, flags);
+        spin_unlock_irq(&chip->substream_lock);
         return HRTIMER_NORESTART;
 }
 
@@ -167,7 +145,7 @@ static void pcsp_start_playing(struct snd_pcsp *chip)
         atomic_set(&chip->timer_active, 1);
         chip->thalf = 0;
 
-        tasklet_schedule(&pcsp_start_timer_tasklet);
+        hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
 }
 
 static void pcsp_stop_playing(struct snd_pcsp *chip)
@@ -270,7 +248,11 @@ static struct snd_pcm_hardware snd_pcsp_playback = {
         .info = (SNDRV_PCM_INFO_INTERLEAVED |
                  SNDRV_PCM_INFO_HALF_DUPLEX |
                  SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
-        .formats = SNDRV_PCM_FMTBIT_U8,
+        .formats = (SNDRV_PCM_FMTBIT_U8
+#if DMIX_WANTS_S16
+                    | SNDRV_PCM_FMTBIT_S16_LE
+#endif
+            ),
         .rates = SNDRV_PCM_RATE_KNOT,
         .rate_min = PCSP_DEFAULT_SRATE,
         .rate_max = PCSP_DEFAULT_SRATE,
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 64a695fef74e..caeb0f57fcca 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,7 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
         uinfo->value.enumerated.items = chip->max_treble + 1;
         if (uinfo->value.enumerated.item > chip->max_treble)
                 uinfo->value.enumerated.item = chip->max_treble;
-        sprintf(uinfo->value.enumerated.name, "%d", PCSP_RATE());
+        sprintf(uinfo->value.enumerated.name, "%d",
+                        PCSP_CALC_RATE(uinfo->value.enumerated.item));
         return 0;
 }
 
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 91d14224f6b3..73d4572d136b 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -925,7 +925,7 @@ static unsigned char als4000_saved_regs[] = {
 static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
 {
         unsigned char *val = chip->saved_regs;
-        snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return);
+        snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return);
         for (; num_regs; num_regs--)
                 *val++ = snd_sbmixer_read(chip, *regs++);
 }
@@ -933,7 +933,7 @@ static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
 static void restore_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
 {
         unsigned char *val = chip->saved_regs;
-        snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return);
+        snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return);
         for (; num_regs; num_regs--)
                 snd_sbmixer_write(chip, *regs++, *val++);
 }
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 857008bb7167..3be2dc1025b5 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -79,7 +79,7 @@ config SOUND_TRIDENT
 
 config SOUND_MSNDCLAS
         tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey"
-        depends on SOUND_PRIME && (m || !STANDALONE)
+        depends on SOUND_PRIME && (m || !STANDALONE) && ISA
         help
           Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or
           Monterey (not for the Pinnacle or Fiji).
@@ -143,7 +143,7 @@ config MSNDCLAS_IO
 
 config SOUND_MSNDPIN
         tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji"
-        depends on SOUND_PRIME && (m || !STANDALONE)
+        depends on SOUND_PRIME && (m || !STANDALONE) && ISA
         help
           Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji.
           See <file:Documentation/sound/oss/MultiSound> for important information
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 2da89810ca10..1292dcee072d 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1971,6 +1971,9 @@ static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd
 
         val = ac97->regs[AC97_AD_MISC];
         ucontrol->value.integer.value[0] = !(val & AC97_AD198X_LOSEL);
+        if (ac97->spec.ad18xx.lo_as_master)
+                ucontrol->value.integer.value[0] =
+                        !ucontrol->value.integer.value[0];
         return 0;
 }
 
@@ -1979,8 +1982,10 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd
         struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
         unsigned short val;
 
