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-rw-r--r--sound/core/pcm.c5
-rw-r--r--sound/core/rawmidi.c42
-rw-r--r--sound/drivers/dummy.c4
-rw-r--r--sound/drivers/pcsp/pcsp_lib.c65
-rw-r--r--sound/drivers/pcsp/pcsp_mixer.c2
-rw-r--r--sound/oss/dmasound/dmasound_core.c4
-rw-r--r--sound/oss/sb_common.c4
-rw-r--r--sound/oss/sb_ess.c2
-rw-r--r--sound/parisc/harmony.c6
-rw-r--r--sound/pci/Kconfig1
-rw-r--r--sound/pci/ali5451/ali5451.c2
-rw-r--r--sound/pci/hda/hda_intel.c13
-rw-r--r--sound/pci/hda/patch_conexant.c16
-rw-r--r--sound/pci/hda/patch_realtek.c18
-rw-r--r--sound/pci/hda/patch_sigmatel.c68
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/pci/via82xx.c59
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf.c21
-rw-r--r--sound/pcmcia/vx/vxpocket.c21
-rw-r--r--sound/sh/aica.c1
-rw-r--r--sound/soc/omap/Kconfig13
-rw-r--r--sound/soc/omap/omap-pcm.c8
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c17
-rw-r--r--sound/soc/s3c24xx/s3c64xx-i2s.c2
-rw-r--r--sound/soc/soc-core.c11
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/usb/caiaq/audio.c16
-rw-r--r--sound/usb/caiaq/device.c2
-rw-r--r--sound/usb/usbaudio.h2
29 files changed, 314 insertions, 119 deletions
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 0c1440121c2..c69c60b2a48 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -953,11 +953,12 @@ static int snd_pcm_dev_register(struct snd_device *device)
struct snd_pcm_substream *substream;
struct snd_pcm_notify *notify;
char str[16];
- struct snd_pcm *pcm = device->device_data;
+ struct snd_pcm *pcm;
struct device *dev;
- if (snd_BUG_ON(!pcm || !device))
+ if (snd_BUG_ON(!device || !device->device_data))
return -ENXIO;
+ pcm = device->device_data;
mutex_lock(&register_mutex);
err = snd_pcm_add(pcm);
if (err) {
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index c0adc14c91f..70d6f25ba52 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -248,7 +248,8 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice,
list_for_each_entry(substream, &s->substreams, list) {
if (substream->opened) {
if (stream == SNDRV_RAWMIDI_STREAM_INPUT ||
- !(mode & SNDRV_RAWMIDI_LFLG_APPEND))
+ !(mode & SNDRV_RAWMIDI_LFLG_APPEND) ||
+ !substream->append)
continue;
}
if (subdevice < 0 || subdevice == substream->number) {
@@ -266,17 +267,21 @@ static int open_substream(struct snd_rawmidi *rmidi,
{
int err;
- err = snd_rawmidi_runtime_create(substream);
- if (err < 0)
- return err;
- err = substream->ops->open(substream);
- if (err < 0)
- return err;
- substream->opened = 1;
- if (substream->use_count++ == 0)
+ if (substream->use_count == 0) {
+ err = snd_rawmidi_runtime_create(substream);
+ if (err < 0)
+ return err;
+ err = substream->ops->open(substream);
+ if (err < 0) {
+ snd_rawmidi_runtime_free(substream);
+ return err;
+ }
+ substream->opened = 1;
substream->active_sensing = 0;
- if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
- substream->append = 1;
+ if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
+ substream->append = 1;
+ }
+ substream->use_count++;
rmidi->streams[substream->stream].substream_opened++;
return 0;
}
@@ -297,27 +302,27 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode,
SNDRV_RAWMIDI_STREAM_INPUT,
mode, &sinput);
if (err < 0)
- goto __error;
+ return err;
}
if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
err = assign_substream(rmidi, subdevice,
SNDRV_RAWMIDI_STREAM_OUTPUT,
mode, &soutput);
if (err < 0)
- goto __error;
+ return err;
}
if (sinput) {
err = open_substream(rmidi, sinput, mode);
if (err < 0)
- goto __error;
+ return err;
}
if (soutput) {
err = open_substream(rmidi, soutput, mode);
if (err < 0) {
if (sinput)
close_substream(rmidi, sinput, 0);
- goto __error;
+ return err;
}
}
@@ -325,13 +330,6 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode,
rfile->input = sinput;
rfile->output = soutput;
return 0;
-
- __error:
- if (sinput && sinput->runtime)
- snd_rawmidi_runtime_free(sinput);
- if (soutput && soutput->runtime)
- snd_rawmidi_runtime_free(soutput);
- return err;
}
/* called from sound/core/seq/seq_midi.c */
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 6ba066c41d2..252e04ce602 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -165,7 +165,7 @@ MODULE_PARM_DESC(enable, "Enable this dummy soundcard.");
module_param_array(pcm_devs, int, NULL, 0444);
MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver.");
module_param_array(pcm_substreams, int, NULL, 0444);
-MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver.");
+MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver.");
//module_param_array(midi_devs, int, NULL, 0444);
//MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver.");
module_param(fake_buffer, bool, 0444);
@@ -808,8 +808,6 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy)
unsigned int idx;
int err;
- if (snd_BUG_ON(!dummy))
- return -EINVAL;
spin_lock_init(&dummy->mixer_lock);
strcpy(card->mixername, "Dummy Mixer");
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 84cc2658c05..e1145ac6e90 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -39,25 +39,20 @@ static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0);
/* write the port and returns the next expire time in ns;
* called at the trigger-start and in hrtimer callback
*/
-static unsigned long pcsp_timer_update(struct hrtimer *handle)
+static u64 pcsp_timer_update(struct snd_pcsp *chip)
{
unsigned char timer_cnt, val;
u64 ns;
struct snd_pcm_substream *substream;
struct snd_pcm_runtime *runtime;
- struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
unsigned long flags;
if (chip->thalf) {
outb(chip->val61, 0x61);
chip->thalf = 0;
- if (!atomic_read(&chip->timer_active))
- return 0;
return chip->ns_rem;
}
- if (!atomic_read(&chip->timer_active))
- return 0;
substream = chip->playback_substream;
if (!substream)
return 0;
@@ -88,24 +83,17 @@ static unsigned long pcsp_timer_update(struct hrtimer *handle)
return ns;
}
-enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+static void pcsp_pointer_update(struct snd_pcsp *chip)
{
- struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
struct snd_pcm_substream *substream;
- int periods_elapsed, pointer_update;
size_t period_bytes, buffer_bytes;
- unsigned long ns;
+ int periods_elapsed;
unsigned long flags;
- pointer_update = !chip->thalf;
- ns = pcsp_timer_update(handle);
- if (!ns)
- return HRTIMER_NORESTART;
-
/* update the playback position */
substream = chip->playback_substream;
if (!substream)
- return HRTIMER_NORESTART;
+ return;
period_bytes = snd_pcm_lib_period_bytes(substream);
buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
@@ -134,6 +122,26 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
if (periods_elapsed)
tasklet_schedule(&pcsp_pcm_tasklet);
+}
+
+enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+{
+ struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
+ int pointer_update;
+ u64 ns;
+
+ if (!atomic_read(&chip->timer_active) || !chip->playback_substream)
+ return HRTIMER_NORESTART;
+
+ pointer_update = !chip->thalf;
+ ns = pcsp_timer_update(chip);
+ if (!ns) {
+ printk(KERN_WARNING "PCSP: unexpected stop\n");
+ return HRTIMER_NORESTART;
+ }
+
+ if (pointer_update)
+ pcsp_pointer_update(chip);
hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
@@ -142,8 +150,6 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
static int pcsp_start_playing(struct snd_pcsp *chip)
{
- unsigned long ns;
-
#if PCSP_DEBUG
printk(KERN_INFO "PCSP: start_playing called\n");
#endif
@@ -159,11 +165,7 @@ static int pcsp_start_playing(struct snd_pcsp *chip)
atomic_set(&chip->timer_active, 1);
chip->thalf = 0;
- ns = pcsp_timer_update(&pcsp_chip.timer);
- if (!ns)
- return -EIO;
-
- hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL);
+ hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
return 0;
}
@@ -232,21 +234,22 @@ static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream)
static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+ pcsp_sync_stop(chip);
+ chip->playback_ptr = 0;
+ chip->period_ptr = 0;
+ chip->fmt_size =
+ snd_pcm_format_physical_width(substream->runtime->format) >> 3;
+ chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
#if PCSP_DEBUG
printk(KERN_INFO "PCSP: prepare called, "
- "size=%zi psize=%zi f=%zi f1=%i\n",
+ "size=%zi psize=%zi f=%zi f1=%i fsize=%i\n",
snd_pcm_lib_buffer_bytes(substream),
snd_pcm_lib_period_bytes(substream),
snd_pcm_lib_buffer_bytes(substream) /
snd_pcm_lib_period_bytes(substream),
- substream->runtime->periods);
+ substream->runtime->periods,
+ chip->fmt_size);
#endif
- pcsp_sync_stop(chip);
- chip->playback_ptr = 0;
- chip->period_ptr = 0;
- chip->fmt_size =
- snd_pcm_format_physical_width(substream->runtime->format) >> 3;
- chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
return 0;
}
