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-rw-r--r--sound/oss/pas2_card.c5
-rw-r--r--sound/pci/au88x0/au88x0_synth.c2
-rw-r--r--sound/pci/hda/hda_codec.c2
-rw-r--r--sound/pci/hda/patch_realtek.c30
-rw-r--r--sound/soc/codecs/arizona.c9
-rw-r--r--sound/soc/codecs/arizona.h18
-rw-r--r--sound/soc/codecs/cs4271.c6
-rw-r--r--sound/soc/codecs/wm5102.c48
-rw-r--r--sound/soc/soc-core.c35
-rw-r--r--sound/soc/soc-pcm.c1
-rw-r--r--sound/usb/midi.c4
-rw-r--r--sound/usb/quirks-table.h24
-rw-r--r--sound/usb/quirks.c16
-rw-r--r--sound/usb/usbaudio.h1
14 files changed, 160 insertions, 41 deletions
diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c
index dabf8a871dcc..7004e24d209f 100644
--- a/sound/oss/pas2_card.c
+++ b/sound/oss/pas2_card.c
@@ -333,6 +333,11 @@ static void __init attach_pas_card(struct address_info *hw_config)
{
char temp[100];
+ if (pas_model < 0 ||
+ pas_model >= ARRAY_SIZE(pas_model_names)) {
+ printk(KERN_ERR "pas2 unrecognized model.\n");
+ return;
+ }
sprintf(temp,
"%s rev %d", pas_model_names[(int) pas_model],
pas_read(0x2789));
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index 2805e34bd41d..8bef47311e45 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -58,7 +58,7 @@ static void vortex_wt_setdsout(vortex_t * vortex, u32 wt, int en)
if (en)
temp |= (1 << (wt & 0x1f));
else
- temp &= (1 << ~(wt & 0x1f));
+ temp &= ~(1 << (wt & 0x1f));
hwwrite(vortex->mmio, WT_DSREG((wt >= 0x20) ? 1 : 0), temp);
}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8353c77536ac..b8fb0a5adb9b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2531,7 +2531,7 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char * const texts[] = {
- "Off", "On", "Follow Master"
+ "On", "Off", "Follow Master"
};
unsigned int index;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6ee34593774a..71ae23dd7103 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5992,6 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec,
spec->automute_hook = alc269_quanta_automute;
}
+/* update mute-LED according to the speaker mute state via mic1 VREF pin */
+static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ unsigned int pinval = AC_PINCTL_IN_EN + (enabled ?
+ AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80);
+ snd_hda_set_pin_ctl_cache(codec, 0x18, pinval);
+}
+
+static void alc269_fixup_mic1_mute(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ switch (action) {
+ case ALC_FIXUP_ACT_BUILD:
+ spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook;
+ snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true);
+ /* fallthru */
+ case ALC_FIXUP_ACT_INIT:
+ snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+ break;
+ }
+}
+
/* update mute-LED according to the speaker mute state via mic2 VREF pin */
static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled)
{
@@ -6043,6 +6067,7 @@ enum {
ALC269_FIXUP_DMIC,
ALC269VB_FIXUP_AMIC,
ALC269VB_FIXUP_DMIC,
+ ALC269_FIXUP_MIC1_MUTE_LED,
ALC269_FIXUP_MIC2_MUTE_LED,
ALC269_FIXUP_INV_DMIC,
ALC269_FIXUP_LENOVO_DOCK,
@@ -6171,6 +6196,10 @@ static const struct alc_fixup alc269_fixups[] = {
{ }
},
},
+ [ALC269_FIXUP_MIC1_MUTE_LED] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_mic1_mute,
+ },
[ALC269_FIXUP_MIC2_MUTE_LED] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc269_fixup_mic2_mute,
@@ -6215,6 +6244,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
+ SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index adf397b9d0e6..1d8bb5917594 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
mode = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- mode = 1;
- break;
case SND_SOC_DAIFMT_I2S:
mode = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- mode = 3;
- break;
default:
arizona_aif_err(dai, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
- ARIZONA_AIF1_RATE_MASK, 8);
+ ARIZONA_AIF1_RATE_MASK,
+ 8 << ARIZONA_AIF1_RATE_SHIFT);
break;
default:
arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 41dae1ed3b71..4deebeb07177 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -34,15 +34,15 @@
#define ARIZONA_FLL_SRC_MCLK1 0
#define ARIZONA_FLL_SRC_MCLK2 1
-#define ARIZONA_FLL_SRC_SLIMCLK 2
-#define ARIZONA_FLL_SRC_FLL1 3
-#define ARIZONA_FLL_SRC_FLL2 4
-#define ARIZONA_FLL_SRC_AIF1BCLK 5
