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-rw-r--r--sound/pci/hda/hda_generic.c6
-rw-r--r--sound/pci/hda/patch_realtek.c11
-rw-r--r--sound/soc/codecs/cs42l52.c5
-rw-r--r--sound/soc/codecs/dmic.c17
-rw-r--r--sound/soc/codecs/sgtl5000.c18
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c22
-rw-r--r--sound/soc/generic/simple-card.c2
-rw-r--r--sound/soc/soc-core.c76
-rw-r--r--sound/soc/soc-jack.c2
-rw-r--r--sound/soc/tegra/tegra30_i2s.c2
-rw-r--r--sound/usb/6fire/midi.c16
-rw-r--r--sound/usb/6fire/midi.h6
-rw-r--r--sound/usb/6fire/pcm.c41
-rw-r--r--sound/usb/6fire/pcm.h2
-rw-r--r--sound/usb/mixer.c1
-rw-r--r--sound/usb/quirks.c6
16 files changed, 121 insertions, 112 deletions
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8e77cbbad871..e3c7ba8d7582 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1,
}
#define nid_has_mute(codec, nid, dir) \
- check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE)
+ check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))
#define nid_has_volume(codec, nid, dir) \
check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS)
@@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
if (enable)
val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
}
- if (caps & AC_AMPCAP_MUTE) {
+ if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
if (!enable)
val |= HDA_AMP_MUTE;
}
@@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec,
{
unsigned int mask = 0xff;
- if (caps & AC_AMPCAP_MUTE) {
+ if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL))
mask &= ~0x80;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8bd226149868..f303cd898515 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1031,6 +1031,7 @@ enum {
ALC880_FIXUP_GPIO2,
ALC880_FIXUP_MEDION_RIM,
ALC880_FIXUP_LG,
+ ALC880_FIXUP_LG_LW25,
ALC880_FIXUP_W810,
ALC880_FIXUP_EAPD_COEF,
ALC880_FIXUP_TCL_S700,
@@ -1089,6 +1090,14 @@ static const struct hda_fixup alc880_fixups[] = {
{ }
}
},
+ [ALC880_FIXUP_LG_LW25] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x0181344f }, /* line-in */
+ { 0x1b, 0x0321403f }, /* headphone */
+ { }
+ }
+ },
[ALC880_FIXUP_W810] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -1341,6 +1350,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = {
SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG),
SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG),
SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG),
+ SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25),
SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700),
/* Below is the copied entries from alc880_quirks.c.
@@ -4329,6 +4339,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
+ SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 987f728718c5..be2ba1b6fe4a 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
+static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0);
+
static const unsigned int limiter_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("Beep Pitch", beep_pitch_enum),
SOC_ENUM("Beep on Time", beep_ontime_enum),
SOC_ENUM("Beep off Time", beep_offtime_enum),
- SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+ SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL,
+ 0, 0x07, 0x1f, beep_tlv),
SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 66967ba6f757..b2090b2a5e2d 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{"DMIC AIF", NULL, "DMic"},
};
-static int dmic_probe(struct snd_soc_codec *codec)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
- ARRAY_SIZE(dmic_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
- snd_soc_dapm_new_widgets(dapm);
-
- return 0;
-}
-
static struct snd_soc_codec_driver soc_dmic = {
- .probe = dmic_probe,
+ .dapm_widgets = dmic_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
};
static int dmic_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 6c8a9e7bee25..760e8bfeacaa 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
static int power_vag_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
+ const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
@@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
break;
case SND_SOC_DAPM_PRE_PMD:
- snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
- SGTL5000_VAG_POWERUP, 0);
- msleep(400);
+ /*
+ * Don't clear VAG_POWERUP, when both DAC and ADC are
+ * operational to prevent inadvertently starving the
+ * other one of them.
