Merge commit 'v2.6.35-rc6' into devicetree/next

Conflicts:
	arch/sparc/kernel/prom_64.c

15 files changed, 204 insertions, 82 deletions

diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index e89991e..3b44134 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -941,11 +941,11 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao,
 
 }
 
-static long outstream_get_space_available(struct hpi_hostbuffer_status
+static u32 outstream_get_space_available(struct hpi_hostbuffer_status
 	*status)
 {
-	return status->size_in_bytes - ((long)(status->host_index) -
-		(long)(status->dSP_index));
+	return status->size_in_bytes - (status->host_index -
+		status->dSP_index);
 }
 
 static void outstream_write(struct hpi_adapter_obj *pao,
@@ -954,7 +954,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 	struct hpi_hw_obj *phw = pao->priv;
 	struct bus_master_interface *interface = phw->p_interface_buffer;
 	struct hpi_hostbuffer_status *status;
-	long space_available;
+	u32 space_available;
 
 	if (!phw->outstream_host_buffer_size[phm->obj_index]) {
 		/* there  is no BBM buffer, write via message */
@@ -1007,7 +1007,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 	}
 
 	space_available = outstream_get_space_available(status);
-	if (space_available < (long)phm->u.d.u.data.data_size) {
+	if (space_available < phm->u.d.u.data.data_size) {
 		phr->error = HPI_ERROR_INVALID_DATASIZE;
 		return;
 	}
@@ -1018,7 +1018,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 		&& hpios_locked_mem_valid(&phw->outstream_host_buffers[phm->
 				obj_index])) {
 		u8 *p_bbm_data;
-		long l_first_write;
+		u32 l_first_write;
 		u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
 
 		if (hpios_locked_mem_get_virt_addr(&phw->
@@ -1248,9 +1248,9 @@ static void instream_start(struct hpi_adapter_obj *pao,
 	hw_message(pao, phm, phr);
 }
 
-static long instream_get_bytes_available(struct hpi_hostbuffer_status *status)
+static u32 instream_get_bytes_available(struct hpi_hostbuffer_status *status)
 {
-	return (long)(status->dSP_index) - (long)(status->host_index);
+	return status->dSP_index - status->host_index;
 }
 
 static void instream_read(struct hpi_adapter_obj *pao,
@@ -1259,9 +1259,9 @@ static void instream_read(struct hpi_adapter_obj *pao,
 	struct hpi_hw_obj *phw = pao->priv;
 	struct bus_master_interface *interface = phw->p_interface_buffer;
 	struct hpi_hostbuffer_status *status;
-	long data_available;
+	u32 data_available;
 	u8 *p_bbm_data;
-	long l_first_read;
+	u32 l_first_read;
 	u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
 
 	if (!phw->instream_host_buffer_size[phm->obj_index]) {
@@ -1272,7 +1272,7 @@ static void instream_read(struct hpi_adapter_obj *pao,
 
