Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Add model=6530g option
  ALSA: hda - Acer Inspire 6530G model for Realtek ALC888
  ALSA: snd_usb_caiaq: fix legacy input streaming
  ASoC: Kill BUS_ID_SIZE
  ALSA: HDA - Correct trivial typos in comments.
  ALSA: HDA - Name-fixes in code (tagra/targa)
  ALSA: HDA - Add pci-quirk for MSI MS-7350 motherboard.
  ALSA: hda - Fix memory leak at codec creation

5 files changed, 109 insertions, 38 deletions

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 562403a..462e2ce 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -972,8 +972,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 			snd_hda_codec_read(codec, nid, 0,
 					   AC_VERB_GET_SUBSYSTEM_ID, 0);
 	}
-	if (bus->modelname)
-		codec->modelname = kstrdup(bus->modelname, GFP_KERNEL);
 
 	/* power-up all before initialization */
 	hda_set_power_state(codec,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d22b260..bf4b78a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -224,6 +224,7 @@ enum {
 	ALC883_ACER,
 	ALC883_ACER_ASPIRE,
 	ALC888_ACER_ASPIRE_4930G,
+	ALC888_ACER_ASPIRE_6530G,
 	ALC888_ACER_ASPIRE_8930G,
 	ALC883_MEDION,
 	ALC883_MEDION_MD2,
@@ -970,7 +971,7 @@ static void alc_automute_pin(struct hda_codec *codec)
 	}
 }
 
-#if 0 /* it's broken in some acses -- temporarily disabled */
+#if 0 /* it's broken in some cases -- temporarily disabled */
 static void alc_mic_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1170,7 +1171,7 @@ static int alc_subsystem_id(struct hda_codec *codec,
 
 	/* invalid SSID, check the special NID pin defcfg instead */
 	/*
-	 * 31~30	: port conetcivity
+	 * 31~30	: port connectivity
 	 * 29~21	: reserve
 	 * 20		: PCBEEP input
 	 * 19~16	: Check sum (15:1)
@@ -1471,6 +1472,25 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
 };
 
 /*
+ * ALC888 Acer Aspire 6530G model
+ */
+
+static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+/* Bias voltage on for external mic port */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+/* 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},
+	{ }
+};
+
+/*
  * ALC889 Acer Aspire 8930G model
  */
 
@@ -1544,6 +1564,25 @@ static struct hda_input_mux alc888_2_capture_sources[2] = {
 	}
 };
 
+static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
+	/* Interal mic only available on one ADC */
+	{
+		.num_items = 3,
+		.items = {
+			{ "Ext Mic", 0x0 },
+			{ "CD", 0x4 },
+			{ "Int Mic", 0xb },
+		},
+	},
+	{
+		.num_items = 2,
+		.items = {
+			{ "Ext Mic", 0x0 },
+			{ "CD", 0x4 },
+		},
+	}
+};
+
 static struct hda_input_mux alc889_capture_sources[3] = {
 	/* Digital mic only available on first "ADC" */
 	{
@@ -6347,7 +6386,7 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = {
 };
 
 /*
- * macbook pro ALC885 can switch LineIn to LineOut without loosing Mic
+ * macbook pro ALC885 can switch LineIn to LineOut without losing Mic
  */
 
 /*
@@ -7047,7 +7086,7 @@ static struct hda_verb alc882_auto_init_verbs[] = {
 #define alc882_loopbacks	alc880_loopbacks
 #endif
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc882_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc882_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc882_pcm_digital_playback	alc880_pcm_digital_playback
@@ -8068,7 +8107,7 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_tagra_mixer[] = {
+static struct snd_kcontrol_new alc883_targa_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -8088,7 +8127,7 @@ static struct snd_kcontrol_new alc883_tagra_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = {
+static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -8153,6 +8192,19 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	{ } /* end */
 };
 
+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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -8417,7 +8469,7 @@ static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc883_tagra_verbs[] = {
+static struct hda_verb alc883_targa_verbs[] = {
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
@@ -8626,8 +8678,8 @@ static void alc883_medion_md2_init_hook(struct hda_codec *codec)
 }
 
