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authorRussell King <rmk@dyn-67.arm.linux.org.uk>2009-09-12 12:02:26 +0100
committerRussell King <rmk+kernel@arm.linux.org.uk>2009-09-12 12:02:26 +0100
commitddd559b13f6d2fe3ad68c4b3f5235fd3c2eae4e3 (patch)
treed827bca3fc825a0ac33efbcd493713be40fcc812 /sound
parentcf7a2b4fb6a9b86779930a0a123b0df41aa9208f (diff)
parentf17a1f06d2fa93f4825be572622eb02c4894db4e (diff)
downloadmrst-s0i3-test-ddd559b13f6d2fe3ad68c4b3f5235fd3c2eae4e3.tar.gz
mrst-s0i3-test-ddd559b13f6d2fe3ad68c4b3f5235fd3c2eae4e3.tar.xz
mrst-s0i3-test-ddd559b13f6d2fe3ad68c4b3f5235fd3c2eae4e3.zip
Merge branch 'devel-stable' into devel
Conflicts: MAINTAINERS arch/arm/mm/fault.c
Diffstat (limited to 'sound')
-rw-r--r--sound/aoa/core/gpio-pmf.c4
-rw-r--r--sound/arm/pxa2xx-pcm-lib.c2
-rw-r--r--sound/core/pcm_lib.c36
-rw-r--r--sound/core/seq/Makefile7
-rw-r--r--sound/isa/gus/gus_pcm.c4
-rw-r--r--sound/oss/aedsp16.c9
-rw-r--r--sound/oss/mpu401.c2
-rw-r--r--sound/pci/ca0106/ca0106_main.c4
-rw-r--r--sound/pci/ctxfi/ctamixer.c14
-rw-r--r--sound/pci/ctxfi/ctdaio.c4
-rw-r--r--sound/pci/ctxfi/ctsrc.c7
-rw-r--r--sound/pci/hda/hda_codec.c6
-rw-r--r--sound/pci/hda/hda_eld.c4
-rw-r--r--sound/pci/hda/patch_analog.c2
-rw-r--r--sound/pci/hda/patch_realtek.c43
-rw-r--r--sound/pci/hda/patch_sigmatel.c11
-rw-r--r--sound/pci/riptide/riptide.c7
-rw-r--r--sound/soc/codecs/tlv320aic3x.c11
-rw-r--r--sound/soc/codecs/wm8988.c4
-rw-r--r--sound/soc/s3c24xx/s3c24xx-ac97.h6
-rw-r--r--sound/usb/Kconfig1
-rw-r--r--sound/usb/caiaq/audio.c1
-rw-r--r--sound/usb/caiaq/device.c8
-rw-r--r--sound/usb/caiaq/device.h1
-rw-r--r--sound/usb/usbaudio.c14
-rw-r--r--sound/usb/usbmixer.c25
26 files changed, 164 insertions, 73 deletions
diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 5ca2220eac7..1dd0c28d1fb 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -182,6 +182,10 @@ static int pmf_set_notify(struct gpio_runtime *rt,
if (!old && notify) {
irq_client = kzalloc(sizeof(struct pmf_irq_client),
GFP_KERNEL);
+ if (!irq_client) {
+ err = -ENOMEM;
+ goto out_unlock;
+ }
irq_client->data = notif;
irq_client->handler = pmf_handle_notify_irq;
irq_client->owner = THIS_MODULE;
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 108b643229b..6205f37d547 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -75,7 +75,7 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
- if (rtd && rtd->params)
+ if (rtd && rtd->params && rtd->params->drcmr)
*rtd->params->drcmr = 0;
snd_pcm_set_runtime_buffer(substream, NULL);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 333e4dd2945..72cfd47af6b 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -233,6 +233,18 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
xrun(substream);
return -EPIPE;
}
+ if (xrun_debug(substream, 8)) {
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, "
+ "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+ name, (unsigned int)pos,
+ (unsigned int)runtime->period_size,
+ (unsigned int)runtime->buffer_size,
+ (unsigned long)old_hw_ptr,
+ (unsigned long)runtime->hw_ptr_base,
+ (unsigned long)runtime->hw_ptr_interrupt);
+ }
hw_base = runtime->hw_ptr_base;
new_hw_ptr = hw_base + pos;
hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;
@@ -244,18 +256,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
delta = new_hw_ptr - hw_ptr_interrupt;
}
if (delta < 0) {
- delta += runtime->buffer_size;
+ if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr)
+ delta += runtime->buffer_size;