 	status = &interface->instream_host_buffer_status[phm->obj_index];
 	data_available = instream_get_bytes_available(status);
-	if (data_available < (long)phm->u.d.u.data.data_size) {
+	if (data_available < phm->u.d.u.data.data_size) {
 		phr->error = HPI_ERROR_INVALID_DATASIZE;
 		return;
 	}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a3d638c..ba2098d 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -784,6 +784,9 @@ static int read_pin_defaults(struct hda_codec *codec)
 		pin->nid = nid;
 		pin->cfg = snd_hda_codec_read(codec, nid, 0,
 					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		pin->ctrl = snd_hda_codec_read(codec, nid, 0,
+					       AC_VERB_GET_PIN_WIDGET_CONTROL,
+					       0);
 	}
 	return 0;
 }
@@ -912,15 +915,38 @@ static void restore_pincfgs(struct hda_codec *codec)
 void snd_hda_shutup_pins(struct hda_codec *codec)
 {
 	int i;
+	/* don't shut up pins when unloading the driver; otherwise it breaks
+	 * the default pin setup at the next load of the driver
+	 */
+	if (codec->bus->shutdown)
+		return;
 	for (i = 0; i < codec->init_pins.used; i++) {
 		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
 		/* use read here for syncing after issuing each verb */
 		snd_hda_codec_read(codec, pin->nid, 0,
 				   AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
 	}
+	codec->pins_shutup = 1;
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
+/* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
+static void restore_shutup_pins(struct hda_codec *codec)
+{
+	int i;
+	if (!codec->pins_shutup)
+		return;
+	if (codec->bus->shutdown)
+		return;
+	for (i = 0; i < codec->init_pins.used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+		snd_hda_codec_write(codec, pin->nid, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL,
+				    pin->ctrl);
+	}
+	codec->pins_shutup = 0;
+}
+
 static void init_hda_cache(struct hda_cache_rec *cache,
 			   unsigned int record_size);
 static void free_hda_cache(struct hda_cache_rec *cache);
@@ -2907,6 +2933,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 			    codec->afg ? codec->afg : codec->mfg,
 			    AC_PWRST_D0);
 	restore_pincfgs(codec); /* restore all current pin configs */
+	restore_shutup_pins(codec);
 	hda_exec_init_verbs(codec);
 	if (codec->patch_ops.resume)
 		codec->patch_ops.resume(codec);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 49e939e..5991d14 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -821,6 +821,7 @@ struct hda_codec {
 	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
 					    * (e.g. Conexant codecs)
 					    */
+	unsigned int pins_shutup:1;	/* pins are shut up */
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
@@ -897,7 +898,9 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec);
 /* the struct for codec->pin_configs */
 struct hda_pincfg {
 	hda_nid_t nid;
-	unsigned int cfg;
+	unsigned char ctrl;	/* current pin control value */
+	unsigned char pad;	/* reserved */
+	unsigned int cfg;	/* default configuration */
 };
 
 unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fc767b6..ff614dd8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1268,8 +1268,10 @@ static int alc_auto_parse_customize_define(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	ass = codec->subsystem_id & 0xffff;
-	if (ass != codec->bus->pci->subsystem_device && (ass & 1))
+	if (ass != codec->bus->pci->subsystem_device && (ass & 1)) {
+		spec->cdefine.enable_pcbeep = 1; /* assume always enabled */
 		goto do_sku;
+	}
 
 	nid = 0x1d;
 	if (codec->vendor_id == 0x10ec0260)
@@ -2547,7 +2549,7 @@ static struct snd_kcontrol_new alc_beep_mixer[] = {
 static int alc_build_controls(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	struct snd_kcontrol *kctl;
+	struct snd_kcontrol *kctl = NULL;
 	struct snd_kcontrol_new *knew;
 	int i, j, err;
 	unsigned int u;
@@ -2619,16 +2621,18 @@ static int alc_build_controls(struct hda_codec *codec)
 	}
 
 	/* assign Capture Source enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
-	if (!kctl)
-		kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
-	for (i = 0; kctl && i < kctl->count; i++) {
-		hda_nid_t *nids = spec->capsrc_nids;
-		if (!nids)
-			nids = spec->adc_nids;
-		err = snd_hda_add_nid(codec, kctl, i, nids[i]);
-		if (err < 0)
-			return err;
+	if (spec->capsrc_nids || spec->adc_nids) {
+		kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
+		if (!kctl)
+			kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
+		for (i = 0; kctl && i < kctl->count; i++) {
+			hda_nid_t *nids = spec->capsrc_nids;
+			if (!nids)
+				nids = spec->adc_nids;
+			err = snd_hda_add_nid(codec, kctl, i, nids[i]);
+			if (err < 0)
+				return err;
+		}
 	}
 	if (spec->cap_mixer) {
 		const char *kname = kctl ? kctl->id.name : NULL;
@@ -6948,7 +6952,7 @@ static struct hda_input_mux mb5_capture_source = {
 	.num_items = 3,
 	.items = {
 		{ "Mic", 0x1 },
-		{ "Line", 0x2 },
+		{ "Line", 0x7 },
 		{ "CD", 0x4 },
 	},
 };
@@ -7469,8 +7473,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, 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("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
@@ -7853,10 +7857,9 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
 	{ }
 };
 
