From 5a6048e1fa8443341b981227f4bcaeae2250fdbb Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jarkko.nikula@bitmer.com>
Date: Fri, 23 Sep 2011 11:19:13 +0300
Subject: [PATCH] --- yaml --- r: 264143 b: refs/heads/master c:
 34c869855a28af3c6031d29e2267b5a3cce0d67d h: refs/heads/master i:   264141:
 292acc9a2a2058d340fa1afd10653b9f4e286a70   264139:
 a3cc34dba1c2ba0776c94c2465e50973f4498c55   264135:
 3998a62b20aebfe6caaea0ea30d3187cddace9da   264127:
 3bc04219f909fc9701ab83a3a3d19f8433ecda58 v: v3

---
 [refs]                               |  2 +-
 trunk/sound/aoa/fabrics/layout.c     |  2 +-
 trunk/sound/core/pcm_lib.c           | 33 ++++---------
 trunk/sound/core/timer.c             |  5 --
 trunk/sound/oss/pas2_pcm.c           |  8 ++--
 trunk/sound/oss/pss.c                |  6 +--
 trunk/sound/pci/Kconfig              | 10 ++--
 trunk/sound/pci/ac97/ac97_patch.c    |  1 -
 trunk/sound/pci/asihpi/hpicmn.c      |  5 +-
 trunk/sound/pci/azt3328.c            | 11 ++---
 trunk/sound/pci/fm801.c              | 15 ++----
 trunk/sound/pci/hda/alc268_quirks.c  | 36 +++++++-------
 trunk/sound/pci/hda/alc269_quirks.c  |  7 +++
 trunk/sound/pci/hda/hda_codec.c      |  6 +--
 trunk/sound/pci/hda/hda_eld.c        | 31 +++++-------
 trunk/sound/pci/hda/patch_cirrus.c   | 10 ++--
 trunk/sound/pci/hda/patch_conexant.c | 57 ++++++++--------------
 trunk/sound/pci/hda/patch_realtek.c  | 71 ++++++----------------------
 trunk/sound/pci/hda/patch_sigmatel.c |  4 --
 trunk/sound/pci/hda/patch_via.c      |  2 +-
 trunk/sound/pci/rme9652/hdspm.c      | 19 +++-----
 trunk/sound/soc/omap/omap-mcbsp.c    |  6 +++
 trunk/sound/usb/caiaq/audio.c        | 37 +++------------
 trunk/sound/usb/caiaq/device.h       |  1 -
 trunk/sound/usb/caiaq/input.c        |  2 +-
 trunk/sound/usb/card.c               |  7 +--
 trunk/sound/usb/endpoint.c           |  2 +-
 trunk/sound/usb/mixer.c              | 28 +++++------
 trunk/sound/usb/mixer.h              |  1 -
 trunk/sound/usb/quirks-table.h       | 40 ----------------
 trunk/sound/usb/quirks.c             |  2 +-
 31 files changed, 149 insertions(+), 318 deletions(-)

diff --git a/[refs] b/[refs]
index 5ba3fba98068..fe24ce91ff4c 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: 61a6a108d15213f5ee06332e1e7766d3860e4453
+refs/heads/master: 34c869855a28af3c6031d29e2267b5a3cce0d67d
diff --git a/trunk/sound/aoa/fabrics/layout.c b/trunk/sound/aoa/fabrics/layout.c
index 552b97afbca5..3fd1a7e24928 100644
--- a/trunk/sound/aoa/fabrics/layout.c
+++ b/trunk/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 	sdev->pcmid = -1;
 	list_del(&ldev->list);
 	layouts_list_items--;
-	kfree(ldev);
  outnodev:
  	of_node_put(sound);
  	layout_device = NULL;
+ 	kfree(ldev);
 	return -ENODEV;
 }
 
diff --git a/trunk/sound/core/pcm_lib.c b/trunk/sound/core/pcm_lib.c
index 62e90b862a0d..86d0caf91b35 100644
--- a/trunk/sound/core/pcm_lib.c
+++ b/trunk/sound/core/pcm_lib.c
@@ -1761,10 +1761,6 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 	snd_pcm_uframes_t avail = 0;
 	long wait_time, tout;
 
