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-rw-r--r--sound/core/pcm_native.c8
-rw-r--r--sound/core/sound.c4
-rw-r--r--sound/core/sound_oss.c2
-rw-r--r--sound/oss/coproc.h2
-rw-r--r--sound/oss/v_midi.h5
-rw-r--r--sound/pci/atiixp.c1
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c10
-rw-r--r--sound/pci/echoaudio/echoaudio.c4
-rw-r--r--sound/pci/hda/hda_intel.c25
-rw-r--r--sound/pci/hda/patch_realtek.c127
-rw-r--r--sound/pci/hda/patch_sigmatel.c61
-rw-r--r--sound/pci/riptide/riptide.c2
-rw-r--r--sound/soc/codecs/tlv320aic23.c2
-rw-r--r--sound/soc/codecs/wm8350.c2
-rw-r--r--sound/soc/codecs/wm8903.c3
-rw-r--r--sound/soc/omap/Makefile2
-rw-r--r--sound/soc/omap/omap3pandora.c1
-rw-r--r--sound/sound_core.c2
-rw-r--r--sound/usb/caiaq/midi.h2
-rw-r--r--sound/usb/usbaudio.c553
-rw-r--r--sound/usb/usbaudio.h90
-rw-r--r--sound/usb/usbmidi.c10
-rw-r--r--sound/usb/usbmixer.c112
-rw-r--r--sound/usb/usbquirks.h77
-rw-r--r--sound/usb/usx2y/us122l.c6
25 files changed, 747 insertions, 366 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 17935746eb1..87288762403 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -316,10 +316,10 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
if (!params->info)
params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES;
if (!params->fifo_size) {
- if (snd_mask_min(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT]) ==
- snd_mask_max(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT]) &&
- snd_mask_min(&params->masks[SNDRV_PCM_HW_PARAM_CHANNELS]) ==
- snd_mask_max(&params->masks[SNDRV_PCM_HW_PARAM_CHANNELS])) {
+ m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ if (snd_mask_min(m) == snd_mask_max(m) &&
+ snd_interval_min(i) == snd_interval_max(i)) {
changed = substream->ops->ioctl(substream,
SNDRV_PCM_IOCTL1_FIFO_SIZE, params);
if (changed < 0)
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 7872a02f6ca..563d1967a0a 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void)
unregister_chrdev(major, "alsa");
}
-module_init(alsa_sound_init)
-module_exit(alsa_sound_exit)
+subsys_initcall(alsa_sound_init);
+module_exit(alsa_sound_exit);
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 7fe12264ff8..0c164e5e432 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
default:
return -EINVAL;
}
- if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+ if (minor < 0 || minor >= SNDRV_OSS_MINORS)
return -EINVAL;
return minor;
}
diff --git a/sound/oss/coproc.h b/sound/oss/coproc.h
index 7306346e9ac..7bec21bbdd8 100644
--- a/sound/oss/coproc.h
+++ b/sound/oss/coproc.h
@@ -4,7 +4,7 @@
*/
/*
- * Coprocessor access types
+ * Coprocessor access types
*/
#define COPR_CUSTOM 0x0001 /* Custom applications */
#define COPR_MIDI 0x0002 /* MIDI (MPU-401) emulation */
diff --git a/sound/oss/v_midi.h b/sound/oss/v_midi.h
index 1b86cb45c60..08e2185ee81 100644
--- a/sound/oss/v_midi.h
+++ b/sound/oss/v_midi.h
@@ -2,9 +2,9 @@ typedef struct vmidi_devc {
int dev;
/* State variables */
- int opened;
+ int opened;
spinlock_t lock;
-
+
/* MIDI fields */
int my_mididev;
int pair_mididev;
@@ -12,4 +12,3 @@ typedef struct vmidi_devc {
int intr_active;
void (*midi_input_intr) (int dev, unsigned char data);
} vmidi_devc;
-
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 81e2bfc1125..49d572a7b23 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -297,6 +297,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = {
MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
static struct snd_pci_quirk atiixp_quirks[] __devinitdata = {
+ SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0),
SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0),
{ } /* terminator */
};
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 56fcf00c0e2..3f99a5e8528 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -2238,11 +2238,11 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97)
/* set the desired CODEC mode */
if (ac97->num == CS46XX_PRIMARY_CODEC_INDEX) {
- snd_printdd("cs46xx: CODOEC1 mode %04x\n",0x0);
- snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x0);
+ snd_printdd("cs46xx: CODEC1 mode %04x\n", 0x0);
+ snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x0);
} else if (ac97->num == CS46XX_SECONDARY_CODEC_INDEX) {
- snd_printdd("cs46xx: CODOEC2 mode %04x\n",0x3);
- snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x3);
+ snd_printdd("cs46xx: CODEC2 mode %04x\n", 0x3);
+ snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x3);
} else {
snd_BUG(); /* should never happen ... */
}
@@ -2266,7 +2266,7 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97)
return;
/* test if we can write to the record gain volume register */
- snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x8a05);
+ snd_ac97_write(ac97, AC97_REC_GAIN, 0x8a05);
if ((err = snd_ac97_read(ac97, AC97_REC_GAIN)) == 0x8a05)
return;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 2783ce6c236..8dab82d7d19 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1864,7 +1864,9 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
/* The hardware doesn't tell us which substream caused the irq,
thus we have to check all running substreams. */
for (ss = 0; ss < DSP_MAXPIPES; ss++) {
- if ((substream = chip->substream[ss])) {
+ substream = chip->substream[ss];
+ if (substream && ((struct audiopipe *)substream->runtime->
+ private_data)->state == PIPE_STATE_STARTED) {
period = pcm_pointer(substream) /
substream->runtime->period_size;
if (period != chip->last_period[ss]) {
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ac05bef7c2e..8d8e0b5aa24 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -426,6 +426,7 @@ struct azx {
/* flags */
int position_fix;
+ int poll_count;
unsigned int running :1;
unsigned int initialized :1;
unsigned int single_cmd :1;
@@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = {
#define get_azx_dev(substream) (substream->runtime->private_data)
static int azx_acquire_irq(struct azx *chip, int do_disconnect);
-
+static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
/*
* Interface for HD codec
*/
@@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
{
struct azx *chip = bus->private_data;
unsigned long timeout;
+ int do_poll = 0;
again:
timeout = jiffies + msecs_to_jiffies(1000);
for (;;) {
- if (chip->polling_mode) {
+ if (chip->polling_mode || do_poll) {
spin_lock_irq(&chip->reg_lock);
azx_update_rirb(chip);
spin_unlock_irq(&chip->reg_lock);
@@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
if (!chip->rirb.cmds[addr]) {
smp_rmb();
bus->rirb_error = 0;
+
+ if (!do_poll)
+ chip->poll_count = 0;
return chip->rirb.res[addr]; /* the last value */
}
if (time_after(jiffies, timeout))
@@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
}
}
+ if (!chip->polling_mode && chip->poll_count < 2) {
+ snd_printdd(SFX "azx_get_response timeout, "
+ "polling the codec once: last cmd=0x%08x\n",
+ chip->last_cmd[addr]);
+ do_poll = 1;
+ chip->poll_count++;
+ goto again;
+ }
+
+
if (!chip->polling_mode) {
snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
"switching to polling mode: last cmd=0x%08x\n",
@@ -1878,6 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
if (!bdl_pos_adj[chip->dev_index])
return 1; /* no delayed ack */
+ if (WARN_ONCE(!azx_dev->period_bytes,
+ "hda-intel: zero azx_dev->period_bytes"))
+ return 0; /* this shouldn't happen! */
if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
return 0; /* NG - it's below the period boundary */
return 1; /* OK, it's fine */
@@ -2043,7 +2061,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
{
if (request_irq(chip->pci->irq, azx_interrupt,
chip->msi ? 0 : IRQF_SHARED,
- "HDA Intel", chip)) {
+ "hda_intel", chip)) {
printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
"disabling device\n", chip->pci->irq);
if (do_disconnect)
@@ -2332,6 +2350,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
*/
static struct snd_pci_quirk msi_black_list[] __devinitdata = {
SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
+ SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 141ff446104..f628c33d80b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1097,6 +1097,16 @@ static void alc889_coef_init(struct hda_codec *codec)
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
}
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+ if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+ return;
+ if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ on ? 2 : 0);
+}
+
static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
unsigned int tmp;
@@ -1114,25 +1124,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case ALC_INIT_DEFAULT:
switch (codec->vendor_id) {
case 0x10ec0260:
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
- snd_hda_codec_write(codec, 0x10, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
+ set_eapd(codec, 0x0f, 1);
+ set_eapd(codec, 0x10, 1);
break;
case 0x10ec0262:
case 0x10ec0267:
case 0x10ec0268:
case 0x10ec0269:
+ case 0x10ec0270:
case 0x10ec0272:
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
case 0x10ec0862:
case 0x10ec0889:
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
- snd_hda_codec_write(codec, 0x15, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
+ set_eapd(codec, 0x14, 1);
+ set_eapd(codec, 0x15, 1);
break;
}
switch (codec->vendor_id) {
@@ -1234,6 +1241,8 @@ static void alc_init_auto_mic(struct hda_codec *codec)
return; /* invalid entry */
}
}
+ if (!ext || !fixed)
+ return;
if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
return; /* no unsol support */
snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
@@ -4917,6 +4926,49 @@ static void fixup_automic_adc(struct hda_codec *codec)
spec->auto_mic = 0; /* disable auto-mic to be sure */
}
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+ int i;
+
+ /* search for the input pin; there must be only one */
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (spec->autocfg.input_pins[i]) {
+ pin = spec->autocfg.input_pins[i];
+ break;
+ }
+ }
+ if (!pin)
+ return;
+
+ /* set the default connection to that pin */
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ hda_nid_t cap = spec->capsrc_nids ?