-#define ARIZONA_FLL_SRC_AIF2BCLK 6
-#define ARIZONA_FLL_SRC_AIF3BCLK 7
-#define ARIZONA_FLL_SRC_AIF1LRCLK 8
-#define ARIZONA_FLL_SRC_AIF2LRCLK 9
-#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+#define ARIZONA_FLL_SRC_SLIMCLK 3
+#define ARIZONA_FLL_SRC_FLL1 4
+#define ARIZONA_FLL_SRC_FLL2 5
+#define ARIZONA_FLL_SRC_AIF1BCLK 8
+#define ARIZONA_FLL_SRC_AIF2BCLK 9
+#define ARIZONA_FLL_SRC_AIF3BCLK 10
+#define ARIZONA_FLL_SRC_AIF1LRCLK 12
+#define ARIZONA_FLL_SRC_AIF2LRCLK 13
+#define ARIZONA_FLL_SRC_AIF3LRCLK 14
#define ARIZONA_MIXER_VOL_MASK 0x00FE
#define ARIZONA_MIXER_VOL_SHIFT 1
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 4f1127935fdf..ac8742a1f25a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec)
struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
int ret;
int gpio_nreset = -EINVAL;
- int amutec_eq_bmutec = 0;
+ bool amutec_eq_bmutec = false;
#ifdef CONFIG_OF
if (of_match_device(cs4271_dt_ids, codec->dev)) {
gpio_nreset = of_get_named_gpio(codec->dev->of_node,
"reset-gpio", 0);
- if (!of_get_property(codec->dev->of_node,
+ if (of_get_property(codec->dev->of_node,
"cirrus,amutec-eq-bmutec", NULL))
- amutec_eq_bmutec = 1;
+ amutec_eq_bmutec = true;
}
#endif
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 688ade080589..7a9048dad1cd 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,6 +36,9 @@
struct wm5102_priv {
struct arizona_priv core;
struct arizona_fll fll[2];
+
+ unsigned int spk_ena:2;
+ unsigned int spk_ena_pending:1;
};
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
};
+static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+ if (arizona->rev < 1)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (!wm5102->spk_ena) {
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ wm5102->spk_ena_pending = true;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ if (wm5102->spk_ena_pending) {
+ msleep(75);
+ snd_soc_write(codec, 0x4f5, 0xda);
+ wm5102->spk_ena_pending = false;
+ wm5102->spk_ena++;
+ }
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ wm5102->spk_ena--;
+ if (!wm5102->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ if (!wm5102->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x0da);
+ break;
+ }
+
+ return 0;
+}
+
+
ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 91d592ff67b7..2370063b5824 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card,
INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
ret = device_add(rtd->dev);
if (ret < 0) {
+ /* calling put_device() here to free the rtd->dev */
+ put_device(rtd->dev);
dev_err(card->dev,
"ASoC: failed to register runtime device: %d\n", ret);
return ret;
@@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
/* unregister the rtd device */
if (rtd->dev_registered) {
device_remove_file(rtd->dev, &dev_attr_codec_reg);
- device_del(rtd->dev);
+ device_unregister(rtd->dev);
rtd->dev_registered = 0;
}
@@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
platform_max = mc->platform_max;
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- uinfo->count = 1;
+ uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = platform_max - min;
@@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
+ unsigned int rreg = mc->rreg;
unsigned int shift = mc->shift;
int min = mc->min;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
unsigned int val, val_mask;
+ int ret;
val = ((ucontrol->value.integer.value[0] + min) & mask);
if (invert)
@@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
val_mask = mask << shift;
val = val << shift;
- return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+ ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
+ if (ret != 0)
+ return ret;
+
+ if (snd_soc_volsw_is_stereo(mc)) {
+ val = ((ucontrol->value.integer.value[1] + min) & mask);
+ if (invert)
+ val = max - val;
+ val_mask = mask << shift;
+ val = val << shift;
+
+ ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val);
+ }
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
@@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
+ unsigned int rreg = mc->rreg;
unsigned int shift = mc->shift;
int min = mc->min;
int max = mc->max;
@@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
ucontrol->value.integer.value[0] =
ucontrol->value.integer.value[0] - min;