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 199b0337714..903bc846763 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -72,7 +72,7 @@ static int pcsp_treble_put(struct snd_kcontrol *kcontrol,
if (treble != chip->treble) {
chip->treble = treble;
#if PCSP_DEBUG
- printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE());
+ printk(KERN_INFO "PCSP: rate set to %li\n", PCSP_RATE());
#endif
changed = 1;
}
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index 793b7f47843..3f3c3f71db4 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -219,7 +219,9 @@ static int shared_resources_initialised;
* Mid level stuff
*/
-struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED };
+struct sound_settings dmasound = {
+ .lock = __SPIN_LOCK_UNLOCKED(dmasound.lock)
+};
static inline void sound_silence(void)
{
diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c
index 77d0e5efda7..ce4db49291f 100644
--- a/sound/oss/sb_common.c
+++ b/sound/oss/sb_common.c
@@ -157,7 +157,7 @@ static void sb_intr (sb_devc *devc)
break;
default:
- /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */
+ /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */
;
}
}
@@ -177,7 +177,7 @@ static void sb_intr (sb_devc *devc)
break;
default:
- /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */
+ /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */
;
}
}
diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c
index 180e95c87e3..51a3d381a59 100644
--- a/sound/oss/sb_ess.c
+++ b/sound/oss/sb_ess.c
@@ -782,7 +782,7 @@ printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode);
break;
default:;
- /* printk(KERN_WARN "ESS: Unexpected interrupt\n"); */
+ /* printk(KERN_WARNING "ESS: Unexpected interrupt\n"); */
}
}
diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c
index e924492df21..f47f9e226b0 100644
--- a/sound/parisc/harmony.c
+++ b/sound/parisc/harmony.c
@@ -624,6 +624,9 @@ snd_harmony_pcm_init(struct snd_harmony *h)
struct snd_pcm *pcm;
int err;
+ if (snd_BUG_ON(!h))
+ return -EINVAL;
+
harmony_disable_interrupts(h);
err = snd_pcm_new(h->card, "harmony", 0, 1, 1, &pcm);
@@ -865,11 +868,12 @@ snd_harmony_mixer_reset(struct snd_harmony *h)
static int __devinit
snd_harmony_mixer_init(struct snd_harmony *h)
{
- struct snd_card *card = h->card;
+ struct snd_card *card;
int idx, err;
if (snd_BUG_ON(!h))
return -EINVAL;
+ card = h->card;
strcpy(card->mixername, "Harmony Gain control interface");
for (idx = 0; idx < HARMONY_CONTROLS; idx++) {
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index fb5ee3cc396..75c602b5b13 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -259,7 +259,6 @@ config SND_CS5530
config SND_CS5535AUDIO
tristate "CS5535/CS5536 Audio"
- depends on X86 && !X86_64
select SND_PCM
select SND_AC97_CODEC
help
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index b458d208720..aaf4da68969 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -973,7 +973,7 @@ static void snd_ali_free_voice(struct snd_ali * codec,
void *private_data;
snd_ali_printk("free_voice: channel=%d\n",pvoice->number);
- if (pvoice == NULL || !pvoice->use)
+ if (!pvoice->use)
return;
snd_ali_clear_voices(codec, pvoice->number, pvoice->number);
spin_lock_irq(&codec->voice_alloc);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c9ad182e1b4..6517f589d01 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -722,9 +722,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
chip->last_cmd[addr]);
chip->single_cmd = 1;
bus->response_reset = 0;
- /* re-initialize CORB/RIRB */
+ /* release CORB/RIRB */
azx_free_cmd_io(chip);
- azx_init_cmd_io(chip);
+ /* disable unsolicited responses */
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL);
return -1;
}
@@ -865,7 +866,9 @@ static int azx_reset(struct azx *chip)
}
/* Accept unsolicited responses */
- azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL);
+ if (!chip->single_cmd)
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) |
+ ICH6_GCTL_UNSOL);
/* detect codecs */
if (!chip->codec_mask) {
@@ -980,7 +983,8 @@ static void azx_init_chip(struct azx *chip)
azx_int_enable(chip);
/* initialize the codec command I/O */
- azx_init_cmd_io(chip);
+ if (!chip->single_cmd)
+ azx_init_cmd_io(chip);
/* program the position buffer */
azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
@@ -2674,6 +2678,7 @@ static struct pci_device_id azx_ids[] = {
{ PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3fbbc8c01e7..905859d4f4d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -110,6 +110,7 @@ struct conexant_spec {
