From ec4e86ba0662ed85f3b3a38fb220dc51d951da84 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 16 Oct 2008 08:02:41 +0200
Subject: ALSA: hda - Fix PCM type of Nvidia HDMI devices

Added the missing PCM type for Nvidia HDMI devices so that they point
the right device number.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 1a65775d28e..2eed2c8b98d 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -116,6 +116,7 @@ static int nvhdmi_build_pcms(struct hda_codec *codec)
 	codec->pcm_info = info;
 
 	info->name = "NVIDIA HDMI";
+	info->pcm_type = HDA_PCM_TYPE_HDMI;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback;
 
 	return 0;
-- 
cgit v1.2.3-70-g09d2


From 2f5ad54ea6e2e38156bfb889964deee991f3087a Mon Sep 17 00:00:00 2001
From: Arjan van de Ven <arjan@linux.intel.com>
Date: Sun, 28 Sep 2008 16:20:09 -0700
Subject: pci: use pci_ioremap_bar() in sound/

Use the newly introduced pci_ioremap_bar() function in sound/.
pci_ioremap_bar() just takes a pci device and a bar number, with the goal
of making it really hard to get wrong, while also having a central place
to stick sanity checks.

Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ad1889.c        | 2 +-
 sound/pci/atiixp.c        | 2 +-
 sound/pci/atiixp_modem.c  | 2 +-
 sound/pci/au88x0/au88x0.c | 3 +--
 sound/pci/bt87x.c         | 3 +--
 sound/pci/cs4281.c        | 4 ++--
 sound/pci/cs5530.c        | 2 +-
 sound/pci/hda/hda_intel.c | 2 +-
 sound/pci/mixart/mixart.c | 3 +--
 9 files changed, 10 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 92f3a976ef2..a7f38e63303 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card,
 		goto free_and_ret;
 
 	chip->bar = pci_resource_start(pci, 0);
-	chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0));
+	chip->iobase = pci_ioremap_bar(pci, 0);
 	if (chip->iobase == NULL) {
 		printk(KERN_ERR PFX "unable to reserve region.\n");
 		err = -EBUSY;
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 085a52b8c80..226fe8237d3 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
 		return err;
 	}
 	chip->addr = pci_resource_start(pci, 0);
-	chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0));
+	chip->remap_addr = pci_ioremap_bar(pci, 0);
 	if (chip->remap_addr == NULL) {
 		snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
 		snd_atiixp_free(chip);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 2f106306c7f..0e6e5cc1c50 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
 		return err;
 	}
 	chip->addr = pci_resource_start(pci, 0);
-	chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0));
+	chip->remap_addr = pci_ioremap_bar(pci, 0);
 	if (chip->remap_addr == NULL) {
 		snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
 		snd_atiixp_free(chip);
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 68368e49007..a36d4d1fd41 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
 	if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0)
 		goto regions_out;
 
-	chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
-	                             pci_resource_len(pci, 0));
+	chip->mmio = pci_ioremap_bar(pci, 0);
 	if (!chip->mmio) {
 		printk(KERN_ERR "MMIO area remap failed.\n");
 		err = -ENOMEM;
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 3aa8d973540..1aa1c040254 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card,
 		pci_disable_device(pci);
 		return err;
 	}
-	chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
-				     pci_resource_len(pci, 0));
+	chip->mmio = pci_ioremap_bar(pci, 0);
 	if (!chip->mmio) {
 		snd_printk(KERN_ERR "cannot remap io memory\n");
 		err = -ENOMEM;
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index ef9308f7c45..192e7842e18 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card,
 	chip->ba0_addr = pci_resource_start(pci, 0);
 	chip->ba1_addr = pci_resource_start(pci, 1);
 
-	chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0));
-	chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1));
+	chip->ba0 = pci_ioremap_bar(pci, 0);
+	chip->ba1 = pci_ioremap_bar(pci, 1);
 	if (!chip->ba0 || !chip->ba1) {
 		snd_cs4281_free(chip);
 		return -ENOMEM;
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index 7ff8b68e997..4d9378d8120 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card,
 	}
 	chip->pci_base = pci_resource_start(pci, 0);
 
-	mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0));
+	mem = pci_ioremap_bar(pci, 0);
 	if (mem == NULL) {
 		kfree(chip);
 		pci_disable_device(pci);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9f316c1b279..f080f8ce0ec 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2158,7 +2158,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	chip->addr = pci_resource_start(pci, 0);
-	chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0));
+	chip->remap_addr = pci_ioremap_bar(pci, 0);
 	if (chip->remap_addr == NULL) {
 		snd_printk(KERN_ERR SFX "ioremap error\n");
 		err = -ENXIO;
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 2d0dce649a6..ae7601f353a 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci,
 	}
 	for (i = 0; i < 2; i++) {
 		mgr->mem[i].phys = pci_resource_start(pci, i);
-		mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys,
-						   pci_resource_len(pci, i));
+		mgr->mem[i].virt = pci_ioremap_bar(pci, i);
 		if (!mgr->mem[i].virt) {
 		        printk(KERN_ERR "unable to remap resource 0x%lx\n",
 			       mgr->mem[i].phys);
-- 
cgit v1.2.3-70-g09d2


From 863b45180ef541a1990e4986d30fb7a93022a733 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 21 Oct 2008 17:01:47 +0200
Subject: ALSA: hda - Fix conflicting volume controls on ALC260

ALC260 auto-parsing mode may create multiple controls for the same volume
widget (0x08 and 0x09) depending on the pin.  For example, Front and
Headphone volumes may control the same volume, just the latter one wins.

