From 5dbd5ec6e1cf2e49128025d80813a275744a7ac5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 29 Mar 2010 09:16:24 +0200
Subject: ALSA: hda - Fix invalid bit values passed to
 snd_hda_codec_amp_stereo()

The mask and value parameters passed to snd_hda_codec_amp_stereo()
should be 8-bit values for mute and volume.  Passing AMP_IN_MUTE() is
wrong, which is found in many places in patch_realtek.c as a left-over
from the conversion to snd_hda_codec_amp_stereo().

Reported-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 52 +++++++++++++++++++++----------------------
 1 file changed, 26 insertions(+), 26 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a23444e9e7..bc55c1e96df 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12459,11 +12459,11 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
 	unsigned char bits;
 
 	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
@@ -13482,11 +13482,11 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
 	unsigned char bits;
 
 	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 
 	snd_hda_codec_write(codec, 0x20, 0,
 			AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13511,11 +13511,11 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
 	/* Check port replicator headphone socket */
 	present |= snd_hda_jack_detect(codec, 0x1a);
 
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 
 	snd_hda_codec_write(codec, 0x20, 0,
 			AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13646,11 +13646,11 @@ static void alc269_speaker_automute(struct hda_codec *codec)
 	unsigned char bits;
 
 	present = snd_hda_jack_detect(codec, nid);
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 /* unsolicited event for HP jack sensing */
@@ -17115,9 +17115,9 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec)
 	present = snd_hda_jack_detect(codec, 0x21);
 	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
@@ -17128,13 +17128,13 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
 	present = snd_hda_jack_detect(codec, 0x21);
 	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
@@ -17145,13 +17145,13 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
 	present = snd_hda_jack_detect(codec, 0x15);
 	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc662_f5z_speaker_automute(struct hda_codec *codec)
@@ -17190,14 +17190,14 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
 
 	if (present1 || present2) {
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
 	} else {
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), 0);
+					 HDA_AMP_MUTE, 0);
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), 0);
+					 HDA_AMP_MUTE, 0);
 	}
 }
 
-- 
cgit v1.2.3


From 6694635d3ae1b038d7a0e38b80637db867c7c8e2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 29 Mar 2010 17:21:45 +0200
Subject: ALSA: hda - Fix ADC/MUX assignment of ALC269 codec

ALC269 codec has a few different variants, and each of them may have
different ADC and MUX widgets.  For example, one model has ADC 0x08
with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or
0x24.  The difference of ADC appears usually as the capability of
the digital mic pin (0x12), and the current driver sometimes misses
the internal mic pin due to the mismatching ADC.

This patch adds a bit more clever way to find the matching ADC instead
of the static list.  Now the driver checks all active input pins and
fills only the ADC/MUX's that contain all of them.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 95 ++++++++++++++++++++++++++++++++++++-------
 1 file changed, 80 insertions(+), 15 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bc55c1e96df..22aea7b089c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4984,6 +4984,69 @@ static void set_capture_mixer(struct hda_codec *codec)
 	}
 }
 
+/* fill adc_nids (and capsrc_nids) containing all active input pins */
+static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
+				 int num_nids)
+{
+	struct alc_spec *spec = codec->spec;
+	int n;
+	hda_nid_t fallback_adc = 0, fallback_cap = 0;
+
+	for (n = 0; n < num_nids; n++) {
+		hda_nid_t adc, cap;
+		hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+		int nconns, i, j;
+
+		adc = nids[n];
+		if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN)
+			continue;
+		cap = adc;
+		nconns = snd_hda_get_connections(codec, cap, conn,
+						 ARRAY_SIZE(conn));
+		if (nconns == 1) {
+			cap = conn[0];
+			nconns = snd_hda_get_connections(codec, cap, conn,
+							 ARRAY_SIZE(conn));
+		}
+		if (nconns <= 0)
+			continue;
+		if (!fallback_adc) {
+			fallback_adc = adc;
+			fallback_cap = cap;
+		}
+		for (i = 0; i < AUTO_PIN_LAST; i++) {
+			hda_nid_t nid = spec->autocfg.input_pins[i];
+			if (!nid)
+				continue;
+			for (j = 0; j < nconns; j++) {
+				if (conn[j] == nid)
+					break;
+			}
+			if (j >= nconns)
+				break;
+		}
+		if (i >= AUTO_PIN_LAST) {
+			int num_adcs = spec->num_adc_nids;
+			spec->private_adc_nids[num_adcs] = adc;
+			spec->private_capsrc_nids[num_adcs] = cap;
+			spec->num_adc_nids++;
+			spec->adc_nids = spec->private_adc_nids;
+			if (adc != cap)
+				spec->capsrc_nids = spec->private_capsrc_nids;
+		}
+	}
+	if (!spec->num_adc_nids) {
+		printk(KERN_WARNING "hda_codec: %s: no valid ADC found;"
+		       " using fallback 0x%x\n", fallback_adc);
+		spec->private_adc_nids[0] = fallback_adc;
+		spec->adc_nids = spec->private_adc_nids;
+		if (fallback_adc != fallback_cap) {
+			spec->private_capsrc_nids[0] = fallback_cap;
+			spec->capsrc_nids = spec->private_adc_nids;
+		}
+	}
+}
+
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 #define set_beep_amp(spec, nid, idx, dir) \
 	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
@@ -13333,9 +13396,9 @@ static hda_nid_t alc269vb_capsrc_nids[1] = {
 	0x22,
 };
 