-        val = !ucontrol->value.integer.value[0]
-                ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
+        val = !ucontrol->value.integer.value[0];
+        if (ac97->spec.ad18xx.lo_as_master)
+                val = !val;
+        val = val ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
         return snd_ac97_update_bits(ac97, AC97_AD_MISC,
                                     AC97_AD198X_LOSEL | AC97_AD198X_HPSEL, val);
 }
@@ -2031,7 +2036,7 @@ static void ad1888_update_jacks(struct snd_ac97 *ac97)
 {
         unsigned short val = 0;
         /* clear LODIS if shared jack is to be used for Surround out */
-        if (is_shared_linein(ac97))
+        if (!ac97->spec.ad18xx.lo_as_master && is_shared_linein(ac97))
                 val |= (1 << 12);
         /* clear CLDIS if shared jack is to be used for C/LFE out */
         if (is_shared_micin(ac97))
@@ -2067,9 +2072,13 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
 
 static int patch_ad1888_specific(struct snd_ac97 *ac97)
 {
-        /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
-        snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Master Surround Playback");
-        snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback");
+        if (!ac97->spec.ad18xx.lo_as_master) {
+                /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
+                snd_ac97_rename_vol_ctl(ac97, "Master Playback",
+                                        "Master Surround Playback");
+                snd_ac97_rename_vol_ctl(ac97, "Headphone Playback",
+                                        "Master Playback");
+        }
         return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls));
 }
 
@@ -2088,16 +2097,27 @@ static int patch_ad1888(struct snd_ac97 * ac97)
         
         patch_ad1881(ac97);
         ac97->build_ops = &patch_ad1888_build_ops;
-        /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
-        /* it seems that most vendors connect line-out connector to headphone out of AC'97 */
+
+        /*
+         * LO can be used as a real line-out on some devices,
+         * and we need to revert the front/surround mixer switches
+         */
+        if (ac97->subsystem_vendor == 0x1043 &&
+            ac97->subsystem_device == 0x1193) /* ASUS A9T laptop */
+                ac97->spec.ad18xx.lo_as_master = 1;
+
+        misc = snd_ac97_read(ac97, AC97_AD_MISC);
         /* AD-compatible mode */
         /* Stereo mutes enabled */
-        misc = snd_ac97_read(ac97, AC97_AD_MISC);
-        snd_ac97_write_cache(ac97, AC97_AD_MISC, misc |
-                             AC97_AD198X_LOSEL |
-                             AC97_AD198X_HPSEL |
-                             AC97_AD198X_MSPLT |
-                             AC97_AD198X_AC97NC);
+        misc |= AC97_AD198X_MSPLT | AC97_AD198X_AC97NC;
+        if (!ac97->spec.ad18xx.lo_as_master)
+                /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
+                /* it seems that most vendors connect line-out connector to
+                 * headphone out of AC'97
+                 */
+                misc |= AC97_AD198X_LOSEL | AC97_AD198X_HPSEL;
+
+        snd_ac97_write_cache(ac97, AC97_AD_MISC, misc);
         ac97->flags |= AC97_STEREO_MUTES;
         return 0;
 }
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 56f87cd33c19..3f00ddf450f8 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -316,6 +316,8 @@ static int __devinit snd_aw2_create(struct snd_card *card,
                 return -ENOMEM;
         }
 
+        /* (2) initialization of the chip hardware */
+        snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
 
         if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
                         IRQF_SHARED, "Audiowerk2", chip)) {
@@ -329,8 +331,6 @@ static int __devinit snd_aw2_create(struct snd_card *card,
         }
         chip->irq = pci->irq;
 
-        /* (2) initialization of the chip hardware */
-        snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
         err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
         if (err < 0) {
                 free_irq(chip->irq, (void *)chip);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index abde5b901884..548c9cc81af5 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1818,13 +1818,6 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
         }
         emu->port = pci_resource_start(pci, 0);
 