This patch adds a proper check of the existing of the volume control
and avoid the doulbed creation of the same volume controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 22 ++++++++++++++--------
 1 file changed, 14 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e72707cb60a..ef4955c73c8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4996,7 +4996,7 @@ static struct hda_verb alc260_test_init_verbs[] = {
  */
 
 static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
-					const char *pfx)
+					const char *pfx, int *vol_bits)
 {
 	hda_nid_t nid_vol;
 	unsigned long vol_val, sw_val;
@@ -5018,10 +5018,14 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
 	} else
 		return 0; /* N/A */
 
-	snprintf(name, sizeof(name), "%s Playback Volume", pfx);
-	err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
-	if (err < 0)
-		return err;
+	if (!(*vol_bits & (1 << nid_vol))) {
+		/* first control for the volume widget */
+		snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+		err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
+		if (err < 0)
+			return err;
+		*vol_bits |= (1 << nid_vol);
+	}
 	snprintf(name, sizeof(name), "%s Playback Switch", pfx);
 	err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
 	if (err < 0)
@@ -5035,6 +5039,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 {
 	hda_nid_t nid;
 	int err;
+	int vols = 0;
 
 	spec->multiout.num_dacs = 1;
 	spec->multiout.dac_nids = spec->private_dac_nids;
@@ -5042,21 +5047,22 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 
 	nid = cfg->line_out_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Front");
+		err = alc260_add_playback_controls(spec, nid, "Front", &vols);
 		if (err < 0)
 			return err;
 	}
 
 	nid = cfg->speaker_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Speaker");
+		err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
 		if (err < 0)
 			return err;
 	}
 
 	nid = cfg->hp_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Headphone");
+		err = alc260_add_playback_controls(spec, nid, "Headphone",
+						   &vols);
 		if (err < 0)
 			return err;
 	}
-- 
cgit v1.2.3-70-g09d2


From da74ae3e32374755e0fbdfed4074cf839a82f615 Mon Sep 17 00:00:00 2001
From: Harvey Harrison <harvey.harrison@gmail.com>
Date: Tue, 21 Oct 2008 20:28:04 -0700
Subject: ALSA: hda - correct bracketing in spdif test in patch_sigmatel.c

Noticed by sparse:
sound/pci/hda/patch_sigmatel.c:1285:43: warning: dubious: !x & y

Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Acked-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a2ac7205d45..788fdc6f326 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1282,7 +1282,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			return err;
 		spec->multiout.share_spdif = 1;
 	}
-	if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) {
+	if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
 		if (err < 0)
 			return err;
-- 
cgit v1.2.3-70-g09d2


From e044c39ae258678d6ebb09fccb2a0fdf7ec51847 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 27 Oct 2008 16:56:24 +0100
Subject: ALSA: hda - Restore default pin configs for realtek codecs

Some machines have broken BIOS resume that doesn't restore the default
pin configuration properly, which results in a wrong detection of HP
pin.  This causes a silent speaker output due to missing HP detection.
Related bug: Novell bug#406101
	https://bugzilla.novell.com/show_bug.cgi?id=406101

This patch fixes the issue by saving/restoring the default pin configs
by the driver itself.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 77 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 77 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ef4955c73c8..4eceab9bd10 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -307,6 +307,13 @@ struct alc_spec {
 	/* for PLL fix */
 	hda_nid_t pll_nid;
 	unsigned int pll_coef_idx, pll_coef_bit;
+
+#ifdef SND_HDA_NEEDS_RESUME
+#define ALC_MAX_PINS	16
+	unsigned int num_pins;
+	hda_nid_t pin_nids[ALC_MAX_PINS];
+	unsigned int pin_cfgs[ALC_MAX_PINS];
+#endif
 };
 
 /*
@@ -2778,6 +2785,64 @@ static void alc_free(struct hda_codec *codec)
 	codec->spec = NULL; /* to be sure */
 }
 
+#ifdef SND_HDA_NEEDS_RESUME
+static void store_pin_configs(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid, end_nid;
+
+	end_nid = codec->start_nid + codec->num_nodes;
+	for (nid = codec->start_nid; nid < end_nid; nid++) {
+		unsigned int wid_caps = get_wcaps(codec, nid);
+		unsigned int wid_type =
+			(wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		if (wid_type != AC_WID_PIN)
+			continue;
+		if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids))
+			break;
+		spec->pin_nids[spec->num_pins] = nid;
+		spec->pin_cfgs[spec->num_pins] =
+			snd_hda_codec_read(codec, nid, 0,
+					   AC_VERB_GET_CONFIG_DEFAULT, 0);
+		spec->num_pins++;
+	}
+}
+
+static void resume_pin_configs(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->num_pins; i++) {
+		hda_nid_t pin_nid = spec->pin_nids[i];
+		unsigned int pin_config = spec->pin_cfgs[i];
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+				    pin_config & 0x000000ff);
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+				    (pin_config & 0x0000ff00) >> 8);
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+				    (pin_config & 0x00ff0000) >> 16);
+		snd_hda_codec_write(codec, pin_nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+				    pin_config >> 24);
+	}
+}
+
+static int alc_resume(struct hda_codec *codec)
+{
+	resume_pin_configs(codec);
+	codec->patch_ops.init(codec);
+	snd_hda_codec_resume_amp(codec);
+	snd_hda_codec_resume_cache(codec);
+	return 0;
+}
+#else
+#define store_pin_configs(codec)
+#endif
+
 /*
  */
 static struct hda_codec_ops alc_patch_ops = {
@@ -2786,6 +2851,9 @@ static struct hda_codec_ops alc_patch_ops = {
 	.init = alc_init,
 	.free = alc_free,
 	.unsol_event = alc_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+	.resume = alc_resume,
+#endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	.check_power_status = alc_check_power_status,
 #endif
@@ -3832,6 +3900,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -5250,6 +5319,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	}
 	spec->num_mixers++;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -10313,6 +10383,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -11447,6 +11518,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -12230,6 +12302,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	spec->mixers[spec->num_mixers] = alc269_capture_mixer;
 	spec->num_mixers++;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -13316,6 +13389,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	spec->mixers[spec->num_mixers] = alc861_capture_mixer;
 	spec->num_mixers++;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -14427,6 +14501,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	store_pin_configs(codec);
 	return 1;
 }
 
@@ -16258,6 +16333,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 
 	spec->mixers[spec->num_mixers] = alc662_capture_mixer;
 	spec->num_mixers++;
+
+	store_pin_configs(codec);
 	return 1;
 }
 