-/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24),
- *       not a mux!
- */
+static hda_nid_t alc269_adc_candidates[] = {
+	0x08, 0x09, 0x07,
+};
 
 #define alc269_modes		alc260_modes
 #define alc269_capture_source	alc880_lg_lw_capture_source
@@ -13842,7 +13905,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int err;
 	static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
-	hda_nid_t real_capsrc_nids;
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc269_ignore);
@@ -13866,18 +13928,19 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) {
 		add_verb(spec, alc269vb_init_verbs);
-		real_capsrc_nids = alc269vb_capsrc_nids[0];
 		alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21);
 	} else {
 		add_verb(spec, alc269_init_verbs);
-		real_capsrc_nids = alc269_capsrc_nids[0];
 		alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
 	}
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
+	fillup_priv_adc_nids(codec, alc269_adc_candidates,
+			     sizeof(alc269_adc_candidates));
+
 	/* set default input source */
-	snd_hda_codec_write_cache(codec, real_capsrc_nids,
+	snd_hda_codec_write_cache(codec, spec->capsrc_nids[0],
 				  0, AC_VERB_SET_CONNECT_SEL,
 				  spec->input_mux->items[0].index);
 
@@ -14156,14 +14219,16 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc269_pcm_digital_playback;
 	spec->stream_digital_capture = &alc269_pcm_digital_capture;
 
-	if (!is_alc269vb) {
-		spec->adc_nids = alc269_adc_nids;
-		spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
-		spec->capsrc_nids = alc269_capsrc_nids;
-	} else {
-		spec->adc_nids = alc269vb_adc_nids;
-		spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
-		spec->capsrc_nids = alc269vb_capsrc_nids;
+	if (!spec->adc_nids) { /* wasn't filled automatically? use default */
+		if (!is_alc269vb) {
+			spec->adc_nids = alc269_adc_nids;
+			spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
+			spec->capsrc_nids = alc269_capsrc_nids;
+		} else {
+			spec->adc_nids = alc269vb_adc_nids;
+			spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
+			spec->capsrc_nids = alc269vb_capsrc_nids;
+		}
 	}
 
 	if (!spec->cap_mixer)
-- 
cgit v1.2.3


From 1f85d72d2c9c9a1d6d32cf325936bc224ad5d591 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 30 Mar 2010 07:48:05 +0200
Subject: ALSA: hda - Add missing printk argument in previous patch

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 22aea7b089c..ca93c4cc144 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5037,7 +5037,8 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
 	}
 	if (!spec->num_adc_nids) {
 		printk(KERN_WARNING "hda_codec: %s: no valid ADC found;"
-		       " using fallback 0x%x\n", fallback_adc);
+		       " using fallback 0x%x\n",
+		       codec->chip_name, fallback_adc);
 		spec->private_adc_nids[0] = fallback_adc;
 		spec->adc_nids = spec->private_adc_nids;
 		if (fallback_adc != fallback_cap) {
-- 
cgit v1.2.3


From b8e80cf386419453678b01bef830f53445ebb15d Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Tue, 30 Mar 2010 13:29:28 -0400
Subject: ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981

BugLink: https://launchpad.net/bugs/551606

The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_ad1981() for all models using the Thinkpad
quirk.

Reported-by: Jane Silber
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e6d1bdff1b6..af34606c30c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1896,6 +1896,14 @@ static int patch_ad1981(struct hda_codec *codec)
 	case AD1981_THINKPAD:
 		spec->mixers[0] = ad1981_thinkpad_mixers;
 		spec->input_mux = &ad1981_thinkpad_capture_source;
+		/* set the upper-limit for mixer amp to 0dB for avoiding the
+		 * possible damage by overloading
+		 */
+		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
+					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+					  (1 << AC_AMPCAP_MUTE_SHIFT));
 		break;
 	case AD1981_TOSHIBA:
 		spec->mixers[0] = ad1981_hp_mixers;
-- 
cgit v1.2.3


From 3815595e78d2baae6feb866e737f92d8ef48b337 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 4 Apr 2010 12:14:03 +0200
Subject: ALSA: hda - Add MSI blacklist for Aopen MZ915-M

The device needs MSI disablement.  Added to the quirk list.

Reported-by: Harald Dunkel <harri@afaics.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4bb90675f70..f8fd586ae02 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2362,6 +2362,7 @@ static struct snd_pci_quirk msi_black_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
 	SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */
 	SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */
+	SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */
 	{}
 };
 
-- 
cgit v1.2.3


From a0fd4345f928d72a56e27b23e4cd28c94bf36be5 Mon Sep 17 00:00:00 2001
From: Julia Lawall <julia@diku.dk>
Date: Fri, 2 Apr 2010 14:47:59 +0200
Subject: ALSA: echoaudio - Eliminate use after free

Use the call to snd_card_free in the error handling code at the end of the
function, as in the other error cases.