+ spec->capsrc_nids[i] : spec->adc_nids[i];
+ int idx;
+
+ idx = get_connection_index(codec, cap, pin);
+ if (idx < 0)
+ continue;
+ /* use only this ADC */
+ if (spec->capsrc_nids)
+ spec->capsrc_nids += i;
+ spec->adc_nids += i;
+ spec->num_adc_nids = 1;
+ /* select or unmute this route */
+ if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+ snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+ HDA_AMP_MUTE, 0);
+ } else {
+ snd_hda_codec_write_cache(codec, cap, 0,
+ AC_VERB_SET_CONNECT_SEL, idx);
+ }
+ return;
+ }
+}
+
static void set_capture_mixer(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -4929,14 +4981,15 @@ static void set_capture_mixer(struct hda_codec *codec)
alc_capture_mixer3 },
};
if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
- int mux;
- if (spec->auto_mic) {
- mux = 0;
+ int mux = 0;
+ if (spec->auto_mic)
fixup_automic_adc(codec);
- } else if (spec->input_mux && spec->input_mux->num_items > 1)
- mux = 1;
- else
- mux = 0;
+ else if (spec->input_mux) {
+ if (spec->input_mux->num_items > 1)
+ mux = 1;
+ else if (spec->input_mux->num_items == 1)
+ fixup_single_adc(codec);
+ }
spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
}
}
@@ -7201,8 +7254,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 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("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7603,6 +7656,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -7787,6 +7841,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
}
+static void alc885_mb5_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+
+}
+
+static void alc885_mb5_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Headphone insertion or removal. */
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc885_mb5_automute(codec);
+}
+
static void alc885_imac91_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -9233,6 +9308,8 @@ static struct alc_config_preset alc882_presets[] = {
.input_mux = &mb5_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc885_mb5_unsol_event,
+ .init_hook = alc885_mb5_automute,
},
[ALC885_MACPRO] = {
.mixers = { alc882_macpro_mixer },
@@ -9510,6 +9587,7 @@ static struct alc_config_preset alc882_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
+ .const_channel_count = 6,
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
@@ -10420,7 +10498,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+ spec->autocfg.speaker_pins[0] = 0x14;
}
static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -11310,7 +11388,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
}
#define alc262_auto_create_input_ctls \
- alc880_auto_create_input_ctls
+ alc882_auto_create_input_ctls
/*
* generic initialization of ADC, input mixers and output mixers
@@ -11849,9 +11927,9 @@ static struct alc_config_preset alc262_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
- .unsol_event = alc_automute_amp_unsol_event,
+ .unsol_event = alc_sku_unsol_event,
.setup = alc262_hp_t5735_setup,
- .init_hook = alc_automute_amp,
+ .init_hook = alc_inithook,
},
[ALC262_HP_RP5700] = {
.mixers = { alc262_hp_rp5700_mixer },
@@ -12605,6 +12683,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
dac = 0x02;
break;
case 0x15:
+ case 0x21:
dac = 0x03;
break;
default:
@@ -14991,6 +15070,8 @@ static int patch_alc861(struct hda_codec *codec)
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
+ if (!spec->cap_mixer)
+ set_capture_mixer(codec);
set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
spec->vmaster_nid = 0x03;
@@ -15633,7 +15714,7 @@ static struct alc_config_preset alc861vd_presets[] = {
static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+ return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0);
}
@@ -17391,7 +17472,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
ALC662_3ST_6ch_DIG),
- SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
+ SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
ALC662_3ST_6ch_DIG),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 117919aa17f..dbffb5b5c69 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4737,6 +4737,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
}
}
+static int hp_blike_system(u32 subsystem_id);
+
+static void set_hp_led_gpio(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ switch (codec->vendor_id) {
+ case 0x111d7608:
+ /* GPIO 0 */
+ spec->gpio_led = 0x01;
+ break;
+ case 0x111d7600:
+ case 0x111d7601:
+ case 0x111d7602:
+ case 0x111d7603:
+ /* GPIO 3 */
+ spec->gpio_led = 0x08;
+ break;
+ }
+}
+
/*
* This method searches for the mute LED GPIO configuration
* provided as OEM string in SMBIOS. The format of that string
@@ -4748,6 +4768,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
*
* So, HP B-series like systems may have HP_Mute_LED_0 (current models)
* or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings
+ *
+ *
+ * The dv-series laptops don't seem to have the HP_Mute_LED* strings in
+ * SMBIOS - at least the ones I have seen do not have them - which include
+ * my own system (HP Pavilion dv6-1110ax) and my cousin's
+ * HP Pavilion dv9500t CTO.
+ * Need more information on whether it is true across the entire series.
+ * -- kunal
*/
static int find_mute_led_gpio(struct hda_codec *codec)
{
@@ -4758,28 +4786,27 @@ static int find_mute_led_gpio(struct hda_codec *codec)
while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
NULL, dev))) {
if (sscanf(dev->name, "HP_Mute_LED_%d_%d",
- &spec->gpio_led_polarity,
- &spec->gpio_led) == 2) {
+ &spec->gpio_led_polarity,
+ &spec->gpio_led) == 2) {
spec->gpio_led = 1 << spec->gpio_led;
return 1;
}
if (sscanf(dev->name, "HP_Mute_LED_%d",
- &spec->gpio_led_polarity) == 1) {
- switch (codec->vendor_id) {
- case 0x111d7608:
- /* GPIO 0 */
- spec->gpio_led = 0x01;
- return 1;
- case 0x111d7600:
- case 0x111d7601:
- case 0x111d7602:
- case 0x111d7603:
- /* GPIO 3 */
- spec->gpio_led = 0x08;
- return 1;
- }
+ &spec->gpio_led_polarity) == 1) {
+ set_hp_led_gpio(codec);
+ return 1;
}
}
+
+ /*
+ * Fallback case - if we don't find the DMI strings,
+ * we statically set the GPIO - if not a B-series system.