@@ -9485,6 +9488,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
 	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
+	SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
 	/* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
 	 * so apparently no perfect solution yet
 	 */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 31ac553..5da30eb 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -83,8 +83,8 @@ config SND_SOC_ALL_CODECS
 
 config SND_SOC_WM_HUBS
 	tristate
-	default y if SND_SOC_WM8993=y
-	default m if SND_SOC_WM8993=m
+	default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
+	default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m
 
 config SND_SOC_AC97_CODEC
 	tristate
diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c
index 1072621..9d1df262 100644
--- a/sound/soc/codecs/wm8727.c
+++ b/sound/soc/codecs/wm8727.c
@@ -127,6 +127,8 @@ static __devinit int wm8727_platform_probe(struct platform_device *pdev)
 		goto err_codec;
 	}
 
+	return 0;
+
 err_codec:
 	snd_soc_unregister_codec(codec);
 err:
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 7e4a627..4e212ed 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -94,7 +94,6 @@ SOC_DAPM_SINGLE("Bypass Switch", WM8776_OUTMUX, 2, 1, 0),
 
 static const struct snd_soc_dapm_widget wm8776_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("AUX"),
-SND_SOC_DAPM_INPUT("AUX"),
 
 SND_SOC_DAPM_INPUT("AIN1"),
 SND_SOC_DAPM_INPUT("AIN2"),
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 0417dae..19ad590 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -885,7 +885,6 @@ static int wm8988_register(struct wm8988_priv *wm8988,
 	ret = snd_soc_register_dai(&wm8988_dai);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		snd_soc_unregister_codec(codec);
 		goto err_codec;
 	}
 
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 79f0f4a..d395509 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -612,7 +612,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
 								NUMDMA_MASK);
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
 				((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
 	}
 
 	if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -623,7 +622,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
 								NUMDMA_MASK);
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
 				((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
 	}
 }
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3396a0d..ec4acac 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -683,20 +683,15 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 
 	/* clock inversion (CKG2) */
 	data = 0;
-	switch (SH_FSI_INVERSION_MASK & flags) {
-	case SH_FSI_LRM_INV:
-		data = 1 << 12;
-		break;
-	case SH_FSI_BRM_INV:
-		data = 1 << 8;
-		break;
-	case SH_FSI_LRS_INV:
-		data = 1 << 4;
-		break;
-	case SH_FSI_BRS_INV:
-		data = 1 << 0;
-		break;
-	}
+	if (SH_FSI_LRM_INV & flags)
+		data |= 1 << 12;
+	if (SH_FSI_BRM_INV & flags)
+		data |= 1 << 8;
+	if (SH_FSI_LRS_INV & flags)
+		data |= 1 << 4;
+	if (SH_FSI_BRS_INV & flags)
+		data |= 1 << 0;
+
 	fsi_reg_write(fsi, CKG2, data);
 
 	/* do fmt, di fmt */
@@ -726,15 +721,15 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 		break;
 	case SH_FSI_FMT_TDM:
 		msg = "TDM";
-		data = CR_FMT(CR_TDM) | (fsi->chan - 1);
 		fsi->chan = is_play ?
 			SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+		data = CR_FMT(CR_TDM) | (fsi->chan - 1);
 		break;
 	case SH_FSI_FMT_TDM_DELAY:
 		msg = "TDM Delay";
-		data = CR_FMT(CR_TDM_D) | (fsi->chan - 1);
 		fsi->chan = is_play ?
 			SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+		data = CR_FMT(CR_TDM_D) | (fsi->chan - 1);
 		break;
 	default:
 		dev_err(dai->dev, "unknown format.\n");
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index b7aadd6..b585511 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -103,7 +103,8 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i
 	ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0),
 			      UAC2_CS_CUR,
 			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
-			      UAC2_CX_CLOCK_SELECTOR << 8, selector_id << 8,
+			      UAC2_CX_CLOCK_SELECTOR << 8,
+			      snd_usb_ctrl_intf(chip) | (selector_id << 8),
 			      &buf, sizeof(buf), 1000);
 
 	if (ret < 0)
@@ -120,7 +121,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
 