-- 
cgit v1.2.3-70-g09d2


From 5695ff44160e62d9193c0201706853bcfe2a077f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Oct 2008 15:39:26 +0100
Subject: ALSA: hda - Add another HP model for AD1884A

Added a quirk entry for another HP mobile device with AD1884A codec.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2b00c4afdf9..d3fd432cb3e 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3860,6 +3860,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
 
 static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
+	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
 	{}
 };
-- 
cgit v1.2.3-70-g09d2


From 8199de3b29d7f22f3163fcf03f7a61644c93c2ea Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 28 Oct 2008 14:50:13 +0000
Subject: ALSA: hda: Add HDA vendor ID for Wolfson Microelectronics

Add Wolfson Microelectronics to the HDA vendor ID table.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 6447754ae56..ba1ab737b55 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -64,6 +64,7 @@ static struct hda_vendor_id hda_vendor_ids[] = {
 	{ 0x14f1, "Conexant" },
 	{ 0x17e8, "Chrontel" },
 	{ 0x1854, "LG" },
+	{ 0x1aec, "Wolfson Microelectronics" },
 	{ 0x434d, "C-Media" },
 	{ 0x8384, "SigmaTel" },
 	{} /* terminator */
-- 
cgit v1.2.3-70-g09d2


From c1e99bd9d6a0c82f4fa9078ed1f570e57b2f3944 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Oct 2008 08:03:42 +0100
Subject: ALSA: hda - Fix SPDIF mute on IDT/STAC codecs

The SPDIF mute switch code seems broken.  It doesn't set unmute bits
properly.  Also it contains the duplicated lines (merge error?) to be
cleaned up.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 788fdc6f326..df9b0bc7f87 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -566,10 +566,8 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 			nid = codec->slave_dig_outs[smux_idx - 1];
 		if (spec->cur_smux[smux_idx] == smux->num_items - 1)
 			val = AMP_OUT_MUTE;
-		if (smux_idx == 0)
-			nid = spec->multiout.dig_out_nid;
 		else
-			nid = codec->slave_dig_outs[smux_idx - 1];
+			val = AMP_OUT_UNMUTE;
 		/* un/mute SPDIF out */
 		snd_hda_codec_write_cache(codec, nid, 0,
 			AC_VERB_SET_AMP_GAIN_MUTE, val);
-- 
cgit v1.2.3-70-g09d2


From 0cbf00980f0fc4cc064a15ab3dfce19b5fae9130 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Oct 2008 16:18:25 +0100
Subject: ALSA: hda - Add reboot notifier

The current snd-hda-intel driver seems blocking the power-off on some
devices like eeepc.  Although this is likely a BIOS problem, we can add
a workaround by disabling IRQ lines before power-off operation.
This patch adds the reboot notifier to achieve it.

The detailed problem description is found in bug#11889:
    http://bugme.linux-foundation.org/show_bug.cgi?id=11889

Tested-by: Luiz Fernando N. Capitulino <lcapitulino@mandriva.com.br>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 29 +++++++++++++++++++++++++++++
 1 file changed, 29 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9f316c1b279..c04ac38e2b2 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -45,6 +45,7 @@
 #include <linux/slab.h>
 #include <linux/pci.h>
 #include <linux/mutex.h>
+#include <linux/reboot.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include "hda_codec.h"
@@ -397,6 +398,9 @@ struct azx {
 
 	/* for pending irqs */
 	struct work_struct irq_pending_work;
+
+	/* reboot notifier (for mysterious hangup problem at power-down) */
+	struct notifier_block reboot_notifier;
 };
 
 /* driver types */
@@ -1978,6 +1982,28 @@ static int azx_resume(struct pci_dev *pci)
 #endif /* CONFIG_PM */
 
 
+/*
+ * reboot notifier for hang-up problem at power-down
+ */
+static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf)
+{
+	struct azx *chip = container_of(nb, struct azx, reboot_notifier);
+	azx_stop_chip(chip);
+	return NOTIFY_OK;
+}
+
+static void azx_notifier_register(struct azx *chip)
+{
+	chip->reboot_notifier.notifier_call = azx_halt;
+	register_reboot_notifier(&chip->reboot_notifier);
+}
+
+static void azx_notifier_unregister(struct azx *chip)
+{
+	if (chip->reboot_notifier.notifier_call)
+		unregister_reboot_notifier(&chip->reboot_notifier);
+}
+
 /*
  * destructor
  */
@@ -1985,6 +2011,8 @@ static int azx_free(struct azx *chip)
 {
 	int i;
 
+	azx_notifier_unregister(chip);
+
 	if (chip->initialized) {
 		azx_clear_irq_pending(chip);
 		for (i = 0; i < chip->num_streams; i++)
@@ -2348,6 +2376,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	pci_set_drvdata(pci, card);
 	chip->running = 1;
 	power_down_all_codecs(chip);
+	azx_notifier_register(chip);
 
 	dev++;
 	return err;
-- 
cgit v1.2.3-70-g09d2


From 4605b718e8b8f0dd3d811ddf90f630fd0835b7bf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 31 Oct 2008 14:18:24 +0100
Subject: ALSA: hda - Disable broken mic auto-muting in Realtek codes

The recent addition of automatic mic-muting is broken in some cases.
The code assumes that the pin nids <= 0x18, but the digital pins can
be less than 0x18.
Also, it assumes the front-mic being the internal mic, but it depends
on the hardware implementation actually.