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

// <smpl>
@@
expression E,E2;
@@

snd_card_free(E)
...
(
  E = E2
|
* E
)
// </smpl>

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

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 8dab82d7d19..668a5ec0449 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2184,10 +2184,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
 			goto ctl_error;
 #endif
 
-	if ((err = snd_card_register(card)) < 0) {
-		snd_card_free(card);
+	err = snd_card_register(card);
+	if (err < 0)
 		goto ctl_error;
-	}
 	snd_printk(KERN_INFO "Card registered: %s\n", card->longname);
 
 	pci_set_drvdata(pci, chip);
-- 
cgit v1.2.3


From d12841827a6de120199609dadb6ff4ec99bd90ea Mon Sep 17 00:00:00 2001
From: Tony Vroon <tony@linx.net>
Date: Mon, 5 Apr 2010 16:30:43 +0100
Subject: ALSA: hda - Enable amplifiers on Acer Inspire 6530G

After more tests it appears that EAPD needs to be enabled
on both the 0x14 and 0x15 NIDs to enable the main speaker
and headphone amplifiers. The maximum volume setting is
now equal to what the machine achieves under other operating
systems.
Disabling Front or LFE playback triggers EAPD and disables
the amplifier. As such, these two playback switches have
been removed from the mixer.

Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ca93c4cc144..547206296d7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1621,6 +1621,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
  */
 
 static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+/* Route to built-in subwoofer as well as speakers */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 /* Bias voltage on for external mic port */
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
 /* Front Mic: set to PIN_IN (empty by default) */
@@ -1632,10 +1637,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
 /* Enable speaker output */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 /* Enable headphone output */
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
@@ -8462,9 +8469,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 
 static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-- 
cgit v1.2.3


From f9700d5a4575e7fb343df10a1d29d425e4b81082 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 5 Apr 2010 23:25:13 +0200
Subject: ALSA: hda - Fix a wrong array range check in patch_realtek.c

The commit 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 introduced a wrong
comparision for the array range check, which effectively skips the whole
initialization of DAC connections.  Fixed now.

Reference: bko#15689
	https://bugzilla.kernel.org/show_bug.cgi?id=15689

Reported-by: Adrian Ulrich <kernel@blinkenlights.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 547206296d7..c7730dbb9dd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10110,13 +10110,12 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
 	int idx;
 
 	alc_set_pin_output(codec, nid, pin_type);
+	if (dac_idx >= spec->multiout.num_dacs)
+		return;
 	if (spec->multiout.dac_nids[dac_idx] == 0x25)
 		idx = 4;
-	else {
-		if (spec->multiout.num_dacs >= dac_idx)
-			return;
+	else
 		idx = spec->multiout.dac_nids[dac_idx] - 2;
-	}
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
 
 }
-- 
cgit v1.2.3


From b0cc58a25d04160d39a80e436847eaa2fbc5aa09 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <error27@gmail.com>
Date: Tue, 6 Apr 2010 19:31:26 +0300
Subject: ALSA: mixart: range checking proc file

The original code doesn't take into consideration that the value of
MIXART_BA0_SIZE - pos can be less than zero which would lead to a large
unsigned value for "count".

Also I moved the check that read size is a multiple of 4 bytes below
the code that adjusts "count".

Signed-off-by: Dan Carpenter <error27@gmail.com>
Cc: <stable@kernel.org>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/mixart/mixart.c | 24 ++++++++++++++----------
 1 file changed, 14 insertions(+), 10 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 7e8e7da592a..ea4256b08a3 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1161,13 +1161,15 @@ static long snd_mixart_BA0_read(struct snd_info_entry *entry, void *file_private
 				unsigned long count, unsigned long pos)
 {
 	struct mixart_mgr *mgr = entry->private_data;
+	unsigned long maxsize;
 
-	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
-	if(count <= 0)
+	if (pos >= MIXART_BA0_SIZE)
 		return 0;
-	if(pos + count > MIXART_BA0_SIZE)
-		count = (long)(MIXART_BA0_SIZE - pos);
-	if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count))
+	maxsize = MIXART_BA0_SIZE - pos;
+	if (count > maxsize)
+		count = maxsize;
+	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+	if (copy_to_user_fromio(buf, MIXART_MEM(mgr, pos), count))
 		return -EFAULT;
 	return count;
 }
@@ -1180,13 +1182,15 @@ static long snd_mixart_BA1_read(struct snd_info_entry *entry, void *file_private
 				unsigned long count, unsigned long pos)
 {
 	struct mixart_mgr *mgr = entry->private_data;
+	unsigned long maxsize;
 
-	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
-	if(count <= 0)
+	if (pos > MIXART_BA1_SIZE)
 		return 0;
-	if(pos + count > MIXART_BA1_SIZE)
-		count = (long)(MIXART_BA1_SIZE - pos);
-	if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count))
+	maxsize = MIXART_BA1_SIZE - pos;
+	if (count > maxsize)
+		count = maxsize;
+	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+	if (copy_to_user_fromio(buf, MIXART_REG(mgr, pos), count))
 		return -EFAULT;
 	return count;
 }
-- 
cgit v1.2.3


From 7ad7b218f4aae4f395b3b4cef261572556bbd20a Mon Sep 17 00:00:00 2001
From: Maurus Cuelenaere <mcuelenaere@gmail.com>
Date: Tue, 6 Apr 2010 18:12:52 +0200
Subject: ALSA: hda: Add support for Medion WIM2160