-	init_waitqueue_entry(&wait, current);
-	set_current_state(TASK_INTERRUPTIBLE);
-	add_wait_queue(&runtime->tsleep, &wait);
-
 	if (runtime->no_period_wakeup)
 		wait_time = MAX_SCHEDULE_TIMEOUT;
 	else {
@@ -1775,32 +1771,16 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 		}
 		wait_time = msecs_to_jiffies(wait_time * 1000);
 	}
-
+	init_waitqueue_entry(&wait, current);
+	add_wait_queue(&runtime->tsleep, &wait);
 	for (;;) {
 		if (signal_pending(current)) {
 			err = -ERESTARTSYS;
 			break;
 		}
-
-		/*
-		 * We need to check if space became available already
-		 * (and thus the wakeup happened already) first to close
-		 * the race of space already having become available.
-		 * This check must happen after been added to the waitqueue
-		 * and having current state be INTERRUPTIBLE.
-		 */
-		if (is_playback)
-			avail = snd_pcm_playback_avail(runtime);
-		else
-			avail = snd_pcm_capture_avail(runtime);
-		if (avail >= runtime->twake)
-			break;
 		snd_pcm_stream_unlock_irq(substream);
-
-		tout = schedule_timeout(wait_time);
-
+		tout = schedule_timeout_interruptible(wait_time);
 		snd_pcm_stream_lock_irq(substream);
-		set_current_state(TASK_INTERRUPTIBLE);
 		switch (runtime->status->state) {
 		case SNDRV_PCM_STATE_SUSPENDED:
 			err = -ESTRPIPE;
@@ -1826,9 +1806,14 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 			err = -EIO;
 			break;
 		}
+		if (is_playback)
+			avail = snd_pcm_playback_avail(runtime);
+		else
+			avail = snd_pcm_capture_avail(runtime);
+		if (avail >= runtime->twake)
+			break;
 	}
  _endloop:
-	set_current_state(TASK_RUNNING);
 	remove_wait_queue(&runtime->tsleep, &wait);
 	*availp = avail;
 	return err;
diff --git a/trunk/sound/core/timer.c b/trunk/sound/core/timer.c
index 67ebf1c21c04..7c1cbf0a0dc4 100644
--- a/trunk/sound/core/timer.c
+++ b/trunk/sound/core/timer.c
@@ -328,8 +328,6 @@ int snd_timer_close(struct snd_timer_instance *timeri)
 		mutex_unlock(&register_mutex);
 	} else {
 		timer = timeri->timer;
-		if (snd_BUG_ON(!timer))
-			goto out;
 		/* wait, until the active callback is finished */
 		spin_lock_irq(&timer->lock);
 		while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
@@ -355,7 +353,6 @@ int snd_timer_close(struct snd_timer_instance *timeri)
 		}
 		mutex_unlock(&register_mutex);
 	}
- out:
 	if (timeri->private_free)
 		timeri->private_free(timeri);
 	kfree(timeri->owner);
@@ -534,8 +531,6 @@ int snd_timer_stop(struct snd_timer_instance *timeri)
 	if (err < 0)
 		return err;
 	timer = timeri->timer;
-	if (!timer)
-		return -EINVAL;
 	spin_lock_irqsave(&timer->lock, flags);
 	timeri->cticks = timeri->ticks;
 	timeri->pticks = 0;
diff --git a/trunk/sound/oss/pas2_pcm.c b/trunk/sound/oss/pas2_pcm.c
index 6f13ab4afc6b..8f7d175767a2 100644
--- a/trunk/sound/oss/pas2_pcm.c
+++ b/trunk/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@ static int pcm_set_speed(int arg)
 
 	if (pcm_channels & 2)
 	{
-		foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
-		arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
+		foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
+		arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
 	}
 	else
 	{
-		foo = (PIT_TICK_RATE + (arg / 2)) / arg;
-		arg = (PIT_TICK_RATE + (foo / 2)) / foo;
+		foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
+		arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
 	}
 
 	pcm_speed = arg;
diff --git a/trunk/sound/oss/pss.c b/trunk/sound/oss/pss.c
index 2fc0624024b5..9b800ce5100e 100644
--- a/trunk/sound/oss/pss.c
+++ b/trunk/sound/oss/pss.c
@@ -673,8 +673,7 @@ static void configure_nonsound_components(void)
 
 	if (pss_cdrom_port == -1) {	/* If cdrom port enablation wasn't requested */
 		printk(KERN_INFO "PSS: CDROM port not enabled.\n");
-	} else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) {
-		pss_cdrom_port = -1;
+	} else if (check_region(pss_cdrom_port, 2)) {
 		printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
 	} else {
 		set_io_base(devc, CONF_CDROM, pss_cdrom_port);
@@ -1233,8 +1232,7 @@ static void __exit cleanup_pss(void)
 		if(pssmpu)
 			unload_pss_mpu(&cfg_mpu);
 		unload_pss(&cfg);
-	} else if (pss_cdrom_port != -1)
-		release_region(pss_cdrom_port, 2);
+	}
 
 	if(!pss_keep_settings)	/* Keep hardware settings if asked */
 	{
diff --git a/trunk/sound/pci/Kconfig b/trunk/sound/pci/Kconfig
index 88168044375f..50abf5bf8e09 100644
--- a/trunk/sound/pci/Kconfig
+++ b/trunk/sound/pci/Kconfig
@@ -1,10 +1,5 @@
 # ALSA PCI drivers
 
-config SND_TEA575X
-	tristate
-	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
-	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
 menuconfig SND_PCI
 	bool "PCI sound devices"
 	depends on PCI
@@ -568,6 +563,11 @@ config SND_FM801_TEA575X_BOOL
 	  FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
 	  SF64-PCR) into the snd-fm801 driver.
 
+config SND_TEA575X
+	tristate
+	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
 source "sound/pci/hda/Kconfig"
 
 config SND_HDSP
diff --git a/trunk/sound/pci/ac97/ac97_patch.c b/trunk/sound/pci/ac97/ac97_patch.c
index a872d0a82976..200c9a1d48b7 100644
--- a/trunk/sound/pci/ac97/ac97_patch.c
+++ b/trunk/sound/pci/ac97/ac97_patch.c
@@ -1909,7 +1909,6 @@ static unsigned int ad1981_jacks_whitelist[] = {
 	0x103c0944, /* HP nc6220 */
 	0x103c0934, /* HP nc8220 */
 	0x103c006d, /* HP nx9105 */
-	0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
 	0x17340088, /* FSC Scenic-W */
 	0 /* end */
 };
diff --git a/trunk/sound/pci/asihpi/hpicmn.c b/trunk/sound/pci/asihpi/hpicmn.c
index bd47521b24ec..65b7ca13115b 100644
--- a/trunk/sound/pci/asihpi/hpicmn.c
+++ b/trunk/sound/pci/asihpi/hpicmn.c
@@ -631,12 +631,13 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
 	if (!p_cache)
 		return NULL;
 