 /* toggle speaker-output according to the hp-jack state */
-#define alc883_tagra_init_hook		alc882_targa_init_hook
-#define alc883_tagra_unsol_event	alc882_targa_unsol_event
+#define alc883_targa_init_hook		alc882_targa_init_hook
+#define alc883_targa_unsol_event	alc882_targa_unsol_event
 
 static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
 {
@@ -8957,7 +9009,7 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
 #define alc883_loopbacks	alc880_loopbacks
 #endif
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc883_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc883_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc883_pcm_analog_alt_capture	alc880_pcm_analog_alt_capture
@@ -8978,6 +9030,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 	[ALC883_ACER]		= "acer",
 	[ALC883_ACER_ASPIRE]	= "acer-aspire",
 	[ALC888_ACER_ASPIRE_4930G]	= "acer-aspire-4930g",
+	[ALC888_ACER_ASPIRE_6530G]	= "acer-aspire-6530g",
 	[ALC888_ACER_ASPIRE_8930G]	= "acer-aspire-8930g",
 	[ALC883_MEDION]		= "medion",
 	[ALC883_MEDION_MD2]	= "medion-md2",
@@ -9021,7 +9074,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
-		ALC888_ACER_ASPIRE_4930G),
+		ALC888_ACER_ASPIRE_6530G),
 	/* default Acer -- disabled as it causes more problems.
 	 *    model=auto should work fine now
 	 */
@@ -9069,6 +9122,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
@@ -9165,8 +9219,8 @@ static struct alc_config_preset alc883_presets[] = {
 		.input_mux = &alc883_capture_source,
 	},
 	[ALC883_TARGA_DIG] = {
-		.mixers = { alc883_tagra_mixer, alc883_chmode_mixer },
-		.init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
+		.mixers = { alc883_targa_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.dig_out_nid = ALC883_DIGOUT_NID,
@@ -9174,12 +9228,12 @@ static struct alc_config_preset alc883_presets[] = {
 		.channel_mode = alc883_3ST_6ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc883_tagra_unsol_event,
-		.init_hook = alc883_tagra_init_hook,
+		.unsol_event = alc883_targa_unsol_event,
+		.init_hook = alc883_targa_init_hook,
 	},
 	[ALC883_TARGA_2ch_DIG] = {
-		.mixers = { alc883_tagra_2ch_mixer},
-		.init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
+		.mixers = { alc883_targa_2ch_mixer},
+		.init_verbs = { alc883_init_verbs, alc883_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.adc_nids = alc883_adc_nids_alt,
@@ -9188,13 +9242,13 @@ static struct alc_config_preset alc883_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc883_tagra_unsol_event,
-		.init_hook = alc883_tagra_init_hook,
+		.unsol_event = alc883_targa_unsol_event,
+		.init_hook = alc883_targa_init_hook,
 	},
 	[ALC883_TARGA_8ch_DIG] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
-				alc883_tagra_verbs },
+				alc883_targa_verbs },
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
@@ -9206,8 +9260,8 @@ static struct alc_config_preset alc883_presets[] = {
 		.channel_mode = alc883_4ST_8ch_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
-		.unsol_event = alc883_tagra_unsol_event,
-		.init_hook = alc883_tagra_init_hook,
+		.unsol_event = alc883_targa_unsol_event,
+		.init_hook = alc883_targa_init_hook,
 	},
 	[ALC883_ACER] = {
 		.mixers = { alc883_base_mixer },
@@ -9255,6 +9309,24 @@ static struct alc_config_preset alc883_presets[] = {
 		.unsol_event = alc_automute_amp_unsol_event,
 		.init_hook = alc888_acer_aspire_4930g_init_hook,
 	},
+	[ALC888_ACER_ASPIRE_6530G] = {
+		.mixers = { alc888_acer_aspire_6530_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+				alc888_acer_aspire_6530g_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.num_mux_defs =
+			ARRAY_SIZE(alc888_2_capture_sources),
+		.input_mux = alc888_acer_aspire_6530_sources,
+		.unsol_event = alc_automute_amp_unsol_event,
+		.init_hook = alc888_acer_aspire_4930g_init_hook,
+	},
 	[ALC888_ACER_ASPIRE_8930G] = {
 		.mixers = { alc888_base_mixer,
 				alc883_chmode_mixer },
@@ -9361,7 +9433,7 @@ static struct alc_config_preset alc883_presets[] = {
 		.init_hook = alc888_lenovo_ms7195_front_automute,
 	},
 	[ALC883_HAIER_W66] = {
-		.mixers = { alc883_tagra_2ch_mixer},
+		.mixers = { alc883_targa_2ch_mixer},
 		.init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
@@ -11131,7 +11203,7 @@ static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
 #define alc262_loopbacks	alc880_loopbacks
 #endif
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc262_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc262_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc262_pcm_digital_playback	alc880_pcm_digital_playback
@@ -12286,7 +12358,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
 			    AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2);
 }
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc268_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc268_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc268_pcm_analog_alt_capture	alc880_pcm_analog_alt_capture
@@ -13197,7 +13269,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
 #define alc269_loopbacks	alc880_loopbacks
 #endif
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc269_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc269_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc269_pcm_digital_playback	alc880_pcm_digital_playback
@@ -14059,7 +14131,7 @@ static void alc861_toshiba_unsol_event(struct hda_codec *codec,
 		alc861_toshiba_automute(codec);
 }
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc861_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc861_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc861_pcm_digital_playback	alc880_pcm_digital_playback
@@ -14582,7 +14654,7 @@ static hda_nid_t alc861vd_dac_nids[4] = {
 