unsigned int dell_automute;
unsigned int port_d_mode;
+ unsigned char ext_mic_bias;
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
#define CXT5066_SPDIF_OUT 0x21
+/* OLPC's microphone port is DC coupled for use with external sensors,
+ * therefore we use a 50% mic bias in order to center the input signal with
+ * the DC input range of the codec. */
+#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
+
static struct hda_channel_mode cxt5066_modes[1] = {
{ 2, NULL },
};
@@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
/* toggle input of built-in and mic jack appropriately */
static void cxt5066_automic(struct hda_codec *codec)
{
- static struct hda_verb ext_mic_present[] = {
+ struct conexant_spec *spec = codec->spec;
+ struct hda_verb ext_mic_present[] = {
/* enable external mic, port B */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
/* switch to external mic input */
{0x17, AC_VERB_SET_CONNECT_SEL, 0},
@@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
/* Port B: external microphone */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS},
/* Port C: internal microphone */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2325,6 +2332,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
CXT5066_LAPTOP),
SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
CXT5066_DELL_LAPTOP),
+ SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
{}
};
@@ -2352,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->input_mux = &cxt5066_capture_source;
spec->port_d_mode = PIN_HP;
+ spec->ext_mic_bias = PIN_VREF80;
spec->num_init_verbs = 1;
spec->init_verbs[0] = cxt5066_init_verbs;
@@ -2383,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->port_d_mode = 0;
+ spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS;
/* no S/PDIF out */
spec->multiout.dig_out_nid = 0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c08ca660dab..daf6975b0c2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -965,6 +965,8 @@ static void alc_automute_pin(struct hda_codec *codec)
unsigned int nid = spec->autocfg.hp_pins[0];
int i;
+ if (!nid)
+ return;
pincap = snd_hda_query_pin_caps(codec, nid);
if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
@@ -4682,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- spec->slave_dig_outs[i - 1] = dig_nid;
- if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
+ spec->slave_dig_outs[i - 1] = dig_nid;
}
}
if (spec->autocfg.dig_in_pin)
@@ -6247,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
- SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
@@ -9811,9 +9813,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- spec->slave_dig_outs[i - 1] = dig_nid;
- if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
+ spec->slave_dig_outs[i - 1] = dig_nid;
}
}
if (spec->autocfg.dig_in_pin)
@@ -11458,6 +11460,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+ SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
@@ -12602,7 +12605,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
- SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
+ SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
+ "Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
/* almost compatible with toshiba but with optional digital outs;
* auto-probing seems working fine
*/
@@ -17374,7 +17378,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
/* create playback/capture controls for input pins */
#define alc662_auto_create_input_ctls \
- alc880_auto_create_input_ctls
+ alc882_auto_create_input_ctls
static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 66c0876bf73..8eb6508cd99 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -28,6 +28,7 @@
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/pci.h>
+#include <linux/dmi.h>
#include <sound/core.h>
#include <sound/asoundef.h>
#include <sound/jack.h>
@@ -1693,6 +1694,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"DFI LanParty", STAC_92HD71BXX_REF),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
"HP dv4-1222nr", STAC_HP_DV4_1222NR),
+ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720,
+ "HP", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
"HP", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
@@ -4665,6 +4668,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
}
}
+static int hp_bseries_system(u32 subsystem_id)
+{
+ switch (subsystem_id) {
+ case 0x103c307e:
+ case 0x103c307f:
+ case 0x103c3080:
+ case 0x103c3081:
+ case 0x103c1722:
