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-rw-r--r--sound/core/jack.c2
-rw-r--r--sound/core/oss/mixer_oss.c3
-rw-r--r--sound/core/oss/pcm_oss.c4
-rw-r--r--sound/core/oss/rate.c2
-rw-r--r--sound/core/sgbuf.c7
-rw-r--r--sound/isa/opl3sa2.c18
-rw-r--r--sound/oss/dmasound/dmasound_atari.c16
-rw-r--r--sound/pci/aw2/aw2-alsa.c2
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c1
-rw-r--r--sound/pci/hda/hda_hwdep.c15
-rw-r--r--sound/pci/hda/hda_intel.c49
-rw-r--r--sound/pci/hda/patch_realtek.c4
-rw-r--r--sound/pci/hda/patch_sigmatel.c8
-rw-r--r--sound/pci/mixart/mixart.c1
-rw-r--r--sound/pci/oxygen/virtuoso.c17
-rw-r--r--sound/pci/pcxhr/pcxhr.h12
-rw-r--r--sound/usb/usbaudio.c20
-rw-r--r--sound/usb/usbmidi.c1
18 files changed, 103 insertions, 79 deletions
diff --git a/sound/core/jack.c b/sound/core/jack.c
index dd4a12dc09aa..077a85262c1c 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -47,7 +47,7 @@ static int snd_jack_dev_register(struct snd_device *device)
int err;
snprintf(jack->name, sizeof(jack->name), "%s %s",
- card->longname, jack->id);
+ card->shortname, jack->id);
jack->input_dev->name = jack->name;
/* Default to the sound card device. */
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 4690b8b5681f..e570649184e2 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -692,6 +692,9 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer,
snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME)
snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) {
+ snd_mixer_oss_put_volume1_vol(fmixer, pslot,
+ slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) {
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 0a1798eafb0b..699d2890535c 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -2872,7 +2872,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
setup = kmalloc(sizeof(*setup), GFP_KERNEL);
if (! setup) {
buffer->error = -ENOMEM;
- mutex_lock(&pstr->oss.setup_mutex);
+ mutex_unlock(&pstr->oss.setup_mutex);
return;
}
if (pstr->oss.setup_list == NULL)
@@ -2886,7 +2886,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
if (! template.task_name) {
kfree(setup);
buffer->error = -ENOMEM;
- mutex_lock(&pstr->oss.setup_mutex);
+ mutex_unlock(&pstr->oss.setup_mutex);
return;
}
}
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index a466443c4a26..2fa9299a440d 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin,
while (dst_frames1 > 0) {
S1 = S2;
if (src_frames1-- > 0) {
- S1 = *src;
+ S2 = *src;
src += src_step;
}
if (pos & ~R_MASK) {
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index d4564edd61d7..4e7ec2b49873 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -38,6 +38,10 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
if (! sgbuf)
return -EINVAL;
+ if (dmab->area)
+ vunmap(dmab->area);
+ dmab->area = NULL;
+
tmpb.dev.type = SNDRV_DMA_TYPE_DEV;
tmpb.dev.dev = sgbuf->dev;
for (i = 0; i < sgbuf->pages; i++) {
@@ -48,9 +52,6 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT;
snd_dma_free_pages(&tmpb);
}
- if (dmab->area)
- vunmap(dmab->area);
- dmab->area = NULL;
kfree(sgbuf->table);
kfree(sgbuf->page_table);
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 58c972b2af03..b848d1001864 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -550,21 +550,27 @@ static int __devinit snd_opl3sa2_mixer(struct snd_card *card)
#ifdef CONFIG_PM
static int snd_opl3sa2_suspend(struct snd_card *card, pm_message_t state)
{
- struct snd_opl3sa2 *chip = card->private_data;
+ if (card) {
+ struct snd_opl3sa2 *chip = card->private_data;
- snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
- chip->wss->suspend(chip->wss);
- /* power down */
- snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ chip->wss->suspend(chip->wss);
+ /* power down */
+ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3);
+ }
return 0;
}
static int snd_opl3sa2_resume(struct snd_card *card)
{
- struct snd_opl3sa2 *chip = card->private_data;
+ struct snd_opl3sa2 *chip;
int i;
+ if (!card)
+ return 0;
+
+ chip = card->private_data;
/* power up */
snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0);
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 57d9f154c88b..38931f2f6967 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -847,23 +847,23 @@ static int __init AtaIrqInit(void)
