Merge commit 'v2.6.26' into bkl-removal

19 files changed, 185 insertions, 131 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index b2a2db47aff..4247406160e 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 65b66fa6f97..62e057a9465 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -272,8 +272,9 @@ 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);
@@ -282,9 +283,6 @@ 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 379bcb07446..602b58e3b55 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 54a1f9036c6..1899cf0685b 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 f07cc1ee1fe..1d661f795e8 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 ac6238e9351..e341f3f83b6 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 64a695fef74..caeb0f57fcc 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 91d14224f6b..73d4572d136 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 857008bb716..3be2dc1025b 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 2da89810ca1..1292dcee072 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 56f87cd33c1..3f00ddf450f 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 abde5b90188..548c9cc81af 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 e0a605adde4..a99e86d7427 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 c73ce074a6e..6ef57fbfb6e 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 6d4df45e81e..b0a2a262ece 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 393f7fd2b1b..a4f44a00bae 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 52b1d81a26f..e7e43524f8c 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 cc0cddadd58..6facac5aed9 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/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
index e97d8b2ac16..a972f77bd78 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);