+ */
+ if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) &
+ mask) != mask) {
+ snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_VAG_POWERUP, 0);
+ msleep(400);
+ }
break;
default:
break;
@@ -388,7 +398,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),
SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",
SGTL5000_CHIP_ANA_ADC_CTRL,
- 8, 2, 0, capture_6db_attenuate),
+ 8, 1, 0, capture_6db_attenuate),
SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
SOC_DOUBLE_TLV("Headphone Playback Volume",
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 17df4e32feac..2ed57d4aa445 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate)
return -EINVAL;
}
-static int aic32x4_add_widgets(struct snd_soc_codec *codec)
-{
- snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
- ARRAY_SIZE(aic32x4_dapm_widgets));
-
- snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
- ARRAY_SIZE(aic32x4_dapm_routes));
-
- snd_soc_dapm_new_widgets(&codec->dapm);
- return 0;
-}
-
static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
@@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
}
aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
- ARRAY_SIZE(aic32x4_snd_controls));
- aic32x4_add_widgets(codec);
/*
* Workaround: for an unknown reason, the ADC needs to be powered up
@@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
.suspend = aic32x4_suspend,
.resume = aic32x4_resume,
.set_bias_level = aic32x4_set_bias_level,
+
+ .controls = aic32x4_snd_controls,
+ .num_controls = ARRAY_SIZE(aic32x4_snd_controls),
+ .dapm_widgets = aic32x4_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+ .dapm_routes = aic32x4_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
};
static int aic32x4_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6cf8355a8542..8c49147db84c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
static struct platform_driver asoc_simple_card = {
.driver = {
.name = "asoc-simple-card",
+ .owner = THIS_MODULE,
},
.probe = asoc_simple_card_probe,
.remove = asoc_simple_card_remove,
@@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = {
module_platform_driver(asoc_simple_card);
+MODULE_ALIAS("platform:asoc-simple-card");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("ASoC Simple Sound Card");
MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d82ee386eab5..d476f752e3f1 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -47,8 +47,6 @@
#define NAME_SIZE 32
-static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
-
#ifdef CONFIG_DEBUG_FS
struct dentry *snd_soc_debugfs_root;
EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
@@ -530,6 +528,15 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
}
#endif
+static void codec2codec_close_delayed_work(struct work_struct *work)
+{
+ /* Currently nothing to do for c2c links
+ * Since c2c links are internal nodes in the DAPM graph and
+ * don't interface with the outside world or application layer
+ * we don't have to do any special handling on close.
+ */
+}
+
#ifdef CONFIG_PM_SLEEP
/* powers down audio subsystem for suspend */
int snd_soc_suspend(struct device *dev)
@@ -1223,9 +1230,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
}
rtd->card = card;
- /* Make sure all DAPM widgets are instantiated */
- snd_soc_dapm_new_widgets(&codec->dapm);
-
/* machine controls, routes and widgets are not prefixed */
temp = codec->name_prefix;
codec->name_prefix = NULL;
@@ -1428,6 +1432,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
return ret;
}
} else {
+ INIT_DELAYED_WORK(&rtd->delayed_work,
+ codec2codec_close_delayed_work);
+
/* link the DAI widgets */
play_w = codec_dai->playback_widget;
capture_w = cpu_dai->capture_widget;
@@ -1718,8 +1725,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
card->num_dapm_routes);
- snd_soc_dapm_new_widgets(&card->dapm);
-
for (i = 0; i < card->num_links; i++) {
dai_link = &card->dai_link[i];
dai_fmt = dai_link->dai_fmt;
@@ -1798,12 +1803,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
}
- snd_soc_dapm_new_widgets(&card->dapm);
-
if (card->fully_routed)
list_for_each_entry(codec, &card->codec_dev_list, card_list)
snd_soc_dapm_auto_nc_codec_pins(codec);
+ snd_soc_dapm_new_widgets(&card->dapm);
+
ret = snd_card_register(card->snd_card);
if (ret < 0) {
dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
@@ -2541,59 +2546,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double);
/**
- * snd_soc_info_enum_ext - external enumerated single mixer info callback
- * @kcontrol: mixer control
- * @uinfo: control element information
- *
- * Callback to provide information about an external enumerated
- * single mixer.