of events. So all we need to keep the music playing is
to provide the sound hardware with new data upon
an interrupt from timer A. */
- mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */
- mfp.tim_dt_a = 1; /* Cause interrupt after first event. */
- mfp.tim_ct_a = 8; /* Turn on event counting. */
+ st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */
+ st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */
+ st_mfp.tim_ct_a = 8; /* Turn on event counting. */
/* Register interrupt handler. */
if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound",
AtaInterrupt))
return 0;
- mfp.int_en_a |= 0x20; /* Turn interrupt on. */
- mfp.int_mk_a |= 0x20;
+ st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */
+ st_mfp.int_mk_a |= 0x20;
return 1;
}
#ifdef MODULE
static void AtaIrqCleanUp(void)
{
- mfp.tim_ct_a = 0; /* stop timer */
- mfp.int_en_a &= ~0x20; /* turn interrupt off */
+ st_mfp.tim_ct_a = 0; /* stop timer */
+ st_mfp.int_en_a &= ~0x20; /* turn interrupt off */
free_irq(IRQ_MFP_TIMA, AtaInterrupt);
}
#endif /* MODULE */
@@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void)
is_falcon = 0;
} else
return -ENODEV;
- if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0)
+ if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0)
return dmasound_init();
else {
printk("DMA sound driver: Timer A interrupt already in use\n");
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3f00ddf450f8..c7c54e7748e9 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
static struct pci_device_id snd_aw2_ids[] = {
- {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID,
+ {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
0, 0, 0},
{0}
};
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7958006a1d66..101a1c13a20d 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0151_chip = 1,
.spk71 = 1,
.spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
.driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]",
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 482fb0304ca9..4ae51dcb81af 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -277,18 +277,19 @@ static ssize_t init_verbs_store(struct device *dev,
{
struct snd_hwdep *hwdep = dev_get_drvdata(dev);
struct hda_codec *codec = hwdep->private_data;
- char *p;
- struct hda_verb verb, *v;
+ struct hda_verb *v;
+ int nid, verb, param;
- verb.nid = simple_strtoul(buf, &p, 0);
- verb.verb = simple_strtoul(p, &p, 0);
- verb.param = simple_strtoul(p, &p, 0);
- if (!verb.nid || !verb.verb || !verb.param)
+ if (sscanf(buf, "%i %i %i", &nid, &verb, &param) != 3)
+ return -EINVAL;
+ if (!nid || !verb)
return -EINVAL;
v = snd_array_new(&codec->init_verbs);
if (!v)
return -ENOMEM;
- *v = verb;
+ v->nid = nid;
+ v->verb = verb;
+ v->param = param;
return count;
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 11e791b965f6..f3b5723c2859 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1947,16 +1947,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
return 0;
}
-static int azx_resume_early(struct pci_dev *pci)
-{
- return pci_restore_state(pci);
-}
-
static int azx_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct azx *chip = card->private_data;
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
if (pci_enable_device(pci) < 0) {
printk(KERN_ERR "hda-intel: pci_enable_device failed, "
"disabling device\n");
@@ -2062,26 +2059,31 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
{
const struct snd_pci_quirk *q;
- /* Check VIA HD Audio Controller exist */
- if (chip->pci->vendor == PCI_VENDOR_ID_VIA &&
- chip->pci->device == VIA_HDAC_DEVICE_ID) {
+ switch (fix) {
+ case POS_FIX_LPIB:
+ case POS_FIX_POSBUF:
+ return fix;
+ }
+
+ /* Check VIA/ATI HD Audio Controller exist */
+ switch (chip->driver_type) {
+ case AZX_DRIVER_VIA:
+ case AZX_DRIVER_ATI:
chip->via_dmapos_patch = 1;
/* Use link position directly, avoid any transfer problem. */
return POS_FIX_LPIB;
}
chip->via_dmapos_patch = 0;
- if (fix == POS_FIX_AUTO) {
- q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
- if (q) {
- printk(KERN_INFO
- "hda_intel: position_fix set to %d "
- "for device %04x:%04x\n",