-	p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
-				  GFP_KERNEL);
+	p_cache->p_info =
+		kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
 	if (!p_cache->p_info) {
 		kfree(p_cache);
 		return NULL;
 	}
+	memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
 	p_cache->cache_size_in_bytes = size_in_bytes;
 	p_cache->control_count = control_count;
 	p_cache->p_cache = p_dsp_control_buffer;
diff --git a/trunk/sound/pci/azt3328.c b/trunk/sound/pci/azt3328.c
index 579fc0dce128..e4d76a270c9f 100644
--- a/trunk/sound/pci/azt3328.c
+++ b/trunk/sound/pci/azt3328.c
@@ -2625,19 +2625,16 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	int err;
 
 	snd_azf3328_dbgcallenter();
-	if (dev >= SNDRV_CARDS) {
-		err = -ENODEV;
-		goto out;
-	}
+	if (dev >= SNDRV_CARDS)
+		return -ENODEV;
 	if (!enable[dev]) {
 		dev++;
-		err = -ENOENT;
-		goto out;
+		return -ENOENT;
 	}
 
 	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 	if (err < 0)
-		goto out;
+		return err;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/trunk/sound/pci/fm801.c b/trunk/sound/pci/fm801.c
index 32b02d906703..f9123f09e83e 100644
--- a/trunk/sound/pci/fm801.c
+++ b/trunk/sound/pci/fm801.c
@@ -68,7 +68,6 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
 module_param_array(tea575x_tuner, int, NULL, 0444);
 MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
 
-#define TUNER_DISABLED		(1<<3)
 #define TUNER_ONLY		(1<<4)
 #define TUNER_TYPE_MASK		(~TUNER_ONLY & 0xFFFF)
 
@@ -1151,8 +1150,7 @@ static int snd_fm801_free(struct fm801 *chip)
 
       __end_hw:
 #ifdef CONFIG_SND_FM801_TEA575X_BOOL
-	if (!(chip->tea575x_tuner & TUNER_DISABLED))
-		snd_tea575x_exit(&chip->tea);
+	snd_tea575x_exit(&chip->tea);
 #endif
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1238,6 +1236,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
 		if (snd_tea575x_init(&chip->tea)) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
+			snd_fm801_free(chip);
 			return -ENODEV;
 		}
 	} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1252,15 +1251,11 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		}
 		if (tea575x_tuner == 4) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			chip->tea575x_tuner = TUNER_DISABLED;
+			snd_fm801_free(chip);
+			return -ENODEV;
 		}
 	}
-	if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
-		strlcpy(chip->tea.card,
-			snd_fm801_tea575x_gpios[(tea575x_tuner &
-						 TUNER_TYPE_MASK) - 1].name,
-			sizeof(chip->tea.card));
-	}
+	strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
 #endif
 
 	*rchip = chip;
diff --git a/trunk/sound/pci/hda/alc268_quirks.c b/trunk/sound/pci/hda/alc268_quirks.c
index 2e5876ce71fe..be58bf2f3aec 100644
--- a/trunk/sound/pci/hda/alc268_quirks.c
+++ b/trunk/sound/pci/hda/alc268_quirks.c
@@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
 
 static const struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
-		.cap_mixer = alc268_capture_nosrc_mixer,
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc267_quanta_il1_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_3ST] = {
-		.mixers = { alc268_base_mixer, alc268_beep_mixer },
-		.cap_mixer = alc268_capture_alt_mixer,
+		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
 		.dac_nids = alc268_dac_nids,
@@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.input_mux = &alc268_capture_source,
 	},
 	[ALC268_TOSHIBA] = {
-		.mixers = { alc268_toshiba_mixer, alc268_beep_mixer },
-		.cap_mixer = alc268_capture_alt_mixer,
+		.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER] = {
-		.mixers = { alc268_acer_mixer, alc268_beep_mixer },
-		.cap_mixer = alc268_capture_alt_mixer,
+		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_DMIC] = {
-		.mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer },
-		.cap_mixer = alc268_capture_alt_mixer,
+		.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -561,8 +561,9 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_ASPIRE_ONE] = {
-		.mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer},
-		.cap_mixer = alc268_capture_nosrc_mixer,
+		.mixers = { alc268_acer_aspire_one_mixer,
+			    alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_aspire_one_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -578,8 +579,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_DELL] = {
-		.mixers = { alc268_dell_mixer, alc268_beep_mixer},
-		.cap_mixer = alc268_capture_nosrc_mixer,
+		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_dell_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -595,8 +596,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ZEPTO] = {
-		.mixers = { alc268_base_mixer, alc268_beep_mixer },
-		.cap_mixer = alc268_capture_alt_mixer,
+		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+			    alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -615,8 +616,7 @@ static const struct alc_config_preset alc268_presets[] = {
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC268_TEST] = {
-		.mixers = { alc268_test_mixer },
-		.cap_mixer = alc268_capture_mixer,
+		.mixers = { alc268_test_mixer, alc268_capture_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_volume_init_verbs,
 				alc268_beep_init_verbs },
diff --git a/trunk/sound/pci/hda/alc269_quirks.c b/trunk/sound/pci/hda/alc269_quirks.c
index 5ac0e2162a46..14fdcf29b154 100644
--- a/trunk/sound/pci/hda/alc269_quirks.c
+++ b/trunk/sound/pci/hda/alc269_quirks.c
@@ -531,10 +531,17 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
+	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
+		      ALC269_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
+		      ALC269_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
 	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
 	SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
diff --git a/trunk/sound/pci/hda/hda_codec.c b/trunk/sound/pci/hda/hda_codec.c
index f3aefef37216..3e7850c238c3 100644
--- a/trunk/sound/pci/hda/hda_codec.c
+++ b/trunk/sound/pci/hda/hda_codec.c
@@ -579,13 +579,9 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 		return -1;
 	}
 	recursive++;
-	for (i = 0; i < nums; i++) {
-		unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
-		if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
-			continue;
+	for (i = 0; i < nums; i++)
 		if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
 			return i;
-	}
 	return -1;
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
diff --git a/trunk/sound/pci/hda/hda_eld.c b/trunk/sound/pci/hda/hda_eld.c
index c34f730f4815..28ce17d09c33 100644
--- a/trunk/sound/pci/hda/hda_eld.c
+++ b/trunk/sound/pci/hda/hda_eld.c
@@ -144,17 +144,25 @@ static int cea_sampling_frequencies[8] = {
 	SNDRV_PCM_RATE_192000,	/* 7: 192000Hz */
 };
 