+ case 0x103c1723:
+ case 0x103c1724:
+ case 0x103c1725:
+ case 0x103c1726:
+ case 0x103c1727:
+ case 0x103c1728:
+ case 0x103c1729:
+ return 1;
+ }
+ return 0;
+}
+
#ifdef CONFIG_PROC_FS
static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid)
@@ -4754,6 +4777,11 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
else
spec->gpio_data |= spec->gpio_led; /* white */
+ if (hp_bseries_system(codec->subsystem_id)) {
+ /* LED state is inverted on these systems */
+ spec->gpio_data ^= spec->gpio_led;
+ }
+
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir,
spec->gpio_data);
@@ -5243,6 +5271,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
+ unsigned int pin_cfg;
int err = 0;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5426,6 +5455,45 @@ again:
break;
}
+ if (hp_bseries_system(codec->subsystem_id)) {
+ pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f);
+ if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
+ get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER ||
+ get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) {
+ /* It was changed in the BIOS to just satisfy MS DTM.
+ * Lets turn it back into slaved HP
+ */
+ pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE))
+ | (AC_JACK_HP_OUT <<
+ AC_DEFCFG_DEVICE_SHIFT);
+ pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC
+ | AC_DEFCFG_SEQUENCE)))
+ | 0x1f;
+ snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg);
+ }
+ }
+
+ if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) {
+ const struct dmi_device *dev = NULL;
+ while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
+ NULL, dev))) {
+ if (strcmp(dev->name, "HP_Mute_LED_1")) {
+ switch (codec->vendor_id) {
+ case 0x111d7608:
+ spec->gpio_led = 0x01;
+ break;
+ case 0x111d7600:
+ case 0x111d7601:
+ case 0x111d7602:
+ case 0x111d7603:
+ spec->gpio_led = 0x08;
+ break;
+ }
+ break;
+ }
+ }
+ }
+
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
spec->gpio_mask |= spec->gpio_led;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 754867ed478..aac20fb4aad 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1950,6 +1950,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
},
{
.subvendor = 0x104d,
+ .subdevice = 0x8144,
+ .name = "Sony",
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x104d,
.subdevice = 0x8197,
.name = "Sony S1XP",
.type = AC97_TUNE_INV_EAPD
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 91683a34903..8a332d2f615 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -386,6 +386,7 @@ struct via82xx {
struct snd_pcm *pcms[2];
struct snd_rawmidi *rmidi;
+ struct snd_kcontrol *dxs_controls[4];
struct snd_ac97_bus *ac97_bus;
struct snd_ac97 *ac97;
@@ -1216,9 +1217,9 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
/*
- * open callback for playback on via686 and via823x DSX
+ * open callback for playback on via686
*/
-static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
+static int snd_via686_playback_open(struct snd_pcm_substream *substream)
{
struct via82xx *chip = snd_pcm_substream_chip(substream);
struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number];
@@ -1230,6 +1231,32 @@ static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
}
/*
+ * open callback for playback on via823x DXS
+ */
+static int snd_via8233_playback_open(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev;
+ unsigned int stream;
+ int err;
+
+ viadev = &chip->devs[chip->playback_devno + substream->number];
+ if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0)
+ return err;
+ stream = viadev->reg_offset / 0x10;
+ if (chip->dxs_controls[stream]) {
+ chip->playback_volume[stream][0] = 0;
+ chip->playback_volume[stream][1] = 0;
+ chip->dxs_controls[stream]->vd[0].access &=
+ ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ &chip->dxs_controls[stream]->id);
+ }
+ return 0;
+}
+
+/*
* open callback for playback on via823x multi-channel
*/
static int snd_via8233_multi_open(struct snd_pcm_substream *substream)
@@ -1302,10 +1329,26 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream)
return 0;
}
+static int snd_via8233_playback_close(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ unsigned int stream;
+
+ stream = viadev->reg_offset / 0x10;
+ if (chip->dxs_controls[stream]) {
+ chip->dxs_controls[stream]->vd[0].access |=
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO,
+ &chip->dxs_controls[stream]->id);
+ }
+ return snd_via82xx_pcm_close(substream);
+}
