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Message-ID: <s5h392m1728.wl%tiwai@suse.de>
Date:	Thu, 13 Sep 2012 13:43:59 +0200
From:	Takashi Iwai <tiwai@...e.de>
To:	Linus Torvalds <torvalds@...ux-foundation.org>
Cc:	linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 3.6-rc6

Linus,

The following changes since commit 2e4a263ca80a203ac6109f5932722a716c265395:

  ALSA: snd-usb: fix cross-interface streaming devices (2012-08-31 21:04:53 +0200)

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus

for you to fetch changes up to 3737e2be505d872bf2b3c1cd4151b2d2b413d7b5:

  ALSA: ice1724: Use linear scale for AK4396 volume control. (2012-09-12 16:17:41 +0200)

----------------------------------------------------------------
Sound fixes for 3.6-rc6

Just a few small / trivial regression fixes at this time.

----------------------------------------------------------------
Catalin Iacob (1):
      ALSA: hda_intel: add position_fix quirk for Asus K53E

Dan Carpenter (1):
      ALSA: compress_core: fix open flags test in snd_compr_open()

Matteo Frigo (1):
      ALSA: ice1724: Use linear scale for AK4396 volume control.

Takashi Iwai (3):
      ALSA: hda - Fix missing Master volume for STAC9200/925x
      ALSA: usb-audio: Fix bogus error messages for delay accounting
      ALSA: hda - Fix Oops at codec reset/reconfig

 sound/core/compress_offload.c    | 8 +++-----
 sound/pci/hda/hda_codec.c        | 2 +-
 sound/pci/hda/hda_intel.c        | 1 +
 sound/pci/hda/patch_sigmatel.c   | 2 +-
 sound/pci/ice1712/prodigy_hifi.c | 3 ++-
 sound/usb/pcm.c                  | 6 ++++++
 6 files changed, 14 insertions(+), 8 deletions(-)

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index ec2118d..eb60cb8 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f)
 	int maj = imajor(inode);
 	int ret;
 
-	if (f->f_flags & O_WRONLY)
+	if ((f->f_flags & O_ACCMODE) == O_WRONLY)
 		dirn = SND_COMPRESS_PLAYBACK;
-	else if (f->f_flags & O_RDONLY)
+	else if ((f->f_flags & O_ACCMODE) == O_RDONLY)
 		dirn = SND_COMPRESS_CAPTURE;
-	else {
-		pr_err("invalid direction\n");
+	else
 		return -EINVAL;
-	}
 
 	if (maj == snd_major)
 		compr = snd_lookup_minor_data(iminor(inode),
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index f25c24c..1c65cc5 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2353,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	}
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
+	memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
 	snd_hda_jack_tbl_clear(codec);
 	codec->proc_widget_hook = NULL;
 	codec->spec = NULL;
@@ -2368,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	codec->num_pcms = 0;
 	codec->pcm_info = NULL;
 	codec->preset = NULL;
-	memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
 	codec->slave_dig_outs = NULL;
 	codec->spdif_status_reset = 0;
 	module_put(codec->owner);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 60882c6..228cdf9 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2701,6 +2701,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6f806d3..3d4722f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1075,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = {
 
 static const char * const slave_pfxs[] = {
 	"Front", "Surround", "Center", "LFE", "Side",
-	"Headphone", "Speaker", "IEC958",
+	"Headphone", "Speaker", "IEC958", "PCM",
 	NULL
 };
 
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 764cc93..075d5aa 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 }
 
 static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
 
 static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
     {
@@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
 	.info = ak4396_dac_vol_info,
 	.get = ak4396_dac_vol_get,
 	.put = ak4396_dac_vol_put,
-	.tlv = { .p = db_scale_wm_dac },
+	.tlv = { .p = ak4396_db_scale },
     },
 };
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index fd5e982..f782ce1 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1140,6 +1140,12 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
 	int processed = urb->transfer_buffer_length / stride;
 	int est_delay;
 
+	/* ignore the delay accounting when procssed=0 is given, i.e.
+	 * silent payloads are procssed before handling the actual data
+	 */
+	if (!processed)
+		return;
+
 	spin_lock_irqsave(&subs->lock, flags);
 	est_delay = snd_usb_pcm_delay(subs, runtime->rate);
 	/* update delay with exact number of samples played */
--
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