-        if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
-                        "EMU10K1", emu)) {
-                err = -EBUSY;
-                goto error;
-        }
-        emu->irq = pci->irq;
-
         emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT;
         if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
                                 32 * 1024, &emu->ptb_pages) < 0) {
@@ -1887,6 +1880,14 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
         emu->fx8010.etram_pages.area = NULL;
         emu->fx8010.etram_pages.bytes = 0;
 
+        /* irq handler must be registered after I/O ports are activated */
+        if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
+                        "EMU10K1", emu)) {
+                err = -EBUSY;
+                goto error;
+        }
+        emu->irq = pci->irq;
+
         /*
          *  Init to 0x02109204 :
          *  Clock accuracy    = 0     (1000ppm)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0a605adde42..a99e86d74278 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2858,6 +2858,7 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = {
 static struct snd_pci_quirk ad1988_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
         SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
+        SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
         {}
 };
 
@@ -3643,33 +3644,17 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
         { } /* end */
 };
 
-static struct hda_input_mux ad1884a_mobile_capture_source = {
-        .num_items = 2,
-        .items = {
-                { "Mic", 0x1 }, /* port-C */
-                { "Mix", 0x3 },
-        },
-};
-
 static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
         HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
         HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
         HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
         HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-        HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
         HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
         HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
         HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
         HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-        {
-                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-                .name = "Capture Source",
-                .info = ad198x_mux_enum_info,
-                .get = ad198x_mux_enum_get,
-                .put = ad198x_mux_enum_put,
-        },
         { } /* end */
 };
 
@@ -3686,14 +3671,31 @@ static void ad1884a_hp_automute(struct hda_codec *codec)
                             present ? 0x00 : 0x02);
 }
 
+/* switch to external mic if plugged */
+static void ad1884a_hp_automic(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_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
+                            present ? 0 : 1);
+}
+
 #define AD1884A_HP_EVENT                0x37
+#define AD1884A_MIC_EVENT               0x36
 
 /* unsolicited event for HP jack sensing */
 static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-        if ((res >> 26) != AD1884A_HP_EVENT)
-                return;
-        ad1884a_hp_automute(codec);
+        switch (res >> 26) {
+        case AD1884A_HP_EVENT:
+                ad1884a_hp_automute(codec);
+                break;
+        case AD1884A_MIC_EVENT:
+                ad1884a_hp_automic(codec);
+                break;
+        }
 }
 
 /* initialize jack-sensing, too */
@@ -3701,6 +3703,7 @@ static int ad1884a_hp_init(struct hda_codec *codec)
 {
         ad198x_init(codec);
         ad1884a_hp_automute(codec);
+        ad1884a_hp_automic(codec);
         return 0;
 }
 
@@ -3714,10 +3717,15 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
         /* Port-F pin */
         {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
         {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* Port-C pin - internal mic-in */
+        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+        {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
         /* analog mix */
         {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
         /* unsolicited event for pin-sense */
         {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
         { } /* end */
 };
 
@@ -3877,7 +3885,6 @@ static int patch_ad1884a(struct hda_codec *codec)
                 spec->mixers[0] = ad1884a_mobile_mixers;
                 spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
                 spec->multiout.dig_out_nid = 0;
-                spec->input_mux = &ad1884a_mobile_capture_source;
                 codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
                 codec->patch_ops.init = ad1884a_hp_init;
                 break;
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index c73ce074a6ea..6ef57fbfb6eb 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -611,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = {
 