+
/* via686 playback callbacks */
static struct snd_pcm_ops snd_via686_playback_ops = {
- .open = snd_via82xx_playback_open,
+ .open = snd_via686_playback_open,
.close = snd_via82xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_via82xx_hw_params,
@@ -1331,8 +1374,8 @@ static struct snd_pcm_ops snd_via686_capture_ops = {
/* via823x DSX playback callbacks */
static struct snd_pcm_ops snd_via8233_playback_ops = {
- .open = snd_via82xx_playback_open,
- .close = snd_via82xx_pcm_close,
+ .open = snd_via8233_playback_open,
+ .close = snd_via8233_playback_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_via82xx_hw_params,
.hw_free = snd_via82xx_hw_free,
@@ -1709,8 +1752,9 @@ static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
.device = 0,
/* .subdevice set later */
.name = "PCM Playback Volume",
- .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
- SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE,
.info = snd_via8233_dxs_volume_info,
.get = snd_via8233_dxs_volume_get,
.put = snd_via8233_dxs_volume_put,
@@ -1948,6 +1992,7 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip)
err = snd_ctl_add(chip->card, kctl);
if (err < 0)
return err;
+ chip->dxs_controls[i] = kctl;
}
}
}
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 7dea74b71cf..64b859925c0 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -217,20 +217,25 @@ static void snd_pdacf_detach(struct pcmcia_device *link)
* configuration callback
*/
-#define CS_CHECK(fn, ret) \
-do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0)
-
static int pdacf_config(struct pcmcia_device *link)
{
struct snd_pdacf *pdacf = link->priv;
- int last_fn, last_ret;
+ int ret;
snd_printdd(KERN_DEBUG "pdacf_config called\n");
link->conf.ConfigIndex = 0x5;
- CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io));
- CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq));
- CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf));
+ ret = pcmcia_request_io(link, &link->io);
+ if (ret)
+ goto failed;
+
+ ret = pcmcia_request_irq(link, &link->irq);
+ if (ret)
+ goto failed;
+
+ ret = pcmcia_request_configuration(link, &link->conf);
+ if (ret)
+ goto failed;
if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0)
goto failed;
@@ -238,8 +243,6 @@ static int pdacf_config(struct pcmcia_device *link)
link->dev_node = &pdacf->node;
return 0;
-cs_failed:
- cs_error(link, last_fn, last_ret);
failed:
pcmcia_disable_device(link);
return -ENODEV;
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 7445cc8a47d..1492744ad67 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -213,14 +213,11 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq
* configuration callback
*/
-#define CS_CHECK(fn, ret) \
-do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0)
-
static int vxpocket_config(struct pcmcia_device *link)
{
struct vx_core *chip = link->priv;
struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip;
- int last_fn, last_ret;
+ int ret;
snd_printdd(KERN_DEBUG "vxpocket_config called\n");
@@ -235,9 +232,17 @@ static int vxpocket_config(struct pcmcia_device *link)
strcpy(chip->card->driver, vxp440_hw.name);
}
- CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io));
- CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq));
- CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf));
+ ret = pcmcia_request_io(link, &link->io);
+ if (ret)
+ goto failed;
+
+ ret = pcmcia_request_irq(link, &link->irq);
+ if (ret)
+ goto failed;
+
+ ret = pcmcia_request_configuration(link, &link->conf);
+ if (ret)
+ goto failed;
chip->dev = &handle_to_dev(link);
snd_card_set_dev(chip->card, chip->dev);
@@ -248,8 +253,6 @@ static int vxpocket_config(struct pcmcia_device *link)
link->dev_node = &vxp->node;
return 0;
-cs_failed:
- cs_error(link, last_fn, last_ret);
failed:
pcmcia_disable_device(link);
return -ENODEV;
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 583a3693df7..a0df401ebb9 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -49,6 +49,7 @@ MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>");
MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}");
+MODULE_FIRMWARE("aica_firmware.bin");
/* module parameters */
#define CARD_NAME "AICA"
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 2dee9839be8..653a362425d 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA
select SND_OMAP_SOC_MCBSP
select SND_SOC_CX20442
help
- Say Y if you want to add support for SoC audio on Amstrad Delta.
