16 files changed, 205 insertions, 83 deletions

diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index e89991ea3543..3b4413448226 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 a3d638c8c1fd..ba2098d20ccc 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 49e939e7e5cd..5991d14e1ec0 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 fc767b6b4785..ff614dd824c1 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 31ac5538fe7e..5da30eb6ad00 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 1072621e93fd..9d1df2628136 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 7e4a627b4c7e..4e212ed62ea6 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 0417dae32e6f..19ad590ca0b3 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 79f0f4ad242c..d3955096d872 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/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
index 22208b373fb9..e1ec6d91ea38 100644
--- a/sound/soc/fsl/mpc5200_dma.h
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -73,7 +73,7 @@ struct psc_dma {
 };
 
 /* Utility for retrieving psc_dma_stream structure from a substream */
-inline struct psc_dma_stream *
+static inline struct psc_dma_stream *
 to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma)
 {
         if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3396a0db06ba..ec4acac49ebd 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 b7aadd614c70..b5855114667e 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 9593b91452b9..6f6596cf2b19 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 5367cd1e52d9..30364aba79cc 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 a6b0e51b3a9a..09bd943c43bf 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 a060d005e209..736d134cc03c 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",