+ */
+ if (!hp_blike_system(codec->subsystem_id)) {
+ set_hp_led_gpio(codec);
+ spec->gpio_led_polarity = 1;
+ return 1;
+ }
}
return 0;
}
@@ -5603,6 +5630,8 @@ again:
spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
+ snd_printdd("Found board config: %d\n", spec->board_config);
+
switch (spec->board_config) {
case STAC_HP_M4:
/* enable internal microphone */
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index bb08a2855fc..960a227eb65 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate)
rptr.retwords[2] != M &&
rptr.retwords[3] != N &&
i++ < MAX_WRITE_RETRY);
- if (i == MAX_WRITE_RETRY) {
+ if (i > MAX_WRITE_RETRY) {
snd_printdd("sent samplerate %d: %d failed\n",
*intdec, rate);
return -EIO;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a9dc5fb5477..da589d8664d 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
u16 reg;
/* Sync reg_cache with the hardware */
- for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
+ for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ebbf11b653a..718ef912e75 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
iface |= 0x3 << 8;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x3 << 8; /* lg not sure which mode */
+ iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce5515e3f2b..3595bd57c4e 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
struct i2c_client *i2c = codec->control_data;
int i;
u16 *reg_cache = codec->reg_cache;
- u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+ u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
GFP_KERNEL);
/* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
if (tmp_cache[i] != reg_cache[i])
snd_soc_write(codec, i, tmp_cache[i]);
+ kfree(tmp_cache);
} else {
dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
}
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3db8a6c523f..19283e5edfb 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 71b2c161158..68980c19a3b 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+ {"PCM DAC", NULL, "APLL Enable"},
{"Headphone Amplifier", NULL, "PCM DAC"},
{"Line Out", NULL, "PCM DAC"},
{"Headphone Jack", NULL, "Headphone Amplifier"},
diff --git a/sound/sound_core.c b/sound/sound_core.c
index dbca7c909a3..7c2d677a2df 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void)
class_destroy(sound_class);
}
-module_init(init_soundcore);
+subsys_initcall(init_soundcore);
module_exit(cleanup_soundcore);
diff --git a/sound/usb/caiaq/midi.h b/sound/usb/caiaq/midi.h
index 9d16db027fc..380f984babc 100644
--- a/sound/usb/caiaq/midi.h
+++ b/sound/usb/caiaq/midi.h
@@ -3,6 +3,6 @@
int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *dev);
void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len);
-void snd_usb_caiaq_midi_output_done(struct urb* urb);
+void snd_usb_caiaq_midi_output_done(struct urb *urb);
#endif /* CAIAQ_MIDI_H */
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 7ad8089b233..20b656e9f90 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -46,6 +46,9 @@
#include <linux/usb.h>
#include <linux/moduleparam.h>
#include <linux/mutex.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/ch9.h>
+
#include <sound/core.h>
#include <sound/info.h>
#include <sound/pcm.h>
@@ -596,7 +599,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs,
if (subs->transfer_done >= runtime->period_size) {
subs->transfer_done -= runtime->period_size;
period_elapsed = 1;
- if (subs->fmt_type == USB_FORMAT_TYPE_II) {
+ if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
if (subs->transfer_done > 0) {
/* FIXME: fill-max mode is not
* supported yet */
@@ -1104,7 +1107,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
u->packets = (i + 1) * total_packs / subs->nurbs
- i * total_packs / subs->nurbs;
u->buffer_size = maxsize * u->packets;
- if (subs->fmt_type == USB_FORMAT_TYPE_II)
+ if (subs->fmt_type == UAC_FORMAT_TYPE_II)
u->packets++; /* for transfer delimiter */
u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
if (!u->urb)
@@ -1180,7 +1183,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned
if (i >= fp->nr_rates)
continue;
}
- attr = fp->ep_attr & EP_ATTR_MASK;
+ attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE;
if (! found) {
found = fp;
cur_attr = attr;
@@ -1192,14 +1195,14 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned
* M-audio audiophile USB.
*/
if (attr != cur_attr) {
- if ((attr == EP_ATTR_ASYNC &&
+ if ((attr == USB_ENDPOINT_SYNC_ASYNC &&
subs->direction == SNDRV_PCM_STREAM_PLAYBACK) ||
- (attr == EP_ATTR_ADAPTIVE &&
+ (attr == USB_ENDPOINT_SYNC_ADAPTIVE &&
subs->direction == SNDRV_PCM_STREAM_CAPTURE))
continue;
- if ((cur_attr == EP_ATTR_ASYNC &&
+ if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC &&
subs->direction == SNDRV_PCM_STREAM_PLAYBACK) ||
- (cur_attr == EP_ATTR_ADAPTIVE &&
+ (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE &&
subs->direction == SNDRV_PCM_STREAM_CAPTURE)) {
found = fp;
cur_attr = attr;
@@ -1229,11 +1232,11 @@ static int init_usb_pitch(struct usb_device *dev, int iface,
ep = get_endpoint(alts, 0)->bEndpointAddress;
/* if endpoint has pitch control, enable it */
- if (fmt->attributes & EP_CS_ATTR_PITCH_CONTROL) {
+ if (fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL) {
data[0] = 1;
- if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR,
+ if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
- PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) {
+ UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n",
dev->devnum, iface, ep);
return err;
@@ -1252,21 +1255,21 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface,
ep = get_endpoint(alts, 0)->bEndpointAddress;
/* if endpoint has sampling rate control, set it */
- if (fmt->attributes & EP_CS_ATTR_SAMPLE_RATE) {
+ if (fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE) {
int crate;
data[0] = rate;
data[1] = rate >> 8;
data[2] = rate >> 16;
- if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR,
+ if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
- SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) {
+ UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
dev->devnum, iface, fmt->altsetting, rate, ep);
return err;
}
- if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR,
+ if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN,
- SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) {
+ UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
dev->devnum, iface, fmt->altsetting, ep);
return 0; /* some devices don't support reading */
@@ -1384,9 +1387,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
* descriptors which fool us. if it has only one EP,
* assume it as adaptive-out or sync-in.
*/
- attr = fmt->ep_attr & EP_ATTR_MASK;
- if (((is_playback && attr == EP_ATTR_ASYNC) ||
- (! is_playback && attr == EP_ATTR_ADAPTIVE)) &&
+ attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+ if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
+ (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
altsd->bNumEndpoints >= 2) {
/* check sync-pipe endpoint */
/* ... and check descriptor size before accessing bSynchAddress
@@ -1426,7 +1429,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
}
/* always fill max packet size */
- if (fmt->attributes & EP_CS_ATTR_FILL_MAX)
+ if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)
subs->fill_max = 1;
if ((err = init_usb_pitch(dev, subs->interface, alts, fmt)) < 0)
@@ -1884,7 +1887,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
runtime->hw.channels_min = fp->channels;
if (runtime->hw.channels_max < fp->channels)
runtime->hw.channels_max = fp->channels;
- if (fp->fmt_type == USB_FORMAT_TYPE_II && fp->frame_size > 0) {
+ if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) {
/* FIXME: there might be more than one audio formats... */
runtime->hw.period_bytes_min = runtime->hw.period_bytes_max =
fp->frame_size;
@@ -1959,7 +1962,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
struct snd_usb_substream *subs = &as->substream[direction];
- if (subs->interface >= 0) {
+ if (!as->chip->shutdown && subs->interface >= 0) {
usb_set_interface(subs->dev, subs->interface, 0);
subs->interface = -1;
}
@@ -2118,7 +2121,7 @@ static struct usb_device_id usb_audio_ids [] = {
#include "usbquirks.h"
{ .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS),
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL },
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL },
{ } /* Terminating entry */
};
@@ -2157,7 +2160,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
snd_iprintf(buffer, " Endpoint: %d %s (%s)\n",
fp->endpoint & USB_ENDPOINT_NUMBER_MASK,
fp->endpoint & USB_DIR_IN ? "IN" : "OUT",
- sync_types[(fp->ep_attr & EP_ATTR_MASK) >> 2]);
+ sync_types[(fp->ep_attr & USB_ENDPOINT_SYNCTYPE) >> 2]);
if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) {
snd_iprintf(buffer, " Rates: %d - %d (continuous)\n",
fp->rate_min, fp->rate_max);
@@ -2420,29 +2423,68 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *
* @format: the format tag (wFormatTag)
* @fmt: the format type descriptor
*/
-static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp,
- int format, unsigned char *fmt)
+static int parse_audio_format_i_type(struct snd_usb_audio *chip,
+ struct audioformat *fp,
+ int format, void *_fmt,
+ int protocol)
{
- int pcm_format;
+ int pcm_format, i;
int sample_width, sample_bytes;
+ switch (protocol) {
+ case UAC_VERSION_1: {
+ struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
+ sample_width = fmt->bBitResolution;
+ sample_bytes = fmt->bSubframeSize;
+ break;
+ }
+
+ case UAC_VERSION_2: {
+ struct uac_format_type_i_ext_descriptor *fmt = _fmt;
+ sample_width = fmt->bBitResolution;
+ sample_bytes = fmt->bSubslotSize;
+
+ /*
+ * FIXME
+ * USB audio class v2 devices specify a bitmap of possible
+ * audio formats rather than one fix value. For now, we just
+ * pick one of them and report that as the only possible
+ * value for this setting.