This adds support for the Medion WIM2160 soundcard.
There's no PCI quirk added because it has the same PCI id as the
Medion MD2.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 53 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 53 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c7730dbb9dd..2971e48e50a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -230,6 +230,7 @@ enum {
 	ALC888_ACER_ASPIRE_7730G,
 	ALC883_MEDION,
 	ALC883_MEDION_MD2,
+	ALC883_MEDION_WIM2160,
 	ALC883_LAPTOP_EAPD,
 	ALC883_LENOVO_101E_2ch,
 	ALC883_LENOVO_NB0763,
@@ -8455,6 +8456,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct hda_verb alc883_medion_wim2160_verbs[] = {
+	/* Unmute front mixer */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Set speaker pin to front mixer */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Init headphone pin */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
+	{ } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_medion_wim2160_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1a;
+	spec->autocfg.speaker_pins[0] = 0x15;
+}
+
 static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -9164,6 +9201,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
 	[ALC888_ACER_ASPIRE_7730G]	= "acer-aspire-7730g",
 	[ALC883_MEDION]		= "medion",
 	[ALC883_MEDION_MD2]	= "medion-md2",
+	[ALC883_MEDION_WIM2160]	= "medion-wim2160",
 	[ALC883_LAPTOP_EAPD]	= "laptop-eapd",
 	[ALC883_LENOVO_101E_2ch] = "lenovo-101e",
 	[ALC883_LENOVO_NB0763]	= "lenovo-nb0763",
@@ -9818,6 +9856,21 @@ static struct alc_config_preset alc882_presets[] = {
 		.setup = alc883_medion_md2_setup,
 		.init_hook = alc_automute_amp,
 	},
+	[ALC883_MEDION_WIM2160] = {
+		.mixers = { alc883_medion_wim2160_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+		.adc_nids = alc883_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_automute_amp_unsol_event,
+		.setup = alc883_medion_wim2160_setup,
+		.init_hook = alc_automute_amp,
+	},
 	[ALC883_LAPTOP_EAPD] = {
 		.mixers = { alc883_base_mixer },
 		.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
-- 
cgit v1.2.3


From 531d8791accf1464bc6854ff69d08dd866189d17 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Fri, 9 Apr 2010 10:57:33 +0200
Subject: ALSA: hda - Fix auto-parser of ALC269vb for HP pin NID 0x21

ALC269vb has an alternative HP pin 0x21 in addition.
Fix the parser to recognize it.

Signed-off-by: Kailang Yang <kailang@realtek.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 2971e48e50a..fbbdfbc8a1c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12869,6 +12869,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
+	case 0x21: /* ALC269vb has this pin, too */
 		dac = 0x03;
 		break;
 	default:
-- 
cgit v1.2.3


From 226b1ec8c18bcb6d1aa448a29b2c8aeae1946228 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Fri, 9 Apr 2010 11:01:20 +0200
Subject: ALSA: hda - Fix setup for ALC269vb amic and dmic models

Corrected HP and mic pins for ALC269vb amic and dmic models.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++--------
 1 file changed, 20 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fbbdfbc8a1c..9b58f29833e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13789,19 +13789,19 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec,
 	}
 }
 
-static void alc269_laptop_dmic_setup(struct hda_codec *codec)
+static void alc269_laptop_amic_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
-	spec->int_mic.pin = 0x12;
-	spec->int_mic.mux_idx = 5;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
 	spec->auto_mic = 1;
 }
 
-static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
+static void alc269_laptop_dmic_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
@@ -13809,14 +13809,14 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
-	spec->int_mic.mux_idx = 6;
+	spec->int_mic.mux_idx = 5;
 	spec->auto_mic = 1;
 }
 
-static void alc269_laptop_amic_setup(struct hda_codec *codec)
+static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.hp_pins[0] = 0x21;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
@@ -13825,6 +13825,18 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec)
 	spec->auto_mic = 1;
 }
 
+static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 6;
+	spec->auto_mic = 1;
+}
+
 static void alc269_laptop_inithook(struct hda_codec *codec)
 {
 	alc269_speaker_automute(codec);
@@ -14162,7 +14174,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
 		.unsol_event = alc269_laptop_unsol_event,
-		.setup = alc269_laptop_amic_setup,
+		.setup = alc269vb_laptop_amic_setup,
 		.init_hook = alc269_laptop_inithook,
 	},
 	[ALC269VB_DMIC] = {
-- 
cgit v1.2.3


From 7f311a46916a3be00a1a8e3f1bdf461d08f1d263 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 9 Apr 2010 17:32:23 +0200
Subject: ALSA: hda - Fix initial capture source connections of ALC880/260

The widget connections of ADC of ALC880 and ALC2260 aren't initialized,
thus it might point to invalid pin.  This can be a problem when mode=auto
and there is only one input pin.  Then user can't change the connection
at all.

This patch adds the code to initialize the input pin connection of these
codecs.