-static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
+static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
 					int byte_index)
 {
 	unsigned int val;
 
 	val = snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_GET_HDMI_ELDD, byte_index);
+
 #ifdef BE_PARANOID
 	printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
 #endif
-	return val;
+
+	if ((val & AC_ELDD_ELD_VALID) == 0) {
+		snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
+								byte_index);
+		val = 0;
+	}
+
+	return val & AC_ELDD_ELD_DATA;
 }
 
 #define GRAB_BITS(buf, byte, lowbit, bits) 		\
@@ -336,26 +344,11 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 	if (!buf)
 		return -ENOMEM;
 
-	for (i = 0; i < size; i++) {
-		unsigned int val = hdmi_get_eld_data(codec, nid, i);
-		if (!(val & AC_ELDD_ELD_VALID)) {
-			if (!i) {
-				snd_printd(KERN_INFO
-					   "HDMI: invalid ELD data\n");
-				ret = -EINVAL;
-				goto error;
-			}
-			snd_printd(KERN_INFO
-				  "HDMI: invalid ELD data byte %d\n", i);
-			val = 0;
-		} else
-			val &= AC_ELDD_ELD_DATA;
-		buf[i] = val;
-	}
+	for (i = 0; i < size; i++)
+		buf[i] = hdmi_get_eld_byte(codec, nid, i);
 
 	ret = hdmi_update_eld(eld, buf, size);
 
-error:
 	kfree(buf);
 	return ret;
 }
diff --git a/trunk/sound/pci/hda/patch_cirrus.c b/trunk/sound/pci/hda/patch_cirrus.c
index c45f3e69bcf0..47d6ffc9b5b5 100644
--- a/trunk/sound/pci/hda/patch_cirrus.c
+++ b/trunk/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
 static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 			 unsigned int *idxp)
 {
-	int i, idx;
+	int i;
 	hda_nid_t nid;
 
 	nid = codec->start_nid;
@@ -384,11 +384,9 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		idx = snd_hda_get_conn_index(codec, nid, pin, false);
-		if (idx >= 0) {
-			*idxp = idx;
+		*idxp = snd_hda_get_conn_index(codec, nid, pin, false);
+		if (*idxp >= 0)
 			return nid;
-		}
 	}
 	return 0;
 }
@@ -535,7 +533,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
 		      int index, unsigned int pval, int dir,
 		      struct snd_kcontrol **kctlp)
 {
-	char tmp[44];
+	char tmp[32];
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
 	knew.private_value = pval;
diff --git a/trunk/sound/pci/hda/patch_conexant.c b/trunk/sound/pci/hda/patch_conexant.c
index 7696d05b9356..502fc9499453 100644
--- a/trunk/sound/pci/hda/patch_conexant.c
+++ b/trunk/sound/pci/hda/patch_conexant.c
@@ -3348,8 +3348,6 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
 
 #define MAX_AUTO_DACS	5
 
-#define DAC_SLAVE_FLAG	0x8000	/* filled dac is a slave */
-
 /* fill analog DAC list from the widget tree */
 static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 {
@@ -3372,26 +3370,16 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 /* fill pin_dac_pair list from the pin and dac list */
 static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
 			      int num_pins, hda_nid_t *dacs, int *rest,
-			      struct pin_dac_pair *filled, int nums, 
-			      int type)
+			      struct pin_dac_pair *filled, int type)
 {
-	int i, start = nums;
+	int i, nums;
 