+ * The bit allocation map is in fact compatible to the
+ * wFormatTag of the v1 AS streaming descriptors, which is why
+ * we can simply map the matrix.
+ */
+
+ for (i = 0; i < 5; i++)
+ if (format & (1UL << i)) {
+ format = i + 1;
+ break;
+ }
+
+ break;
+ }
+
+ default:
+ return -EINVAL;
+ }
+
/* FIXME: correct endianess and sign? */
pcm_format = -1;
- sample_width = fmt[6];
- sample_bytes = fmt[5];
+
switch (format) {
- case 0: /* some devices don't define this correctly... */
+ case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */
snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n",
chip->dev->devnum, fp->iface, fp->altsetting);
/* fall-through */
- case USB_AUDIO_FORMAT_PCM:
+ case UAC_FORMAT_TYPE_I_PCM:
if (sample_width > sample_bytes * 8) {
snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n",
chip->dev->devnum, fp->iface, fp->altsetting,
sample_width, sample_bytes);
}
/* check the format byte size */
- switch (fmt[5]) {
+ printk(" XXXXX SAMPLE BYTES %d\n", sample_bytes);
+ switch (sample_bytes) {
case 1:
pcm_format = SNDRV_PCM_FORMAT_S8;
break;
@@ -2463,12 +2505,12 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor
break;
default:
snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n",
- chip->dev->devnum, fp->iface,
- fp->altsetting, sample_width, sample_bytes);
+ chip->dev->devnum, fp->iface, fp->altsetting,
+ sample_width, sample_bytes);
break;
}
break;
- case USB_AUDIO_FORMAT_PCM8:
+ case UAC_FORMAT_TYPE_I_PCM8:
pcm_format = SNDRV_PCM_FORMAT_U8;
/* Dallas DS4201 workaround: it advertises U8 format, but really
@@ -2476,13 +2518,13 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor
if (chip->usb_id == USB_ID(0x04fa, 0x4201))
pcm_format = SNDRV_PCM_FORMAT_S8;
break;
- case USB_AUDIO_FORMAT_IEEE_FLOAT:
+ case UAC_FORMAT_TYPE_I_IEEE_FLOAT:
pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE;
break;
- case USB_AUDIO_FORMAT_ALAW:
+ case UAC_FORMAT_TYPE_I_ALAW:
pcm_format = SNDRV_PCM_FORMAT_A_LAW;
break;
- case USB_AUDIO_FORMAT_MU_LAW:
+ case UAC_FORMAT_TYPE_I_MULAW:
pcm_format = SNDRV_PCM_FORMAT_MU_LAW;
break;
default:
@@ -2496,7 +2538,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor
/*
* parse the format descriptor and stores the possible sample rates
- * on the audioformat table.
+ * on the audioformat table (audio class v1).
*
* @dev: usb device
* @fp: audioformat record
@@ -2504,13 +2546,13 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor
* @offset: the start offset of descriptor pointing the rate type
* (7 for type I and II, 8 for type II)
*/
-static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioformat *fp,
- unsigned char *fmt, int offset)
+static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audioformat *fp,
+ unsigned char *fmt, int offset)
{
int nr_rates = fmt[offset];
if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) {
- snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n",
+ snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
chip->dev->devnum, fp->iface, fp->altsetting);
return -1;
}
@@ -2561,14 +2603,87 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
}
/*
+ * parse the format descriptor and stores the possible sample rates
+ * on the audioformat table (audio class v2).
+ */
+static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
+ struct audioformat *fp,
+ struct usb_host_interface *iface)
+{
+ struct usb_device *dev = chip->dev;
+ unsigned char tmp[2], *data;
+ int i, nr_rates, data_size, ret = 0;
+
+ /* 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,
+ 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000);
+
+ if (ret < 0) {
+ snd_printk(KERN_ERR "unable to retrieve number of sample rates\n");
+ goto err;
+ }
+
+ nr_rates = (tmp[1] << 8) | tmp[0];
+ data_size = 2 + 12 * nr_rates;
+ data = kzalloc(data_size, GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto err;
+ }
+
+ /* 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,
+ 0x0100, chip->clock_id << 8, data, data_size, 1000);
+
+ if (ret < 0) {
+ snd_printk(KERN_ERR "unable to retrieve sample rate range\n");
+ ret = -EINVAL;
+ goto err_free;
+ }
+
+ fp->rate_table = kmalloc(sizeof(int) * 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++;
+ }
+
+err_free:
+ kfree(data);
+err:
+ return ret;
+}
+
+/*
* parse the format type I and III descriptors
*/
-static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp,
- int format, unsigned char *fmt)
+static int parse_audio_format_i(struct snd_usb_audio *chip,
+ struct audioformat *fp,
+ int format, void *_fmt,
+ struct usb_host_interface *iface)
{
- int pcm_format;
+ struct usb_interface_descriptor *altsd = get_iface_desc(iface);
+ struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
+ int protocol = altsd->bInterfaceProtocol;
+ int pcm_format, ret;
- if (fmt[3] == USB_FORMAT_TYPE_III) {
+ if (fmt->bFormatType == UAC_FORMAT_TYPE_III) {
/* FIXME: the format type is really IECxxx
* but we give normal PCM format to get the existing
* apps working...
@@ -2586,34 +2701,57 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *
pcm_format = SNDRV_PCM_FORMAT_S16_LE;
}
} else {
- pcm_format = parse_audio_format_i_type(chip, fp, format, fmt);
+ pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol);
if (pcm_format < 0)
return -1;
}
+
fp->format = pcm_format;
- fp->channels = fmt[4];
+
+ /* gather possible sample rates */
+ /* audio class v1 reports possible sample rates as part of the
+ * proprietary class specific descriptor.
+ * audio class v2 uses class specific EP0 range requests for that.