Instead of complex case-fixes, better to disable the code as now.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4eceab9bd10..a80d57cbc35 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -829,6 +829,7 @@ static void alc_sku_automute(struct hda_codec *codec)
 			    spec->jack_present ? 0 : PIN_OUT);
 }
 
+#if 0 /* it's broken in some acses -- temporarily disabled */
 static void alc_mic_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -849,6 +850,9 @@ static void alc_mic_automute(struct hda_codec *codec)
 	snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic,
 			 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
+#else
+#define alc_mic_automute(codec) /* NOP */
+#endif /* disabled */
 
 /* unsolicited event for HP jack sensing */
 static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -1058,12 +1062,14 @@ do_sku:
 			AC_VERB_SET_UNSOLICITED_ENABLE,
 			AC_USRSP_EN | ALC880_HP_EVENT);
 
+#if 0 /* it's broken in some acses -- temporarily disabled */
 	if (spec->autocfg.input_pins[AUTO_PIN_MIC] &&
 		spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC])
 		snd_hda_codec_write(codec,
 			spec->autocfg.input_pins[AUTO_PIN_MIC], 0,
 			AC_VERB_SET_UNSOLICITED_ENABLE,
 			AC_USRSP_EN | ALC880_MIC_EVENT);
+#endif /* disabled */
 
 	spec->unsol_event = alc_sku_unsol_event;
 }
-- 
cgit v1.2.3-70-g09d2


From ee956e090e114ede6542c76a7465ed6ce118ad52 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 31 Oct 2008 17:16:31 +0100
Subject: ALSA: hda - Add digital-mic for ALC269 auto-probe mode

The digital mic wasn't detected properly for ALC269 auto-probing mode
because of its widget number.  Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++--
 1 file changed, 20 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a80d57cbc35..d327a371595 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12244,8 +12244,26 @@ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
 	return 0;
 }
 
-#define alc269_auto_create_analog_input_ctls \
-	alc880_auto_create_analog_input_ctls
+static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
+						const struct auto_pin_cfg *cfg)
+{
+	int err;
+
+	err = alc880_auto_create_analog_input_ctls(spec, cfg);
+	if (err < 0)
+		return err;
+	/* digital-mic input pin is excluded in alc880_auto_create..()
+	 * because it's under 0x18
+	 */
+	if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
+	    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
+		struct hda_input_mux *imux = &spec->private_imux;
+		imux->items[imux->num_items].label = "Int Mic";
+		imux->items[imux->num_items].index = 0x05;
+		imux->num_items++;
+	}
+	return 0;
+}
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc269_loopbacks	alc880_loopbacks
-- 
cgit v1.2.3-70-g09d2


From 69e50282b726bab75c8050c4836dc89b7eb7bf1a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 3 Nov 2008 10:07:43 +0100
Subject: ALSA: hda - Add a quirk for another Acer Aspire (1025:0090)

Added a quirk for another Acer Aspier laptop (1025:0090) with ALC883
codec.  Reported in Novell bnc#426935:
    https://bugzilla.novell.com/show_bug.cgi?id=426935

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 d327a371595..a4666c96a44 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8414,6 +8414,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
-- 
cgit v1.2.3-70-g09d2


From 6b3ab21ef1ac15db4b053ce0ba8eae0ef9361c8a Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Mon, 3 Nov 2008 08:12:43 -0500
Subject: ALSA: hda: make a STAC_DELL_EQ option

Add support for explicitly enabling the EQ distortion hack for
systems without software biquad support.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 16 ++++++++++------
 1 file changed, 10 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index df9b0bc7f87..e6085915d86 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -69,6 +69,7 @@ enum {
 enum {
 	STAC_92HD73XX_REF,
 	STAC_DELL_M6,
+	STAC_DELL_EQ,
 	STAC_92HD73XX_MODELS
 };
 
@@ -773,9 +774,7 @@ static struct hda_verb dell_eq_core_init[] = {
 };
 
 static struct hda_verb dell_m6_core_init[] = {
-	/* set master volume to max value without distortion
-	 * and direct control */
-	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
+	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* setup audio connections */
 	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
@@ -1600,11 +1599,13 @@ static unsigned int dell_m6_pin_configs[13] = {
 static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_REF]	= ref92hd73xx_pin_configs,
 	[STAC_DELL_M6]	= dell_m6_pin_configs,
+	[STAC_DELL_EQ]	= dell_m6_pin_configs,
 };
 
 static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_REF] = "ref",
 	[STAC_DELL_M6] = "dell-m6",
+	[STAC_DELL_EQ] = "dell-eq",
 };
 
 static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
@@ -4131,12 +4132,17 @@ again:
 			sizeof(stac92hd73xx_dmux));
 
 	switch (spec->board_config) {
-	case STAC_DELL_M6:
+	case STAC_DELL_EQ:
 		spec->init = dell_eq_core_init;
+		/* fallthru */
+	case STAC_DELL_M6:
 		spec->num_smuxes = 0;
 		spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
 		spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
 		spec->num_amps = 1;
+
+		if (!spec->init)
+			spec->init = dell_m6_core_init;
 		switch (codec->subsystem_id) {
 		case 0x1028025e: /* Analog Mics */
 		case 0x1028025f:
@@ -4146,8 +4152,6 @@ again:
 			break;
 		case 0x10280271: /* Digital Mics */
 		case 0x10280272:
-			spec->init = dell_m6_core_init;
-			/* fall-through */
 		case 0x10280254:
 		case 0x10280255:
 			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
-- 
cgit v1.2.3-70-g09d2


From c4dc5071859bf666a5a9d6565f16c51a261a88b7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 4 Nov 2008 13:30:57 +0100
Subject: ALSA: hda - Limit the number of GPIOs show in proc

Limit the number of GPIOs shown in proc.  Otherwise it gets too long
unnecessarily, and hard to analyze.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 743d77922bc..c39af986bff 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -483,6 +483,8 @@ static void print_gpio(struct snd_info_buffer *buffer,
 		    (gpio & AC_GPIO_UNSOLICITED) ? 1 : 0,
 		    (gpio & AC_GPIO_WAKE) ? 1 : 0);
 	max = gpio & AC_GPIO_IO_COUNT;
+	if (!max || max > 8)
+		return;
 	enable = snd_hda_codec_read(codec, nid, 0,
 				    AC_VERB_GET_GPIO_MASK, 0);
 	direction = snd_hda_codec_read(codec, nid, 0,
-- 
cgit v1.2.3-70-g09d2


From 959973b92d3ba235edfa5dcb5df1be1e5d1deba2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 5 Nov 2008 11:30:56 +0100
Subject: ALSA: hda - Add a quirk for MEDION MD96630

Use model=lenovo-ms7195-dig for MEDION MD96630 laptop (17c0:4085)
with ALC888 codec.
Reference: Novell bnc#412548
	https://bugzilla.novell.com/show_bug.cgi?id=412528

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 a4666c96a44..a378c014512 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8469,6 +8469,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
 	SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+	SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
 	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
 	SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
-- 
cgit v1.2.3-70-g09d2


From 65b92e5cbc8acd14ea83190b4d016f765dce6075 Mon Sep 17 00:00:00 2001
From: Michel Marti <mma@objectxp.com>
Date: Sat, 8 Nov 2008 11:33:32 +0100
Subject: ALSA: hda - Add another HP model (6730s) for AD1884A

Added model=laptop for another HP machine (103c:3614) with AD1884A
codec.

