Merge branch 'fix/asoc' into for-linus

diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7c1cbf0a0dc4098326e587ee30b1ed268c460a45..67ebf1c21c043a61dc0b51012f00778612af7b45 100644 (file)
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -328,6 +328,8 @@ int snd_timer_close(struct snd_timer_instance *timeri)
                mutex_unlock(&register_mutex);
        } else {
                timer = timeri->timer;
+               if (snd_BUG_ON(!timer))
+                       goto out;
                /* wait, until the active callback is finished */
                spin_lock_irq(&timer->lock);
                while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
@@ -353,6 +355,7 @@ int snd_timer_close(struct snd_timer_instance *timeri)
                }
                mutex_unlock(&register_mutex);
        }
+ out:
        if (timeri->private_free)
                timeri->private_free(timeri);
        kfree(timeri->owner);
@@ -531,6 +534,8 @@ int snd_timer_stop(struct snd_timer_instance *timeri)
        if (err < 0)
                return err;
        timer = timeri->timer;
+       if (!timer)
+               return -EINVAL;
        spin_lock_irqsave(&timer->lock, flags);
        timeri->cticks = timeri->ticks;
        timeri->pticks = 0;

diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c
index 8f7d175767a21e75b5167bf7eb117295bef3d250..6f13ab4afc6bcf60537adb93936af23ff107db51 100644 (file)
--- a/sound/oss/pas2_pcm.c
+++ b/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@ static int pcm_set_speed(int arg)
 
        if (pcm_channels & 2)
        {
-               foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
-               arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
+               foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
+               arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
        }
        else
        {
-               foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
-               arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
+               foo = (PIT_TICK_RATE + (arg / 2)) / arg;
+               arg = (PIT_TICK_RATE + (foo / 2)) / foo;
        }
 
        pcm_speed = arg;

diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 9b800ce5100ef286864234b308fb646c7d5dc9a1..2fc0624024b5c44e84f60c9e4e20926239b9319a 100644 (file)
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -673,7 +673,8 @@ static void configure_nonsound_components(void)
 
        if (pss_cdrom_port == -1) {     /* If cdrom port enablation wasn't requested */
                printk(KERN_INFO "PSS: CDROM port not enabled.\n");
-       } else if (check_region(pss_cdrom_port, 2)) {
+       } else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) {
+               pss_cdrom_port = -1;
                printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
        } else {
                set_io_base(devc, CONF_CDROM, pss_cdrom_port);
@@ -1232,7 +1233,8 @@ static void __exit cleanup_pss(void)
                if(pssmpu)
                        unload_pss_mpu(&cfg_mpu);
                unload_pss(&cfg);
-       }
+       } else if (pss_cdrom_port != -1)
+               release_region(pss_cdrom_port, 2);
 
        if(!pss_keep_settings)  /* Keep hardware settings if asked */
        {

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5bf8e096039c54f1b351d14493fe8dd9012..88168044375f9d7fba0b41d7567ee30c263e698e 100644 (file)
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
 # ALSA PCI drivers
 
+config SND_TEA575X
+       tristate
+       depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+       default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
 menuconfig SND_PCI
        bool "PCI sound devices"
        depends on PCI
@@ -563,11 +568,6 @@ config SND_FM801_TEA575X_BOOL
          FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
          SF64-PCR) into the snd-fm801 driver.
 
-config SND_TEA575X
-       tristate
-       depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
-       default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
 source "sound/pci/hda/Kconfig"
 
 config SND_HDSP

diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca13115b2ba92d167d89151f502509e0eed4..bd47521b24ec1fa336f0b0a952d2b41c39cc5dd6 100644 (file)
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
        if (!p_cache)
                return NULL;
 
-       p_cache->p_info =
-               kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+       p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
+                                 GFP_KERNEL);
        if (!p_cache->p_info) {
                kfree(p_cache);
                return NULL;
        }
-       memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
        p_cache->cache_size_in_bytes = size_in_bytes;
        p_cache->control_count = control_count;
        p_cache->p_cache = p_dsp_control_buffer;