Reference: Novell bnc#594363
	https://bugzilla.novell.com/show_bug.cgi?id=594363

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++
 1 file changed, 23 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9b58f29833e..8d60b1f25ce 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4809,6 +4809,25 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+static void alc880_auto_init_input_src(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int c;
+
+	for (c = 0; c < spec->num_adc_nids; c++) {
+		unsigned int mux_idx;
+		const struct hda_input_mux *imux;
+		mux_idx = c >= spec->num_mux_defs ? 0 : c;
+		imux = &spec->input_mux[mux_idx];
+		if (!imux->num_items && mux_idx > 0)
+			imux = &spec->input_mux[0];
+		if (imux)
+			snd_hda_codec_write(codec, spec->adc_nids[c], 0,
+					    AC_VERB_SET_CONNECT_SEL,
+					    imux->items[0].index);
+	}
+}
+
 /* parse the BIOS configuration and set up the alc_spec */
 /* return 1 if successful, 0 if the proper config is not found,
  * or a negative error code
@@ -4887,6 +4906,7 @@ static void alc880_auto_init(struct hda_codec *codec)
 	alc880_auto_init_multi_out(codec);
 	alc880_auto_init_extra_out(codec);
 	alc880_auto_init_analog_input(codec);
+	alc880_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -6398,6 +6418,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+#define alc260_auto_init_input_src	alc880_auto_init_input_src
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -6484,6 +6506,7 @@ static void alc260_auto_init(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	alc260_auto_init_multi_out(codec);
 	alc260_auto_init_analog_input(codec);
+	alc260_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
-- 
cgit v1.2.3


From 7fa90e873f520dad5ec58f47340996cda083e875 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 12 Apr 2010 08:49:00 +0200
Subject: ALSA: hda - Enhance fix-up table for Realtek codecs

A few enhancement / fixes for fix-up table of some Realtek codecs:
 - Apply fix-ups only for the auto model
 - Apply additional verbs after normal init verbs
 - Add a debug print to show the fix-up application

This is basically a preliminary work for the next fix for Sony VAIO.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 35 ++++++++++++++++++++++++++++-------
 1 file changed, 28 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8d60b1f25ce..cff57710d1f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1390,22 +1390,31 @@ struct alc_fixup {
 
 static void alc_pick_fixup(struct hda_codec *codec,
 			   const struct snd_pci_quirk *quirk,
-			   const struct alc_fixup *fix)
+			   const struct alc_fixup *fix,
+			   int pre_init)
 {
 	const struct alc_pincfg *cfg;
 
 	quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
 	if (!quirk)
 		return;
-
 	fix += quirk->value;
 	cfg = fix->pins;
-	if (cfg) {
+	if (pre_init && cfg) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+		snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n",
+			    codec->chip_name, quirk->name);
+#endif
 		for (; cfg->nid; cfg++)
 			snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
 	}
-	if (fix->verbs)
+	if (!pre_init && fix->verbs) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+		snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n",
+			    codec->chip_name, quirk->name);
+#endif
 		add_verb(codec->spec, fix->verbs);
+	}
 }
 
 static int alc_read_coef_idx(struct hda_codec *codec,
@@ -10439,7 +10448,8 @@ static int patch_alc882(struct hda_codec *codec)
 		board_config = ALC882_AUTO;
 	}
 
-	alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups);
+	if (board_config == ALC882_AUTO)
+		alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1);
 
 	if (board_config == ALC882_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -10512,6 +10522,9 @@ static int patch_alc882(struct hda_codec *codec)
 	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
+	if (board_config == ALC882_AUTO)
+		alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0);
+
 	spec->vmaster_nid = 0x0c;
 
 	codec->patch_ops = alc_patch_ops;
@@ -15417,7 +15430,8 @@ static int patch_alc861(struct hda_codec *codec)
 		board_config = ALC861_AUTO;
 	}
 
-	alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups);
+	if (board_config == ALC861_AUTO)
+		alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1);
 
 	if (board_config == ALC861_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -15454,6 +15468,9 @@ static int patch_alc861(struct hda_codec *codec)
 
 	spec->vmaster_nid = 0x03;
 
+	if (board_config == ALC861_AUTO)
+		alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0);
+
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC861_AUTO) {
 		spec->init_hook = alc861_auto_init;
@@ -16388,7 +16405,8 @@ static int patch_alc861vd(struct hda_codec *codec)
 		board_config = ALC861VD_AUTO;
 	}
 
-	alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups);
+	if (board_config == ALC861VD_AUTO)
+		alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1);
 
 	if (board_config == ALC861VD_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -16436,6 +16454,9 @@ static int patch_alc861vd(struct hda_codec *codec)
 
 	spec->vmaster_nid = 0x02;
 
+	if (board_config == ALC861VD_AUTO)
+		alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0);
+
 	codec->patch_ops = alc_patch_ops;
 
 	if (board_config == ALC861VD_AUTO)
-- 
cgit v1.2.3


From ff818c24c2af370153646d302d831b69b023816f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 12 Apr 2010 08:59:25 +0200
Subject: ALSA: hda - Add fix-up for Sony VAIO with ALC269

Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF
ground or Hi-Z to make the headphone working.  Other than that, model=auto
works fine, so let's use model=auto with a specific fix-up table.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++++++++++++++-
 1 file changed, 28 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cff57710d1f..4b35176d345 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14077,6 +14077,27 @@ static void alc269_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
+enum {
+	ALC269_FIXUP_SONY_VAIO,
+};
+
+const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = {
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
+	{}
+};
+
+static const struct alc_fixup alc269_fixups[] = {
+	[ALC269_FIXUP_SONY_VAIO] = {
+		.verbs = alc269_sony_vaio_fixup_verbs
+	},
+};
+
+static struct snd_pci_quirk alc269_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+	{}
+};
+
+
 /*
  * configuration and preset
  */
@@ -14136,7 +14157,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = {
 		      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 XTB", 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),
 	SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
@@ -14290,6 +14311,9 @@ static int patch_alc269(struct hda_codec *codec)
 		board_config = ALC269_AUTO;
 	}
 