-	for (i = 0; i < num_pins; i++, nums++) {
+	nums = 0;
+	for (i = 0; i < num_pins; i++) {
 		filled[nums].pin = pins[i];
 		filled[nums].type = type;
 		filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
-		if (filled[nums].dac) 
-			continue;
-		if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
-			filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
-			continue;
-		}
-		if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
-			filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
-			continue;
-		}
-		snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
+		nums++;
 	}
 	return nums;
 }
@@ -3407,19 +3395,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 	rest = fill_cx_auto_dacs(codec, dacs);
 	/* parse all analog output pins */
 	nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
-			  dacs, &rest, spec->dac_info, 0,
-			  AUTO_PIN_LINE_OUT);
-	nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
-			  dacs, &rest, spec->dac_info, nums,
-			  AUTO_PIN_HP_OUT);
-	nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
-			  dacs, &rest, spec->dac_info, nums,
-			  AUTO_PIN_SPEAKER_OUT);
+				  dacs, &rest, spec->dac_info,
+				  AUTO_PIN_LINE_OUT);
+	nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+				  dacs, &rest, spec->dac_info + nums,
+				  AUTO_PIN_HP_OUT);
+	nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+				  dacs, &rest, spec->dac_info + nums,
+				  AUTO_PIN_SPEAKER_OUT);
 	spec->dac_info_filled = nums;
 	/* fill multiout struct */
 	for (i = 0; i < nums; i++) {
 		hda_nid_t dac = spec->dac_info[i].dac;
-		if (!dac || (dac & DAC_SLAVE_FLAG))
+		if (!dac)
 			continue;
 		switch (spec->dac_info[i].type) {
 		case AUTO_PIN_LINE_OUT:
@@ -3874,7 +3862,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
 	}
 	if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
 		cx_auto_check_auto_mic(codec);
-	if (imux->num_items > 1) {
+	if (imux->num_items > 1 && !spec->auto_mic) {
 		for (i = 1; i < imux->num_items; i++) {
 			if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
 				spec->adc_switching = 1;
@@ -4047,8 +4035,6 @@ static void cx_auto_init_output(struct hda_codec *codec)
 		nid = spec->dac_info[i].dac;
 		if (!nid)
 			nid = spec->multiout.dac_nids[0];
-		else if (nid & DAC_SLAVE_FLAG)
-			nid &= ~DAC_SLAVE_FLAG;
 		select_connection(codec, spec->dac_info[i].pin, nid);
 	}
 	if (spec->auto_mute) {
@@ -4181,11 +4167,9 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
 			     hda_nid_t pin, const char *name, int idx)
 {
 	unsigned int caps;
-	if (dac && !(dac & DAC_SLAVE_FLAG)) {
-		caps = query_amp_caps(codec, dac, HDA_OUTPUT);
-		if (caps & AC_AMPCAP_NUM_STEPS)
-			return cx_auto_add_pb_volume(codec, dac, name, idx);
-	}
+	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+	if (caps & AC_AMPCAP_NUM_STEPS)
+		return cx_auto_add_pb_volume(codec, dac, name, idx);
 	caps = query_amp_caps(codec, pin, HDA_OUTPUT);
 	if (caps & AC_AMPCAP_NUM_STEPS)
 		return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4207,7 +4191,8 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
-		hda_nid_t dac = spec->dac_info[i].dac;
+		if (!spec->dac_info[i].dac)
+			continue;
 		type = spec->dac_info[i].type;
 		if (type == AUTO_PIN_LINE_OUT)
 			type = spec->autocfg.line_out_type;
@@ -4226,7 +4211,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 			idx = num_spk++;
 			break;
 		}
-		err = try_add_pb_volume(codec, dac,
+		err = try_add_pb_volume(codec, spec->dac_info[i].dac,
 					spec->dac_info[i].pin,
 					label, idx);
 		if (err < 0)
diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c
index 7a73621a8909..e125c60fe352 100644
--- a/trunk/sound/pci/hda/patch_realtek.c
+++ b/trunk/sound/pci/hda/patch_realtek.c
@@ -168,7 +168,7 @@ struct alc_spec {
 	unsigned int auto_mic_valid_imux:1;	/* valid imux for auto-mic */
 	unsigned int automute:1;	/* HP automute enabled */
 	unsigned int detect_line:1;	/* Line-out detection enabled */
-	unsigned int automute_lines:1;	/* automute line-out as well; NOP when automute_hp_lo isn't set */
+	unsigned int automute_lines:1;	/* automute line-out as well */
 	unsigned int automute_hp_lo:1;	/* both HP and LO available */
 
 	/* other flags */
@@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec)
 	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
 	    spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
 		return;
-	if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
+	if (!spec->automute_lines || !spec->automute)
 		on = 0;
 	else
 		on = spec->jack_present;
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
+	if (!spec->automute)
+		return;
 	spec->jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
 			     spec->autocfg.hp_pins);
-	if (!spec->automute)
-		return;
 	update_speakers(codec);
 }
 
@@ -578,15 +578,11 @@ static void alc_line_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	/* check LO jack only when it's different from HP */
-	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
+	if (!spec->automute || !spec->detect_line)
 		return;
-
 	spec->line_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 			     spec->autocfg.line_out_pins);
-	if (!spec->automute || !spec->detect_line)
-		return;
 	update_speakers(codec);
 }
 
@@ -807,7 +803,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 	if (!spec->automute)
 		val = 0;
-	else if (!spec->automute_hp_lo || !spec->automute_lines)
+	else if (!spec->automute_lines)
 		val = 1;
 	else
 		val = 2;
@@ -828,8 +824,7 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
 		spec->automute = 0;
 		break;
 	case 1:
-		if (spec->automute &&
-		    (!spec->automute_hp_lo || !spec->automute_lines))
+		if (spec->automute && !spec->automute_lines)
 			return 0;
 		spec->automute = 1;
 		spec->automute_lines = 0;
@@ -1325,9 +1320,7 @@ static int alc_subsystem_id(struct hda_codec *codec,
 	 * 15   : 1 --> enable the function "Mute internal speaker
 	 *	        when the external headphone out jack is plugged"
 	 */
-	if (!spec->autocfg.hp_pins[0] &&
-	    !(spec->autocfg.line_out_pins[0] &&
-	      spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
+	if (!spec->autocfg.hp_pins[0]) {
 		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
@@ -1791,7 +1784,6 @@ static const char * const alc_slave_vols[] = {
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
 	"Line-Out Playback Volume",
-	"PCM Playback Volume",
 	NULL,
 };
 
@@ -1806,7 +1798,6 @@ static const char * const alc_slave_sws[] = {
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
 	"Line-Out Playback Switch",
-	"PCM Playback Switch",
 	NULL,
 };
 
@@ -3090,22 +3081,16 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
 static void alc_auto_init_extra_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin, dac;
+	hda_nid_t pin;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin) {
-		dac = spec->multiout.hp_nid;
-		if (!dac)
-			dac = spec->multiout.dac_nids[0];
-		alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
-	}
+	if (pin)
+		alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
+						  spec->multiout.hp_nid);
 	pin = spec->autocfg.speaker_pins[0];
-	if (pin) {
-		dac = spec->multiout.extra_out_nid[0];
-		if (!dac)
-			dac = spec->multiout.dac_nids[0];
-		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
-	}
+	if (pin)
+		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+					spec->multiout.extra_out_nid[0]);
 }
 
