Merge branch 'for-linus' into for-next

Back-merged for refactoring beep stuff.

21 files changed, 166 insertions, 60 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index ce6581c..4499bd9 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -912,7 +912,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     models depending on the codec chip.  The list of available models
     is found in HD-Audio-Models.txt
 
-    The model name "genric" is treated as a special case.  When this
+    The model name "generic" is treated as a special case.  When this
     model is given, the driver uses the generic codec parser without
     "codec-patch".  It's sometimes good for testing and debugging.
 
diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html
index d9776cf..9663b45 100644
--- a/Documentation/sound/alsa/seq_oss.html
+++ b/Documentation/sound/alsa/seq_oss.html
@@ -285,7 +285,7 @@ sample data.
 <H4>
 7.2.4 Close Callback</H4>
 The <TT>close</TT> callback is called when this device is closed by the
-applicaion. If any private data was allocated in open callback, it must
+application. If any private data was allocated in open callback, it must
 be released in the close callback. The deletion of ALSA port should be
 done here, too. This callback must not be NULL.
 <H4>
diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c
index 066f5f3..c390886 100644
--- a/sound/core/seq/oss/seq_oss_event.c
+++ b/sound/core/seq/oss/seq_oss_event.c
@@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev
 static int
 note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
 {
-	struct seq_oss_synthinfo *info = &dp->synths[dev];
+	struct seq_oss_synthinfo *info;
+
+	if (!snd_seq_oss_synth_is_valid(dp, dev))
+		return -ENXIO;
+
+	info = &dp->synths[dev];
 	switch (info->arg.event_passing) {
 	case SNDRV_SEQ_OSS_PROCESS_EVENTS:
 		if (! info->ch || ch < 0 || ch >= info->nr_voices) {
@@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st
 static int
 note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
 {
-	struct seq_oss_synthinfo *info = &dp->synths[dev];
+	struct seq_oss_synthinfo *info;
+
+	if (!snd_seq_oss_synth_is_valid(dp, dev))
+		return -ENXIO;
+
+	info = &dp->synths[dev];
 	switch (info->arg.event_passing) {
 	case SNDRV_SEQ_OSS_PROCESS_EVENTS:
 		if (! info->ch || ch < 0 || ch >= info->nr_voices) {
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 160b1bd..24d44b2 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -290,10 +290,10 @@ int snd_seq_timer_open(struct snd_seq_queue *q)
 			tid.device = SNDRV_TIMER_GLOBAL_SYSTEM;
 			err = snd_timer_open(&t, str, &tid, q->queue);
 		}
-		if (err < 0) {
-			snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
-			return err;
-		}
+	}
+	if (err < 0) {
+		snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
+		return err;
 	}
 	t->callback = snd_seq_timer_interrupt;
 	t->callback_data = q;
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 55c9d8c..02f90b4 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -213,7 +213,10 @@ static int slave_put(struct snd_kcontrol *kcontrol,
 	}
 	if (!changed)
 		return 0;
-	return slave_put_val(slave, ucontrol);
+	err = slave_put_val(slave, ucontrol);
+	if (err < 0)
+		return err;
+	return 1;
 }
 
 static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index 30bcfe4..4ff60a6 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -545,6 +545,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
 		case MIDI_PGM_CHANGE:
 			if (seq_mode == SEQ_2)
 			{
+				if (chn > 15)
+					break;
+
 				synth_devs[dev]->chn_info[chn].pgm_num = p1;
 				if ((int) dev >= num_synths)
 					synth_devs[dev]->set_instr(dev, chn, p1);
@@ -596,6 +599,9 @@ static void seq_chn_common_event(unsigned char *event_rec)
 		case MIDI_PITCH_BEND:
 			if (seq_mode == SEQ_2)
 			{
+				if (chn > 15)
+					break;
+
 				synth_devs[dev]->chn_info[chn].bender_value = w14;
 
 				if ((int) dev < num_synths)
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 3536b07..0aabfed 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -2549,7 +2549,7 @@ static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi,
 
 static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 {
-	struct snd_card *card = asihpi->card;
+	struct snd_card *card;
 	unsigned int idx = 0;
 	unsigned int subindex = 0;
 	int err;
@@ -2557,6 +2557,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 
 	if (snd_BUG_ON(!asihpi))
 		return -EINVAL;
+	card = asihpi->card;
 	strcpy(card->mixername, "Asihpi Mixer");
 
 	err =
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 39a5106..5868c61 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -494,7 +494,7 @@ static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid)
 
 int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid)
 {
-	return get_num_conns(codec, nid) & AC_CLIST_LENGTH;
+	return snd_hda_get_raw_connections(codec, nid, NULL, 0);
 }
 