 	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			      UAC2_CS_CONTROL_CLOCK_VALID << 8, source_id << 8,
+			      UAC2_CS_CONTROL_CLOCK_VALID << 8,
+			      snd_usb_ctrl_intf(chip) | (source_id << 8),
 			      &data, sizeof(data), 1000);
 
 	if (err < 0) {
@@ -269,7 +271,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 	data[3] = rate >> 24;
 	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+				   UAC2_CS_CONTROL_SAM_FREQ << 8,
+				   snd_usb_ctrl_intf(chip) | (clock << 8),
 				   data, sizeof(data), 1000)) < 0) {
 		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
 			   dev->devnum, iface, fmt->altsetting, rate);
@@ -278,7 +281,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 
 	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
 				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+				   UAC2_CS_CONTROL_SAM_FREQ << 8,
+				   snd_usb_ctrl_intf(chip) | (clock << 8),
 				   data, sizeof(data), 1000)) < 0) {
 		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
 			   dev->devnum, iface, fmt->altsetting);
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 9593b91..6f6596c 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -427,6 +427,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 		if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
 			kfree(fp->rate_table);
 			kfree(fp);
+			fp = NULL;
 			continue;
 		}
 
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5367cd1..30364ab 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -206,6 +206,60 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 }
 
 /*
+ * Helper function to walk the array of sample rate triplets reported by
+ * the device. The problem is that we need to parse whole array first to
+ * get to know how many sample rates we have to expect.
+ * Then fp->rate_table can be allocated and filled.
+ */
+static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
+					const unsigned char *data)
+{
+	int i, nr_rates = 0;
+
+	fp->rates = fp->rate_min = fp->rate_max = 0;
+
+	for (i = 0; i < nr_triplets; i++) {
+		int min = combine_quad(&data[2 + 12 * i]);
+		int max = combine_quad(&data[6 + 12 * i]);
+		int res = combine_quad(&data[10 + 12 * i]);
+		int rate;
+
+		if ((max < 0) || (min < 0) || (res < 0) || (max < min))
+			continue;
+
+		/*
+		 * for ranges with res == 1, we announce a continuous sample
+		 * rate range, and this function should return 0 for no further
+		 * parsing.
+		 */
+		if (res == 1) {
+			fp->rate_min = min;
+			fp->rate_max = max;
+			fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
+			return 0;
+		}
+
+		for (rate = min; rate <= max; rate += res) {
+			if (fp->rate_table)
+				fp->rate_table[nr_rates] = rate;
+			if (!fp->rate_min || rate < fp->rate_min)
+				fp->rate_min = rate;
+			if (!fp->rate_max || rate > fp->rate_max)
+				fp->rate_max = rate;
+			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+
+			nr_rates++;
+
+			/* avoid endless loop */
+			if (res == 0)
+				break;
+		}
+	}
+
+	return nr_rates;
+}
+
+/*
  * parse the format descriptor and stores the possible sample rates
  * on the audioformat table (audio class v2).
  */
@@ -215,13 +269,20 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 {
 	struct usb_device *dev = chip->dev;
 	unsigned char tmp[2], *data;
-	int i, nr_rates, data_size, ret = 0;
+	int nr_triplets, data_size, ret = 0;
 	int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock);
 
+	if (clock < 0) {
+		snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n",
+				__func__, clock);
+		goto err;
+	}
+
 	/* get the number of sample rates first by only fetching 2 bytes */
 	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			      UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
 			      tmp, sizeof(tmp), 1000);
 
 	if (ret < 0) {
@@ -230,8 +291,8 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 		goto err;
 	}
 
