From 707fba3fa76a4c8855552f5d4c1a12430c09bce8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 2 Aug 2012 09:04:39 +0200
Subject: ALSA: hda - Support dock on Lenovo Thinkpad T530 with ALC269VC

Lenovo Thinkpad T530 with ALC269VC codec has a dock port but BIOS
doesn't set up the pins properly.  Enable the pins as well as on
Thinkpad X230 Tablet.

Reported-and-tested-by: Mario <anyc@hadiko.de>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 344b221d2102..b9a5c4503336 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6206,6 +6206,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
-- 
cgit v1.2.3


From fcfb7866af9a5d0280b7e51dd772990c636b7dec Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Aug 2012 12:51:21 +0200
Subject: ALSA: emu10k1: Avoid access to invalid pages when period=1

When period=1, the driver tries to allocate a bit bigger buffer than
requested by the user due to the irq latency tolerance.  This may lead
to accesses over the actually allocated pages.

This patch adds a check of the page index and assigns the silent page
when it's over the given buffer size.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/emu10k1/memory.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 4f502a2bdc3c..0a436626182b 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
 	for (page = blk->first_page; page <= blk->last_page; page++, idx++) {
 		unsigned long ofs = idx << PAGE_SHIFT;
 		dma_addr_t addr;
-		addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+		if (ofs >= runtime->dma_bytes)
+			addr = emu->silent_page.addr;
+		else
+			addr = snd_pcm_sgbuf_get_addr(substream, ofs);
 		if (! is_valid_page(emu, addr)) {
 			printk(KERN_ERR "emu: failure page = %d\n", idx);
 			mutex_unlock(&hdr->block_mutex);
-- 
cgit v1.2.3


From 4407be6ba217514b1bc01488f8b56467d309e416 Mon Sep 17 00:00:00 2001
From: "Philipp A. Mohrenweiser" <phiamo@googlemail.com>
Date: Mon, 6 Aug 2012 13:14:18 +0200
Subject: ALSA: hda - add dock support for Thinkpad T430s

Add a model/fixup string "lenovo-dock", for Thinkpad T430s, to allow
sound in docking station.

Tested on Lenovo T430s with ThinkPad Mini Dock Plus Series 3

Cc: stable@kernel.org
Signed-off-by: Philipp A. Mohrenweiser <phiamo@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b9a5c4503336..bb93f927f7be 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6207,6 +6207,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
-- 
cgit v1.2.3


From 8dfaa573918afa34c8eaf8b2120b2e38cc4f651f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 Aug 2012 14:49:36 +0200
Subject: ALSA: hda - Fix regression of HDMI codec probing

The commit c4bfe94a causes a regression on some codecs at probing.
Since this was just a workaround to shut up a kernel warning, it'd be
better to revert and fix properly.  So we ended up with re-adding the
cleanup callback.

Tested-and-reported-by: Matt Horan <matt@matthoran.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 12 +++++++++---
 1 file changed, 9 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 69b928449789..8f23374fa642 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -877,8 +877,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 	struct hdmi_eld *eld;
 	struct hdmi_spec_per_cvt *per_cvt = NULL;
 
-	hinfo->nid = 0; /* clear the leftover value */
-
 	/* Validate hinfo */
 	pin_idx = hinfo_to_pin_index(spec, hinfo);
 	if (snd_BUG_ON(pin_idx < 0))
@@ -1163,6 +1161,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
 }
 
+static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					     struct hda_codec *codec,
+					     struct snd_pcm_substream *substream)
+{
+	snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+	return 0;
+}
+
 static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 			  struct hda_codec *codec,
 			  struct snd_pcm_substream *substream)
@@ -1202,6 +1208,7 @@ static const struct hda_pcm_ops generic_ops = {
 	.open = hdmi_pcm_open,
 	.close = hdmi_pcm_close,
 	.prepare = generic_hdmi_playback_pcm_prepare,
+	.cleanup = generic_hdmi_playback_pcm_cleanup,
 };
 
 static int generic_hdmi_build_pcms(struct hda_codec *codec)
@@ -1220,7 +1227,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
 		pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
 		pstr->substreams = 1;
 		pstr->ops = generic_ops;
-		pstr->nid = 1; /* FIXME: just for avoiding a debug WARNING */
 		/* other pstr fields are set in open */
 	}
 
-- 
cgit v1.2.3


From c8415a48fcb7a29889f4405d38c57db351e4b50a Mon Sep 17 00:00:00 2001
From: Felix Kaechele <felix@fetzig.org>
Date: Mon, 6 Aug 2012 23:02:01 +0200
Subject: ALSA: hda - add dock support for Thinkpad X230

As with the ThinkPad Models X230 Tablet and T530 the X230 needs a qurik to
correctly set up the pins for the dock port.