Signed-off-by: Michel Marti <mma@objectxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d3fd432cb3e..400df85a445 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3861,6 +3861,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
 static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
+	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
 	{}
 };
-- 
cgit v1.2.3-70-g09d2


From 00c9ddd1d4cc73aa0077f379279d716cb0ab0ba5 Mon Sep 17 00:00:00 2001
From: Tim Blechmann <tim@klingt.org>
Date: Sun, 9 Nov 2008 12:50:52 +0100
Subject: ALSA: HDSP: check for io box before uploading firmware

currently the hdsp driver tries to upload the firmware, even if the
io box is not connected. this patch adds a check for the io box
before trying to upload the firmware.
thus instead of messages complaining about the fifo status and firmware
loading failure, the driver gives a message that no multiface or
digiface is connected.

[A minor coding-style fix by tiwai]

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index d723543bead..d680114b3e4 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5045,6 +5045,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 		/* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
 		ssleep(2);
 
+		err = hdsp_check_for_iobox(hdsp);
+		if (err < 0)
+			return err;
+
 		if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
 #ifdef HDSP_FW_LOADER
 			if ((err = hdsp_request_fw_loader(hdsp)) < 0)
@@ -5057,7 +5061,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 				/* init is complete, we return */
 				return 0;
 #endif
-			/* no iobox connected, we defer initialization */
+			/* we defer initialization */
 			snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n");
 			if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
 				return err;
-- 
cgit v1.2.3-70-g09d2


From 3ae7e2e22900b90bbe1d19454405950893c110be Mon Sep 17 00:00:00 2001
From: Tim Blechmann <tim@klingt.org>
Date: Sat, 8 Nov 2008 14:42:18 +0100
Subject: ALSA: hdsp: check for iobox and upload firmware during ioctl

currently, the error message when trying to run hdspmixer or hdspconf
if the breakout box is not connected is somehow misleading, since it
asks the user to upload the firmware.

this patch adds a test, whether the breakout box is connected and
tries to upload the firmware in the case, that it is not present, e.g.
because of power failures of the breakout box.

[Minor coding-style fixes by tiwai]

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 21 +++++++++++++++++----
 1 file changed, 17 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index d680114b3e4..736246f98ac 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -4548,11 +4548,20 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 {
 	struct hdsp *hdsp = (struct hdsp *)hw->private_data;	
 	void __user *argp = (void __user *)arg;
+	int err;
 
 	switch (cmd) {
 	case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: {
 		struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg;
 
+		err = hdsp_check_for_iobox(hdsp);
+		if (err < 0)
+			return err;
+
+		err = hdsp_check_for_firmware(hdsp, 1);
+		if (err < 0)
+			return err;
+
 		if (!(hdsp->state & HDSP_FirmwareLoaded)) {
 			snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n");
 			return -EINVAL;
@@ -4572,10 +4581,14 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 		unsigned long flags;
 		int i;
 		
-		if (!(hdsp->state & HDSP_FirmwareLoaded)) {
-			snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n");	
-			return -EINVAL;
-		}
+		err = hdsp_check_for_iobox(hdsp);
+		if (err < 0)
+			return err;
+
+		err = hdsp_check_for_firmware(hdsp, 1);
+		if (err < 0)
+			return err;
+
 		spin_lock_irqsave(&hdsp->lock, flags);
 		info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp);
 		info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp);
-- 
cgit v1.2.3-70-g09d2


From 254248313aed7e6ff295ca21a82ca989b1f69c16 Mon Sep 17 00:00:00 2001
From: Travis Place <wishie@wishie.net>
Date: Mon, 10 Nov 2008 17:56:23 +0100
Subject: ALSA: hda - Make the HP EliteBook 8530p use AD1884A model laptop

Added a QUIRK to patch_analog.c for the HP Elitebook 8530p
(IDs 0x103c:0x30e7) to use AD1884A model 'laptop' by default.
Playback and Capture confirmed working.

Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 400df85a445..686c77491de 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3861,6 +3861,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
 static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
+	SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
 	{}
-- 
cgit v1.2.3-70-g09d2


From 355a0ec471377a8a907b79c8fba3f0f09b44656e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Nov 2008 16:46:19 +0100
Subject: ALSA: hda - Add missing analog-mux mixer creation for STAC9200

The creation of analog-mux mixer element is missing in
patch_stac9200() due to the dynamic allocation patch.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e6085915d86..1627756e5c7 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3530,6 +3530,12 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
 	if ((err = stac9200_auto_create_lfe_ctls(codec, &spec->autocfg)) < 0)
 		return err;
 
+	if (spec->num_muxes > 0) {
+		err = stac92xx_auto_create_mux_input_ctls(codec);
+		if (err < 0)
+			return err;
+	}
+
 	if (spec->autocfg.dig_out_pin)
 		spec->multiout.dig_out_nid = 0x05;
 	if (spec->autocfg.dig_in_pin)
-- 
cgit v1.2.3-70-g09d2


From 4f1e6bc3646ab50b8181555ab7e6eeab68b8632a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Nov 2008 16:47:24 +0100
Subject: ALSA: hda - Fix input pin initialization for STAC/IDT codecs

The input pins are sometimes not initialized properly because
of the optimization check of the current pinctl code.

Force to initialize the mic input pins so that they can be set up
properly even if they were in a weird state.  But keep other input
pins if already set up as input, since this could be an extra mic
pin.