- q->value, q->subvendor, q->subdevice);
- return q->value;
- }
+ q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
+ if (q) {
+ printk(KERN_INFO
+ "hda_intel: position_fix set to %d "
+ "for device %04x:%04x\n",
+ q->value, q->subvendor, q->subdevice);
+ return q->value;
}
- return fix;
+ return POS_FIX_AUTO;
}
/*
@@ -2098,6 +2100,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
+ /* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */
+ SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03),
{}
};
@@ -2211,9 +2215,17 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
gcap = azx_readw(chip, GCAP);
snd_printdd("chipset global capabilities = 0x%x\n", gcap);
+ /* ATI chips seems buggy about 64bit DMA addresses */
+ if (chip->driver_type == AZX_DRIVER_ATI)
+ gcap &= ~0x01;
+
/* allow 64bit DMA address if supported by H/W */
if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK))
pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK);
+ else {
+ pci_set_dma_mask(pci, DMA_32BIT_MASK);
+ pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK);
+ }
/* read number of streams from GCAP register instead of using
* hardcoded value
@@ -2468,7 +2480,6 @@ static struct pci_driver driver = {
.remove = __devexit_p(azx_remove),
#ifdef CONFIG_PM
.suspend = azx_suspend,
- .resume_early = azx_resume_early,
.resume = azx_resume,
#endif
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ed8fcbd60003..6c26afcb8262 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7017,6 +7017,7 @@ static int patch_alc882(struct hda_codec *codec)
case 0x106b3e00: /* iMac 24 Aluminium */
board_config = ALC885_IMAC24;
break;
+ case 0x106b00a0: /* MacBookPro3,1 - Another revision */
case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
case 0x106b00a4: /* MacbookPro4,1 */
case 0x106b2c00: /* Macbook Pro rev3 */
@@ -8469,6 +8470,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
ALC888_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
ALC888_ACER_ASPIRE_4930G),
+ SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
+ ALC888_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
@@ -10554,6 +10557,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8027edf3c8f2..6094344fb223 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1207,7 +1207,7 @@ static const char *slave_vols[] = {
"LFE Playback Volume",
"Side Playback Volume",
"Headphone Playback Volume",
- "Headphone Playback Volume",
+ "Headphone2 Playback Volume",
"Speaker Playback Volume",
"External Speaker Playback Volume",
"Speaker2 Playback Volume",
@@ -1221,7 +1221,7 @@ static const char *slave_sws[] = {
"LFE Playback Switch",
"Side Playback Switch",
"Headphone Playback Switch",
- "Headphone Playback Switch",
+ "Headphone2 Playback Switch",
"Speaker Playback Switch",
"External Speaker Playback Switch",
"Speaker2 Playback Switch",
@@ -3516,6 +3516,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
if (! spec->autocfg.line_outs)
return 0; /* can't find valid pin config */
+#if 0 /* FIXME: temporarily disabled */
/* If we have no real line-out pin and multiple hp-outs, HPs should
* be set up as multi-channel outputs.
*/
@@ -3535,6 +3536,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
spec->autocfg.hp_outs = 0;
}
+#endif /* FIXME: temporarily disabled */
if (spec->autocfg.mono_out_pin) {
int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
@@ -4989,7 +4991,7 @@ again:
case STAC_DELL_M4_3:
spec->num_dmics = 1;
spec->num_smuxes = 0;
- spec->num_dmuxes = 0;
+ spec->num_dmuxes = 1;
break;
default:
spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index f23a73577c22..bb162507fe6c 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -607,6 +607,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs,
/* set the format to the board */
err = mixart_set_format(stream, format);
if(err < 0) {
+ mutex_unlock(&mgr->setup_mutex);
return err;
}
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 18c7c91786bc..6c870c12a177 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -26,7 +26,7 @@
* SPI 0 -> 1st PCM1796 (front)
* SPI 1 -> 2nd PCM1796 (surround)
* SPI 2 -> 3rd PCM1796 (center/LFE)
- * SPI 4 -> 4th PCM1796 (back) and EEPROM self-destruct (do not use!)