Signed-off-by: Felix Kaechele <felix@fetzig.org>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bb93f927f7be..daa5103eb936 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6207,6 +6207,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
-- 
cgit v1.2.3


From e9fc83cb2e5877801a255a37ddbc5be996ea8046 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 7 Aug 2012 14:03:29 +0200
Subject: ALSA: hda - remove quirk for Dell Vostro 1015

This computer is confirmed working with model=auto on kernel 3.2.
Also, parsing fails with hda-emu with the current model.

Cc: stable@kernel.org (3.2+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 14361184ae1e..6f538eb3871e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2972,7 +2972,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
-	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
-- 
cgit v1.2.3


From bb10b09a8ea10228ef3e56365fae40f1e24e5589 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 7 Aug 2012 14:03:30 +0200
Subject: ALSA: hda - remove redundant auto quirks for conexant 506x

Now that the auto model is the default, these quirks are redundant
and can be removed.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6f538eb3871e..5e22a8f43d2e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2967,7 +2967,6 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 };
 
 static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO),
 	SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
@@ -2987,14 +2986,10 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
-	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
-	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
-	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
-	SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
 	{}
 };
 
-- 
cgit v1.2.3


From 709aea6b05f8f6c852c293fb5d47a6461373b4dd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Aug 2012 18:09:23 +0200
Subject: ALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y

When CONFIG_SND_VERBOSE_PRINTK=y is set, the debug print in
hda_auto_parser.c looks really ugly like:

  ALSA sound/pci/hda/hda_auto_parser.c:331    mono: mono_out=0x0
  ALSA sound/pci/hda/hda_auto_parser.c:334    dig-out=0x12/0x0
  ALSA sound/pci/hda/hda_auto_parser.c:335    inputs:
  ALSA sound/pci/hda/hda_auto_parser.c:339  Mic=0x11ALSA sound/pci/hda/hda_auto_parser.c:339  Line=0x10
  ALSA sound/pci/hda/hda_auto_parser.c:341
  ALSA sound/pci/hda/hda_auto_parser.c:343    dig-in=0x13

Better to put one item at each line.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 647218d69f68..4f7d2dfcef7b 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	if (cfg->dig_outs)
 		snd_printd("   dig-out=0x%x/0x%x\n",
 			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
-	snd_printd("   inputs:");
+	snd_printd("   inputs:\n");
 	for (i = 0; i < cfg->num_inputs; i++) {
-		snd_printd(" %s=0x%x",
+		snd_printd("     %s=0x%x\n",
 			    hda_get_autocfg_input_label(codec, cfg, i),
 			    cfg->inputs[i].pin);
 	}
-	snd_printd("\n");
 	if (cfg->dig_in_pin)
 		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
 
-- 
cgit v1.2.3


From 012e7eb1e501d0120e0383b81477f63091f5e365 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 8 Aug 2012 08:43:37 +0200
Subject: ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad

The same ID is twice in the quirk table, so the second one is not used.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index daa5103eb936..4f81dd44c837 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6099,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = {
 	[ALC269_FIXUP_PCM_44K] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_pcm_44k,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_QUANTA_MUTE
 	},
 	[ALC269_FIXUP_STEREO_DMIC] = {
 		.type = ALC_FIXUP_FUNC,
@@ -6210,8 +6212,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
+	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 
 #if 0
-- 
cgit v1.2.3


From ed36081350d2ca4f692f04c6a2d24d1e3a339da1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Aug 2012 17:12:52 +0200
Subject: ALSA: hda - Add codec->pcm_format_first flag

Introduced a new flag to set up the PCM stream format at first before
the stream_id and channel tag.  Some codecs (e.g. CA0132) seem
preferring this over stream_id -> format order.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++++++++-----------------
 sound/pci/hda/hda_codec.h |  1 +
 2 files changed, 46 insertions(+), 24 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88a9c20eb7a2..598b9e2d85e6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
 