 static struct snd_pci_quirk cmi9880_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
+        SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL),
         SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
         {} /* terminator */
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6d4df45e81e0..b0a2a262ece2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -853,6 +853,7 @@ do_sku:
                 case 0x10ec0269:
                 case 0x10ec0862:
                 case 0x10ec0662:        
+                case 0x10ec0889:
                         snd_hda_codec_write(codec, 0x14, 0,
                                             AC_VERB_SET_EAPD_BTLENABLE, 2);
                         snd_hda_codec_write(codec, 0x15, 0,
@@ -877,6 +878,7 @@ do_sku:
                 case 0x10ec0883:
                 case 0x10ec0885:
                 case 0x10ec0888:
+                case 0x10ec0889:
                         snd_hda_codec_write(codec, 0x20, 0,
                                             AC_VERB_SET_COEF_INDEX, 7);
                         tmp = snd_hda_codec_read(codec, 0x20, 0,
@@ -940,7 +942,6 @@ do_sku:
                             AC_VERB_SET_UNSOLICITED_ENABLE,
                             AC_USRSP_EN | ALC880_HP_EVENT);
         spec->unsol_event = alc_sku_unsol_event;
-        spec->init_hook = alc_sku_automute;     
 }
 
 /*
@@ -2981,7 +2982,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
         /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
         SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
         SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
-        SND_PCI_QUIRK(0x1043, 0x814e, "ASUS", ALC880_ASUS),
+        SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG),
         SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
         SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
         SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
@@ -7743,6 +7744,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
         SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
         SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+        SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
         SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
@@ -8640,6 +8642,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
 
         {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
         {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+        {}
 };
 
 /* mute/unmute internal speaker according to the hp jack and mute state */
@@ -8757,35 +8760,39 @@ static struct hda_input_mux alc262_HP_D7000_capture_source = {
         },
 };
 
-/* mute/unmute internal speaker according to the hp jack and mute state */
+/* mute/unmute internal speaker according to the hp jacks and mute state */
 static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
 {
         struct alc_spec *spec = codec->spec;
         unsigned int mute;
 
         if (force || !spec->sense_updated) {
-                unsigned int present_int_hp, present_dock_hp;
+                unsigned int present;
                 /* need to execute and sync at first */
                 snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
-                present_int_hp = snd_hda_codec_read(codec, 0x14, 0,
-                                        AC_VERB_GET_PIN_SENSE, 0);
-                snd_hda_codec_read(codec, 0x1B, 0, AC_VERB_SET_PIN_SENSE, 0);
-                present_dock_hp = snd_hda_codec_read(codec, 0x1b, 0,
-                                        AC_VERB_GET_PIN_SENSE, 0);
-                spec->jack_present = (present_int_hp & 0x80000000) != 0;
-                spec->jack_present |= (present_dock_hp & 0x80000000) != 0;
+                /* check laptop HP jack */
+                present = snd_hda_codec_read(codec, 0x14, 0,
+                                             AC_VERB_GET_PIN_SENSE, 0);
+                /* need to execute and sync at first */
+                snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+                /* check docking HP jack */
+                present |= snd_hda_codec_read(codec, 0x1b, 0,
+                                              AC_VERB_GET_PIN_SENSE, 0);
+                if (present & AC_PINSENSE_PRESENCE)
+                        spec->jack_present = 1;
+                else
+                        spec->jack_present = 0;
                 spec->sense_updated = 1;
         }
-        if (spec->jack_present) {
-                /* mute internal speaker */
-                snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-                                         HDA_AMP_MUTE, HDA_AMP_MUTE);
-        } else {
-                /* unmute internal speaker if necessary */
+        /* unmute internal speaker only if both HPs are unplugged and
+         * master switch is on
+         */
+        if (spec->jack_present)
+                mute = HDA_AMP_MUTE;
+        else
                 mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
-                snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
-                                         HDA_AMP_MUTE, mute);
-        }
+        snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                 HDA_AMP_MUTE, mute);
 }
 
 /* unsolicited event for HP jack sensing */
@@ -8797,6 +8804,11 @@ static void alc262_fujitsu_unsol_event(struct hda_codec *codec,
         alc262_fujitsu_automute(codec, 1);
 }
 