- *
- * Returns 0 for success.
- */
-int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = e->max;
-
- if (uinfo->value.enumerated.item > e->max - 1)
- uinfo->value.enumerated.item = e->max - 1;
- strcpy(uinfo->value.enumerated.name,
- e->texts[uinfo->value.enumerated.item]);
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
-
-/**
- * snd_soc_info_volsw_ext - external single mixer info callback
- * @kcontrol: mixer control
- * @uinfo: control element information
- *
- * Callback to provide information about a single external mixer control.
- *
- * Returns 0 for success.
- */
-int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- int max = kcontrol->private_value;
-
- if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- else
- uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = max;
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
-
-/**
* snd_soc_info_volsw - single mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7aa26b5178aa..71358e3b54d9 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
list_add(&(pins[i].list), &jack->pins);
}
- snd_soc_dapm_new_widgets(&jack->codec->card->dapm);
-
/* Update to reflect the last reported status; canned jack
* implementations are likely to set their state before the
* card has an opportunity to associate pins.
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index d04146cad61f..47565fd04505 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
reg = TEGRA30_I2S_CIF_RX_CTRL;
} else {
val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
- reg = TEGRA30_I2S_CIF_RX_CTRL;
+ reg = TEGRA30_I2S_CIF_TX_CTRL;
}
regmap_write(i2s->regmap, reg, val);
diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c
index 26722423330d..f3dd7266c391 100644
--- a/sound/usb/6fire/midi.c
+++ b/sound/usb/6fire/midi.c
@@ -19,6 +19,10 @@
#include "chip.h"
#include "comm.h"
+enum {
+ MIDI_BUFSIZE = 64
+};
+
static void usb6fire_midi_out_handler(struct urb *urb)
{
struct midi_runtime *rt = urb->context;
@@ -156,6 +160,12 @@ int usb6fire_midi_init(struct sfire_chip *chip)
if (!rt)
return -ENOMEM;
+ rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL);
+ if (!rt->out_buffer) {
+ kfree(rt);
+ return -ENOMEM;
+ }
+
rt->chip = chip;
rt->in_received = usb6fire_midi_in_received;
rt->out_buffer[0] = 0x80; /* 'send midi' command */
@@ -169,6 +179,7 @@ int usb6fire_midi_init(struct sfire_chip *chip)
ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance);
if (ret < 0) {
+ kfree(rt->out_buffer);
kfree(rt);
snd_printk(KERN_ERR PREFIX "unable to create midi.\n");
return ret;
@@ -197,6 +208,9 @@ void usb6fire_midi_abort(struct sfire_chip *chip)
void usb6fire_midi_destroy(struct sfire_chip *chip)
{
- kfree(chip->midi);
+ struct midi_runtime *rt = chip->midi;
+
+ kfree(rt->out_buffer);
+ kfree(rt);
chip->midi = NULL;
}
diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h
index c321006e5430..84851b9f5559 100644
--- a/sound/usb/6fire/midi.h
+++ b/sound/usb/6fire/midi.h
@@ -16,10 +16,6 @@
#include "common.h"
-enum {
- MIDI_BUFSIZE = 64
-};
-
struct midi_runtime {
struct sfire_chip *chip;
struct snd_rawmidi *instance;
@@ -32,7 +28,7 @@ struct midi_runtime {
struct snd_rawmidi_substream *out;
struct urb out_urb;
u8 out_serial; /* serial number of out packet */
- u8 out_buffer[MIDI_BUFSIZE];
+ u8 *out_buffer;
int buffer_offset;
void (*in_received)(struct midi_runtime *rt, u8 *data, int length);
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index 3d2551cc10f2..b5eb97fdc842 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -582,6 +582,33 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb,
urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB;
}
+static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt)
+{
+ int i;
+
+ for (i = 0; i < PCM_N_URBS; i++) {
+ rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB
+ * PCM_MAX_PACKET_SIZE, GFP_KERNEL);
+ if (!rt->out_urbs[i].buffer)
+ return -ENOMEM;
+ rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB
+ * PCM_MAX_PACKET_SIZE, GFP_KERNEL);
+ if (!rt->in_urbs[i].buffer)
+ return -ENOMEM;
+ }
+ return 0;
+}
+
+static void usb6fire_pcm_buffers_destroy(struct pcm_runtime *rt)
+{
+ int i;
+
+ for (i = 0; i < PCM_N_URBS; i++) {
+ kfree(rt->out_urbs[i].buffer);
+ kfree(rt->in_urbs[i].buffer);
+ }
+}
+
int usb6fire_pcm_init(struct sfire_chip *chip)
{
int i;
@@ -593,6 +620,13 @@ int usb6fire_pcm_init(struct sfire_chip *chip)
if (!rt)
return -ENOMEM;
+ ret = usb6fire_pcm_buffers_init(rt);
+ if (ret) {
+ usb6fire_pcm_buffers_destroy(rt);
+ kfree(rt);
+ return ret;
+ }
+
rt->chip = chip;
rt->stream_state = STREAM_DISABLED;
rt->rate = ARRAY_SIZE(rates);
@@ -614,6 +648,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip)
ret = snd_pcm_new(chip->card, "DMX6FireUSB", 0, 1, 1, &pcm);
if (ret < 0) {
+ usb6fire_pcm_buffers_destroy(rt);
kfree(rt);
snd_printk(KERN_ERR PREFIX "cannot create pcm instance.\n");
return ret;
@@ -625,6 +660,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip)
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops);
if (ret) {
+ usb6fire_pcm_buffers_destroy(rt);
kfree(rt);
snd_printk(KERN_ERR PREFIX
"error preallocating pcm buffers.\n");
@@ -669,6 +705,9 @@ void usb6fire_pcm_abort(struct sfire_chip *chip)
void usb6fire_pcm_destroy(struct sfire_chip *chip)
{
- kfree(chip->pcm);
+ struct pcm_runtime *rt = chip->pcm;
+
+ usb6fire_pcm_buffers_destroy(rt);
+ kfree(rt);
chip->pcm = NULL;
}
diff --git a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h
index 9b01133ee3fe..f5779d6182c6 100644
--- a/sound/usb/6fire/pcm.h
+++ b/sound/usb/6fire/pcm.h
@@ -32,7 +32,7 @@ struct pcm_urb {
struct urb instance;
struct usb_iso_packet_descriptor packets[PCM_N_PACKETS_PER_URB];
/* END DO NOT SEPARATE */
- u8 buffer[PCM_N_PACKETS_PER_URB * PCM_MAX_PACKET_SIZE];
+ u8 *buffer;
struct pcm_urb *peer;
};
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index d5438083fd6a..95558ef4a7a0 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -888,6 +888,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */
case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */
+ case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */
case USB_ID(0x046d, 0x0991):
/* Most audio usb devices lie about volume resolution.
* Most Logitech webcams have res = 384.
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 1bc45e71f1fe..0df9ede99dfd 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -319,19 +319,19 @@ static int create_auto_midi_quirk(struct snd_usb_audio *chip,
if (altsd->bNumEndpoints < 1)
return -ENODEV;
epd = get_endpoint(alts, 0);
- if (!usb_endpoint_xfer_bulk(epd) ||
+ if (!usb_endpoint_xfer_bulk(epd) &&
!usb_endpoint_xfer_int(epd))
return -ENODEV;
switch (USB_ID_VENDOR(chip->usb_id)) {
case 0x0499: /* Yamaha */
err = create_yamaha_midi_quirk(chip, iface, driver, alts);
- if (err < 0 && err != -ENODEV)
+ if (err != -ENODEV)
return err;
break;
case 0x0582: /* Roland */
err = create_roland_midi_quirk(chip, iface, driver, alts);
- if (err < 0 && err != -ENODEV)
+ if (err != -ENODEV)
return err;
break;
}
generated by cgit v1.2.3 (git 2.39.0) at 2025-01-18 08:35:02 +0300