Reference: Novell bnc#443738
	https://bugzilla.novell.com/show_bug.cgi?id=443738

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++--------
 1 file changed, 12 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 1627756e5c7..a51160106df 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3653,14 +3653,18 @@ static int stac92xx_init(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = cfg->input_pins[i];
 		if (nid) {
-			unsigned int pinctl = snd_hda_codec_read(codec, nid,
-				0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-			/* if PINCTL already set then skip */
-			if (pinctl & AC_PINCAP_IN)
-				continue;
-			pinctl = AC_PINCTL_IN_EN;
-			if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC)
-				pinctl |= stac92xx_get_vref(codec, nid);
+			unsigned int pinctl;
+			if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) {
+				/* for mic pins, force to initialize */
+				pinctl = stac92xx_get_vref(codec, nid);
+			} else {
+				pinctl = snd_hda_codec_read(codec, nid, 0,
+					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+				/* if PINCTL already set then skip */
+				if (pinctl & AC_PINCTL_IN_EN)
+					continue;
+			}
+			pinctl |= AC_PINCTL_IN_EN;
 			stac92xx_auto_set_pinctl(codec, nid, pinctl);
 		}
 	}
-- 
cgit v1.2.3-70-g09d2


From d7a8943635485597ae7c6d554a8ccf3ce5a42d2d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 12 Nov 2008 09:48:04 +0100
Subject: ALSA: hda - Fix IDT/STAC multiple HP detection

Due to the recent change for multiple HP as line-out switch, only
one of the multiple headphons (usually a wrong one) is toggled
and the other pins are still disabled.  This causes the silent output
problem on some Dell laptops.

Also, the hp_switch check is screwed up when a line-in or a mic-in
jack exists.  This is added as an additional output, but hp_switch
check doesn't take it into account.

This patch fixes these issues: simplify hp_switch check by using
the NID instead of bool, and clean up / fix the toggle of HP pins
in unsol event handler code.

Reference: Novell bnc#443267
	https://bugzilla.novell.com/show_bug.cgi?id=443267

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 59 ++++++++++++++++++++++++++++++++----------
 1 file changed, 45 insertions(+), 14 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a51160106df..4300a679cd8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -212,7 +212,7 @@ struct sigmatel_spec {
 	/* i/o switches */
 	unsigned int io_switch[2];
 	unsigned int clfe_swap;
-	unsigned int hp_switch;
+	unsigned int hp_switch; /* NID of HP as line-out */
 	unsigned int aloopback;
 
 	struct hda_pcm pcm_rec[2];	/* PCM information */
@@ -2443,7 +2443,7 @@ static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct sigmatel_spec *spec = codec->spec;
 
-	ucontrol->value.integer.value[0] = spec->hp_switch;
+	ucontrol->value.integer.value[0] = !!spec->hp_switch;
 	return 0;
 }
 
@@ -2452,8 +2452,9 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct sigmatel_spec *spec = codec->spec;
-
-	spec->hp_switch = ucontrol->value.integer.value[0];
+	int nid = kcontrol->private_value;
+ 
+	spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0;
 
 	/* check to be sure that the ports are upto date with
 	 * switch changes
@@ -2862,7 +2863,8 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 	if (cfg->hp_outs > 1) {
 		err = stac92xx_add_control(spec,
 			STAC_CTL_WIDGET_HP_SWITCH,
-			"Headphone as Line Out Switch", 0);
+			"Headphone as Line Out Switch",
+			cfg->hp_pins[cfg->hp_outs - 1]);
 		if (err < 0)
 			return err;
 	}
@@ -3786,11 +3788,30 @@ static int get_hp_pin_presence(struct hda_codec *codec, hda_nid_t nid)
 	return 0;
 }
 
+/* return non-zero if the hp-pin of the given array index isn't
+ * a jack-detection target
+ */
+static int no_hp_sensing(struct sigmatel_spec *spec, int i)
+{
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+
+	/* ignore sensing of shared line and mic jacks */
+	if (spec->line_switch &&
+	    cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_LINE])
+		return 1;
+	if (spec->mic_switch &&
+	    cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_MIC])
+		return 1;
+	/* ignore if the pin is set as line-out */
+	if (cfg->hp_pins[i] == spec->hp_switch)
+		return 1;
+	return 0;
+}
+
 static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int nid = cfg->hp_pins[cfg->hp_outs - 1];
 	int i, presence;
 
 	presence = 0;
@@ -3801,15 +3822,16 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
 	for (i = 0; i < cfg->hp_outs; i++) {
 		if (presence)
 			break;
-		if (spec->hp_switch && cfg->hp_pins[i] == nid)
-			break;
+		if (no_hp_sensing(spec, i))
+			continue;
 		presence = get_hp_pin_presence(codec, cfg->hp_pins[i]);
 	}
 
 	if (presence) {
-		/* disable lineouts, enable hp */
+		/* disable lineouts */
 		if (spec->hp_switch)
-			stac92xx_reset_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+			stac92xx_reset_pinctl(codec, spec->hp_switch,
+					      AC_PINCTL_OUT_EN);
 		for (i = 0; i < cfg->line_outs; i++)
 			stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
 						AC_PINCTL_OUT_EN);
@@ -3821,9 +3843,10 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
 				spec->gpio_dir, spec->gpio_data &
 				~spec->eapd_mask);
 	} else {
-		/* enable lineouts, disable hp */
+		/* enable lineouts */
 		if (spec->hp_switch)
-			stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+			stac92xx_set_pinctl(codec, spec->hp_switch,
+					    AC_PINCTL_OUT_EN);
 		for (i = 0; i < cfg->line_outs; i++)
 			stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
 						AC_PINCTL_OUT_EN);
@@ -3835,8 +3858,16 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
 				spec->gpio_dir, spec->gpio_data |
 				spec->eapd_mask);
 	}
-	if (!spec->hp_switch && cfg->hp_outs > 1 && presence)
-		stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+	/* toggle hp outs */
+	for (i = 0; i < cfg->hp_outs; i++) {
+		unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
+		if (no_hp_sensing(spec, i))
+			continue;
+		if (presence)
+			stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
+		else
+			stac92xx_reset_pinctl(codec, cfg->hp_pins[i], val);
+	}
 } 
 
 static void stac92xx_pin_sense(struct hda_codec *codec, int idx)
-- 
cgit v1.2.3-70-g09d2


From 4d4e9bb339cfcde7811af10859ba1ce2fe3d46b4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 12 Nov 2008 16:45:04 +0100
Subject: ALSA: hda - Add digital beep playback switch for STAC/IDT codecs

The digital beep widget may have no mute control, and always enabling
the beep is ofen pretty annoying, especially on laptops.

