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-rw-r--r--sound/drivers/aloop.c6
-rw-r--r--sound/drivers/mpu401/mpu401_uart.c1
-rw-r--r--sound/pci/ac97/ac97_codec.c2
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.c2
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c9
-rw-r--r--sound/pci/hda/hda_intel.c6
-rw-r--r--sound/pci/hda/hda_proc.c2
-rw-r--r--sound/pci/hda/patch_realtek.c2
-rw-r--r--sound/pci/ice1712/prodigy_hifi.c3
-rw-r--r--sound/soc/codecs/tlv320aic3x.c4
-rw-r--r--sound/soc/codecs/tlv320aic3x.h1
-rw-r--r--sound/soc/codecs/wm8962.c3
-rw-r--r--sound/soc/codecs/wm8994.c2
-rw-r--r--sound/soc/codecs/wm9712.c21
-rw-r--r--sound/usb/clock.c3
-rw-r--r--sound/usb/mixer.c7
-rw-r--r--sound/usb/pcm.c3
-rw-r--r--sound/usb/quirks-table.h53
18 files changed, 117 insertions, 13 deletions
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index a0da775..5eab948 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -119,6 +119,7 @@ struct loopback_pcm {
unsigned int period_size_frac;
unsigned long last_jiffies;
struct timer_list timer;
+ spinlock_t timer_lock;
};
static struct platform_device *devices[SNDRV_CARDS];
@@ -169,6 +170,7 @@ static void loopback_timer_start(struct loopback_pcm *dpcm)
unsigned long tick;
unsigned int rate_shift = get_rate_shift(dpcm);
+ spin_lock(&dpcm->timer_lock);
if (rate_shift != dpcm->pcm_rate_shift) {
dpcm->pcm_rate_shift = rate_shift;
dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size);
@@ -181,12 +183,15 @@ static void loopback_timer_start(struct loopback_pcm *dpcm)
tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps;
dpcm->timer.expires = jiffies + tick;
add_timer(&dpcm->timer);
+ spin_unlock(&dpcm->timer_lock);
}
static inline void loopback_timer_stop(struct loopback_pcm *dpcm)
{
+ spin_lock(&dpcm->timer_lock);
del_timer(&dpcm->timer);
dpcm->timer.expires = 0;
+ spin_unlock(&dpcm->timer_lock);
}
#define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK)
@@ -658,6 +663,7 @@ static int loopback_open(struct snd_pcm_substream *substream)
dpcm->substream = substream;
setup_timer(&dpcm->timer, loopback_timer_function,
(unsigned long)dpcm);
+ spin_lock_init(&dpcm->timer_lock);
cable = loopback->cables[substream->number][dev];
if (!cable) {
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 2af0999..74f5a3d 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -554,6 +554,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
spin_lock_init(&mpu->output_lock);
spin_lock_init(&mpu->timer_lock);
mpu->hardware = hardware;
+ mpu->irq = -1;
if (! (info_flags & MPU401_INFO_INTEGRATED)) {
int res_size = hardware == MPU401_HW_PC98II ? 4 : 2;
mpu->res = request_region(port, res_size, "MPU401 UART");
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 7f4d619..11ccc23 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1271,6 +1271,8 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne
tmp.index = ac97->num;
kctl = snd_ctl_new1(&tmp, ac97);
}
+ if (!kctl)
+ return -ENOMEM;
if (reg >= AC97_PHONE && reg <= AC97_PCM)
set_tlv_db_scale(kctl, db_scale_5bit_12db_max);
else
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 64417a7..d8c670c 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip)
const struct firmware *fw;
int box_type, err;
- if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page))
+ if (snd_BUG_ON(!chip->comm_page))
return -EPERM;
/* See if the ASIC is present and working - only if the DSP is already loaded */
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 15f0161..0800bcc 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1415,6 +1415,15 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.ca0108_chip = 1,
.spk71 = 1,
.emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */
+ /* Tested by Maxim Kachur <mcdebugger@duganet.ru> 17th Oct 2012. */
+ /* This is MAEM8986, 0202 is MAEM8980 */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102,
+ .driver = "Audigy2", .name = "E-mu 1010 PCIe [MAEM8986]",
+ .id = "EMU1010",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .spk71 = 1,
+ .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */
/* Tested by James@superbug.co.uk 8th July 2005. */
/* This is MAEM8810, 0202 is MAEM8820 */
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 981b6fd..c5c9788 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -702,11 +702,13 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
{
struct azx *chip = bus->private_data;
unsigned long timeout;
+ unsigned long loopcounter;
int do_poll = 0;
again:
timeout = jiffies + msecs_to_jiffies(1000);
- for (;;) {
+
+ for (loopcounter = 0;; loopcounter++) {
if (chip->polling_mode || do_poll) {
spin_lock_irq(&chip->reg_lock);
azx_update_rirb(chip);
@@ -722,7 +724,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
}
if (time_after(jiffies, timeout))
break;
- if (bus->needs_damn_long_delay)
+ if (bus->needs_damn_long_delay || loopcounter > 3000)
msleep(2); /* temporary workaround */
else {
udelay(10);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 6fe944a..d0e5dec 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -406,7 +406,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
if (digi1 & AC_DIG1_EMPHASIS)
snd_iprintf(buffer, " Preemphasis");
if (digi1 & AC_DIG1_COPYRIGHT)
- snd_iprintf(buffer, " Copyright");
+ snd_iprintf(buffer, " Non-Copyright");
if (digi1 & AC_DIG1_NONAUDIO)
snd_iprintf(buffer, " Non-Audio");
if (digi1 & AC_DIG1_PROFESSIONAL)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 51412e1..8d288a7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -20132,6 +20132,8 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
+ { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
+ { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
.patch = patch_alc861 },
{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 764cc93..075d5aa 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
}
static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
{
@@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
.info = ak4396_dac_vol_info,
.get = ak4396_dac_vol_get,
.put = ak4396_dac_vol_put,
- .tlv = { .p = db_scale_wm_dac },