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a270c9f13cf137fb68a78898d4d961200c5..579fc0dce12850e8e7cf604039def20abd54b025 100644 (file)
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
        int err;
 
        snd_azf3328_dbgcallenter();
-       if (dev >= SNDRV_CARDS)
-               return -ENODEV;
+       if (dev >= SNDRV_CARDS) {
+               err = -ENODEV;
+               goto out;
+       }
        if (!enable[dev]) {
                dev++;
-               return -ENOENT;
+               err = -ENOENT;
+               goto out;
        }
 
        err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
        if (err < 0)
-               return err;
+               goto out;
 
        strcpy(card->driver, "AZF3328");
        strcpy(card->shortname, "Aztech AZF3328 (PCI168)");

diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
index 14fdcf29b15457062899907e7c9fcf5d8fa60a0e..5ac0e2162a46c2bde510c96ed63804e380e567a6 100644 (file)
--- a/sound/pci/hda/alc269_quirks.c
+++ b/sound/pci/hda/alc269_quirks.c
@@ -531,17 +531,10 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
        SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
        SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
-       SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
        SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
        SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
        SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
        SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
-       SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
-                     ALC269_DMIC),
-       SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
-                     ALC269_DMIC),
-       SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
-       SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
        SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
        SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
        SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60fe3527abafd690c6825c19c57956c8265..9a1aa09f47fe1b9191cfba2f924c96ece962fc5c 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4484,6 +4484,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
        spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
 }
 
+static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
+                                    const struct alc_fixup *fix, int action)
+{
+       int coef;
+
+       if (action != ALC_FIXUP_ACT_INIT)
+               return;
+       /* The digital-mic unit sends PDM (differential signal) instead of
+        * the standard PCM, thus you can't record a valid mono stream as is.
+        * Below is a workaround specific to ALC269 to control the dmic
+        * signal source as mono.
+        */
+       coef = alc_read_coef_idx(codec, 0x07);
+       alc_write_coef_idx(codec, 0x07, coef | 0x80);
+}
+
 enum {
        ALC269_FIXUP_SONY_VAIO,
        ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4494,6 +4510,7 @@ enum {
        ALC275_FIXUP_SONY_HWEQ,
        ALC271_FIXUP_DMIC,
        ALC269_FIXUP_PCM_44K,
+       ALC269_FIXUP_STEREO_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -4556,10 +4573,19 @@ static const struct alc_fixup alc269_fixups[] = {
                .type = ALC_FIXUP_FUNC,
                .v.func = alc269_fixup_pcm_44k,
        },
+       [ALC269_FIXUP_STEREO_DMIC] = {
+               .type = ALC_FIXUP_FUNC,
+               .v.func = alc269_fixup_stereo_dmic,
+       },
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+       SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+       SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
+       SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
+       SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+       SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
        SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
        SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
        SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b59c006227f0f9be4a6349abb8ac4dff6d5..5145b663ef6ee831765b45ace9fe8f482e6a5fe9 100644 (file)
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
        return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 static int stac_vrefout_set(struct hda_codec *codec,
                                        hda_nid_t nid, unsigned int new_vref)
 {
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
 
        return 1;
 }
+#endif
 
 static unsigned int stac92xx_vref_set(struct hda_codec *codec,
                                        hda_nid_t nid, unsigned int new_vref)

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798bf33aea5f0e3f864930a67ce2b860b56f..4ebfbd874c9a81db01fe3961c5d1c65656735ab5 100644 (file)
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
        struct via_spec *spec = codec->spec;
        struct nid_path *path;
        bool check_dac;
-       hda_nid_t pin, dac;
+       hda_nid_t pin, dac = 0;
        int err;
 
        pin = spec->autocfg.speaker_pins[0];

diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67ced905f4815c3b021f15c75d3e7cd1bb92..493e3946756f9e30cecbb700d96cc48bea3df94e 100644 (file)
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1339,6 +1339,10 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
                break;
        case MADIface:
                freq_const = 131072000000000ULL;
+               break;
+       default:
+               snd_BUG();
+               return 0;
        }
 