This patch adds a mixer control "PC Beep Playback Switch" when there
is no mixer amp mute is found, and controls it on software.

Reference: Novell bnc#444572
	https://bugzilla.novell.com/show_bug.cgi?id=444572

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c       |  4 +++
 sound/pci/hda/hda_beep.h       |  1 +
 sound/pci/hda/patch_sigmatel.c | 69 ++++++++++++++++++++++++++++++++++++++----
 3 files changed, 68 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 9b77b3e0fa9..b1796ae1e8f 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -37,6 +37,9 @@ static void snd_hda_generate_beep(struct work_struct *work)
 		container_of(work, struct hda_beep, beep_work);
 	struct hda_codec *codec = beep->codec;
 
+	if (!beep->enabled)
+		return;
+
 	/* generate tone */
 	snd_hda_codec_write_cache(codec, beep->nid, 0,
 			AC_VERB_SET_BEEP_CONTROL, beep->tone);
@@ -115,6 +118,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 	beep->nid = nid;
 	beep->dev = input_dev;
 	beep->codec = codec;
+	beep->enabled = 1;
 	codec->beep = beep;
 
 	INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index de4036e6e71..b9679f081ca 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -31,6 +31,7 @@ struct hda_beep {
 	char phys[32];
 	int tone;
 	int nid;
+	int enabled;
 	struct work_struct beep_work; /* scheduled task for beep event */
 };
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4300a679cd8..1633ef2c654 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2587,8 +2587,10 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
 };
 
 /* add dynamic controls */
-static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type,
-		int idx, const char *name, unsigned long val)
+static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
+				     struct snd_kcontrol_new *ktemp,
+				     int idx, const char *name,
+				     unsigned long val)
 {
 	struct snd_kcontrol_new *knew;
 
@@ -2607,20 +2609,29 @@ static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type,
 	}
 
 	knew = &spec->kctl_alloc[spec->num_kctl_used];
-	*knew = stac92xx_control_templates[type];
+	*knew = *ktemp;
 	knew->index = idx;
 	knew->name = kstrdup(name, GFP_KERNEL);
-	if (! knew->name)
+	if (!knew->name)
 		return -ENOMEM;
 	knew->private_value = val;
 	spec->num_kctl_used++;
 	return 0;
 }
 
+static inline int stac92xx_add_control_idx(struct sigmatel_spec *spec,
+					   int type, int idx, const char *name,
+					   unsigned long val)
+{
+	return stac92xx_add_control_temp(spec,
+					 &stac92xx_control_templates[type],
+					 idx, name, val);
+}
+
 
 /* add dynamic controls */
-static int stac92xx_add_control(struct sigmatel_spec *spec, int type,
-		const char *name, unsigned long val)
+static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type,
+				       const char *name, unsigned long val)
 {
 	return stac92xx_add_control_idx(spec, type, 0, name, val);
 }
@@ -3062,6 +3073,43 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+#define stac92xx_dig_beep_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_dig_beep_switch_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	ucontrol->value.integer.value[0] = codec->beep->enabled;
+	return 0;
+}
+
+static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	int enabled = !!ucontrol->value.integer.value[0];
+	if (codec->beep->enabled != enabled) {
+		codec->beep->enabled = enabled;
+		return 1;
+	}
+	return 0;
+}
+
+static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.info = stac92xx_dig_beep_switch_info,
+	.get = stac92xx_dig_beep_switch_get,
+	.put = stac92xx_dig_beep_switch_put,
+};
+
+static int stac92xx_beep_switch_ctl(struct hda_codec *codec)
+{
+	return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl,
+					 0, "PC Beep Playback Switch", 0);
+}
+#endif
+
 static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -3368,6 +3416,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 	if (spec->digbeep_nid > 0) {
 		hda_nid_t nid = spec->digbeep_nid;
+		unsigned int caps;
 
 		err = stac92xx_auto_create_beep_ctls(codec, nid);
 		if (err < 0)
@@ -3375,6 +3424,14 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		err = snd_hda_attach_beep_device(codec, nid);
 		if (err < 0)
 			return err;
+		/* if no beep switch is available, make its own one */
+		caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+		if (codec->beep &&
+		    !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) {
+			err = stac92xx_beep_switch_ctl(codec);
+			if (err < 0)
+				return err;
+		}
 	}
 #endif
 
-- 
cgit v1.2.3-70-g09d2


From 6a12afb5640217454cad21e8cf9a74b038fa3717 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Nov 2008 13:08:56 +0100
Subject: ALSA: hda - Missing NULL check in hda_beep.c

Added a NULL check of input_allocate_device() in hda_beep.c.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index b1796ae1e8f..3ecd7e797de 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -88,6 +88,10 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 	snprintf(beep->phys, sizeof(beep->phys),
 		"card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
 	input_dev = input_allocate_device();
+	if (!input_dev) {
+		kfree(beep);
+		return -ENOMEM;
+	}
 
 	/* setup digital beep device */
 	input_dev->name = "HDA Digital PCBeep";
-- 
cgit v1.2.3-70-g09d2


From c39555d6edd0142972f577405d5259879ecc1635 Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Fri, 14 Nov 2008 17:46:22 -0500
Subject: ALSA: hda: STAC_VREF_EVENT value change

Changed value for STAC_VREF_EVENT from 0x40 to 0x00 because the
unsol response value is only 6-bits width and the former value
was 1<<6 which is an overrun.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 1633ef2c654..f205570def1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -36,9 +36,11 @@
 #include "hda_beep.h"
 