+ * SPI 4 -> 4th PCM1796 (back)
*
* GPIO 2 -> M0 of CS5381
* GPIO 3 -> M1 of CS5381
@@ -207,12 +207,6 @@ static void xonar_gpio_changed(struct oxygen *chip);
static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
u8 reg, u8 value)
{
- /*
- * We don't want to do writes on SPI 4 because the EEPROM, which shares
- * the same pin, might get confused and broken. We'd better take care
- * that the driver works with the default register values ...
- */
-#if 0
/* maps ALSA channel pair number to SPI output */
static const u8 codec_map[4] = {
0, 1, 2, 4
@@ -223,7 +217,6 @@ static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
(codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) |
OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
(reg << 8) | value);
-#endif
}
static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec,
@@ -757,9 +750,6 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
{
- if (!strncmp(template->name, "Master Playback ", 16))
- /* disable volume/mute because they would require SPI writes */
- return 1;
if (!strncmp(template->name, "CD Capture ", 11))
/* CD in is actually connected to the video in pin */
template->private_value ^= AC97_CD ^ AC97_VIDEO;
@@ -850,8 +840,9 @@ static const struct oxygen_model model_xonar_d2 = {
.dac_volume_min = 0x0f,
.dac_volume_max = 0xff,
.misc_flags = OXYGEN_MISC_MIDI,
- .function_flags = OXYGEN_FUNCTION_SPI,
- .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S,
+ .function_flags = OXYGEN_FUNCTION_SPI |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
};
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 84131a916c92..69d87dee6995 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -97,12 +97,12 @@ struct pcxhr_mgr {
int capture_chips;
int fw_file_set;
int firmware_num;
- int is_hr_stereo:1;
- int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */
- int board_has_analog:1; /* if 0 the board is digital only */
- int board_has_mic:1; /* if 1 the board has microphone input */
- int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
- int mono_capture:1; /* if 1 the board does mono capture */
+ unsigned int is_hr_stereo:1;
+ unsigned int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */
+ unsigned int board_has_analog:1; /* if 0 the board is digital only */
+ unsigned int board_has_mic:1; /* if 1 the board has microphone input */
+ unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
+ unsigned int mono_capture:1; /* if 1 the board does mono capture */
struct snd_dma_buffer hostport;
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 2ab83129d9b0..19e37451c216 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
* build the rate table and bitmap flags
*/
int r, idx;
- unsigned int nonzero_rates = 0;
fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
if (fp->rate_table == NULL) {
@@ -2532,24 +2531,27 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
return -1;
}
- fp->nr_rates = nr_rates;
- fp->rate_min = fp->rate_max = combine_triple(&fmt[8]);
+ fp->nr_rates = 0;
+ fp->rate_min = fp->rate_max = 0;
for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) {
unsigned int rate = combine_triple(&fmt[idx]);
+ if (!rate)
+ continue;
/* C-Media CM6501 mislabels its 96 kHz altsetting */
if (rate == 48000 && nr_rates == 1 &&
- chip->usb_id == USB_ID(0x0d8c, 0x0201) &&
+ (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
+ chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
fp->altsetting == 5 && fp->maxpacksize == 392)
rate = 96000;
- fp->rate_table[r] = rate;
- nonzero_rates |= rate;
- if (rate < fp->rate_min)
+ fp->rate_table[fp->nr_rates] = rate;
+ if (!fp->rate_min || rate < fp->rate_min)
fp->rate_min = rate;
- else if (rate > fp->rate_max)
+ if (!fp->rate_max || rate > fp->rate_max)
fp->rate_max = rate;
fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+ fp->nr_rates++;
}
- if (!nonzero_rates) {
+ if (!fp->nr_rates) {
hwc_debug("All rates were zero. Skipping format!\n");
return -1;
}
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 320641ab5be7..26bad373fe65 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
}
ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.out_interval = 0;
ep_info.out_cables = endpoint->out_cables & 0x5555;
err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
if (err < 0)