if (delta < 0) {
hw_ptr_error(substream,
"Unexpected hw_pointer value "
"(stream=%i, pos=%ld, intr_ptr=%ld)\n",
substream->stream, (long)pos,
(long)hw_ptr_interrupt);
+#if 1
+ /* simply skipping the hwptr update seems more
+ * robust in some cases, e.g. on VMware with
+ * inaccurate timer source
+ */
+ return 0; /* skip this update */
+#else
/* rebase to interrupt position */
hw_base = new_hw_ptr = hw_ptr_interrupt;
/* align hw_base to buffer_size */
hw_base -= hw_base % runtime->buffer_size;
delta = 0;
+#endif
} else {
hw_base += runtime->buffer_size;
if (hw_base >= runtime->boundary)
@@ -344,6 +365,19 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
xrun(substream);
return -EPIPE;
}
+ if (xrun_debug(substream, 16)) {
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, "
+ "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+ name, (unsigned int)pos,
+ (unsigned int)runtime->period_size,
+ (unsigned int)runtime->buffer_size,
+ (unsigned long)old_hw_ptr,
+ (unsigned long)runtime->hw_ptr_base,
+ (unsigned long)runtime->hw_ptr_interrupt);
+ }
+
hw_base = runtime->hw_ptr_base;
new_hw_ptr = hw_base + pos;
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 1bcb360330e..941f64a853e 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -3,10 +3,6 @@
# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
#
-ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
- obj-$(CONFIG_SND_SEQUENCER) += oss/
-endif
-
snd-seq-device-objs := seq_device.o
snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
seq_fifo.o seq_prioq.o seq_timer.o \
@@ -19,7 +15,8 @@ snd-seq-virmidi-objs := seq_virmidi.o
obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o
ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
-obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+ obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+ obj-$(CONFIG_SND_SEQUENCER) += oss/
endif
obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index edb11eefdfe..2dcf45bf729 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -795,13 +795,13 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE))
continue;
/* load real volume - better precision */
- spin_lock_irqsave(&gus->reg_lock, flags);
+ spin_lock(&gus->reg_lock);
snd_gf1_select_voice(gus, pvoice->number);
snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol);
pcmp->final_volume = 1;
- spin_unlock_irqrestore(&gus->reg_lock, flags);
+ spin_unlock(&gus->reg_lock);
}
spin_unlock_irqrestore(&gus->voice_alloc, flags);
return change;
diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c
index 3ee9900ffd7..35b5912cf3f 100644
--- a/sound/oss/aedsp16.c
+++ b/sound/oss/aedsp16.c
@@ -325,8 +325,9 @@
/*
* Size of character arrays that store name and version of sound card
*/
-#define CARDNAMELEN 15 /* Size of the card's name in chars */
-#define CARDVERLEN 2 /* Size of the card's version in chars */
+#define CARDNAMELEN 15 /* Size of the card's name in chars */
+#define CARDVERLEN 10 /* Size of the card's version in chars */
+#define CARDVERDIGITS 2 /* Number of digits in the version */
#if defined(CONFIG_SC6600)
/*
@@ -410,7 +411,7 @@
static int soft_cfg __initdata = 0; /* bitmapped config */
static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */
-static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver:
+static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver:
hi->ver[0] lo->ver[1] */
#if defined(CONFIG_SC6600)
@@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port)
* string is finished.