+	if (board_config == ALC269_AUTO)
+		alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1);
+
 	if (board_config == ALC269_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc269_parse_auto_config(codec);
@@ -14342,6 +14366,9 @@ static int patch_alc269(struct hda_codec *codec)
 		set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
+	if (board_config == ALC269_AUTO)
+		alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0);
+
 	spec->vmaster_nid = 0x02;
 
 	codec->patch_ops = alc_patch_ops;
-- 
cgit v1.2.3


From b331439dfd41dc813b3557ca5927a3a644f35792 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 14 Apr 2010 14:33:57 +0200
Subject: ALSA: hda - Fix control element allocations in VIA codec parser

The commit 5b0cb1d850c26893b1468b3a519433a1b7a176be
    ALSA: hda - add more NID->Control mapping
breaks the control element allocation by returning a wrong value.
Let's fix it.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 9ddc37300f6..be129543898 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -476,7 +476,7 @@ static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec,
 	knew->name = kstrdup(tmpl->name, GFP_KERNEL);
 	if (!knew->name)
 		return NULL;
-	return 0;
+	return knew;
 }
 
 static void via_free_kctls(struct hda_codec *codec)
-- 
cgit v1.2.3


From 3d83e577a8206f0f3822a3840e12f76477142ba2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 14 Apr 2010 14:36:23 +0200
Subject: ALSA: hda - Avoid invalid "Independent HP" control for VIA codecs

Some VIA codecs have no multiple source selection for headphone pins,
thus it's useless (and wrong) to create "Independent HP" control on them.

This patch adds the check of connections to skip the control in such a
case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 39 +++++++++++++++++++++++----------------
 1 file changed, 23 insertions(+), 16 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index be129543898..73453814e09 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1215,14 +1215,13 @@ static struct snd_kcontrol_new via_hp_mixer[2] = {
 	},
 };
 
-static int via_hp_build(struct via_spec *spec)
+static int via_hp_build(struct hda_codec *codec)
 {
+	struct via_spec *spec = codec->spec;
 	struct snd_kcontrol_new *knew;
 	hda_nid_t nid;
-
-	knew = via_clone_control(spec, &via_hp_mixer[0]);
-	if (knew == NULL)
-		return -ENOMEM;
+	int nums;
+	hda_nid_t conn[HDA_MAX_CONNECTIONS];
 
 	switch (spec->codec_type) {
 	case VT1718S:
@@ -1239,6 +1238,14 @@ static int via_hp_build(struct via_spec *spec)
 		break;
 	}
 
+	nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS);
+	if (nums <= 1)
+		return 0;
+
+	knew = via_clone_control(spec, &via_hp_mixer[0]);
+	if (knew == NULL)
+		return -ENOMEM;
+
 	knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
 	knew->private_value = nid;
 
@@ -2561,7 +2568,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -3087,7 +3094,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -3654,7 +3661,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -4140,7 +4147,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -4510,7 +4517,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	return 1;
 }
@@ -4930,7 +4937,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 
@@ -5425,7 +5432,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 
@@ -5781,7 +5788,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	return 1;
 }
@@ -6000,12 +6007,12 @@ static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec,
 
 	/* Line-Out: PortE */
 	err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
-			      "Master Front Playback Volume",
+			      "Front Playback Volume",
 			      HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT));
 	if (err < 0)
 		return err;
 	err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
-			      "Master Front Playback Switch",
+			      "Front Playback Switch",
 			      HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT));
 	if (err < 0)
 		return err;
@@ -6130,7 +6137,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	return 1;
 }
-- 
cgit v1.2.3


From d1501ea844eefdf925f6b711875b4b2b928fddf8 Mon Sep 17 00:00:00 2001
From: Joerg Schirottke <master@kanotix.com>
Date: Thu, 15 Apr 2010 08:37:41 +0200
Subject: ALSA: hda - add a quirk for Clevo M570U laptop

Added the matching model for Clevo laptop M570U.

Signed-off-by: Joerg Schirottke <master@kanotix.com>
Tested-by: Maximilian Gerhard <maxbox@directbox.com>
Cc: <stable@kernel.org>
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 4b35176d345..aad1627f56f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9350,6 +9350,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
 
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
-- 
cgit v1.2.3


From 8815cd030fdd73932a791d1f06194c8db807cde7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 15 Apr 2010 09:02:41 +0200
Subject: ALSA: hda - Add position_fix quirk for Biostar mobo

The Biostar mobo seems to give a wrong DMA position, resulting in
stuttering or skipping sounds on 2.6.34.  Since the commit
7b3a177b0d4f92b3431b8dca777313a07533a710, "ALSA: pcm_lib: fix "something
must be really wrong" condition", makes the position check more strictly,
the DMA position problem is revealed more clearly now.

The fix is to use only LPIB for obtaining the position, i.e. passing
position_fix=1.  This patch adds a static quirk to achieve it as default.