+ */
+ switch (protocol) {
+ case UAC_VERSION_1:
+ fp->channels = fmt->bNrChannels;
+ ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7);
+ break;
+ case UAC_VERSION_2:
+ /* fp->channels is already set in this case */
+ ret = parse_audio_format_rates_v2(chip, fp, iface);
+ break;
+ }
+
if (fp->channels < 1) {
snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n",
chip->dev->devnum, fp->iface, fp->altsetting, fp->channels);
return -1;
}
- return parse_audio_format_rates(chip, fp, fmt, 7);
+
+ return ret;
}
/*
- * prase the format type II descriptor
+ * parse the format type II descriptor
*/
-static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp,
- int format, unsigned char *fmt)
+static int parse_audio_format_ii(struct snd_usb_audio *chip,
+ struct audioformat *fp,
+ int format, void *_fmt,
+ struct usb_host_interface *iface)
{
- int brate, framesize;
+ int brate, framesize, ret;
+ struct usb_interface_descriptor *altsd = get_iface_desc(iface);
+ int protocol = altsd->bInterfaceProtocol;
+
switch (format) {
- case USB_AUDIO_FORMAT_AC3:
+ case UAC_FORMAT_TYPE_II_AC3:
/* FIXME: there is no AC3 format defined yet */
// fp->format = SNDRV_PCM_FORMAT_AC3;
fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */
break;
- case USB_AUDIO_FORMAT_MPEG:
+ case UAC_FORMAT_TYPE_II_MPEG:
fp->format = SNDRV_PCM_FORMAT_MPEG;
break;
default:
@@ -2622,26 +2760,46 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat
fp->format = SNDRV_PCM_FORMAT_MPEG;
break;
}
+
fp->channels = 1;
- brate = combine_word(&fmt[4]); /* fmt[4,5] : wMaxBitRate (in kbps) */
- framesize = combine_word(&fmt[6]); /* fmt[6,7]: wSamplesPerFrame */
- snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize);
- fp->frame_size = framesize;
- return parse_audio_format_rates(chip, fp, fmt, 8); /* fmt[8..] sample rates */
+
+ switch (protocol) {
+ case UAC_VERSION_1: {
+ struct uac_format_type_ii_discrete_descriptor *fmt = _fmt;
+ brate = le16_to_cpu(fmt->wMaxBitRate);
+ framesize = le16_to_cpu(fmt->wSamplesPerFrame);
+ snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize);
+ fp->frame_size = framesize;
+ ret = parse_audio_format_rates_v1(chip, fp, _fmt, 8); /* fmt[8..] sample rates */
+ break;
+ }
+ case UAC_VERSION_2: {
+ struct uac_format_type_ii_ext_descriptor *fmt = _fmt;
+ brate = le16_to_cpu(fmt->wMaxBitRate);
+ framesize = le16_to_cpu(fmt->wSamplesPerFrame);
+ snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize);
+ fp->frame_size = framesize;
+ ret = parse_audio_format_rates_v2(chip, fp, iface);
+ break;
+ }
+ }
+
+ return ret;
}
static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp,
- int format, unsigned char *fmt, int stream)
+ int format, unsigned char *fmt, int stream,
+ struct usb_host_interface *iface)
{
int err;
switch (fmt[3]) {
- case USB_FORMAT_TYPE_I:
- case USB_FORMAT_TYPE_III:
- err = parse_audio_format_i(chip, fp, format, fmt);
+ case UAC_FORMAT_TYPE_I:
+ case UAC_FORMAT_TYPE_III:
+ err = parse_audio_format_i(chip, fp, format, fmt, iface);
break;
- case USB_FORMAT_TYPE_II:
- err = parse_audio_format_ii(chip, fp, format, fmt);
+ case UAC_FORMAT_TYPE_II:
+ err = parse_audio_format_ii(chip, fp, format, fmt, iface);
break;
default:
snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n",
@@ -2659,7 +2817,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp
if (chip->usb_id == USB_ID(0x041e, 0x3000) ||
chip->usb_id == USB_ID(0x041e, 0x3020) ||
chip->usb_id == USB_ID(0x041e, 0x3061)) {
- if (fmt[3] == USB_FORMAT_TYPE_I &&
+ if (fmt[3] == UAC_FORMAT_TYPE_I &&
fp->rates != SNDRV_PCM_RATE_48000 &&
fp->rates != SNDRV_PCM_RATE_96000)
return -1;
@@ -2688,10 +2846,10 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
struct usb_host_interface *alts;
struct usb_interface_descriptor *altsd;
int i, altno, err, stream;
- int format;
+ int format = 0, num_channels = 0;
struct audioformat *fp = NULL;
unsigned char *fmt, *csep;
- int num;
+ int num, protocol;
dev = chip->dev;
@@ -2710,10 +2868,11 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
for (i = 0; i < num; i++) {
alts = &iface->altsetting[i];
altsd = get_iface_desc(alts);
+ protocol = altsd->bInterfaceProtocol;
/* skip invalid one */
if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
- (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING &&
+ (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING &&
altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) ||
altsd->bNumEndpoints < 1 ||
le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0)
@@ -2734,30 +2893,65 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
continue;
/* get audio formats */
- fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL);
- if (!fmt) {
- snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n",
- dev->devnum, iface_no, altno);
- continue;
+ switch (protocol) {
+ case UAC_VERSION_1: {
+ struct uac_as_header_descriptor_v1 *as =
+ snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
+
+ if (!as) {
+ snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+
+ if (as->bLength < sizeof(*as)) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+
+ format = le16_to_cpu(as->wFormatTag); /* remember the format value */
+ break;
}
- if (fmt[0] < 7) {
- snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n",
- dev->devnum, iface_no, altno);
- continue;
+ case UAC_VERSION_2: {
+ struct uac_as_header_descriptor_v2 *as =
+ snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
+
+ if (!as) {
+ snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+
+ if (as->bLength < sizeof(*as)) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+
+ num_channels = as->bNrChannels;
+ format = le32_to_cpu(as->bmFormats);
+
+ break;
}
- format = (fmt[6] << 8) | fmt[5]; /* remember the format value */
+ default:
+ snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n",
+ dev->devnum, iface_no, altno, protocol);
+ continue;
+ }
/* get format type */
- fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE);
+ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE);
if (!fmt) {
- snd_printk(KERN_ERR "%d:%u:%d : no FORMAT_TYPE desc\n",
+ snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n",
dev->devnum, iface_no, altno);
continue;
}
- if (fmt[0] < 8) {
- snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n",
+ if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) ||
+ ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
dev->devnum, iface_no, altno);
continue;
}
@@ -2770,6 +2964,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 &&
fp && fp->altsetting == 1 && fp->channels == 1 &&
fp->format == SNDRV_PCM_FORMAT_S16_LE &&
+ protocol == UAC_VERSION_1 &&
le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
fp->maxpacksize * 2)
continue;
@@ -2778,7 +2973,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
/* Creamware Noah has this descriptor after the 2nd endpoint */
if (!csep && altsd->bNumEndpoints >= 2)
csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
- if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) {
+ if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) {
snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
" class specific endpoint descriptor\n",
dev->devnum, iface_no, altno);
@@ -2798,6 +2993,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
fp->datainterval = parse_datainterval(chip, alts);
fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ /* num_channels is only set for v2 interfaces */
+ fp->channels = num_channels;
if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
* (fp->maxpacksize & 0x7ff);
@@ -2810,12 +3007,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
/* Optoplay sets the sample rate attribute although
* it seems not supporting it in fact.
*/
- fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE;
+ fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE;
break;
case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
/* doesn't set the sample rate attribute, but supports it */
- fp->attributes |= EP_CS_ATTR_SAMPLE_RATE;
+ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE;
break;
case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
@@ -2824,16 +3021,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
* plantronics headset and Griffin iMic have set adaptive-in
* although it's really not...