*/
ver[len++] = ret;
- } while (len < CARDVERLEN);
+ } while (len < CARDVERDIGITS);
sprintf(DSPVersion, "%d.%d", ver[0], ver[1]);
DBG(("success.\n"));
diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c
index 1b2316f35b1..734b8f9e2f7 100644
--- a/sound/oss/mpu401.c
+++ b/sound/oss/mpu401.c
@@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner)
sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name);
else
sprintf(mpu_synth_info[m].name,
- "MPU-401 %d.%d%c Midi interface #%d",
+ "MPU-401 %d.%d%c MIDI #%d",
(int) (devc->version & 0xf0) >> 4,
devc->version & 0x0f,
revision_char,
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index f24bf1ecb36..15e4138bce1 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -325,9 +325,9 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
.rate_max = 192000,
.channels_min = 2,
.channels_max = 2,
- .buffer_bytes_max = ((65536 - 64) * 8),
+ .buffer_bytes_max = 65536 - 128,
.period_bytes_min = 64,
- .period_bytes_max = (65536 - 64),
+ .period_bytes_max = 32768 - 64,
.periods_min = 2,
.periods_max = 2,
.fifo_size = 0,
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index a1db51b3ead..a7f4a671f7b 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr,
/* Allocate mem for amixer resource */
amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
- if (NULL == amixer) {
- err = -ENOMEM;
- return err;
- }
+ if (!amixer)
+ return -ENOMEM;
/* Check whether there are sufficient
* amixer resources to meet request. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
@@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr,
/* Allocate mem for sum resource */
sum = kzalloc(sizeof(*sum), GFP_KERNEL);
- if (NULL == sum) {
- err = -ENOMEM;
- return err;
- }
+ if (!sum)
+ return -ENOMEM;
/* Check whether there are sufficient sum resources to meet request. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 082e35c08c0..deb6cfa7360 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -57,9 +57,9 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = {
struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
[LINEO1] = {.left = 0x40, .right = 0x41},
- [LINEO2] = {.left = 0x70, .right = 0x71},
+ [LINEO2] = {.left = 0x60, .right = 0x61},
[LINEO3] = {.left = 0x50, .right = 0x51},
- [LINEO4] = {.left = 0x60, .right = 0x61},
+ [LINEO4] = {.left = 0x70, .right = 0x71},
[LINEIM] = {.left = 0x45, .right = 0xc5},
[SPDIFOO] = {.left = 0x00, .right = 0x01},
[SPDIFIO] = {.left = 0x05, .right = 0x85},
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index e1c145d8b70..df43a5cd393 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr,
/* Allocate mem for SRCIMP resource */
srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
- if (NULL == srcimp) {
- err = -ENOMEM;
- return err;
- }
+ if (!srcimp)
+ return -ENOMEM;
/* Check whether there are sufficient SRCIMP resources. */
+ err = 0;
spin_lock_irqsave(&mgr->mgr_lock, flags);
for (i = 0; i < desc->msr; i++) {
err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d255de6be..88480c0c58a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -332,6 +332,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
AC_VERB_GET_CONNECT_LIST, i);
range_val = !!(parm & (1 << (shift-1))); /* ranges */
val = parm & mask;
+ if (val == 0) {
+ snd_printk(KERN_WARNING "hda_codec: "
+ "invalid CONNECT_LIST verb %x[%i]:%x\n",
+ nid, i, parm);
+ return 0;
+ }
parm >>= shift;
if (range_val) {
/* ranges between the previous and this one */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index fcad5ec3177..9446a5abea1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -508,7 +508,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
char name[64];
char *sname;
long long val;
- int n;
+ unsigned int n;
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%s %llx", name, &val) != 2)
@@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
sname++;
n = 10 * n + name[4] - '0';
}
- if (n < 0 || n > 31) /* double the CEA limit */
+ if (n >= ELD_MAX_SAD)
continue;
if (!strcmp(sname, "_coding_type"))
e->sad[n].format = val;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index be7d25fa7f3..3da85caf8af 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3754,7 +3754,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
int mute = (!ucontrol->value.integer.value[0] &&
!ucontrol->value.integer.value[1]);
/* toggle GPIO1 according to the mute state */
- snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
mute ? 0x02 : 0x0);
return ret;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bbb9b42e260..b95df5d5dcc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,13 +275,13 @@ struct alc_spec {
*/
unsigned int num_init_verbs;
- char stream_name_analog[16]; /* analog PCM stream */
+ char stream_name_analog[32]; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
struct hda_pcm_stream *stream_analog_alt_playback;