Reported-by: Frank Griffin <ftg@roadrunner.com>
Cc: Eric Piel <Eric.Piel@tremplin-utc.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f8fd586ae02..f669442b7c8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2272,6 +2272,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
 	{}
 };
-- 
cgit v1.2.3


From b7d2526f5c20385894a5e57b1a4292f5a1741f1b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 19 Apr 2010 18:11:29 +0200
Subject: ALSA: hda - Fix resume from StR of HP 2510p with docking-station

When HP laptop with AD1981 codec is suspended and the docking-station
is connected before the resume, the outputs get confused, and wrongly
routed still to the speaker.  This is because of a change in 2.6.34-rc1
ea52bf260ecbb175339af3178c15788df21b7516
    ALSA: hda: Add powerdown for Analog Devices HDA codecs

The problem was the added resume callback that doesn't consider the
modified init hook.  The fix is simply remove the resume callback here
and make the resume normally.  This doesn't change any behavior intended
in the commit above (for shutting down the sound at suspend) but only
fixes the resume.

Reported-and-tested-by: Frans Pop <elendil@planet.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 9 ---------
 1 file changed, 9 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index af34606c30c..e9fdfc4b1c5 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -519,14 +519,6 @@ static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
 	ad198x_power_eapd(codec);
 	return 0;
 }
-
-static int ad198x_resume(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	snd_hda_codec_resume_amp(codec);
-	snd_hda_codec_resume_cache(codec);
-	return 0;
-}
 #endif
 
 static struct hda_codec_ops ad198x_patch_ops = {
@@ -539,7 +531,6 @@ static struct hda_codec_ops ad198x_patch_ops = {
 #endif
 #ifdef SND_HDA_NEEDS_RESUME
 	.suspend = ad198x_suspend,
-	.resume = ad198x_resume,
 #endif
 	.reboot_notify = ad198x_shutup,
 };
-- 
cgit v1.2.3


From aac78daf8f37256283f56820ae858add7139c56c Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Wed, 21 Apr 2010 20:41:52 -0400
Subject: ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio XPS 1645

BugLink: https://launchpad.net/bugs/553002

The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.

This change is necessary for 2.6.32.11 and 2.6.33.2 alike.

Reported-by: Robert Chambers
Tested-by: Robert Chambers
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c4be3fab94e..81ecd9388a8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1607,6 +1607,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 1555", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
 				"Dell Studio 1557", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe,
+				"Dell Studio XPS 1645", STAC_DELL_M6_BOTH),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From 3353541fe533350a22a03e2fb7dc085b35912575 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Thu, 22 Apr 2010 07:15:26 -0400
Subject: ALSA: hda: Use ALC880_F1734 quirk for Fujitsu Siemens AMILO Xi 1526

BugLink: https://launchpad.net/bugs/567494

The OR has verified that the existing model quirk, ALC880_UNIWILL,
is insufficient for audible playback and capture by default. Instead,
the ALC880_F1734 model quirk needs to be used.

This change is necessary for both 2.6.32.11 and 2.6.33.2.

Reported-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Tested-by: Arnaud Malpeyre <amalpeyre@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index aad1627f56f..7404dba16f8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4143,7 +4143,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
 	SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
-	SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL),
+	SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734),
 	SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
 	SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
 	SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
-- 
cgit v1.2.3


From 7efbfd1ae98ef9efe06352e2a1ad83e8c14ceeb1 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Wed, 21 Apr 2010 11:04:06 -0400
Subject: ALSA: snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C

Without this quirk sound stops working after suspend resume. With this quirk,
one still needs to manually unmute the master volume control after a suspend /
/ resume cycle. That is fixed in another patch in this set.

Note that this patch was submitted to the alsa bug tracker a long time ago:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/maestro3.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index b64e78139d6..728de232e09 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -884,6 +884,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = {
 MODULE_DEVICE_TABLE(pci, snd_m3_ids);
 
 static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = {
+	SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c),
 	SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d),
 	SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d),
 	SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03),
-- 
cgit v1.2.3


From 715aa675338ce6e1a3b4f77cf87ea611f93058a8 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Wed, 21 Apr 2010 11:04:08 -0400
Subject: ALSA: snd-meastro3: Ignore spurious HV interrupts during suspend /
 resume

Ignore spurious HV interrupts during suspend / resume, this avoids
mistaking them for a mute button press. This is not very pretty but
it seems the only way to fix the master volume control gets muted
after suspend issue I'm seeing. Note that the es1968 driver is doing
exactly the same.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/maestro3.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 728de232e09..b56e3367678 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -849,6 +849,7 @@ struct snd_m3 {
 	struct snd_kcontrol *master_switch;
 	struct snd_kcontrol *master_volume;
 	struct tasklet_struct hwvol_tq;
+	unsigned int in_suspend;
 
 #ifdef CONFIG_PM
 	u16 *suspend_mem;
@@ -1614,6 +1615,11 @@ static void snd_m3_update_hw_volume(unsigned long private_data)
 	outb(0x88, chip->iobase + SHADOW_MIX_REG_MASTER);
 	outb(0x88, chip->iobase + HW_VOL_COUNTER_MASTER);
 
+	/* Ignore spurious HV interrupts during suspend / resume, this avoids
+	   mistaking them for a mute button press. */
+	if (chip->in_suspend)
+		return;
+
 	if (!chip->master_switch || !chip->master_volume)
 		return;
 
@@ -2425,6 +2431,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state)
 	if (chip->suspend_mem == NULL)
 		return 0;
 
+	chip->in_suspend = 1;
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 	snd_pcm_suspend_all(chip->pcm);
 	snd_ac97_suspend(chip->ac97);
@@ -2498,6 +2505,7 @@ static int m3_resume(struct pci_dev *pci)
 	snd_m3_hv_init(chip);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	chip->in_suspend = 0;
 	return 0;
 }
 #endif /* CONFIG_PM */
-- 
cgit v1.2.3


From 0e0280dc2b0c7395a880d25544b47f3e3e3f79db Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Wed, 21 Apr 2010 19:55:43 -0400
Subject: ALSA: hda: Use LPIB quirk for DG965OT board version AAD63733-203

BugLink: https://launchpad.net/bugs/459083

The OR has verified with 2.6.32.11 and the latest alsa-driver stable
daily snapshot that position_fix=1 is necessary for the external mic
to work and for PulseAudio not to crash constantly.

