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: mixart: range checking proc file
  ALSA: hda - Fix a wrong array range check in patch_realtek.c
  ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
  ALSA: hda - Enable amplifiers on Acer Inspire 6530G
  ASoC: Only do WM8994 bias off transition from standby
  ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
  ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
  ASoC: Support second DC servo readback method for wm_hubs
  ASoC: Avoid wraparound in wm_hubs DC servo correction
  ALSA: echoaudio - Eliminate use after free
  ALSA: i2c: cleanup: change parameter to pointer
  ALSA: hda - Add MSI blacklist for Aopen MZ915-M
  ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
  ALSA: hda - Update document about MSI and interrupts
  ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
  ALSA: hda - Add missing printk argument in previous patch
  ASoC: Fix passing platform_data to ac97 bus users and fix a leak
  ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
  ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
  ASoC: wm8994: playback => capture

5 files changed, 142 insertions, 60 deletions

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);
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 */
 	{}
 };
 
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;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a23444e9e7..c7730dbb9dd 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},
 	{ }
 };
 
@@ -4984,6 +4991,70 @@ 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",
+		       codec->chip_name, 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))
@@ -8398,9 +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),
@@ -10041,13 +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);
 
 }
@@ -12459,11 +12527,11 @@ 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,
@@ -13333,9 +13401,9 @@ 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
@@ -13482,11 +13550,11 @@ 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 +13579,11 @@ 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 +13714,11 @@ 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 */
@@ -13842,7 +13910,6 @@ 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 +13933,19 @@ 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 +14224,16 @@ 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)
@@ -17115,9 +17185,9 @@ 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 +17198,13 @@ 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 +17215,13 @@ 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 +17260,14 @@ 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);
 	}
 }
 
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 55e9315d4cc..3be8f97c8bc 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1162,13 +1162,15 @@ 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;
 }
@@ -1181,13 +1183,15 @@ 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;
 }