        return div_u64(freq_const, period);
@@ -1356,16 +1360,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
 
        switch (hdspm->io_type) {
        case MADIface:
-         n = 131072000000000ULL;  /* 125 MHz */
-         break;
+               n = 131072000000000ULL;  /* 125 MHz */
+               break;
        case MADI:
        case AES32:
-         n = 110069313433624ULL;  /* 105 MHz */
-         break;
+               n = 110069313433624ULL;  /* 105 MHz */
+               break;
        case RayDAT:
        case AIO:
-         n = 104857600000000ULL;  /* 100 MHz */
-         break;
+               n = 104857600000000ULL;  /* 100 MHz */
+               break;
+       default:
+               snd_BUG();
+               return;
        }
 
        n = div_u64(n, rate);

diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index d0d493ca28ae2ec5693af9899d0199a0d4433a38..aa52b3e13bb5348c81f6e0b74cede4cd90960455 100644 (file)
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -614,6 +614,7 @@ static void read_completed(struct urb *urb)
        struct snd_usb_caiaqdev *dev;
        struct urb *out;
        int frame, len, send_it = 0, outframe = 0;
+       size_t offset = 0;
 
        if (urb->status || !info)
                return;
@@ -634,7 +635,8 @@ static void read_completed(struct urb *urb)
                len = urb->iso_frame_desc[outframe].actual_length;
                out->iso_frame_desc[outframe].length = len;
                out->iso_frame_desc[outframe].actual_length = 0;
-               out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+               out->iso_frame_desc[outframe].offset = offset;
+               offset += len;
 
                if (len > 0) {
                        spin_lock(&dev->spinlock);
@@ -650,7 +652,7 @@ static void read_completed(struct urb *urb)
        }
 
        if (send_it) {
-               out->number_of_packets = FRAMES_PER_URB;
+               out->number_of_packets = outframe;
                out->transfer_flags = URB_ISO_ASAP;
                usb_submit_urb(out, GFP_ATOMIC);
        }

diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 4432ef7a70a9ad184066d4d0b63ff594c442acdf..a213813487bd77b02211597a8448a9dde44d32ef 100644 (file)
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] =  { KEY_C, KEY_B, KEY_A };
 static unsigned short keycode_rk2[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
                                         KEY_5, KEY_6, KEY_7 };
 static unsigned short keycode_rk3[] =  { KEY_1, KEY_2, KEY_3, KEY_4,
-                                        KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
+                                        KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
 
 static unsigned short keycode_kore[] = {
        KEY_FN_F1,      /* "menu"               */

diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7c0d21ecd821b9661f78a8e59beaa30adb0b0c25..7d46e482375d1dce4247f9805d81b6fc1c6568ba 100644 (file)
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -352,7 +352,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                        continue;
                }
                if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
-                   ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
+                   ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) {
                        snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
                                   dev->devnum, iface_no, altno);
                        continue;

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c22fa76e363ae5699e9885cc548fdc28d25607b3..c04d7c71ac88836c32ee3fd3e533472de573c953 100644 (file)
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1191,6 +1191,11 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 
        if (state->mixer->protocol == UAC_VERSION_1) {
                csize = hdr->bControlSize;
+               if (!csize) {
+                       snd_printdd(KERN_ERR "usbaudio: unit %u: "
+                                   "invalid bControlSize == 0\n", unitid);
+                       return -EINVAL;
+               }
                channels = (hdr->bLength - 7) / csize - 1;
                bmaControls = hdr->bmaControls;
        } else {
@@ -1934,15 +1939,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
        struct mixer_build state;
        int err;
        const struct usbmix_ctl_map *map;
-       struct usb_host_interface *hostif;
        void *p;
 
-       hostif = mixer->chip->ctrl_intf;
        memset(&state, 0, sizeof(state));
        state.chip = mixer->chip;
        state.mixer = mixer;
-       state.buffer = hostif->extra;
-       state.buflen = hostif->extralen;
+       state.buffer = mixer->hostif->extra;
+       state.buflen = mixer->hostif->extralen;
 