+/* update the stream-id if changed */
+static void update_pcm_stream_id(struct hda_codec *codec,
+				 struct hda_cvt_setup *p, hda_nid_t nid,
+				 u32 stream_tag, int channel_id)
+{
+	unsigned int oldval, newval;
+
+	if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
+		oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
+		newval = (stream_tag << 4) | channel_id;
+		if (oldval != newval)
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_CHANNEL_STREAMID,
+					    newval);
+		p->stream_tag = stream_tag;
+		p->channel_id = channel_id;
+	}
+}
+
+/* update the format-id if changed */
+static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p,
+			      hda_nid_t nid, int format)
+{
+	unsigned int oldval;
+
+	if (p->format_id != format) {
+		oldval = snd_hda_codec_read(codec, nid, 0,
+					    AC_VERB_GET_STREAM_FORMAT, 0);
+		if (oldval != format) {
+			msleep(1);
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_STREAM_FORMAT,
+					    format);
+		}
+		p->format_id = format;
+	}
+}
+
 /**
  * snd_hda_codec_setup_stream - set up the codec for streaming
  * @codec: the CODEC to set up
@@ -1400,7 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 {
 	struct hda_codec *c;
 	struct hda_cvt_setup *p;
-	unsigned int oldval, newval;
 	int type;
 	int i;
 
@@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 	p = get_hda_cvt_setup(codec, nid);
 	if (!p)
 		return;
-	/* update the stream-id if changed */
-	if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
-		oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
-		newval = (stream_tag << 4) | channel_id;
-		if (oldval != newval)
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_CHANNEL_STREAMID,
-					    newval);
-		p->stream_tag = stream_tag;
-		p->channel_id = channel_id;
-	}
-	/* update the format-id if changed */
-	if (p->format_id != format) {
-		oldval = snd_hda_codec_read(codec, nid, 0,
-					    AC_VERB_GET_STREAM_FORMAT, 0);
-		if (oldval != format) {
-			msleep(1);
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_STREAM_FORMAT,
-					    format);
-		}
-		p->format_id = format;
-	}
+
+	if (codec->pcm_format_first)
+		update_pcm_format(codec, p, nid, format);
+	update_pcm_stream_id(codec, p, nid, stream_tag, channel_id);
+	if (!codec->pcm_format_first)
+		update_pcm_format(codec, p, nid, format);
+
 	p->active = 1;
 	p->dirty = 0;
 
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index c422d330ca54..7fbc1bcaf1a9 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -861,6 +861,7 @@ struct hda_codec {
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
 	unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
 	unsigned int no_jack_detect:1;	/* Machine has no jack-detection */
+	unsigned int pcm_format_first:1; /* PCM format must be set first */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	int power_transition;	/* power-state in transition */
-- 
cgit v1.2.3


From 55cf87fe0e9783e25f442be1e48b8319d86131ea Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Aug 2012 17:15:55 +0200
Subject: ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index d0d3540e39e7..2685590925ff 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -988,12 +988,12 @@ static void ca0132_config(struct hda_codec *codec)
 
 	/* Mic-in */
 	spec->input_pins[0] = 0x12;
-	spec->input_labels[0] = "Mic-In";
+	spec->input_labels[0] = "Mic";
 	spec->adcs[0] = 0x07;
 
 	/* Line-In */
 	spec->input_pins[1] = 0x11;
-	spec->input_labels[1] = "Line-In";
+	spec->input_labels[1] = "Line";
 	spec->adcs[1] = 0x08;
 	spec->num_inputs = 2;
 }
-- 
cgit v1.2.3


From 27ebeb0b1b5bb26908e485a7e8bd2ec30366ffef Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Aug 2012 17:20:18 +0200
Subject: ALSA: hda - Use the standard PCM ops for CA0132

Now with the workaround using codec->pcm_format_first flag, we can
clean up the home-baked codes in patch_ca0132.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 142 +++++++++----------------------------------
 1 file changed, 29 insertions(+), 113 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 2685590925ff..31512a0f1d07 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -464,50 +464,17 @@ exit:
 }
 