*/
- fp->ep_attr &= ~EP_ATTR_MASK;
+ fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- fp->ep_attr |= EP_ATTR_ADAPTIVE;
+ fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE;
else
- fp->ep_attr |= EP_ATTR_SYNC;
+ fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
break;
}
/* ok, let's parse further... */
- if (parse_audio_format(chip, fp, format, fmt, stream) < 0) {
+ if (parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
kfree(fp->rate_table);
kfree(fp);
continue;
@@ -2875,6 +3072,65 @@ static void snd_usb_stream_disconnect(struct list_head *head)
}
}
+static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int interface)
+{
+ struct usb_device *dev = chip->dev;
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ struct usb_interface *iface = usb_ifnum_to_if(dev, interface);
+
+ if (!iface) {
+ snd_printk(KERN_ERR "%d:%u:%d : does not exist\n",
+ dev->devnum, ctrlif, interface);
+ return -EINVAL;
+ }
+
+ if (usb_interface_claimed(iface)) {
+ snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n",
+ dev->devnum, ctrlif, interface);
+ return -EINVAL;
+ }
+
+ alts = &iface->altsetting[0];
+ altsd = get_iface_desc(alts);
+ if ((altsd->bInterfaceClass == USB_CLASS_AUDIO ||
+ altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) &&
+ altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) {
+ int err = snd_usbmidi_create(chip->card, iface,
+ &chip->midi_list, NULL);
+ if (err < 0) {
+ snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n",
+ dev->devnum, ctrlif, interface);
+ return -EINVAL;
+ }
+ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+
+ return 0;
+ }
+
+ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
+ altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
+ altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING) {
+ snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n",
+ dev->devnum, ctrlif, interface, altsd->bInterfaceClass);
+ /* skip non-supported classes */
+ return -EINVAL;
+ }
+
+ if (snd_usb_get_speed(dev) == USB_SPEED_LOW) {
+ snd_printk(KERN_ERR "low speed audio streaming not supported\n");
+ return -EINVAL;
+ }
+
+ if (! parse_audio_endpoints(chip, interface)) {
+ usb_set_interface(dev, interface, 0); /* reset the current interface */
+ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
/*
* parse audio control descriptor and create pcm/midi streams
*/
@@ -2882,67 +3138,81 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
{
struct usb_device *dev = chip->dev;
struct usb_host_interface *host_iface;
- struct usb_interface *iface;
- unsigned char *p1;
- int i, j;
+ struct usb_interface_descriptor *altsd;
+ void *control_header;
+ int i, protocol;
/* find audiocontrol interface */
host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0];
- if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) {
- snd_printk(KERN_ERR "cannot find HEADER\n");
- return -EINVAL;
- }
- if (! p1[7] || p1[0] < 8 + p1[7]) {
- snd_printk(KERN_ERR "invalid HEADER\n");
+ control_header = snd_usb_find_csint_desc(host_iface->extra,
+ host_iface->extralen,
+ NULL, UAC_HEADER);
+ altsd = get_iface_desc(host_iface);
+ protocol = altsd->bInterfaceProtocol;
+
+ if (!control_header) {
+ snd_printk(KERN_ERR "cannot find UAC_HEADER\n");
return -EINVAL;
}
- /*
- * parse all USB audio streaming interfaces
- */
- for (i = 0; i < p1[7]; i++) {
- struct usb_host_interface *alts;
- struct usb_interface_descriptor *altsd;
- j = p1[8 + i];
- iface = usb_ifnum_to_if(dev, j);
- if (!iface) {
- snd_printk(KERN_ERR "%d:%u:%d : does not exist\n",
- dev->devnum, ctrlif, j);
- continue;
- }
- if (usb_interface_claimed(iface)) {
- snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, j);
- continue;
+ switch (protocol) {
+ case UAC_VERSION_1: {
+ struct uac_ac_header_descriptor_v1 *h1 = control_header;
+
+ if (!h1->bInCollection) {
+ snd_printk(KERN_INFO "skipping empty audio interface (v1)\n");
+ return -EINVAL;
}
- alts = &iface->altsetting[0];
- altsd = get_iface_desc(alts);
- if ((altsd->bInterfaceClass == USB_CLASS_AUDIO ||
- altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) &&
- altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) {
- int err = snd_usbmidi_create(chip->card, iface,
- &chip->midi_list, NULL);
- if (err < 0) {
- snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j);
- continue;
- }
- usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
- continue;
+
+ if (h1->bLength < sizeof(*h1) + h1->bInCollection) {
+ snd_printk(KERN_ERR "invalid UAC_HEADER (v1)\n");
+ return -EINVAL;
}
- if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
- altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
- altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) {
- snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, j, altsd->bInterfaceClass);
- /* skip non-supported classes */
- continue;
+
+ for (i = 0; i < h1->bInCollection; i++)
+ snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]);
+
+ break;
+ }
+
+ case UAC_VERSION_2: {
+ struct uac_clock_source_descriptor *cs;
+ struct usb_interface_assoc_descriptor *assoc =
+ usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
+
+ if (!assoc) {
+ snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n");
+ return -EINVAL;
}
- if (snd_usb_get_speed(dev) == USB_SPEED_LOW) {
- snd_printk(KERN_ERR "low speed audio streaming not supported\n");
- continue;
+
+ /* FIXME: for now, we expect there is at least one clock source
+ * descriptor and we always take the first one.
+ * We should properly support devices with multiple clock sources,
+ * clock selectors and sample rate conversion units. */
+
+ cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen,
+ NULL, UAC_CLOCK_SOURCE);
+
+ if (!cs) {
+ snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n");
+ return -EINVAL;
}
- if (! parse_audio_endpoints(chip, j)) {
- usb_set_interface(dev, j, 0); /* reset the current interface */
- usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+
+ chip->clock_id = cs->bClockID;
+
+ for (i = 0; i < assoc->bInterfaceCount; i++) {
+ int intf = assoc->bFirstInterface + i;
+
+ if (intf != ctrlif)
+ snd_usb_create_stream(chip, ctrlif, intf);
}
+
+ break;
+ }
+
+ default:
+ snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol);
+ return -EINVAL;
}
return 0;
@@ -3033,7 +3303,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
static const struct audioformat ua_format = {
.format = SNDRV_PCM_FORMAT_S24_3LE,
.channels = 2,
- .fmt_type = USB_FORMAT_TYPE_I,
+ .fmt_type = UAC_FORMAT_TYPE_I,
.altsetting = 1,
.altset_idx = 1,
.rates = SNDRV_PCM_RATE_CONTINUOUS,
@@ -3554,7 +3824,6 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
ifnum = get_iface_desc(alts)->bInterfaceNumber;
id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
le16_to_cpu(dev->descriptor.idProduct));
-
if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum)
goto __err_val;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 96c558a76ba..42c299cbf63 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -21,93 +21,6 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
-
-/*
- */
-
-#define USB_SUBCLASS_AUDIO_CONTROL 0x01
-#define USB_SUBCLASS_AUDIO_STREAMING 0x02
-#define USB_SUBCLASS_MIDI_STREAMING 0x03
-#define USB_SUBCLASS_VENDOR_SPEC 0xff
-
-#define HEADER 0x01
-#define INPUT_TERMINAL 0x02
-#define OUTPUT_TERMINAL 0x03
-#define MIXER_UNIT 0x04
-#define SELECTOR_UNIT 0x05
-#define FEATURE_UNIT 0x06
-#define PROCESSING_UNIT 0x07
-#define EXTENSION_UNIT 0x08
-
-#define AS_GENERAL 0x01
-#define FORMAT_TYPE 0x02
-#define FORMAT_SPECIFIC 0x03
-
-#define EP_GENERAL 0x01
-
-#define MS_GENERAL 0x01
-#define MIDI_IN_JACK 0x02
-#define MIDI_OUT_JACK 0x03
-
-/* endpoint attributes */
-#define EP_ATTR_MASK 0x0c
-#define EP_ATTR_ASYNC 0x04
-#define EP_ATTR_ADAPTIVE 0x08