 /**
@@ -517,9 +517,6 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 	hda_nid_t prev_nid;
 	int null_count = 0;
 
-	if (snd_BUG_ON(!conn_list || max_conns <= 0))
-		return -EINVAL;
-
 	parm = get_num_conns(codec, nid);
 	if (!parm)
 		return 0;
@@ -545,7 +542,8 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 					  AC_VERB_GET_CONNECT_LIST, 0);
 		if (parm == -1 && codec->bus->rirb_error)
 			return -EIO;
-		conn_list[0] = parm & mask;
+		if (conn_list)
+			conn_list[0] = parm & mask;
 		return 1;
 	}
 
@@ -580,14 +578,20 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
 				continue;
 			}
 			for (n = prev_nid + 1; n <= val; n++) {
+				if (conn_list) {
+					if (conns >= max_conns)
+						return -ENOSPC;
+					conn_list[conns] = n;
+				}
+				conns++;
+			}
+		} else {
+			if (conn_list) {
 				if (conns >= max_conns)
 					return -ENOSPC;
-				conn_list[conns++] = n;
+				conn_list[conns] = val;
 			}
-		} else {
-			if (conns >= max_conns)
-				return -ENOSPC;
-			conn_list[conns++] = val;
+			conns++;
 		}
 		prev_nid = val;
 	}
@@ -3331,6 +3335,8 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec,
 		return -EBUSY;
 	}
 	spdif = snd_array_new(&codec->spdif_out);
+	if (!spdif)
+		return -ENOMEM;
 	for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
 		kctl = snd_ctl_new1(dig_mix, codec);
 		if (!kctl)
@@ -3428,11 +3434,16 @@ static struct snd_kcontrol_new spdif_share_sw = {
 int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
 				  struct hda_multi_out *mout)
 {
+	struct snd_kcontrol *kctl;
+
 	if (!mout->dig_out_nid)
 		return 0;
+
+	kctl = snd_ctl_new1(&spdif_share_sw, mout);
+	if (!kctl)
+		return -ENOMEM;
 	/* ATTENTION: here mout is passed as private_data, instead of codec */
-	return snd_hda_ctl_add(codec, mout->dig_out_nid,
-			      snd_ctl_new1(&spdif_share_sw, mout));
+	return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl);
 }
 EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw);
 
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index db02c1e..0792b57 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2298,6 +2298,11 @@ static int dspxfr_one_seg(struct hda_codec *codec,
 	hda_frame_size_words = ((sample_rate_div == 0) ? 0 :
 			(num_chans * sample_rate_mul / sample_rate_div));
 
+	if (hda_frame_size_words == 0) {
+		snd_printdd(KERN_ERR "frmsz zero\n");
+		return -EINVAL;
+	}
+
 	buffer_size_words = min(buffer_size_words,
 				(unsigned int)(UC_RANGE(chip_addx, 1) ?
 				65536 : 32768));
@@ -2308,8 +2313,7 @@ static int dspxfr_one_seg(struct hda_codec *codec,
 		   chip_addx, hda_frame_size_words, num_chans,
 		   sample_rate_mul, sample_rate_div, buffer_size_words);
 
-	if ((buffer_addx == NULL) || (hda_frame_size_words == 0) ||
-	    (buffer_size_words < hda_frame_size_words)) {
+	if (buffer_size_words < hda_frame_size_words) {
 		snd_printdd(KERN_ERR "dspxfr_one_seg:failed\n");
 		return -EINVAL;
 	}
@@ -3235,7 +3239,7 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val)
 	struct ca0132_spec *spec = codec->spec;
 	unsigned int tmp;
 
-	if (!dspload_is_loaded(codec))
+	if (spec->dsp_state != DSP_DOWNLOADED)
 		return 0;
 
 	/* if CrystalVoice if off, vipsource should be 0 */
@@ -4263,11 +4267,12 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
  */
 static void ca0132_setup_defaults(struct hda_codec *codec)
 {
+	struct ca0132_spec *spec = codec->spec;
 	unsigned int tmp;
 	int num_fx;
 	int idx, i;
 
-	if (!dspload_is_loaded(codec))
+	if (spec->dsp_state != DSP_DOWNLOADED)
 		return;
 
 	/* out, in effects + voicefx */
@@ -4347,12 +4352,16 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
 		return false;
 
 	dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
-	dspload_image(codec, dsp_os_image, 0, 0, true, 0);
+	if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
+		pr_err("ca0132 dspload_image failed.\n");
+		goto exit_download;
+	}
+
 	dsp_loaded = dspload_wait_loaded(codec);
 
+exit_download:
 	release_firmware(fw_entry);
 