 /* dac_nids for ALC660vd are in a different order - according to
  * Realtek's driver.
- * This should probably tesult in a different mixer for 6stack models
+ * This should probably result in a different mixer for 6stack models
  * of ALC660vd codecs, but for now there is only 3stack mixer
  * - and it is the same as in 861vd.
  * adc_nids in ALC660vd are (is) the same as in 861vd
@@ -15027,7 +15099,7 @@ static void alc861vd_dallas_init_hook(struct hda_codec *codec)
 #define alc861vd_loopbacks	alc880_loopbacks
 #endif
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc861vd_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc861vd_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc861vd_pcm_digital_playback	alc880_pcm_digital_playback
@@ -15206,7 +15278,7 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
 	hda_nid_t pin;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin) /* connect to front and  use dac 0 */
+	if (pin) /* connect to front and use dac 0 */
 		alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
 	pin = spec->autocfg.speaker_pins[0];
 	if (pin)
@@ -16669,7 +16741,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = {
 #endif
 
 
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
 #define alc662_pcm_analog_playback	alc880_pcm_analog_playback
 #define alc662_pcm_analog_capture	alc880_pcm_analog_capture
 #define alc662_pcm_digital_playback	alc880_pcm_digital_playback
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index fa33661..938a58a 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -297,9 +297,9 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 static bool filter(struct dma_chan *chan, void *param)
 {
 	struct txx9aclc_dmadata *dmadata = param;
-	char devname[BUS_ID_SIZE + 2];
+	char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */
 
-	sprintf(devname, "%s.%d", dmadata->dma_res->name,
+	snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name,
 		(int)dmadata->dma_res->start);
 	if (strcmp(dev_name(chan->device->dev), devname) == 0) {
 		chan->private = &dmadata->dma_slave;
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index b144513..8f9b60c 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -199,8 +199,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
 		dev->period_out_count[index] = BYTES_PER_SAMPLE + 1;
 		dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
 	} else {
-		dev->period_in_count[index] = BYTES_PER_SAMPLE;
-		dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE;
+		int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2;
+		dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos;
+		dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos;
 	}
 
 	if (dev->streaming)
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 2240624..0e5db71 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"