-#define EP_ATTR_SYNC 0x0c
-
-/* cs endpoint attributes */
-#define EP_CS_ATTR_SAMPLE_RATE 0x01
-#define EP_CS_ATTR_PITCH_CONTROL 0x02
-#define EP_CS_ATTR_FILL_MAX 0x80
-
-/* Audio Class specific Request Codes */
-
-#define SET_CUR 0x01
-#define GET_CUR 0x81
-#define SET_MIN 0x02
-#define GET_MIN 0x82
-#define SET_MAX 0x03
-#define GET_MAX 0x83
-#define SET_RES 0x04
-#define GET_RES 0x84
-#define SET_MEM 0x05
-#define GET_MEM 0x85
-#define GET_STAT 0xff
-
-/* Terminal Control Selectors */
-
-#define COPY_PROTECT_CONTROL 0x01
-
-/* Endpoint Control Selectors */
-
-#define SAMPLING_FREQ_CONTROL 0x01
-#define PITCH_CONTROL 0x02
-
-/* Format Types */
-#define USB_FORMAT_TYPE_I 0x01
-#define USB_FORMAT_TYPE_II 0x02
-#define USB_FORMAT_TYPE_III 0x03
-
-/* type I */
-#define USB_AUDIO_FORMAT_PCM 0x01
-#define USB_AUDIO_FORMAT_PCM8 0x02
-#define USB_AUDIO_FORMAT_IEEE_FLOAT 0x03
-#define USB_AUDIO_FORMAT_ALAW 0x04
-#define USB_AUDIO_FORMAT_MU_LAW 0x05
-
-/* type II */
-#define USB_AUDIO_FORMAT_MPEG 0x1001
-#define USB_AUDIO_FORMAT_AC3 0x1002
-
-/* type III */
-#define USB_AUDIO_FORMAT_IEC1937_AC3 0x2001
-#define USB_AUDIO_FORMAT_IEC1937_MPEG1_LAYER1 0x2002
-#define USB_AUDIO_FORMAT_IEC1937_MPEG2_NOEXT 0x2003
-#define USB_AUDIO_FORMAT_IEC1937_MPEG2_EXT 0x2004
-#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER1_LS 0x2005
-#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER23_LS 0x2006
-
-
/* maximum number of endpoints per interface */
#define MIDI_MAX_ENDPOINTS 2
@@ -129,6 +42,9 @@ struct snd_usb_audio {
int num_interfaces;
int num_suspended_intf;
+ /* for audio class v2 */
+ int clock_id;
+
struct list_head pcm_list; /* list of pcm streams */
int pcm_devs;
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index b2da478a0fa..2c59afd9961 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -46,6 +46,8 @@
#include <linux/timer.h>
#include <linux/usb.h>
#include <linux/wait.h>
+#include <linux/usb/audio.h>
+
#include <sound/core.h>
#include <sound/control.h>
#include <sound/rawmidi.h>
@@ -1540,7 +1542,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
if (hostif->extralen >= 7 &&
ms_header->bLength >= 7 &&
ms_header->bDescriptorType == USB_DT_CS_INTERFACE &&
- ms_header->bDescriptorSubtype == HEADER)
+ ms_header->bDescriptorSubtype == UAC_HEADER)
snd_printdd(KERN_INFO "MIDIStreaming version %02x.%02x\n",
ms_header->bcdMSC[1], ms_header->bcdMSC[0]);
else
@@ -1556,7 +1558,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
if (hostep->extralen < 4 ||
ms_ep->bLength < 4 ||
ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
- ms_ep->bDescriptorSubtype != MS_GENERAL)
+ ms_ep->bDescriptorSubtype != UAC_MS_GENERAL)
continue;
if (usb_endpoint_dir_out(ep)) {
if (endpoints[epidx].out_ep) {
@@ -1768,9 +1770,9 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi,
cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2;
cs_desc += cs_desc[0]) {
if (cs_desc[1] == USB_DT_CS_INTERFACE) {
- if (cs_desc[2] == MIDI_IN_JACK)
+ if (cs_desc[2] == UAC_MIDI_IN_JACK)
endpoint->in_cables = (endpoint->in_cables << 1) | 1;
- else if (cs_desc[2] == MIDI_OUT_JACK)
+ else if (cs_desc[2] == UAC_MIDI_OUT_JACK)
endpoint->out_cables = (endpoint->out_cables << 1) | 1;
}
}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 35b4830fb0c..8e8f871b74c 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -32,6 +32,8 @@
#include <linux/slab.h>
#include <linux/string.h>
#include <linux/usb.h>
+#include <linux/usb/audio.h>
+
#include <sound/core.h>
#include <sound/control.h>
#include <sound/hwdep.h>
@@ -284,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un
p = NULL;
while ((p = snd_usb_find_desc(state->buffer, state->buflen, p,
USB_DT_CS_INTERFACE)) != NULL) {
- if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT && p[3] == unit)
+ if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC_EXTENSION_UNIT_V1 && p[3] == unit)
return p;
}
return NULL;
@@ -405,14 +407,14 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali
static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value)
{
- return get_ctl_value(cval, GET_CUR, validx, value);
+ return get_ctl_value(cval, UAC_GET_CUR, validx, value);
}
/* channel = 0: master, 1 = first channel */
static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval,
int channel, int *value)
{
- return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value);
+ return get_ctl_value(cval, UAC_GET_CUR, (cval->control << 8) | channel, value);
}
static int get_cur_mix_value(struct usb_mixer_elem_info *cval,
@@ -466,14 +468,14 @@ static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali
static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value)
{
- return set_ctl_value(cval, SET_CUR, validx, value);
+ return set_ctl_value(cval, UAC_SET_CUR, validx, value);
}
static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,
int index, int value)
{
int err;
- err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel,
+ err = set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel,
value);
if (err < 0)
return err;
@@ -603,13 +605,13 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm
if (term_only)
return 0;
switch (iterm->type >> 16) {
- case SELECTOR_UNIT:
+ case UAC_SELECTOR_UNIT:
strcpy(name, "Selector"); return 8;
- case PROCESSING_UNIT:
+ case UAC_PROCESSING_UNIT_V1:
strcpy(name, "Process Unit"); return 12;
- case EXTENSION_UNIT:
+ case UAC_EXTENSION_UNIT_V1:
strcpy(name, "Ext Unit"); return 8;
- case MIXER_UNIT:
+ case UAC_MIXER_UNIT:
strcpy(name, "Mixer"); return 5;
default:
return sprintf(name, "Unit %d", iterm->id);
@@ -648,22 +650,22 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
while ((p1 = find_audio_control_unit(state, id)) != NULL) {
term->id = id;
switch (p1[2]) {
- case INPUT_TERMINAL:
+ case UAC_INPUT_TERMINAL:
term->type = combine_word(p1 + 4);
term->channels = p1[7];
term->chconfig = combine_word(p1 + 8);
term->name = p1[11];
return 0;
- case FEATURE_UNIT:
+ case UAC_FEATURE_UNIT:
id = p1[4];
break; /* continue to parse */
- case MIXER_UNIT:
+ case UAC_MIXER_UNIT:
term->type = p1[2] << 16; /* virtual type */
term->channels = p1[5 + p1[4]];
term->chconfig = combine_word(p1 + 6 + p1[4]);
term->name = p1[p1[0] - 1];
return 0;
- case SELECTOR_UNIT:
+ case UAC_SELECTOR_UNIT:
/* call recursively to retrieve the channel info */
if (check_input_term(state, p1[5], term) < 0)
return -ENODEV;
@@ -671,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
term->id = id;
term->name = p1[9 + p1[0] - 1];
return 0;
- case PROCESSING_UNIT:
- case EXTENSION_UNIT:
+ case UAC_PROCESSING_UNIT_V1:
+ case UAC_EXTENSION_UNIT_V1:
if (p1[6] == 1) {
id = p1[7];
break; /* continue to parse */
@@ -750,23 +752,23 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
break;
}
}
- if (get_ctl_value(cval, GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 ||
- get_ctl_value(cval, GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
+ if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 ||
+ get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n",
cval->id, cval->mixer->ctrlif, cval->control, cval->id);
return -EINVAL;
}
- if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) {
+ if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) {
cval->res = 1;
} else {
int last_valid_res = cval->res;
while (cval->res > 1) {
- if (set_ctl_value(cval, SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0)
+ if (set_ctl_value(cval, UAC_SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0)
break;
cval->res /= 2;
}
- if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0)
+ if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0)
cval->res = last_valid_res;
}
if (cval->res == 0)
@@ -1086,29 +1088,30 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
*
* most of controlls are defined here.