        /* check the mapping table */
        for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1955,7 +1958,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
        }
 
        p = NULL;
-       while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
+       while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
+                                           p, UAC_OUTPUT_TERMINAL)) != NULL) {
                if (mixer->protocol == UAC_VERSION_1) {
                        struct uac1_output_terminal_descriptor *desc = p;
 
@@ -2162,17 +2166,15 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
 /* create the handler for the optional status interrupt endpoint */
 static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
 {
-       struct usb_host_interface *hostif;
        struct usb_endpoint_descriptor *ep;
        void *transfer_buffer;
        int buffer_length;
        unsigned int epnum;
 
-       hostif = mixer->chip->ctrl_intf;
        /* we need one interrupt input endpoint */
-       if (get_iface_desc(hostif)->bNumEndpoints < 1)
+       if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
                return 0;
-       ep = get_endpoint(hostif, 0);
+       ep = get_endpoint(mixer->hostif, 0);
        if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
                return 0;
 
@@ -2202,7 +2204,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
        };
        struct usb_mixer_interface *mixer;
        struct snd_info_entry *entry;
-       struct usb_host_interface *host_iface;
        int err;
 
        strcpy(chip->card->mixername, "USB Mixer");
@@ -2219,8 +2220,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
                return -ENOMEM;
        }
 
-       host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
-       switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+       mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+       switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
        case UAC_VERSION_1:
        default:
                mixer->protocol = UAC_VERSION_1;

diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ae1a14dcfe82cf51753da7265bcdaf6ce95969f4..81b2d8a32fb0f82e33475fabf6aa890ca7a675f4 100644 (file)
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -3,6 +3,7 @@
 
 struct usb_mixer_interface {
        struct snd_usb_audio *chip;
+       struct usb_host_interface *hostif;
        struct list_head list;
        unsigned int ignore_ctl_error;
        struct urb *urb;

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index dba0b7f11c546fa2a9ac71b4ccfaff4ae607a564..a42e3ef3832d709d11b1653bc8385d20c5a8271f 100644 (file)
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"),
                }
        }
 },
+{
+       USB_DEVICE(0x0582, 0x0130),
+       .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+               /* .vendor_name = "BOSS", */
+               /* .product_name = "MICRO BR-80", */
+               .ifnum = QUIRK_ANY_INTERFACE,
+               .type = QUIRK_COMPOSITE,
+               .data = (const struct snd_usb_audio_quirk[]) {
+                       {
+                               .ifnum = 0,
+                               .type = QUIRK_IGNORE_INTERFACE
+                       },
+                       {
+                               .ifnum = 1,
+                               .type = QUIRK_AUDIO_STANDARD_INTERFACE
+                       },
+                       {
+                               .ifnum = 2,
+                               .type = QUIRK_AUDIO_STANDARD_INTERFACE
+                       },
+                       {
+                               .ifnum = 3,
+                               .type = QUIRK_MIDI_FIXED_ENDPOINT,
+                               .data = & (const struct snd_usb_midi_endpoint_info) {
+                                       .out_cables = 0x0001,
+                                       .in_cables  = 0x0001
+                               }
+                       },
+                       {
+                               .ifnum = -1
+                       }
+               }
+       }
+},
 
 /* Guillemot devices */
 {
@@ -2417,6 +2451,12 @@ YAMAHA_DEVICE(0x7010, "UB99"),
        .idProduct = 0x1020,
 },
 
+/* KeithMcMillen Stringport */
+{
+       USB_DEVICE(0x1f38, 0x0001),
+       .bInterfaceClass = USB_CLASS_AUDIO,
+},
+
 /* Miditech devices */
 {
        USB_DEVICE(0x4752, 0x0011),

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 77762c99afbee9286365944029727001b444b8d3..81e07d842581771ac82d4a3107f2a54cd69663e3 100644 (file)
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -426,7 +426,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
  */
 static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
 {
-       int err, reg;
+       int err  = 0, reg;
        int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
 
        for (reg = 0; reg < ARRAY_SIZE(val); reg++) {