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

32 files changed, 377 insertions, 206 deletions

diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index fff62cc8607c..971a84a4fa77 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -70,7 +70,7 @@ static int snd_ak4113_dev_free(struct snd_device *device)
 }
 
 int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
-		ak4113_write_t *write, const unsigned char pgm[5],
+		ak4113_write_t *write, const unsigned char *pgm,
 		void *private_data, struct ak4113 **r_ak4113)
 {
 	struct ak4113 *chip;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 8dab82d7d19d..668a5ec04499 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 4bb90675f70f..f8fd586ae024 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 e6d1bdff1b6e..af34606c30c3 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 9a23444e9e7a..c7730dbb9ddb 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 55e9315d4ccd..3be8f97c8bc0 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;
 }
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 9ef6b96373f5..3e6628c8e665 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = params_buffer_bytes(params);
 
-	prtd->params = rtd->dai->cpu_dai->dma_data;
+	prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
 
 	prtd->dma_buffer = runtime->dma_addr;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index e588e63f18d2..0b59806905d1 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	ssc_p->dma_params[dir] = dma_params;
 
 	/*
-	 * The cpu_dai->dma_data field is only used to communicate the
-	 * appropriate DMA parameters to the pcm driver hw_params()
+	 * The snd_soc_pcm_stream->dma_data field is only used to communicate
+	 * the appropriate DMA parameters to the pcm driver hw_params()
 	 * function.  It should not be used for other purposes
 	 * as it is common to all substreams.
 	 */
-	rtd->dai->cpu_dai->dma_data = dma_params;
+	snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params);
 
 	channels = params_channels(params);
 
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index fd101d450d56..1f5e57a4bb7a 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -81,9 +81,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 static int ac97_soc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_codec *codec;
 	struct snd_ac97_bus *ac97_bus;
 	struct snd_ac97_template ac97_template;
+	int i;
 	int ret = 0;
 
 	printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
@@ -103,12 +105,6 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
-	if (ret < 0) {
-		printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n");
-		goto err;
-	}
-
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0)
@@ -124,6 +120,13 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto bus_err;
 
+	for (i = 0; i < card->num_links; i++) {
+		if (card->dai_link[i].codec_dai->ac97_control) {
+			snd_ac97_dev_add_pdata(codec->ac97,
+				card->dai_link[i].cpu_dai->ac97_pdata);
+		}
+	}
+
 	return 0;
 
 bus_err:
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 8d1c63754be4..9da0724cd47a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3008,34 +3008,39 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		/* Switch over to startup biases */
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-				    WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
-				    WM8994_VMID_BUF_ENA |
-				    WM8994_VMID_RAMP_MASK,
-				    WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
-				    WM8994_VMID_BUF_ENA |
-				    (1 << WM8994_VMID_RAMP_SHIFT));
-
-		/* Disable main biases */
-		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
-				    WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0);
+		if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+			/* Switch over to startup biases */
+			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+					    WM8994_BIAS_SRC |
+					    WM8994_STARTUP_BIAS_ENA |
+					    WM8994_VMID_BUF_ENA |
+					    WM8994_VMID_RAMP_MASK,
+					    WM8994_BIAS_SRC |
+					    WM8994_STARTUP_BIAS_ENA |
+					    WM8994_VMID_BUF_ENA |
+					    (1 << WM8994_VMID_RAMP_SHIFT));
 
-		/* Discharge line */
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
-				    WM8994_LINEOUT1_DISCH |
-				    WM8994_LINEOUT2_DISCH,
-				    WM8994_LINEOUT1_DISCH |
-				    WM8994_LINEOUT2_DISCH);
+			/* Disable main biases */
+			snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
+					    WM8994_BIAS_ENA |
+					    WM8994_VMID_SEL_MASK, 0);
 
-		msleep(5);
+			/* Discharge line */
+			snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+					    WM8994_LINEOUT1_DISCH |
+					    WM8994_LINEOUT2_DISCH,
+					    WM8994_LINEOUT1_DISCH |
+					    WM8994_LINEOUT2_DISCH);
 
-		/* Switch off startup biases */
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-				    WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
-				    WM8994_VMID_BUF_ENA |
-				    WM8994_VMID_RAMP_MASK, 0);
+			msleep(5);
 
+			/* Switch off startup biases */
+			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+					    WM8994_BIAS_SRC |
+					    WM8994_STARTUP_BIAS_ENA |
+					    WM8994_VMID_BUF_ENA |
+					    WM8994_VMID_RAMP_MASK, 0);
+		}
 		break;
 	}
 	codec->bias_level = level;
@@ -3402,7 +3407,7 @@ struct snd_soc_dai wm8994_dai[] = {
 			.rates = WM8994_RATES,
 			.formats = WM8994_FORMATS,
 		},
-		.playback = {
+		.capture = {
 			.stream_name = "AIF3 Capture",
 			.channels_min = 2,
 			.channels_max = 2,
@@ -3731,11 +3736,12 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 	case 3:
 		wm8994->hubs.dcs_codes = -5;
 		wm8994->hubs.hp_startup_mode = 1;
+		wm8994->hubs.dcs_readback_mode = 1;
 		break;
 	default:
+		wm8994->hubs.dcs_readback_mode = 1;
 		break;
 	}
-			   
 