+ .tlv = { .p = ak4396_db_scale },
},
};
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 789453d..0b08bb7 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -949,9 +949,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
}
found:
- data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
- snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
- data | (pll_p << PLLP_SHIFT));
+ snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p);
snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG,
pll_r << PLLR_SHIFT);
snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT);
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 06a1978..16d9999 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -166,6 +166,7 @@
/* PLL registers bitfields */
#define PLLP_SHIFT 0
+#define PLLP_MASK 7
#define PLLQ_SHIFT 3
#define PLLR_SHIFT 0
#define PLLJ_SHIFT 2
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index c850e3d..f16f587 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2890,6 +2890,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
/* VMID 2*250k */
snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
WM8962_VMID_SEL_MASK, 0x100);
+
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ msleep(100);
break;
case SND_SOC_BIAS_OFF:
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 2194912..1f7616d 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2127,7 +2127,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- bclk_rate = params_rate(params) * 2;
+ bclk_rate = params_rate(params) * 4;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
bclk_rate *= 16;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 90117f8..520a20e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -144,7 +144,7 @@ SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1),
SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
-SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
@@ -270,7 +270,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]);
/* Mic select */
static const struct snd_kcontrol_new wm9712_mic_src_controls =
-SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
/* diff select */
static const struct snd_kcontrol_new wm9712_diff_sel_controls =
@@ -289,7 +289,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectl_controls),
SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectr_controls),
-SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
&wm9712_mic_src_controls),
SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
&wm9712_diff_sel_controls),
@@ -317,6 +319,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
SND_SOC_DAPM_OUTPUT("MONOOUT"),
SND_SOC_DAPM_OUTPUT("HPOUTL"),
@@ -377,6 +380,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = {
{"Mic PGA", NULL, "MIC1"},
{"Mic PGA", NULL, "MIC2"},
+ /* microphones */
+ {"Differential Mic", NULL, "MIC1"},
+ {"Differential Mic", NULL, "MIC2"},
+ {"Left Mic Select Source", "Mic 1", "MIC1"},
+ {"Left Mic Select Source", "Mic 2", "MIC2"},
+ {"Left Mic Select Source", "Stereo", "MIC1"},
+ {"Left Mic Select Source", "Differential", "Differential Mic"},
+ {"Right Mic Select Source", "Mic 1", "MIC1"},
+ {"Right Mic Select Source", "Mic 2", "MIC2"},
+ {"Right Mic Select Source", "Stereo", "MIC2"},
+ {"Right Mic Select Source", "Differential", "Differential Mic"},
+
/* left capture selector */
{"Left Capture Select", "Mic", "MIC1"},
{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 075195e..f0ff776 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -111,7 +111,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
return 0;
/* If a clock source can't tell us whether it's valid, we assume it is */
- if (!uac2_control_is_readable(cs_desc->bmControls, UAC2_CS_CONTROL_CLOCK_VALID))
+ if (!uac2_control_is_readable(cs_desc->bmControls,
+ UAC2_CS_CONTROL_CLOCK_VALID - 1))
return 1;
err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 0de7cbd..9363a8c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1246,6 +1246,13 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
/* disable non-functional volume control */
master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME);
break;
+ case USB_ID(0x1130, 0xf211):
+ snd_printk(KERN_INFO
+ "usbmixer: volume control quirk for Tenx TP6911 Audio Headset\n");
+ /* disable non-functional volume control */
+ channels = 0;
+ break;
+
}
if (channels > 0)
first_ch_bits = snd_usb_combine_bytes(bmaControls + csize, csize);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index b8dcbf4..506c0fa 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -670,6 +670,9 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime,
int count = 0, needs_knot = 0;
int err;
+ kfree(subs->rate_list.list);
+ subs->rate_list.list = NULL;
+
list_for_each_entry(fp, &subs->fmt_list, list) {
if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)
return 0;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 0b2ae8e..7ccffb2 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2581,6 +2581,59 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
+/* Microsoft XboxLive Headset/Xbox Communicator */
+{
+ USB_DEVICE(0x045e, 0x0283),
+ .bInterfaceClass = USB_CLASS_PER_INTERFACE,
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "Microsoft",
+ .product_name = "XboxLive Headset/Xbox Communicator",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = &(const struct snd_usb_audio_quirk[]) {
+ {
+ /* playback */
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels = 1,
+ .iface = 0,
+ .altsetting = 0,
+ .altset_idx = 0,
+ .attributes = 0,
+ .endpoint = 0x04,
+ .ep_attr = 0x05,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 22050,
+ .rate_max = 22050
+ }
+ },
+ {
+ /* capture */
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels = 1,
+ .iface = 1,
+ .altsetting = 0,
+ .altset_idx = 0,
+ .attributes = 0,
+ .endpoint = 0x85,
+ .ep_attr = 0x05,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 16000,
+ .rate_max = 16000
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+
{
/*
* Some USB MIDI devices don't have an audio control interface,
generated by cgit v1.1 at 2025-08-25 10:02:13 +0000