struct hda_pcm_stream *stream_analog_alt_capture;
- char stream_name_digital[16]; /* digital PCM stream */
+ char stream_name_digital[32]; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
@@ -4505,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
&dig_nid, 1);
if (err < 0)
continue;
+ if (dig_nid > 0x7f) {
+ printk(KERN_ERR "alc880_auto: invalid dig_nid "
+ "connection 0x%x for NID 0x%x\n", dig_nid,
+ spec->autocfg.dig_out_pins[i]);
+ continue;
+ }
if (!i)
spec->multiout.dig_out_nid = dig_nid;
else {
@@ -10625,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
alc262_lenovo_3000_automute(codec, 1);
}
+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx, long *valp)
+{
+ int i, change = 0;
+
+ for (i = 0; i < 2; i++, valp++)
+ change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+ return change;
+}
+
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -10633,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
+ change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_fujitsu_automute(codec, 0);
return change;
@@ -10674,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp ? 0 : HDA_AMP_MUTE);
-
+ change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
if (change)
alc262_lenovo_3000_automute(codec, 0);
return change;
@@ -11848,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change;
- change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[0] ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[1] ? 0 : HDA_AMP_MUTE);
+ change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
if (change)
alc268_acer_automute(codec, 0);
return change;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 41b5b3a18c1..5383d8cff88 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1809,6 +1809,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
"Dell Studio 1537", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0,
"Dell Studio 17", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
+ "Dell Studio 1555", STAC_DELL_M6_DMIC),
{} /* terminator */
};
@@ -2378,6 +2380,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
"Dell Vostro 1500", STAC_9205_DELL_M42),
/* Gateway */
+ SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
{} /* terminator */
};
@@ -4065,7 +4068,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
jack->nid = nid;
jack->type = type;
- sprintf(name, "%s at %s %s Jack",
+ snprintf(name, sizeof(name), "%s at %s %s Jack",
snd_hda_get_jack_type(def_conf),
snd_hda_get_jack_connectivity(def_conf),
snd_hda_get_jack_location(def_conf));
@@ -5854,6 +5857,8 @@ static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
};
static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+ SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0,
+ "Sony VAIO F/S", STAC_9872_VAIO),
{} /* terminator */
};
@@ -5866,6 +5871,8 @@ static int patch_stac9872(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+ spec->pin_nids = stac9872_pin_nids;
spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
stac9872_models,
@@ -5877,8 +5884,6 @@ static int patch_stac9872(struct hda_codec *codec)
stac92xx_set_config_regs(codec,
stac9872_brd_tbl[spec->board_config]);
- spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
- spec->pin_nids = stac9872_pin_nids;
spec->multiout.dac_nids = spec->dac_nids;
spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
spec->adc_nids = stac9872_adc_nids;
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 235a71e5ac8..b5ca02e2038 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2197,9 +2197,12 @@ static int __init alsa_card_riptide_init(void)
if (err < 0)
return err;
#if defined(SUPPORT_JOYSTICK)
- pci_register_driver(&joystick_driver);
+ err = pci_register_driver(&joystick_driver);
+ /* On failure unregister formerly registered audio driver */
+ if (err < 0)
+ pci_unregister_driver(&driver);
#endif
- return 0;
+ return err;
}
static void __exit alsa_card_riptide_exit(void)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index ab099f48248..cb0d1bf34b5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
u16 pll_d = 1;
+ u8 reg;
/* select data word length */
data =
@@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
pll_q &= 0xf;
aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
- } else
+ /* disable PLL if it is bypassed */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE);
+
+ } else {
aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+ /* enable PLL when it is used */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE);
+ }
/* Route Left DAC to left channel input and
* right DAC to right channel input */
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index c05f71803aa..8c0fdf84aac 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -1037,14 +1037,14 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi)
codec->control_data = spi;
codec->dev = &spi->dev;
- spi->dev.driver_data = wm8988;