 /*
@@ -4499,22 +4484,6 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
 	spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
 }
 
-static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
-				     const struct alc_fixup *fix, int action)
-{
-	int coef;
-
-	if (action != ALC_FIXUP_ACT_INIT)
-		return;
-	/* The digital-mic unit sends PDM (differential signal) instead of
-	 * the standard PCM, thus you can't record a valid mono stream as is.
-	 * Below is a workaround specific to ALC269 to control the dmic
-	 * signal source as mono.
-	 */
-	coef = alc_read_coef_idx(codec, 0x07);
-	alc_write_coef_idx(codec, 0x07, coef | 0x80);
-}
-
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4525,7 +4494,6 @@ enum {
 	ALC275_FIXUP_SONY_HWEQ,
 	ALC271_FIXUP_DMIC,
 	ALC269_FIXUP_PCM_44K,
-	ALC269_FIXUP_STEREO_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -4588,19 +4556,10 @@ static const struct alc_fixup alc269_fixups[] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_pcm_44k,
 	},
-	[ALC269_FIXUP_STEREO_DMIC] = {
-		.type = ALC_FIXUP_FUNC,
-		.v.func = alc269_fixup_stereo_dmic,
-	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
-	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
diff --git a/trunk/sound/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index 987e3cf71a0b..aa376b59c006 100644
--- a/trunk/sound/pci/hda/patch_sigmatel.c
+++ b/trunk/sound/pci/hda/patch_sigmatel.c
@@ -673,7 +673,6 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-#ifdef CONFIG_SND_HDA_POWER_SAVE
 static int stac_vrefout_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
 {
@@ -697,7 +696,6 @@ static int stac_vrefout_set(struct hda_codec *codec,
 
 	return 1;
 }
-#endif
 
 static unsigned int stac92xx_vref_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
@@ -5630,7 +5628,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	switch (codec->vendor_id) {
 	case 0x111d76d1:
 	case 0x111d76d9:
-	case 0x111d76df:
 	case 0x111d76e5:
 	case 0x111d7666:
 	case 0x111d7667:
@@ -6574,7 +6571,6 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
-	{ .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
diff --git a/trunk/sound/pci/hda/patch_via.c b/trunk/sound/pci/hda/patch_via.c
index 4ebfbd874c9a..84d8798bf33a 100644
--- a/trunk/sound/pci/hda/patch_via.c
+++ b/trunk/sound/pci/hda/patch_via.c
@@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
 	struct via_spec *spec = codec->spec;
 	struct nid_path *path;
 	bool check_dac;
-	hda_nid_t pin, dac = 0;
+	hda_nid_t pin, dac;
 	int err;
 
 	pin = spec->autocfg.speaker_pins[0];
diff --git a/trunk/sound/pci/rme9652/hdspm.c b/trunk/sound/pci/rme9652/hdspm.c
index 493e3946756f..6edc67ced905 100644
--- a/trunk/sound/pci/rme9652/hdspm.c
+++ b/trunk/sound/pci/rme9652/hdspm.c
@@ -1339,10 +1339,6 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
 		break;
 	case MADIface:
 		freq_const = 131072000000000ULL;
-		break;
-	default:
-		snd_BUG();
-		return 0;
 	}
 
 	return div_u64(freq_const, period);
@@ -1360,19 +1356,16 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
 
 	switch (hdspm->io_type) {
 	case MADIface:
-		n = 131072000000000ULL;  /* 125 MHz */
-		break;
+	  n = 131072000000000ULL;  /* 125 MHz */
+	  break;
 	case MADI:
 	case AES32:
-		n = 110069313433624ULL;  /* 105 MHz */
-		break;
+	  n = 110069313433624ULL;  /* 105 MHz */
+	  break;
 	case RayDAT:
 	case AIO:
-		n = 104857600000000ULL;  /* 100 MHz */
-		break;
-	default:
-		snd_BUG();
-		return;
+	  n = 104857600000000ULL;  /* 100 MHz */
+	  break;
 	}
 
 	n = div_u64(n, rate);
diff --git a/trunk/sound/soc/omap/omap-mcbsp.c b/trunk/sound/soc/omap/omap-mcbsp.c
index ebcc2d4d2b18..478d60778453 100644
--- a/trunk/sound/soc/omap/omap-mcbsp.c
+++ b/trunk/sound/soc/omap/omap-mcbsp.c
@@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int err = 0;
 
+	if (mcbsp_data->active)
+		if (freq == mcbsp_data->in_freq)
+			return 0;
+		else
+			return -EBUSY;
+
 	/* The McBSP signal muxing functions are only available on McBSP1 */
 	if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR ||
 	    clk_id == OMAP_MCBSP_CLKR_SRC_CLKX ||
diff --git a/trunk/sound/usb/caiaq/audio.c b/trunk/sound/usb/caiaq/audio.c
index 2cf87f5afed4..d0d493ca28ae 100644
--- a/trunk/sound/usb/caiaq/audio.c
+++ b/trunk/sound/usb/caiaq/audio.c
@@ -139,12 +139,8 @@ static void stream_stop(struct snd_usb_caiaqdev *dev)
 
 	for (i = 0; i < N_URBS; i++) {
 		usb_kill_urb(dev->data_urbs_in[i]);
-
-		if (test_bit(i, &dev->outurb_active_mask))
-			usb_kill_urb(dev->data_urbs_out[i]);
+		usb_kill_urb(dev->data_urbs_out[i]);
 	}
-
-	dev->outurb_active_mask = 0;
 }
 
 static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
@@ -616,9 +612,8 @@ static void read_completed(struct urb *urb)
 {
 	struct snd_usb_caiaq_cb_info *info = urb->context;
 	struct snd_usb_caiaqdev *dev;
-	struct urb *out = NULL;
-	int i, frame, len, send_it = 0, outframe = 0;
-	size_t offset = 0;
+	struct urb *out;
+	int frame, len, send_it = 0, outframe = 0;
 
 	if (urb->status || !info)
 		return;
@@ -628,17 +623,7 @@ static void read_completed(struct urb *urb)
 	if (!dev->streaming)
 		return;
 