 #define NUM_CONTROL_ALLOC	32
+
+#define STAC_VREF_EVENT		0x00
+#define STAC_INSERT_EVENT	0x10
 #define STAC_PWR_EVENT		0x20
 #define STAC_HP_EVENT		0x30
-#define STAC_VREF_EVENT		0x40
 
 enum {
 	STAC_REF,
-- 
cgit v1.2.3-70-g09d2


From d6f35e3f0da14a5eb02114708d88ae2f59dc8283 Mon Sep 17 00:00:00 2001
From: Julia Lawall <julia@diku.dk>
Date: Fri, 14 Nov 2008 19:08:18 +0100
Subject: ALSA: sound/pci/pcxhr/pcxhr.c: introduce missing kfree and
 pci_disable_device

Error handling code following a kzalloc should free the allocated data.
The error handling code is adjusted to call pci_disable_device(pci); as
well, as done later in the function

The semantic match that finds the problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,l;
position p1,p2;
expression *ptr != NULL;
@@

(
if ((x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...)) == NULL) S
|
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
)
<... when != x
     when != if (...) { <+...x...+> }
x->f = E
...>
(
 return \(0\|<+...x...+>\|ptr\);
|
 return@p2 ...;
)

@script:python@
p1 << r.p1;
p2 << r.p2;
@@

print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 0e06c6c9fcc..73de6e989b3 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1229,8 +1229,11 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id
 		return -ENOMEM;
 	}
 
-	if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST))
+	if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) {
+		kfree(mgr);
+		pci_disable_device(pci);
 		return -ENODEV;
+	}
 	card_name = pcxhr_board_params[pci_id->driver_data].board_name;
 	mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips;
 	mgr->capture_chips  = pcxhr_board_params[pci_id->driver_data].capture_chips;
-- 
cgit v1.2.3-70-g09d2


From 8e5f262bfcd90c041160a491a238661ebbb584a4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 15 Nov 2008 19:28:54 +0100
Subject: ALSA: hda - Check model type instead of SSID in patch_92hd71bxx()

Check board preset model instead of codec->subsystem_id in
patch_92hd71bxx() so that other hardwares configured via the model
option work like the given model.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f205570def1..0df6f979f2a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4488,8 +4488,8 @@ again:
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		break;
 	case 0x111d7608: /* 5 Port with Analog Mixer */
-		switch (codec->subsystem_id) {
-		case 0x103c361a:
+		switch (spec->board_config) {
+		case STAC_HP_M4:
 			/* Enable VREF power saving on GPIO1 detect */
 			snd_hda_codec_write(codec, codec->afg, 0,
 				AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02);
-- 
cgit v1.2.3-70-g09d2


From 41c3b648bd4cdc34fd1918e288f8afe78903432e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 18 Nov 2008 10:45:15 +0100
Subject: ALSA: hda - Fix GPIO initialization in patch_stac92hd71bxx()

Fixed the GPIO mask and co initialization in patch_stac92hd71bxx()
so that the gpio_maks for HP_M4 model is set properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 0df6f979f2a..b254e90b47c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4478,6 +4478,13 @@ again:
 		stac92xx_set_config_regs(codec);
 	}
 
+	if (spec->board_config > STAC_92HD71BXX_REF) {
+		/* GPIO0 = EAPD */
+		spec->gpio_mask = 0x01;
+		spec->gpio_dir = 0x01;
+		spec->gpio_data = 0x01;
+	}
+
 	switch (codec->vendor_id) {
 	case 0x111d76b6: /* 4 Port without Analog Mixer */
 	case 0x111d76b7:
@@ -4537,13 +4544,6 @@ again:
 	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
 
-	if (spec->board_config > STAC_92HD71BXX_REF) {
-		/* GPIO0 = EAPD */
-		spec->gpio_mask = 0x01;
-		spec->gpio_dir = 0x01;
-		spec->gpio_data = 0x01;
-	}
-
 	spec->powerdown_adcs = 1;
 	spec->digbeep_nid = 0x26;
 	spec->mux_nids = stac92hd71bxx_mux_nids;
-- 
cgit v1.2.3-70-g09d2


From 80bf272468a8b63f5550304363e55c1ba06c13db Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 18 Nov 2008 10:48:41 +0100
Subject: ALSA: hda - Add quirks for HP Pavilion DV models

Added the quirk entries for HP Pavilion DV5 and DV7 with model=hp-m4.

Reference: Novell bnc#445321, bnc#445161
	https://bugzilla.novell.com/show_bug.cgi?id=445321
	https://bugzilla.novell.com/show_bug.cgi?id=445161

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b254e90b47c..25d4bf8e422 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1688,6 +1688,10 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
+		      "HP dv5", STAC_HP_M4),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
+		      "HP dv7", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
 				"unknown HP", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
-- 
cgit v1.2.3-70-g09d2


From c5d08bb567874e410210cf7d21a25f28fcceb0a7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 18 Nov 2008 10:55:36 +0100
Subject: ALSA: hda - Fix resume of GPIO unsol event for STAC/IDT

Use cached write for setting the GPIO unsolicited event mask to be
restored properly at resume.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 25d4bf8e422..9563b5bbb27 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4502,7 +4502,7 @@ again:
 		switch (spec->board_config) {
 		case STAC_HP_M4:
 			/* Enable VREF power saving on GPIO1 detect */
-			snd_hda_codec_write(codec, codec->afg, 0,
+			snd_hda_codec_write_cache(codec, codec->afg, 0,
 				AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02);
 			snd_hda_codec_write_cache(codec, codec->afg, 0,
 					AC_VERB_SET_UNSOLICITED_ENABLE,
@@ -4895,7 +4895,7 @@ static int patch_stac9205(struct hda_codec *codec)
 		stac92xx_set_config_reg(codec, 0x20, 0x1c410030);
 
 		/* Enable unsol response for GPIO4/Dock HP connection */
-		snd_hda_codec_write(codec, codec->afg, 0,
+		snd_hda_codec_write_cache(codec, codec->afg, 0,
 			AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10);
 		snd_hda_codec_write_cache(codec, codec->afg, 0,
 					  AC_VERB_SET_UNSOLICITED_ENABLE,
-- 
cgit v1.2.3-70-g09d2