+ dev_set_drvdata(&spi->dev, wm8988);
return wm8988_register(wm8988);
}
static int __devexit wm8988_spi_remove(struct spi_device *spi)
{
- struct wm8988_priv *wm8988 = spi->dev.driver_data;
+ struct wm8988_priv *wm8988 = dev_get_drvdata(&spi->dev);
wm8988_unregister(wm8988);
diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h
index a96dcadf28b..e96f941a810 100644
--- a/sound/soc/s3c24xx/s3c24xx-ac97.h
+++ b/sound/soc/s3c24xx/s3c24xx-ac97.h
@@ -20,12 +20,6 @@
#define AC_CMD_ADDR(x) (x << 16)
#define AC_CMD_DATA(x) (x & 0xffff)
-#ifdef CONFIG_CPU_S3C2440
-#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97
-#else
-#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97
-#endif
-
extern struct snd_soc_dai s3c2443_ac97_dai[];
#endif /*S3C24XXAC97_H_*/
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 523aec188cc..73525c048e7 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -48,6 +48,7 @@ config SND_USB_CAIAQ
* Native Instruments Kore Controller
* Native Instruments Kore Controller 2
* Native Instruments Audio Kontrol 1
+ * Native Instruments Audio 2 DJ
* Native Instruments Audio 4 DJ
* Native Instruments Audio 8 DJ
* Native Instruments Guitar Rig Session I/O
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 8f9b60c5d74..121af0644fd 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
dev->samplerates |= SNDRV_PCM_RATE_192000;
/* fall thru */
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
dev->samplerates |= SNDRV_PCM_RATE_88200;
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index de38108f0b2..83e6c1312d4 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,13 +35,14 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
"{Native Instruments, Kore Controller},"
"{Native Instruments, Kore Controller 2},"
"{Native Instruments, Audio Kontrol 1},"
+ "{Native Instruments, Audio 2 DJ},"
"{Native Instruments, Audio 4 DJ},"
"{Native Instruments, Audio 8 DJ},"
"{Native Instruments, Session I/O},"
@@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = {
.idVendor = USB_VID_NATIVEINSTRUMENTS,
.idProduct = USB_PID_AUDIO4DJ
},
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_AUDIO2DJ
+ },
{ /* terminator */ }
};
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index ece73514854..44e3edf88be 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -10,6 +10,7 @@
#define USB_PID_KORECONTROLLER 0x4711
#define USB_PID_KORECONTROLLER2 0x4712
#define USB_PID_AK1 0x0815
+#define USB_PID_AUDIO2DJ 0x041c
#define USB_PID_AUDIO4DJ 0x0839
#define USB_PID_AUDIO8DJ 0x1978
#define USB_PID_SESSIONIO 0x1915
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index c7b902358b7..44b9cdc8a83 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2661,7 +2661,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
struct usb_interface_descriptor *altsd;
int i, altno, err, stream;
int format;
- struct audioformat *fp;
+ struct audioformat *fp = NULL;
unsigned char *fmt, *csep;
int num;
@@ -2734,6 +2734,18 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
continue;
}
+ /*
+ * Blue Microphones workaround: The last altsetting is identical
+ * with the previous one, except for a larger packet size, but
+ * is actually a mislabeled two-channel setting; ignore it.
+ */
+ 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 &&
+ le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
+ fp->maxpacksize * 2)
+ continue;
+
csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
/* Creamware Noah has this descriptor after the 2nd endpoint */
if (!csep && altsd->bNumEndpoints >= 2)
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 4bd3a7a0edc..ec9cdf98692 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -990,20 +990,35 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
break;
}
- /* quirk for UDA1321/N101 */
- /* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */
- /* is not very clear from datasheets */
- /* I hope that the min value is -15360 for newer firmware --jk */
+ /* volume control quirks */
switch (state->chip->usb_id) {
case USB_ID(0x0471, 0x0101):
case USB_ID(0x0471, 0x0104):
case USB_ID(0x0471, 0x0105):
case USB_ID(0x0672, 0x1041):
+ /* quirk for UDA1321/N101.
+ * note that detection between firmware 2.1.1.7 (N101)
+ * and later 2.1.1.21 is not very clear from datasheets.
+ * I hope that the min value is -15360 for newer firmware --jk
+ */
if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
cval->min == -15616) {
- snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n");
+ snd_printk(KERN_INFO
+ "set volume quirk for UDA1321/N101 chip\n");
cval->max = -256;
}
+ break;
+
+ case USB_ID(0x046d, 0x09a4):
+ if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+ snd_printk(KERN_INFO
+ "set volume quirk for QuickCam E3500\n");
+ cval->min = 6080;
+ cval->max = 8768;
+ cval->res = 192;
+ }
+ break;
+
}
snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",