-
 	return dsp_loaded;
 }
 
@@ -4363,16 +4372,13 @@ static void ca0132_download_dsp(struct hda_codec *codec)
 #ifndef CONFIG_SND_HDA_CODEC_CA0132_DSP
 	return; /* NOP */
 #endif
-	spec->dsp_state = DSP_DOWNLOAD_INIT;
 
-	if (spec->dsp_state == DSP_DOWNLOAD_INIT) {
-		chipio_enable_clocks(codec);
-		spec->dsp_state = DSP_DOWNLOADING;
-		if (!ca0132_download_dsp_images(codec))
-			spec->dsp_state = DSP_DOWNLOAD_FAILED;
-		else
-			spec->dsp_state = DSP_DOWNLOADED;
-	}
+	chipio_enable_clocks(codec);
+	spec->dsp_state = DSP_DOWNLOADING;
+	if (!ca0132_download_dsp_images(codec))
+		spec->dsp_state = DSP_DOWNLOAD_FAILED;
+	else
+		spec->dsp_state = DSP_DOWNLOADED;
 
 	if (spec->dsp_state == DSP_DOWNLOADED)
 		ca0132_set_dsp_msr(codec, true);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 72ebb8a..60d08f6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -506,6 +506,8 @@ static int patch_cs420x(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 
+	spec->gen.automute_hook = cs_automute;
+
 	snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
 			   cs420x_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
@@ -893,6 +895,8 @@ static int patch_cs4210(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 
+	spec->gen.automute_hook = cs_automute;
+
 	snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl,
 			   cs421x_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7d941ef..d0100a8 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1142,7 +1142,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 	}
 
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	return 0;
 }
@@ -1921,7 +1921,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 	}
 
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	return 0;
 }
@@ -3099,7 +3099,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 	}
 
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	return 0;
 }
@@ -3191,11 +3191,17 @@ static int cx_auto_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
+static void cx_auto_free(struct hda_codec *codec)
+{
+	snd_hda_detach_beep_device(codec);
+	snd_hda_gen_free(codec);
+}
+
 static const struct hda_codec_ops cx_auto_patch_ops = {
 	.build_controls = cx_auto_build_controls,
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = snd_hda_gen_init,
-	.free = snd_hda_gen_free,
+	.free = cx_auto_free,
 	.unsol_event = snd_hda_jack_unsol_event,
 #ifdef CONFIG_PM
 	.check_power_status = snd_hda_gen_check_power_status,
@@ -3390,7 +3396,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	codec->patch_ops = cx_auto_patch_ops;
 	if (spec->beep_amp)
-		snd_hda_attach_beep_device(codec, spec->beep_amp);
+		snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp));
 
 	/* Some laptops with Conexant chips show stalls in S3 resume,
 	 * which falls into the single-cmd mode.
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f772585..c0bf155 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3182,6 +3182,7 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0290:
 		spec->codec_variant = ALC269_TYPE_ALC280;
 		break;
+	case 0x10ec0233:
 	case 0x10ec0282:
 	case 0x10ec0283:
 		spec->codec_variant = ALC269_TYPE_ALC282;
@@ -3881,6 +3882,7 @@ static int patch_alc680(struct hda_codec *codec)
  */
 static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
+	{ .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
 	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
 	{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
 	{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d57c81e..3566731 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
 	return 0;
 }
 
+/* check whether a built-in speaker is included in parsed pins */
+static bool has_builtin_speaker(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t *nid_pin;
+	int nids, i;
+
+	if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) {
+		nid_pin = spec->gen.autocfg.line_out_pins;
+		nids = spec->gen.autocfg.line_outs;
+	} else {
+		nid_pin = spec->gen.autocfg.speaker_pins;
+		nids = spec->gen.autocfg.speaker_outs;
+	}
+
+	for (i = 0; i < nids; i++) {
+		unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]);
+		if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT)
+			return true;
+	}
+	return false;
+}
+
 /*
  * PC beep controls
  */
@@ -3891,6 +3914,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 		return err;
 	}
 
+	/* Don't GPIO-mute speakers if there are no internal speakers, because
+	 * the GPIO might be necessary for Headphone
+	 */
+	if (spec->eapd_switch && !has_builtin_speaker(codec))
+		spec->eapd_switch = 0;
+
 	codec->proc_widget_hook = stac92hd7x_proc_hook;
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 2ffdc35..806407a 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2594,6 +2594,8 @@ static int snd_ice1712_create(struct snd_card *card,
 	snd_ice1712_proc_init(ice);
 	synchronize_irq(pci->irq);
 