-	/* find an unused output urb that is unused */
-	for (i = 0; i < N_URBS; i++)
-		if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) {
-			out = dev->data_urbs_out[i];
-			break;
-		}
-
-	if (!out) {
-		log("Unable to find an output urb to use\n");
-		goto requeue;
-	}
+	out = dev->data_urbs_out[info->index];
 
 	/* read the recently received packet and send back one which has
 	 * the same layout */
@@ -649,8 +634,7 @@ static void read_completed(struct urb *urb)
 		len = urb->iso_frame_desc[outframe].actual_length;
 		out->iso_frame_desc[outframe].length = len;
 		out->iso_frame_desc[outframe].actual_length = 0;
-		out->iso_frame_desc[outframe].offset = offset;
-		offset += len;
+		out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
 
 		if (len > 0) {
 			spin_lock(&dev->spinlock);
@@ -666,15 +650,11 @@ static void read_completed(struct urb *urb)
 	}
 
 	if (send_it) {
-		out->number_of_packets = outframe;
+		out->number_of_packets = FRAMES_PER_URB;
 		out->transfer_flags = URB_ISO_ASAP;
 		usb_submit_urb(out, GFP_ATOMIC);
-	} else {
-		struct snd_usb_caiaq_cb_info *oinfo = out->context;
-		clear_bit(oinfo->index, &dev->outurb_active_mask);
 	}
 
-requeue:
 	/* re-submit inbound urb */
 	for (frame = 0; frame < FRAMES_PER_URB; frame++) {
 		urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
@@ -696,8 +676,6 @@ static void write_completed(struct urb *urb)
 		dev->output_running = 1;
 		wake_up(&dev->prepare_wait_queue);
 	}
-
-	clear_bit(info->index, &dev->outurb_active_mask);
 }
 
 static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
@@ -849,9 +827,6 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	if (!dev->data_cb_info)
 		return -ENOMEM;
 
-	dev->outurb_active_mask = 0;
-	BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8));
-
 	for (i = 0; i < N_URBS; i++) {
 		dev->data_cb_info[i].dev = dev;
 		dev->data_cb_info[i].index = i;
diff --git a/trunk/sound/usb/caiaq/device.h b/trunk/sound/usb/caiaq/device.h
index 3f9c6339ae90..b2b310194ffa 100644
--- a/trunk/sound/usb/caiaq/device.h
+++ b/trunk/sound/usb/caiaq/device.h
@@ -96,7 +96,6 @@ struct snd_usb_caiaqdev {
 	int input_panic, output_panic, warned;
 	char *audio_in_buf, *audio_out_buf;
 	unsigned int samplerates, bpp;
-	unsigned long outurb_active_mask;
 
 	struct snd_pcm_substream *sub_playback[MAX_STREAMS];
 	struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/trunk/sound/usb/caiaq/input.c b/trunk/sound/usb/caiaq/input.c
index a213813487bd..4432ef7a70a9 100644
--- a/trunk/sound/usb/caiaq/input.c
+++ b/trunk/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] =  { KEY_C, KEY_B, KEY_A };
 static unsigned short keycode_rk2[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
 					 KEY_5, KEY_6, KEY_7 };
 static unsigned short keycode_rk3[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
-					 KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
+					 KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
 
 static unsigned short keycode_kore[] = {
 	KEY_FN_F1,      /* "menu"               */
diff --git a/trunk/sound/usb/card.c b/trunk/sound/usb/card.c
index d8f2bf401458..781d9e61adfb 100644
--- a/trunk/sound/usb/card.c
+++ b/trunk/sound/usb/card.c
@@ -530,11 +530,8 @@ snd_usb_audio_probe(struct usb_device *dev,
 	return chip;
 
  __error:
-	if (chip) {
-		if (!chip->num_interfaces)
-			snd_card_free(chip->card);
-		chip->probing = 0;
-	}
+	if (chip && !chip->num_interfaces)
+		snd_card_free(chip->card);
 	mutex_unlock(&register_mutex);
  __err_val:
 	return NULL;
diff --git a/trunk/sound/usb/endpoint.c b/trunk/sound/usb/endpoint.c
index 7d46e482375d..7c0d21ecd821 100644
--- a/trunk/sound/usb/endpoint.c
+++ b/trunk/sound/usb/endpoint.c
@@ -352,7 +352,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 			continue;
 		}
 		if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
-		    ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) {
+		    ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
 			snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
 				   dev->devnum, iface_no, altno);
 			continue;
diff --git a/trunk/sound/usb/mixer.c b/trunk/sound/usb/mixer.c
index cdd19d7fe500..c22fa76e363a 100644
--- a/trunk/sound/usb/mixer.c
+++ b/trunk/sound/usb/mixer.c
@@ -152,7 +152,6 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p,
 	if (p && p->dB) {
 		cval->dBmin = p->dB->min;
 		cval->dBmax = p->dB->max;
-		cval->initialized = 1;
 	}
 }
 