 /*
- * PCM stuffs
+ * PCM callbacks
  */
-static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid,
-				 u32 stream_tag,
-				 int channel_id, int format)
+static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				    struct hda_codec *codec,
+				    struct snd_pcm_substream *substream)
 {
-	unsigned int oldval, newval;
-
-	if (!nid)
-		return;
-
-	snd_printdd("ca0132_setup_stream: "
-		"NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
-		nid, stream_tag, channel_id, format);
-
-	/* update the format-id if changed */
-	oldval = snd_hda_codec_read(codec, nid, 0,
-				    AC_VERB_GET_STREAM_FORMAT,
-				    0);
-	if (oldval != format) {
-		msleep(20);
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_STREAM_FORMAT,
-				    format);
-	}
-
-	oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
-	newval = (stream_tag << 4) | channel_id;
-	if (oldval != newval) {
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_CHANNEL_STREAMID,
-				    newval);
-	}
-}
-
-static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
-{
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+	struct ca0132_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+					     hinfo);
 }
 
-/*
- * PCM callbacks
- */
 static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
 			unsigned int stream_tag,
@@ -515,10 +482,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+						stream_tag, format, substream);
 }
 
 static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
@@ -526,92 +491,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dacs[0]);
-
-	return 0;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 }
 
 /*
  * Digital out
  */
-static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			unsigned int stream_tag,
-			unsigned int format,
-			struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format);
-
-	return 0;
-}
-
-static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			struct snd_pcm_substream *substream)
-{
-	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dig_out);
-
-	return 0;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
 }
 
-/*
- * Analog capture
- */
-static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
 			unsigned int stream_tag,
 			unsigned int format,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->adcs[substream->number],
-			     stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+					     stream_tag, format, substream);
 }
 
-static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			struct snd_pcm_substream *substream)
-{
-	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->adcs[substream->number]);
-
-	return 0;
-}
-
-/*
- * Digital capture
- */
-static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
-			unsigned int stream_tag,
-			unsigned int format,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
 }
 
-static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dig_in);
-
-	return 0;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
 
 /*
@@ -621,6 +539,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.ops = {
+		.open = ca0132_playback_pcm_open,
 		.prepare = ca0132_playback_pcm_prepare,
 		.cleanup = ca0132_playback_pcm_cleanup
 	},
@@ -630,10 +549,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	.ops = {
-		.prepare = ca0132_capture_pcm_prepare,
-		.cleanup = ca0132_capture_pcm_cleanup
-	},
 };
 
 static struct hda_pcm_stream ca0132_pcm_digital_playback = {
@@ -641,6 +556,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.ops = {
+		.open = ca0132_dig_playback_pcm_open,
+		.close = ca0132_dig_playback_pcm_close,
 		.prepare = ca0132_dig_playback_pcm_prepare,
 		.cleanup = ca0132_dig_playback_pcm_cleanup
 	},
@@ -650,10 +567,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	.ops = {
-		.prepare = ca0132_dig_capture_pcm_prepare,
-		.cleanup = ca0132_dig_capture_pcm_cleanup
-	},
 };
 
 static int ca0132_build_pcms(struct hda_codec *codec)
@@ -961,6 +874,9 @@ static void ca0132_config(struct hda_codec *codec)
 	struct ca0132_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 
+	codec->pcm_format_first = 1;
+	codec->no_sticky_stream = 1;
+
 	/* line-outs */
 	cfg->line_outs = 1;
 	cfg->line_out_pins[0] = 0x0b; /* front */
-- 
cgit v1.2.3


From 8e13fc1c5f694a6ae4032c7f94103c137136733f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Aug 2012 17:26:54 +0200
Subject: ALSA: hda - Add missing SPDIF I/O setup for CA0132

CA0132 driver had some codes to handle the S/PDIF I/O, but the actual
setups of pins and converters were missing.  Now the pins are added.

Also, fixed a few points triggering invalid codec verbs and mixer
elements since the digital I/O audio widgets on CA0132 have no amp.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 20 ++++++++++++++------
 1 file changed, 14 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 31512a0f1d07..9c0ec0a55bef 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_UNMUTE);
 	}
-	if (dac)
+	if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP))
 		snd_hda_codec_write(codec, dac, 0,
 				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
 }
@@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_IN_UNMUTE(0));
 	}
-	if (adc)
+	if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP))
 		snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
 				    AMP_IN_UNMUTE(0));
 }
@@ -841,18 +841,16 @@ static int ca0132_build_controls(struct hda_codec *codec)
 						    spec->dig_out);
 		if (err < 0)
 			return err;
-		err = add_out_volume(codec, spec->dig_out, "IEC958");
+		err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
 		if (err < 0)
 			return err;
+		/* spec->multiout.share_spdif = 1; */
 	}
 