+static void alc262_fujitsu_init_hook(struct hda_codec *codec)
+{
+        alc262_fujitsu_automute(codec, 1);
+}
+
 /* bind volumes of both NID 0x0c and 0x0d */
 static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
         .ops = &snd_hda_bind_vol,
@@ -9570,6 +9582,7 @@ static struct alc_config_preset alc262_presets[] = {
                 .channel_mode = alc262_modes,
                 .input_mux = &alc262_fujitsu_capture_source,
                 .unsol_event = alc262_fujitsu_unsol_event,
+                .init_hook = alc262_fujitsu_init_hook,
         },
         [ALC262_HP_BPC] = {
                 .mixers = { alc262_HP_BPC_mixer },
@@ -10500,6 +10513,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
         SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
         SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
+        SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
         SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
         SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
         SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 393f7fd2b1be..a4f44a00bae8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -840,7 +840,7 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 static struct snd_kcontrol_new stac925x_mixer[] = {
         STAC_INPUT_SOURCE(1),
         HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
-        HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
         HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT),
         { } /* end */
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 52b1d81a26f7..e7e43524f8c7 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -447,6 +447,23 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = {
         },
 };
 
+static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
+        .substreams = 1,
+        .channels_min = 2,
+        .channels_max = 8,
+        .nid = 0x10, /* NID to query formats and rates */
+        /* We got noisy outputs on the right channel on VT1708 when
+         * 24bit samples are used.  Until any workaround is found,
+         * disable the 24bit format, so far.
+         */
+        .formats = SNDRV_PCM_FMTBIT_S16_LE,
+        .ops = {
+                .open = via_playback_pcm_open,
+                .prepare = via_playback_pcm_prepare,
+                .cleanup = via_playback_pcm_cleanup
+        },
+};
+
 static struct hda_pcm_stream vt1708_pcm_analog_capture = {
         .substreams = 2,
         .channels_min = 2,
@@ -899,6 +916,9 @@ static int patch_vt1708(struct hda_codec *codec)
         
         spec->stream_name_analog = "VT1708 Analog";
         spec->stream_analog_playback = &vt1708_pcm_analog_playback;
+        /* disable 32bit format on VT1708 */
+        if (codec->vendor_id == 0x11061708)
+                spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback;
         spec->stream_analog_capture = &vt1708_pcm_analog_capture;
 
         spec->stream_name_digital = "VT1708 Digital";
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index cc0cddadd589..6facac5aed90 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -936,11 +936,13 @@ static int add_controls(struct oxygen *chip,
 
         for (i = 0; i < count; ++i) {
                 template = controls[i];
-                err = chip->model->control_filter(&template);
-                if (err < 0)
-                        return err;
-                if (err == 1)
-                        continue;
+                if (chip->model->control_filter) {
+                        err = chip->model->control_filter(&template);
+                        if (err < 0)
+                                return err;
+                        if (err == 1)
+                                continue;
+                }
                 if (!strcmp(template.name, "Master Playback Volume") &&
                     chip->model->dac_tlv) {
                         template.tlv.p = chip->model->dac_tlv;
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index bc35d00a38f8..1a4260950142 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -41,7 +41,7 @@
 #define DBG(x...)
 #endif
 
-#if defined(CONFIG_ARCH_AT91SAM9260)
+#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
 #define NUM_SSC_DEVICES         1
 #else
 #define NUM_SSC_DEVICES         3
diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
index e97d8b2ac16a..a972f77bd785 100644
--- a/sound/usb/caiaq/caiaq-device.c
+++ b/sound/usb/caiaq/caiaq-device.c
@@ -351,8 +351,8 @@ static struct snd_card* create_card(struct usb_device* usb_dev)
         dev = caiaqdev(card);
         dev->chip.dev = usb_dev;
         dev->chip.card = card;
-        dev->chip.usb_id = USB_ID(usb_dev->descriptor.idVendor,
-                                        usb_dev->descriptor.idProduct);
+        dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor),
+                                  le16_to_cpu(usb_dev->descriptor.idProduct));
         spin_lock_init(&dev->spinlock);
         snd_card_set_dev(card, &usb_dev->dev);