+ if (snd_soc_volsw_is_stereo(mc)) {
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, rreg) >> shift) & mask;
+ if (invert)
+ ucontrol->value.integer.value[1] =
+ max - ucontrol->value.integer.value[1];
+ ucontrol->value.integer.value[1] =
+ ucontrol->value.integer.value[1] - min;
+ }
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d7711fce119b..cf191e6aebbe 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
(be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
continue;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index c183d34842ac..34b9bb7fe87c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -2181,10 +2181,6 @@ int snd_usbmidi_create(struct snd_card *card,
umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
break;
- case QUIRK_MIDI_MBOX2:
- umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops;
- err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
- break;
case QUIRK_MIDI_RAW_BYTES:
umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
/*
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index cdcf6b45e8a8..78e845ec65da 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -50,6 +50,28 @@
}
},
+{
+ /* Creative BT-D1 */
+ USB_DEVICE(0x041e, 0x0005),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x03,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .attributes = 0,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ }
+ }
+},
+
/* Creative/Toshiba Multimedia Center SB-0500 */
{
USB_DEVICE(0x041e, 0x3048),
@@ -2993,7 +3015,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
},
{
.ifnum = 6,
- .type = QUIRK_MIDI_MBOX2,
+ .type = QUIRK_MIDI_MIDIMAN,
.data = &(const struct snd_usb_midi_endpoint_info) {
.out_ep = 0x02,
.out_cables = 0x0001,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index f104c68fe1e0..acc12f004c23 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -306,7 +306,6 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_MIDI_YAMAHA] = create_any_midi_quirk,
[QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk,
[QUIRK_MIDI_NOVATION] = create_any_midi_quirk,
- [QUIRK_MIDI_MBOX2] = create_any_midi_quirk,
[QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk,
[QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
[QUIRK_MIDI_CME] = create_any_midi_quirk,
@@ -528,11 +527,11 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev)
#define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */
#define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */
-int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
+static int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
{
struct usb_host_config *config = dev->actconfig;
int err;
- u8 bootresponse;
+ u8 bootresponse[12];
int fwsize;
int count;
@@ -546,20 +545,20 @@ int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
snd_printd("usb-audio: Sending Digidesign Mbox 2 boot sequence...\n");
count = 0;
- bootresponse = MBOX2_BOOT_LOADING;
- while ((bootresponse == MBOX2_BOOT_LOADING) && (count < 10)) {
+ bootresponse[0] = MBOX2_BOOT_LOADING;
+ while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)) {
msleep(500); /* 0.5 second delay */
snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
/* Control magic - load onboard firmware */
0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012);
- if (bootresponse == MBOX2_BOOT_READY)
+ if (bootresponse[0] == MBOX2_BOOT_READY)
break;
snd_printd("usb-audio: device not ready, resending boot sequence...\n");
count++;
}
- if (bootresponse != MBOX2_BOOT_READY) {
- snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse);
+ if (bootresponse[0] != MBOX2_BOOT_READY) {
+ snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse[0]);
return -ENODEV;
}
@@ -660,7 +659,6 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
return 0; /* keep this altsetting */
}
-
static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip,
int iface, int altno)
{
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index a8172c119796..1ac3fd9cc5a6 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -76,7 +76,6 @@ enum quirk_type {
QUIRK_MIDI_YAMAHA,
QUIRK_MIDI_MIDIMAN,
QUIRK_MIDI_NOVATION,
- QUIRK_MIDI_MBOX2,
QUIRK_MIDI_RAW_BYTES,
QUIRK_MIDI_EMAGIC,
QUIRK_MIDI_CME,
generated by cgit v1.2.3 (git 2.39.0) at 2024-08-16 14:55:31 +0300