+	card->private_data = ice;
+
 	err = pci_request_regions(pci, "ICE1712");
 	if (err < 0) {
 		kfree(ice);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index b8d461d..b82bbf5 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -573,6 +573,13 @@ static const struct reg_default wm5102_sysclk_reva_patch[] = {
 	{ 0x025e, 0x0112 },
 };
 
+static const struct reg_default wm5102_sysclk_revb_patch[] = {
+	{ 0x3081, 0x08FE },
+	{ 0x3083, 0x00ED },
+	{ 0x30C1, 0x08FE },
+	{ 0x30C3, 0x00ED },
+};
+
 static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
 			    struct snd_kcontrol *kcontrol, int event)
 {
@@ -587,6 +594,10 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
 		patch = wm5102_sysclk_reva_patch;
 		patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch);
 		break;
+	default:
+		patch = wm5102_sysclk_revb_patch;
+		patch_size = ARRAY_SIZE(wm5102_sysclk_revb_patch);
+		break;
 	}
 
 	switch (event) {
@@ -755,7 +766,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
 
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
 SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
 	   ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
@@ -767,7 +778,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
 SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index cd17b47..cdeb301 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -213,9 +213,9 @@ ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
 
 SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
 	   ARIZONA_OUT1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
 	   ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("OUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
 	   ARIZONA_OUT3_OSR_SHIFT, 1, 0),
 SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
 	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
@@ -226,9 +226,9 @@ SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
 
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
-SOC_DOUBLE_R("OUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+SOC_DOUBLE_R("HPOUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
@@ -240,10 +240,10 @@ SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L,
 SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("OUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
@@ -260,11 +260,11 @@ SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
 		       ARIZONA_OUTPUT_PATH_CONFIG_1R,
 		       ARIZONA_OUT1L_PGA_VOL_SHIFT,
 		       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
 		       ARIZONA_OUTPUT_PATH_CONFIG_2R,
 		       ARIZONA_OUT2L_PGA_VOL_SHIFT,
 		       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("OUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
 		       ARIZONA_OUTPUT_PATH_CONFIG_3R,
 		       ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ec0efc1..0e8b3aa 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1301,7 +1301,7 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data)
 	if (device_may_wakeup(wm8350->dev))
 		pm_wakeup_event(wm8350->dev, 250);
 
-	schedule_delayed_work(&priv->hpl.work, 200);
+	schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200));
 
 	return IRQ_HANDLED;
 }
@@ -1318,7 +1318,7 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data)
 	if (device_may_wakeup(wm8350->dev))
 		pm_wakeup_event(wm8350->dev, 250);
 
-	schedule_delayed_work(&priv->hpr.work, 200);
+	schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200));
 
 	return IRQ_HANDLED;
 }
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 9bb9273..a64b934 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -53,8 +53,8 @@
  * using 2 wire for device control, so we cache them instead.
  */
 static const struct reg_default wm8960_reg_defaults[] = {
-	{  0x0, 0x0097 },
-	{  0x1, 0x0097 },
+	{  0x0, 0x00a7 },
+	{  0x1, 0x00a7 },
 	{  0x2, 0x0000 },
 	{  0x3, 0x0000 },
 	{  0x4, 0x0000 },
@@ -323,8 +323,8 @@ SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
 SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
 		   wm8960_rin, ARRAY_SIZE(wm8960_rin)),
 
-SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0),
-SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER1, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0),
 
 SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
 SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index c27069d..7299587 100644
--- a/sound/soc/tegra/tegra20_i2s.h
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -121,7 +121,7 @@
 
 #define TEGRA20_I2S_TIMING_NON_SYM_ENABLE		(1 << 12)
 #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
-#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
+#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7ff
 #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK	(TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
 
 /* Fields in TEGRA20_I2S_FIFO_SCR */
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 34dc47b..a294d94 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -110,7 +110,7 @@
 
 #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE		(1 << 12)
 #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT	0
-#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7fff
+#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US	0x7ff
 #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK	(TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT)
 
 /* Fields in TEGRA30_I2S_OFFSET */
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 803953a..2da8ad7 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -244,6 +244,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 			usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
 
 		if (!assoc) {
+			/*
+			 * Firmware writers cannot count to three.  So to find
+			 * the IAD on the NuForce UDH-100, also check the next
+			 * interface.
+			 */
+			struct usb_interface *iface =
+				usb_ifnum_to_if(dev, ctrlif + 1);
+			if (iface &&
+			    iface->intf_assoc &&
+			    iface->intf_assoc->bFunctionClass == USB_CLASS_AUDIO &&
+			    iface->intf_assoc->bFunctionProtocol == UAC_VERSION_2)
+				assoc = iface->intf_assoc;
+		}
+
+		if (!assoc) {
 			snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n");
 			return -EINVAL;
 		}