 	/* Remember if AIFnLRCLK is configured as a GPIO.  This should be
 	 * configured on init - if a system wants to do this dynamically
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 486bdd21a98a..e1f225a3ac46 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -62,21 +62,27 @@ static const char *speaker_mode_text[] = {
 static const struct soc_enum speaker_mode =
 	SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text);
 
-static void wait_for_dc_servo(struct snd_soc_codec *codec)
+static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
 {
 	unsigned int reg;
 	int count = 0;
+	unsigned int val;
+
+	val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1;
+
+	/* Trigger the command */
+	snd_soc_write(codec, WM8993_DC_SERVO_0, val);
 
 	dev_dbg(codec->dev, "Waiting for DC servo...\n");
 
 	do {
 		count++;
 		msleep(1);
-		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0);
+		reg = snd_soc_read(codec, WM8993_DC_SERVO_0);
 		dev_dbg(codec->dev, "DC servo: %x\n", reg);
-	} while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400);
+	} while (reg & op && count < 400);
 
-	if (reg & WM8993_DCS_DATAPATH_BUSY)
+	if (reg & op)
 		dev_err(codec->dev, "Timed out waiting for DC Servo\n");
 }
 
@@ -86,51 +92,58 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
 	struct wm_hubs_data *hubs = codec->private_data;
-	u16 reg, dcs_cfg;
+	u16 reg, reg_l, reg_r, dcs_cfg;
 
 	/* Set for 32 series updates */
 	snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
 			    WM8993_DCS_SERIES_NO_01_MASK,
 			    32 << WM8993_DCS_SERIES_NO_01_SHIFT);
-
-	/* Enable the DC servo.  Write all bits to avoid triggering startup
-	 * or write calibration.
-	 */
-	snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
-			    0xFFFF,
-			    WM8993_DCS_ENA_CHAN_0 |
-			    WM8993_DCS_ENA_CHAN_1 |
-			    WM8993_DCS_TRIG_SERIES_1 |
-			    WM8993_DCS_TRIG_SERIES_0);
-
-	wait_for_dc_servo(codec);
+	wait_for_dc_servo(codec,
+			  WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1);
 
 	/* Apply correction to DC servo result */
 	if (hubs->dcs_codes) {
 		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
 			hubs->dcs_codes);
 
+		/* Different chips in the family support different
+		 * readback methods.
+		 */
+		switch (hubs->dcs_readback_mode) {
+		case 0:
+			reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+				& WM8993_DCS_INTEG_CHAN_0_MASK;;
+			reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
+				& WM8993_DCS_INTEG_CHAN_1_MASK;
+			break;
+		case 1:
+			reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+			reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+				>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+			reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+			break;
+		default:
+			WARN(1, "Unknown DCS readback method");
+			break;
+		}
+
 		/* HPOUT1L */
-		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) &
-			WM8993_DCS_INTEG_CHAN_0_MASK;;
-		reg += hubs->dcs_codes;
-		dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		if (reg_l + hubs->dcs_codes > 0 &&
+		    reg_l + hubs->dcs_codes < 0xff)
+			reg_l += hubs->dcs_codes;
+		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
 		/* HPOUT1R */
-		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) &
-			WM8993_DCS_INTEG_CHAN_1_MASK;
-		reg += hubs->dcs_codes;
-		dcs_cfg |= reg;
+		if (reg_r + hubs->dcs_codes > 0 &&
+		    reg_r + hubs->dcs_codes < 0xff)
+			reg_r += hubs->dcs_codes;
+		dcs_cfg |= reg_r;
 
 		/* Do it */
 		snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
-		snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
-				    WM8993_DCS_TRIG_DAC_WR_0 |
-				    WM8993_DCS_TRIG_DAC_WR_1,
-				    WM8993_DCS_TRIG_DAC_WR_0 |
-				    WM8993_DCS_TRIG_DAC_WR_1);
-
-		wait_for_dc_servo(codec);
+		wait_for_dc_servo(codec,
+				  WM8993_DCS_TRIG_DAC_WR_0 |
+				  WM8993_DCS_TRIG_DAC_WR_1);
 	}
 }
 
@@ -141,10 +154,16 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm_hubs_data *hubs = codec->private_data;
 	int ret;
 
 	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
 
+	/* If we're applying an offset correction then updating the
+	 * callibration would be likely to introduce further offsets. */
+	if (hubs->dcs_codes)
+		return ret;
+
 	/* Only need to do this if the outputs are active */
 	if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1)
 	    & (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA))
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 420104fe9c90..e51c16683589 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -21,6 +21,7 @@ extern const unsigned int wm_hubs_spkmix_tlv[];
 /* This *must* be the first element of the codec->private_data struct */
 struct wm_hubs_data {
 	int dcs_codes;
+	int dcs_readback_mode;
 	int hp_startup_mode;
 };
 
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 62af7e025e7f..adadcd3aa1b1 100644
--- a/