 	if (spec->dig_in) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
 		if (err < 0)
 			return err;
-		err = add_in_volume(codec, spec->dig_in, "IEC958");
-		if (err < 0)
-			return err;
 	}
 	return 0;
 }
@@ -912,6 +910,16 @@ static void ca0132_config(struct hda_codec *codec)
 	spec->input_labels[1] = "Line";
 	spec->adcs[1] = 0x08;
 	spec->num_inputs = 2;
+
+	/* SPDIF I/O */
+	spec->dig_out = 0x05;
+	spec->multiout.dig_out_nid = spec->dig_out;
+	cfg->dig_out_pins[0] = 0x0c;
+	cfg->dig_outs = 1;
+	cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+	spec->dig_in = 0x09;
+	cfg->dig_in_pin = 0x0e;
+	cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
 }
 
 static void ca0132_init_chip(struct hda_codec *codec)
-- 
cgit v1.2.3


From 94c142a160d63edac0e1fca7848960dcf75dd2a9 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 9 Aug 2012 10:56:12 +0200
Subject: ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V

To turn off pin control for the pin was tested, and helped against
this issue.

BugLink: https://bugs.launchpad.net/bugs/1034779
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 80d90cb42853..430771776915 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	vt1708_stop_hp_work(spec);
+
+	if (spec->codec_type == VT1802) {
+		/* Fix pop noise on headphones */
+		int i;
+		for (i = 0; i < spec->autocfg.hp_outs; i++)
+			snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0);
+	}
+
 	return 0;
 }
 #endif
-- 
cgit v1.2.3


From 144dad99ef6ad10c8c8ebe787d08157c4a94201f Mon Sep 17 00:00:00 2001
From: James Ralston <james.d.ralston@intel.com>
Date: Thu, 9 Aug 2012 09:38:59 -0700
Subject: ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH

This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH

Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 9 +++++++++
 1 file changed, 9 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8aced182fd1..60882c62f180 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, CPT},"
 			 "{Intel, PPT},"
 			 "{Intel, LPT},"
+			 "{Intel, LPT_LP},"
 			 "{Intel, HPT},"
 			 "{Intel, PBG},"
 			 "{Intel, SCH},"
@@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
 	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Lynx Point-LP */
+	{ PCI_DEVICE(0x8086, 0x9c20),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Lynx Point-LP */
+	{ PCI_DEVICE(0x8086, 0x9c21),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* Haswell */
 	{ PCI_DEVICE(0x8086, 0x0c0c),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
-- 
cgit v1.2.3


From de64c0ee7dbcbfbbe63bd9ea45783d87babc6452 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Fri, 10 Aug 2012 12:22:58 +0300
Subject: ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read()

This function returns its own error codes instead of normal negative
error codes.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/cs46xx/cs46xx_lib.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index f75f5ffdfdfb..a71d1c14a0f6 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
 
 	if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
 		       codec_index != CS46XX_SECONDARY_CODEC_INDEX))
-		return -EINVAL;
+		return 0xffff;
 
 	chip->active_ctrl(chip, 1);
 
-- 
cgit v1.2.3


From 14bc9c6dc694e2d7930802f7afd275de25ef8394 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 10 Aug 2012 13:29:32 +0200
Subject: ALSA: hda - Fix panned "Beep Playback Switch"

When "Beep Playback Switch" had a different value on left and right
channels (such as muting left but not right, or vice versa), this
could result in the right channel being ignored.

This patch enables beep to be sounding from right channel only, and
also give correct result back to userspace (e g amixer).

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c | 20 ++++++++++++++------
 1 file changed, 14 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 0bc2315b181d..d26ae65b43b7 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -237,10 +237,9 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
-	if (beep) {
+	if (beep && !beep->enabled) {
 		ucontrol->value.integer.value[0] =
-			ucontrol->value.integer.value[1] =
-			beep->enabled;
+			ucontrol->value.integer.value[1] = 0;
 		return 0;
 	}
 	return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
@@ -252,9 +251,18 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
-	if (beep)
-		snd_hda_enable_beep_device(codec,
-					   *ucontrol->value.integer.value);
+	if (beep) {
+		u8 chs = get_amp_channels(kcontrol);
+		int enable = 0;
+		long *valp = ucontrol->value.integer.value;
+		if (chs & 1) {
+			enable |= *valp;
+			valp++;
+		}
+		if (chs & 2)
+			enable |= *valp;
+		snd_hda_enable_beep_device(codec, enable);
+	}
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
-- 
cgit v1.2.3


From e037cb4a54e26b5f55f856e0e7445cfcfb2f3d31 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Fri, 10 Aug 2012 14:11:58 +0200
Subject: ALSA : hda - bug fix on checking the supported power states of a
 codec

The return value of snd_hda_param_read() is -1 for an error, otherwise
it's the supported power states of a codec.