This patch is necessary also for 2.6.32.11 and 2.6.33.2.

Reported-by: <imwithid@yahoo.com>
Tested-by: <imwithid@yahoo.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f669442b7c8..cec68152dcb 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2273,6 +2273,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
 	{}
 };
-- 
cgit v1.2.3


From 5c1bccf645d4ab65e4c7502acb42e8b9afdb5bdc Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Thu, 22 Apr 2010 17:54:45 -0400
Subject: ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio 1558

BugLink: https://launchpad.net/bugs/568600

The OR has verified that the dell-m6 model quirk is necessary for audio
to be audible by default on the Dell Studio XPS 1645.

This change is necessary for 2.6.32.11 and 2.6.33.2 alike.

Reported-by: Andy Ross <andy@plausible.org>
Tested-by: Andy Ross <andy@plausible.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 81ecd9388a8..7fb7d017a34 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1609,6 +1609,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 1557", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe,
 				"Dell Studio XPS 1645", STAC_DELL_M6_BOTH),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413,
+				"Dell Studio 1558", STAC_DELL_M6_BOTH),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From 8dd34ab111dc6ccb35a1a7a59222cb9bb0160e6f Mon Sep 17 00:00:00 2001
From: "Brian J. Tarricone" <brian@tarricone.org>
Date: Sun, 2 May 2010 17:32:10 -0700
Subject: ALSA: hda - fix array indexing while creating inputs for Cirrus
 codecs

This fixes a problem where cards show up as only having a single mixer
element, suppressing all sound output.

Signed-off-by: Brian J. Tarricone <brian@tarricone.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 7de782a5b8f..350ee8ac415 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -766,7 +766,7 @@ static int build_input(struct hda_codec *codec)
 		for (n = 0; n < AUTO_PIN_LAST; n++) {
 			if (!spec->adc_nid[n])
 				continue;
-			err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]);
+			err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]);
 			if (err < 0)
 				return err;
 		}
-- 
cgit v1.2.3


From 4442dd4613fe3795b4c8a5f42fc96b7ffb90d01a Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Mon, 3 May 2010 20:39:31 -0400
Subject: ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F

BugLink: https://launchpad.net/bugs/573284

The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.

Reported-by: Andy Couldrake <acouldrake@googlemail.com>
Tested-by: Andy Couldrake <acouldrake@googlemail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 61682e1d09d..e1323e45f12 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2842,6 +2842,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 		      CXT5066_DELL_LAPTOP),
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
+	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
 	{}
 };
-- 
cgit v1.2.3


From c53666813813a0ea3d0391e1911eefc05a5e6b4f Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Tue, 4 May 2010 22:07:58 -0400
Subject: ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite
 P500-PSPGSC-01800T

BugLink: https://launchpad.net/bugs/549267

The OR verified that using the olpc-xo-1_5 model quirk allows the
headphones to be audible when inserted into the jack. Capture was
also verified to work correctly.

Reported-by: Richard Gagne
Tested-by: Richard Gagne
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e1323e45f12..924c122f16f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2842,6 +2842,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 		      CXT5066_DELL_LAPTOP),
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
+	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
 	{}
-- 
cgit v1.2.3


From 231f50bc0e9735fd1b3fd376a8d3b6a14aee0694 Mon Sep 17 00:00:00 2001
From: Anisse Astier <anisse@astier.eu>
Date: Wed, 28 Apr 2010 18:05:06 +0200
Subject: ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582

Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper
HP and Mic support.

Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 924c122f16f..e2b698b721d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2842,6 +2842,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 		      CXT5066_DELL_LAPTOP),
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
+	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
-- 
cgit v1.2.3


From 8f0f5ff6777104084b4b2e1ae079541c2a6ed6d9 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Wed, 28 Apr 2010 18:00:11 -0400
Subject: ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549
 (Venice)

BugLink: https://launchpad.net/bugs/541802

The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_cxt5045() for all Packard Bell models.

Reported-by: Valombre
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e2b698b721d..56e52071c76 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1195,9 +1195,10 @@ static int patch_cxt5045(struct hda_codec *codec)
 
 	switch (codec->subsystem_id >> 16) {
 	case 0x103c:
+	case 0x1631:
 	case 0x1734:
-		/* HP & Fujitsu-Siemens laptops have really bad sound over 0dB
-		 * on NID 0x17. Fix max PCM level to 0 dB
+		/* HP, Packard Bell, & Fujitsu-Siemens laptops have really bad
+		 * sound over 0dB on NID 0x17. Fix max PCM level to 0 dB
 		 * (originally it has 0x2b steps with 0dB offset 0x14)
 		 */
 		snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
-- 
cgit v1.2.3