-	nr_rates = (tmp[1] << 8) | tmp[0];
-	data_size = 2 + 12 * nr_rates;
+	nr_triplets = (tmp[1] << 8) | tmp[0];
+	data_size = 2 + 12 * nr_triplets;
 	data = kzalloc(data_size, GFP_KERNEL);
 	if (!data) {
 		ret = -ENOMEM;
@@ -241,7 +302,8 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 	/* now get the full information */
 	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			      UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
 			      data, data_size, 1000);
 
 	if (ret < 0) {
@@ -251,26 +313,28 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 		goto err_free;
 	}
 
-	fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
+	/* Call the triplet parser, and make sure fp->rate_table is NULL.
+	 * We just use the return value to know how many sample rates we
+	 * will have to deal with. */
+	kfree(fp->rate_table);
+	fp->rate_table = NULL;
+	fp->nr_rates = parse_uac2_sample_rate_range(fp, nr_triplets, data);
+
+	if (fp->nr_rates == 0) {
+		/* SNDRV_PCM_RATE_CONTINUOUS */
+		ret = 0;
+		goto err_free;
+	}
+
+	fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL);
 	if (!fp->rate_table) {
 		ret = -ENOMEM;
 		goto err_free;
 	}
 
-	fp->nr_rates = 0;
-	fp->rate_min = fp->rate_max = 0;
-
-	for (i = 0; i < nr_rates; i++) {
-		int rate = combine_quad(&data[2 + 12 * i]);
-
-		fp->rate_table[fp->nr_rates] = rate;
-		if (!fp->rate_min || rate < fp->rate_min)
-			fp->rate_min = rate;
-		if (!fp->rate_max || rate > fp->rate_max)
-			fp->rate_max = rate;
-		fp->rates |= snd_pcm_rate_to_rate_bit(rate);
-		fp->nr_rates++;
-	}
+	/* Call the triplet parser again, but this time, fp->rate_table is
+	 * allocated, so the rates will be stored */
+	parse_uac2_sample_rate_range(fp, nr_triplets, data);
 
 err_free:
 	kfree(data);
diff --git a/sound/usb/helper.h b/sound/usb/helper.h
index a6b0e51..09bd943 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -28,5 +28,9 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
 #define snd_usb_get_speed(dev) ((dev)->speed)
 #endif
 
+static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip)
+{
+	return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber;
+}
 
 #endif /* __USBAUDIO_HELPER_H */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index a060d00..736d134 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -297,20 +297,27 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v
 
 static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret)
 {
-	unsigned char buf[14]; /* enough space for one range of 4 bytes */
+	unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */
 	unsigned char *val;
-	int ret;
+	int ret, size;
 	__u8 bRequest;
 
-	bRequest = (request == UAC_GET_CUR) ?
-		UAC2_CS_CUR : UAC2_CS_RANGE;
+	if (request == UAC_GET_CUR) {
+		bRequest = UAC2_CS_CUR;
+		size = sizeof(__u16);
+	} else {
+		bRequest = UAC2_CS_RANGE;
+		size = sizeof(buf);
+	}
+
+	memset(buf, 0, sizeof(buf));
 
 	ret = snd_usb_ctl_msg(cval->mixer->chip->dev,
 			      usb_rcvctrlpipe(cval->mixer->chip->dev, 0),
 			      bRequest,
 			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 			      validx, cval->mixer->ctrlif | (cval->id << 8),
-			      buf, sizeof(buf), 1000);
+			      buf, size, 1000);
 
 	if (ret < 0) {
 		snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
@@ -318,6 +325,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v
 		return ret;
 	}
 
+	/* FIXME: how should we handle multiple triplets here? */
+
 	switch (request) {
 	case UAC_GET_CUR:
 		val = buf;
@@ -1098,6 +1107,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		}
 		break;
 
+	case USB_ID(0x046d, 0x0809):
+	case USB_ID(0x046d, 0x0991):
+	/* Most audio usb devices lie about volume resolution.
+	 * Most Logitech webcams have res = 384.
+	 * Proboly there is some logitech magic behind this number --fishor
+	 */
+		if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+			snd_printk(KERN_INFO
+				"set resolution quirk: cval->res = 384\n");
+			cval->res = 384;
+		}
+		break;
+
 	}
 
 	snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",