@@ -1093,7 +1092,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 				" Switch" : " Volume");
 		if (control == UAC_FU_VOLUME) {
 			check_mapped_dB(map, cval);
-			if (cval->dBmin < cval->dBmax || !cval->initialized) {
+			if (cval->dBmin < cval->dBmax) {
 				kctl->tlv.c = mixer_vol_tlv;
 				kctl->vd[0].access |= 
 					SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1192,11 +1191,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 
 	if (state->mixer->protocol == UAC_VERSION_1) {
 		csize = hdr->bControlSize;
-		if (!csize) {
-			snd_printdd(KERN_ERR "usbaudio: unit %u: "
-				    "invalid bControlSize == 0\n", unitid);
-			return -EINVAL;
-		}
 		channels = (hdr->bLength - 7) / csize - 1;
 		bmaControls = hdr->bmaControls;
 	} else {
@@ -1940,13 +1934,15 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 	struct mixer_build state;
 	int err;
 	const struct usbmix_ctl_map *map;
+	struct usb_host_interface *hostif;
 	void *p;
 
+	hostif = mixer->chip->ctrl_intf;
 	memset(&state, 0, sizeof(state));
 	state.chip = mixer->chip;
 	state.mixer = mixer;
-	state.buffer = mixer->hostif->extra;
-	state.buflen = mixer->hostif->extralen;
+	state.buffer = hostif->extra;
+	state.buflen = hostif->extralen;
 
 	/* check the mapping table */
 	for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1959,8 +1955,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 	}
 
 	p = NULL;
-	while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
-					    p, UAC_OUTPUT_TERMINAL)) != NULL) {
+	while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
 		if (mixer->protocol == UAC_VERSION_1) {
 			struct uac1_output_terminal_descriptor *desc = p;
 
@@ -2167,15 +2162,17 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
 /* create the handler for the optional status interrupt endpoint */
 static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
 {
+	struct usb_host_interface *hostif;
 	struct usb_endpoint_descriptor *ep;
 	void *transfer_buffer;
 	int buffer_length;
 	unsigned int epnum;
 
+	hostif = mixer->chip->ctrl_intf;
 	/* we need one interrupt input endpoint */
-	if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
+	if (get_iface_desc(hostif)->bNumEndpoints < 1)
 		return 0;
-	ep = get_endpoint(mixer->hostif, 0);
+	ep = get_endpoint(hostif, 0);
 	if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
 		return 0;
 
@@ -2205,6 +2202,7 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 	};
 	struct usb_mixer_interface *mixer;
 	struct snd_info_entry *entry;
+	struct usb_host_interface *host_iface;
 	int err;
 
 	strcpy(chip->card->mixername, "USB Mixer");
@@ -2221,8 +2219,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 		return -ENOMEM;
 	}
 
-	mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
-	switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
+	host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+	switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
 	case UAC_VERSION_1:
 	default:
 		mixer->protocol = UAC_VERSION_1;
diff --git a/trunk/sound/usb/mixer.h b/trunk/sound/usb/mixer.h
index 81b2d8a32fb0..ae1a14dcfe82 100644
--- a/trunk/sound/usb/mixer.h
+++ b/trunk/sound/usb/mixer.h
@@ -3,7 +3,6 @@
 
 struct usb_mixer_interface {
 	struct snd_usb_audio *chip;
-	struct usb_host_interface *hostif;
 	struct list_head list;
 	unsigned int ignore_ctl_error;
 	struct urb *urb;
diff --git a/trunk/sound/usb/quirks-table.h b/trunk/sound/usb/quirks-table.h
index a42e3ef3832d..dba0b7f11c54 100644
--- a/trunk/sound/usb/quirks-table.h
+++ b/trunk/sound/usb/quirks-table.h
@@ -1707,40 +1707,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
-{
-	USB_DEVICE(0x0582, 0x0130),
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		/* .vendor_name = "BOSS", */
-		/* .product_name = "MICRO BR-80", */
-		.ifnum = QUIRK_ANY_INTERFACE,
-		.type = QUIRK_COMPOSITE,
-		.data = (const struct snd_usb_audio_quirk[]) {
-			{
-				.ifnum = 0,
-				.type = QUIRK_IGNORE_INTERFACE
-			},
-			{
-				.ifnum = 1,
-				.type = QUIRK_AUDIO_STANDARD_INTERFACE
-			},
-			{
-				.ifnum = 2,
-				.type = QUIRK_AUDIO_STANDARD_INTERFACE
-			},
-			{
-				.ifnum = 3,
-				.type = QUIRK_MIDI_FIXED_ENDPOINT,
-				.data = & (const struct snd_usb_midi_endpoint_info) {
-					.out_cables = 0x0001,
-					.in_cables  = 0x0001
-				}
-			},
-			{
-				.ifnum = -1
-			}
-		}
-	}
-},
 
 /* Guillemot devices */
 {
@@ -2451,12 +2417,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	.idProduct = 0x1020,
 },
 
-/* KeithMcMillen Stringport */
-{
-	USB_DEVICE(0x1f38, 0x0001),
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
-
 /* Miditech devices */
 {
 	USB_DEVICE(0x4752, 0x0011),
diff --git a/trunk/sound/usb/quirks.c b/trunk/sound/usb/quirks.c
index 81e07d842581..77762c99afbe 100644
--- a/trunk/sound/usb/quirks.c
+++ b/trunk/sound/usb/quirks.c
@@ -426,7 +426,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
  */
 static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
 {
-	int err  = 0, reg;
+	int err, reg;
 	int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
 
 	for (reg = 0; reg < ARRAY_SIZE(val); reg++) {