Merge branch 'for-linus' into for-next

Back-merged for refactoring beep stuff.

19 files changed, 164 insertions, 58 deletions

diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c
index 066f5f3e3f4c..c3908862bc8b 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 160b1bd0cd62..24d44b2f61ac 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 55c9d8c8d3c1..02f90b4f8b86 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 30bcfe470f83..4ff60a6427d9 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 3536b076b529..0aabfedeecba 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 39a510699b86..5868c61797bc 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 db02c1e96b08..0792b5725f9c 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 72ebb8a36b13..60d08f669f0c 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 7d941ef54172..d0100a85e189 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 f772585c89ba..c0bf15554507 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 d57c81e79edd..356673106788 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 2ffdc35d5ffd..806407a3973e 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 b8d461db369f..b82bbf584146 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 cd17b477781d..cdeb301da1f6 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 ec0efc1443ba..0e8b3aaf6c8d 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 9bb927325993..a64b93425ae3 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 c27069d24d77..729958713cd4 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 34dc47b9581c..a294d942b9f7 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 803953a9bff3..2da8ad75fd96 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;
 		}