The supported power states is a 32-bit value. Bit 31 will be set to 1
if the codec supports EPSS, thus making "sup" negative. And the bit
28:5 is reserved as "0".
So a negative value other than -1 shall be further checked.

Please refer to High-Definition spec 7.3.4.12 "Supported Power
States", thanks!

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88a9c20eb7a2..629131ad7b8b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3497,7 +3497,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg
 {
 	int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE);
 
-	if (sup < 0)
+	if (sup == -1)
 		return false;
 	if (sup & power_state)
 		return true;
-- 
cgit v1.2.3


From 088c820b732dbfd515fc66d459d5f5777f79b406 Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Mon, 13 Aug 2012 14:11:10 +0800
Subject: ALSA: hda - fix Copyright debug message

As spec said, 1 indicates no copyright is asserted.

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_proc.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 7e46258fc700..6894ec66258c 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
 	if (digi1 & AC_DIG1_EMPHASIS)
 		snd_iprintf(buffer, " Preemphasis");
 	if (digi1 & AC_DIG1_COPYRIGHT)
-		snd_iprintf(buffer, " Copyright");
+		snd_iprintf(buffer, " Non-Copyright");
 	if (digi1 & AC_DIG1_NONAUDIO)
 		snd_iprintf(buffer, " Non-Audio");
 	if (digi1 & AC_DIG1_PROFESSIONAL)
-- 
cgit v1.2.3


From 265d931a9e9a7e290faa5e2145f4b2ebf38ea84c Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Mon, 13 Aug 2012 17:10:46 +0200
Subject: ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch

Some Conexant devices (e g CX20590) have no mute capability on
their Beep widgets.
This patch makes sure we don't try setting mutes on those widgets.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c | 13 +++++++++++--
 1 file changed, 11 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index d26ae65b43b7..0849aac449f2 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -231,15 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
 
+static bool ctl_has_mute(struct snd_kcontrol *kcontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	return query_amp_caps(codec, get_amp_nid(kcontrol),
+			      get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE;
+}
+
 /* get/put callbacks for beep mute mixer switches */
 int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
-	if (beep && !beep->enabled) {
+	if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
 		ucontrol->value.integer.value[0] =
-			ucontrol->value.integer.value[1] = 0;
+			ucontrol->value.integer.value[1] = beep->enabled;
 		return 0;
 	}
 	return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
@@ -263,6 +270,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 			enable |= *valp;
 		snd_hda_enable_beep_device(codec, enable);
 	}
+	if (!ctl_has_mute(kcontrol))
+		return 0;
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
-- 
cgit v1.2.3


From 3bdcff70b6cd049e6f4437b955850f5db83653cc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 14 Aug 2012 17:42:11 +0200
Subject: ALSA: lx6464es: Add a missing error check

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44541

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/lx6464es/lx6464es.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d1ab43706735..5579b08bb35b 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip)
 	/* hardcoded device name & channel count */
 	err = snd_pcm_new(chip->card, (char *)card_name, 0,
 			  1, 1, &pcm);
+	if (err < 0)
+		return err;
 
 	pcm->private_data = chip;
 
-- 
cgit v1.2.3


From 5e68fb3cab23b327e9f15803607e697d7eea1966 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 16 Aug 2012 14:11:09 +0200
Subject: ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx

Instead of blindly initializing a volume knob widget, first check
that there actually is a volume knob widget.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 94040ccf8e8f..ea5775a1a7db 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec)
 	unsigned int gpio;
 	int i;
 
-	snd_hda_sequence_write(codec, spec->init);
+	if (spec->init)
+		snd_hda_sequence_write(codec, spec->init);
 
 	/* power down adcs initially */
 	if (spec->powerdown_adcs)
@@ -5748,7 +5749,6 @@ again:
 		/* fallthru */
 	case 0x111d76b4: /* 6 Port without Analog Mixer */
 	case 0x111d76b5:
-		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92xx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
@@ -5773,7 +5773,6 @@ again:
 			spec->stream_delay = 40; /* 40 milliseconds */
 
