Merge branch 'for-linus' into for-next

Back-merged for refactoring beep stuff.

Documentation/sound/alsa/ALSA-Configuration.txt

···
     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.
 

Documentation/sound/alsa/seq_oss.html

···
 <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>

sound/core/seq/oss/seq_oss_event.c

···
 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) {
···
 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) {

sound/core/seq/seq_timer.c

···
 			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;

sound/core/vmaster.c

···
 	}
 	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,

sound/oss/sequencer.c

···
 		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);
···
 		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)

sound/pci/asihpi/asihpi.c

···
 
 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;
···
 
 	if (snd_BUG_ON(!asihpi))
 		return -EINVAL;
+	card = asihpi->card;
 	strcpy(card->mixername, "Asihpi Mixer");
 
 	err =

sound/pci/hda/hda_codec.c

···
 
 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);
 }
 
 /**
···
 	unsigned int shift, num_elems, mask;
 	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)
···
 					  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;
 	}
 
···
 				continue;
 			}
 			for (n = prev_nid + 1; n <= val; n++) {
-				if (conns >= max_conns)
-					return -ENOSPC;
-				conn_list[conns++] = n;
+				if (conn_list) {
+					if (conns >= max_conns)
+						return -ENOSPC;
+					conn_list[conns] = n;
+				}
+				conns++;
 			}
 		} else {
-			if (conns >= max_conns)
-				return -ENOSPC;
-			conn_list[conns++] = val;
+			if (conn_list) {
+				if (conns >= max_conns)
+					return -ENOSPC;
+				conn_list[conns] = val;
+			}
+			conns++;
 		}
 		prev_nid = val;
 	}
···
 		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)
···
 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);
 

sound/pci/hda/patch_ca0132.c

···
 	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));
···
 		   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;
 	}
···
 	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 */
···
  */
 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 */
···
 		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;
 }
···
 #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);

sound/pci/hda/patch_cirrus.c

···
 	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);
···
 	spec = cs_alloc_spec(codec, CS4210_VENDOR_NID);
 	if (!spec)
 		return -ENOMEM;
+
+	spec->gen.automute_hook = cs_automute;
 
 	snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl,
 			   cs421x_fixups);

sound/pci/hda/patch_conexant.c

···
 	}
 
 	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;
 }
···
 	}
 
 	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;
 }
···
 	}
 
 	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;
 }
···
 	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,
···
 
 	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.

sound/pci/hda/patch_realtek.c

···
 	case 0x10ec0290:
 		spec->codec_variant = ALC269_TYPE_ALC280;
 		break;
+	case 0x10ec0233:
 	case 0x10ec0282:
 	case 0x10ec0283:
 		spec->codec_variant = ALC269_TYPE_ALC282;
···
  */
 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 },

sound/pci/hda/patch_sigmatel.c

···
 	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
  */
···
 		stac_free(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;
 

sound/pci/ice1712/ice1712.c

···
 	snd_ice1712_proc_init(ice);
 	synchronize_irq(pci->irq);
 
+	card->private_data = ice;
+
 	err = pci_request_regions(pci, "ICE1712");
 	if (err < 0) {
 		kfree(ice);

sound/soc/codecs/wm5102.c

···
 	{ 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)
 {
···
 	case 0:
 		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;
 	}
 
···
 
 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),
···
 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,

sound/soc/codecs/wm5110.c

···
 
 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),
···
 
 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),
···
 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,
···
 		       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),
 

sound/soc/codecs/wm8350.c

···
 	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;
 }
···
 	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;
 }

sound/soc/codecs/wm8960.c

···
  * 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 },
···
 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),

sound/soc/tegra/tegra20_i2s.h

···
 
 #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 */

sound/soc/tegra/tegra30_i2s.h

···
 
 #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 */

sound/usb/card.c

···
 			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;
 		}