*/
-static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsigned char *ftr)
+static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr)
{
int channels, i, j;
struct usb_audio_term iterm;
unsigned int master_bits, first_ch_bits;
int err, csize;
+ struct uac_feature_unit_descriptor *ftr = _ftr;
- if (ftr[0] < 7 || ! (csize = ftr[5]) || ftr[0] < 7 + csize) {
- snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid);
+ if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) {
+ snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid);
return -EINVAL;
}
/* parse the source unit */
- if ((err = parse_audio_unit(state, ftr[4])) < 0)
+ if ((err = parse_audio_unit(state, ftr->bSourceID)) < 0)
return err;
/* determine the input source type and name */
- if (check_input_term(state, ftr[4], &iterm) < 0)
+ if (check_input_term(state, ftr->bSourceID, &iterm) < 0)
return -EINVAL;
- channels = (ftr[0] - 7) / csize - 1;
+ channels = (ftr->bLength - 7) / csize - 1;
- master_bits = snd_usb_combine_bytes(ftr + 6, csize);
+ master_bits = snd_usb_combine_bytes(ftr->controls, csize);
/* master configuration quirks */
switch (state->chip->usb_id) {
case USB_ID(0x08bb, 0x2702):
@@ -1119,21 +1122,21 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig
break;
}
if (channels > 0)
- first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize);
+ first_ch_bits = snd_usb_combine_bytes(ftr->controls + csize, csize);
else
first_ch_bits = 0;
/* check all control types */
for (i = 0; i < 10; i++) {
unsigned int ch_bits = 0;
for (j = 0; j < channels; j++) {
- unsigned int mask = snd_usb_combine_bytes(ftr + 6 + csize * (j+1), csize);
+ unsigned int mask = snd_usb_combine_bytes(ftr->controls + csize * (j+1), csize);
if (mask & (1 << i))
ch_bits |= (1 << j);
}
if (ch_bits & 1) /* the first channel must be set (for ease of programming) */
- build_feature_ctl(state, ftr, ch_bits, i, &iterm, unitid);
+ build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid);
if (master_bits & (1 << i))
- build_feature_ctl(state, ftr, 0, i, &iterm, unitid);
+ build_feature_ctl(state, _ftr, 0, i, &iterm, unitid);
}
return 0;
@@ -1736,17 +1739,17 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
}
switch (p1[2]) {
- case INPUT_TERMINAL:
+ case UAC_INPUT_TERMINAL:
return 0; /* NOP */
- case MIXER_UNIT:
+ case UAC_MIXER_UNIT:
return parse_audio_mixer_unit(state, unitid, p1);
- case SELECTOR_UNIT:
+ case UAC_SELECTOR_UNIT:
return parse_audio_selector_unit(state, unitid, p1);
- case FEATURE_UNIT:
+ case UAC_FEATURE_UNIT:
return parse_audio_feature_unit(state, unitid, p1);
- case PROCESSING_UNIT:
+ case UAC_PROCESSING_UNIT_V1:
return parse_audio_processing_unit(state, unitid, p1);
- case EXTENSION_UNIT:
+ case UAC_EXTENSION_UNIT_V1:
return parse_audio_extension_unit(state, unitid, p1);
default:
snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
@@ -1776,11 +1779,11 @@ static int snd_usb_mixer_dev_free(struct snd_device *device)
/*
* create mixer controls
*
- * walk through all OUTPUT_TERMINAL descriptors to search for mixers
+ * walk through all UAC_OUTPUT_TERMINAL descriptors to search for mixers
*/
static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
{
- unsigned char *desc;
+ struct uac_output_terminal_descriptor_v1 *desc;
struct mixer_build state;
int err;
const struct usbmix_ctl_map *map;
@@ -1804,14 +1807,14 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
}
desc = NULL;
- while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) {
- if (desc[0] < 9)
+ while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, UAC_OUTPUT_TERMINAL)) != NULL) {
+ if (desc->bLength < 9)
continue; /* invalid descriptor? */
- set_bit(desc[3], state.unitbitmap); /* mark terminal ID as visited */
- state.oterm.id = desc[3];
- state.oterm.type = combine_word(&desc[4]);
- state.oterm.name = desc[8];
- err = parse_audio_unit(&state, desc[7]);
+ set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */
+ state.oterm.id = desc->bTerminalID;
+ state.oterm.type = le16_to_cpu(desc->wTerminalType);
+ state.oterm.name = desc->iTerminal;
+ err = parse_audio_unit(&state, desc->bSourceID);
if (err < 0)
return err;
}
@@ -2044,7 +2047,7 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer)
}
mixer->rc_setup_packet->bRequestType =
USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE;
- mixer->rc_setup_packet->bRequest = GET_MEM;
+ mixer->rc_setup_packet->bRequest = UAC_GET_MEM;
mixer->rc_setup_packet->wValue = cpu_to_le16(0);
mixer->rc_setup_packet->wIndex = cpu_to_le16(0);
mixer->rc_setup_packet->wLength = cpu_to_le16(len);
@@ -2167,7 +2170,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry,
snd_iprintf(buffer, "%s: ", jacks[i].name);
err = snd_usb_ctl_msg(mixer->chip->dev,
usb_rcvctrlpipe(mixer->chip->dev, 0),
- GET_MEM, USB_DIR_IN | USB_TYPE_CLASS |
+ UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS |
USB_RECIP_INTERFACE, 0,
jacks[i].unitid << 8, buf, 3, 100);
if (err == 3 && (buf[0] == 3 || buf[0] == 6))
@@ -2255,7 +2258,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
};
struct usb_mixer_interface *mixer;
struct snd_info_entry *entry;
- int err;
+ struct usb_host_interface *host_iface;
+ int err, protocol;
strcpy(chip->card->mixername, "USB Mixer");
@@ -2272,6 +2276,16 @@ 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];
+ protocol = host_iface->desc.bInterfaceProtocol;
+
+ /* FIXME! */
+ if (protocol != UAC_VERSION_1) {
+ snd_printk(KERN_WARNING "mixer interface protocol 0x%02x not yet supported\n",
+ protocol);
+ return 0;
+ }
+
if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
(err = snd_usb_mixer_status_create(mixer)) < 0)
goto _error;
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 977d980fb11..2b426c1fd0e 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -91,7 +91,7 @@
.idVendor = 0x046d,
.idProduct = 0x0850,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
},
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
@@ -100,7 +100,7 @@
.idVendor = 0x046d,
.idProduct = 0x08ae,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
},
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
@@ -109,7 +109,7 @@
.idVendor = 0x046d,
.idProduct = 0x08c6,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
},
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
@@ -118,7 +118,7 @@
.idVendor = 0x046d,
.idProduct = 0x08f0,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
},
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
@@ -127,7 +127,7 @@
.idVendor = 0x046d,
.idProduct = 0x08f5,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
},
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
@@ -136,7 +136,7 @@
.idVendor = 0x046d,
.idProduct = 0x08f6,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
},
{
USB_DEVICE(0x046d, 0x0990),
@@ -301,7 +301,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.iface = 1,
.altsetting = 1,
.altset_idx = 1,
- .attributes = EP_CS_ATTR_FILL_MAX,
+ .attributes = UAC_EP_CS_ATTR_FILL_MAX,
.endpoint = 0x81,
.ep_attr = 0x05,
.rates = SNDRV_PCM_RATE_CONTINUOUS,
@@ -2078,7 +2078,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-950Q",
@@ -2092,7 +2092,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-950Q",
@@ -2106,7 +2106,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-950Q",
@@ -2120,7 +2120,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-950Q",
@@ -2134,7 +2134,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-950Q",
@@ -2148,7 +2148,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-950Q",
@@ -2162,7 +2162,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-950Q",
@@ -2176,7 +2176,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "Hauppauge",
.product_name = "HVR-850",
@@ -2185,6 +2185,51 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
+/* Digidesign Mbox */
+{
+ /* Thanks to Clemens Ladisch <clemens@ladisch.de> */
+ USB_DEVICE(0x0dba, 0x1000),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "Digidesign",
+ .product_name = "MBox",
+ .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_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S24_3BE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
+ .endpoint = 0x02,
+ .ep_attr = 0x01,
+ .maxpacksize = 0x130,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .nr_rates = 2,
+ .rate_table = (unsigned int[]) {
+ 44100, 48000
+ }
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+
+ }
+},
+
{
/*
* Some USB MIDI devices don't have an audio control interface,
@@ -2193,7 +2238,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.match_flags = USB_DEVICE_ID_MATCH_INT_CLASS |
USB_DEVICE_ID_MATCH_INT_SUBCLASS,
.bInterfaceClass = USB_CLASS_AUDIO,
- .bInterfaceSubClass = USB_SUBCLASS_MIDI_STREAMING,
+ .bInterfaceSubClass = USB_SUBCLASS_MIDISTREAMING,
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
.ifnum = QUIRK_ANY_INTERFACE,
.type = QUIRK_MIDI_STANDARD_INTERFACE
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index 91bb29666d2..44deb21b177 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -16,6 +16,8 @@
* Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
+#include <linux/usb.h>
+#include <linux/usb/audio.h>
#include <sound/core.h>
#include <sound/hwdep.h>
#include <sound/pcm.h>
@@ -315,9 +317,9 @@ static int us122l_set_sample_rate(struct usb_device *dev, int rate)
data[0] = rate;
data[1] = rate >> 8;
data[2] = rate >> 16;
- err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR,
+ err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
- SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000);
+ UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000);
if (err < 0)
snd_printk(KERN_ERR "%d: cannot set freq %d to ep 0x%x\n",
dev->devnum, rate, ep);