 		/* disable VSW */
-		spec->init = stac92hd71bxx_core_init;
 		unmute_init++;
 		snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
 		snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
@@ -5788,7 +5787,6 @@ again:
 
 		/* fallthru */
 	default:
-		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92xx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
@@ -5796,6 +5794,9 @@ again:
 		break;
 	}
 
+	if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB)
+		spec->init = stac92hd71bxx_core_init;
+
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
 		snd_hda_sequence_write_cache(codec, unmute_init);
 
-- 
cgit v1.2.3


From 4d8ce1c9966663bad69e738952179f3cc52710bf Mon Sep 17 00:00:00 2001
From: Julia Lawall <Julia.Lawall@lip6.fr>
Date: Sun, 19 Aug 2012 09:02:56 +0200
Subject: ALSA: sound/pci/ctxfi/ctatc.c: fix error return code

Initialize err before returning on failure, as done elsewhere in the
function.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctatc.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 8e40262d4117..2f6e9c762d3f 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
 	atc_connect_resources(atc);
 
 	atc->timer = ct_timer_new(atc);
-	if (!atc->timer)
+	if (!atc->timer) {
+		err = -ENOMEM;
 		goto error1;
+	}
 
 	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops);
 	if (err < 0)
-- 
cgit v1.2.3


From ae970eb45d8a1ea4506be23c3f776225b9575d0e Mon Sep 17 00:00:00 2001
From: Julia Lawall <Julia.Lawall@lip6.fr>
Date: Sun, 19 Aug 2012 09:02:55 +0200
Subject: ALSA: sound/pci/sis7019.c: fix error return code

Initialize rc before returning on failure, as done elsewhere in the
function.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/sis7019.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 512434efcc31..805ab6e9a78f 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card,
 	if (rc)
 		goto error_out_cleanup;
 
-	if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
-			sis)) {
+	rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
+			 sis);
+	if (rc) {
 		dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq);
 		goto error_out_cleanup;
 	}
-- 
cgit v1.2.3


From b17cbdd85f84c8323189416da6e9701d2793b0e5 Mon Sep 17 00:00:00 2001
From: Julia Lawall <Julia.Lawall@lip6.fr>
Date: Sun, 19 Aug 2012 09:02:54 +0200
Subject: ALSA: sound/pci/rme9652/hdspm.c: fix error return code

Convert a nonnegative error return code to a negative one, as returned
elsewhere in the function.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
 { ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
    when != &ret
*if(...)
{
  ... when != ret = e2
      when forall
 return ret;
}

// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdspm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index b8ac8710f47f..b12308b5ba2a 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
 		snd_printk(KERN_ERR "HDSPM: "
 				"unable to kmalloc Mixer memory of %d Bytes\n",
 				(int)sizeof(struct hdspm_mixer));
-		return err;
+		return -ENOMEM;
 	}
 
 	hdspm->port_names_in = NULL;
-- 
cgit v1.2.3


From c41999a23929f30808bae6009d8065052d4d73fd Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Mon, 20 Aug 2012 11:17:00 +0200
Subject: ALSA: hda - don't create dysfunctional mixer controls for ca0132

It's possible that these amps are settable somehow, e g through
secret codec verbs, but for now, don't create the controls (as
they won't be working anyway, and cause errors in amixer).

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1038651
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 9c0ec0a55bef..49750a96d649 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
 	int type = dir ? HDA_INPUT : HDA_OUTPUT;
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+	if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) {
+		snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid);
+		return 0;
+	}
 	sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
 	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
 }
@@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
 	int type = dir ? HDA_INPUT : HDA_OUTPUT;
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+	if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) {
+		snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid);
+		return 0;
+	}
 	sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
 	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
 }
-- 
cgit v1.2.3


From 535b6c51fe8293c88ce919cdfc4390c67a1cb6d1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Aug 2012 21:25:22 +0200
Subject: ALSA: hda - Fix leftover codec->power_transition

When the codec turn-on operation is canceled by the immediate
power-on, the driver left the power_transition flag as is.
This caused the persistent avoidance of power-save behavior.

Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c3077d5dec6e..f560051a949e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -4454,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 	 * then there is no need to go through power up here.
 	 */
 	if (codec->power_on) {
+		if (codec->power_transition < 0)
+			codec->power_transition = 0;
 		spin_unlock(&codec->power_lock);
 		return;
 	}
-- 
cgit v1.2.3