+ Say Y if you want to add support for SoC audio device connected to
+ a handset and a speakerphone found on Amstrad E3 (Delta) videophone.
+
+ Note that in order to get those devices fully supported, you have to
+ build the kernel with standard serial port driver included and
+ configured for at least 4 ports. Then, from userspace, you must load
+ a line discipline #19 on the modem (ttyS3) serial line. The simplest
+ way to achieve this is to install util-linux-ng and use the included
+ ldattach utility. This can be started automatically from udev,
+ a simple rule like this one should do the trick (it does for me):
+ ACTION=="add", KERNEL=="controlC0", \
+ RUN+="/usr/sbin/ldattach 19 /dev/ttyS3"
config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 5735945788b..6a829eef2a4 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
else
omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
- omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
- omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+ if (!(cpu_class_is_omap1())) {
+ omap_set_dma_src_burst_mode(prtd->dma_ch,
+ OMAP_DMA_DATA_BURST_16);
+ omap_set_dma_dest_burst_mode(prtd->dma_ch,
+ OMAP_DMA_DATA_BURST_16);
+ }
return 0;
}
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 5cbbdc80fde..1f35c6fcf5f 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -75,11 +75,19 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
{
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
dma_addr_t pos = prtd->dma_pos;
+ unsigned int limit;
int ret;
pr_debug("Entered %s\n", __func__);
- while (prtd->dma_loaded < prtd->dma_limit) {
+ if (s3c_dma_has_circular()) {
+ limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period;
+ } else
+ limit = prtd->dma_limit;
+
+ pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit);
+
+ while (prtd->dma_loaded < limit) {
unsigned long len = prtd->dma_period;
pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
@@ -123,7 +131,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
snd_pcm_period_elapsed(substream);
spin_lock(&prtd->lock);
- if (prtd->state & ST_RUNNING) {
+ if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) {
prtd->dma_loaded--;
s3c24xx_pcm_enqueue(substream);
}
@@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
printk(KERN_ERR "failed to get dma channel\n");
return ret;
}
+
+ /* use the circular buffering if we have it available. */
+ if (s3c_dma_has_circular())
+ s3c2410_dma_setflags(prtd->params->channel,
+ S3C2410_DMAF_CIRCULAR);
}
s3c2410_dma_set_buffdone_fn(prtd->params->channel,
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 3c06c401d0f..105a77eeded 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -220,6 +220,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
goto err;
}
+ clk_enable(i2s->iis_cclk);
+
ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
if (ret)
goto err_clk;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7ff04ad2a97..0a1b2f64bbe 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -834,6 +834,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume_device);
#define soc_resume NULL
#endif
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct platform_device *pdev = container_of(card->dev,
@@ -877,6 +880,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
ac97 = 1;
}
+ for (i = 0; i < card->num_links; i++) {
+ if (!card->dai_link[i].codec_dai->ops)
+ card->dai_link[i].codec_dai->ops = &null_dai_ops;
+ }
+
/* If we have AC97 in the system then don't wait for the
* codec. This will need revisiting if we have to handle
* systems with mixed AC97 and non-AC97 parts. Only check for
@@ -2329,9 +2337,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
return 0;
}
-static struct snd_soc_dai_ops null_dai_ops = {
-};
-
/**
* snd_soc_register_dai - Register a DAI with the ASoC core
*
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8de6f9dec4a..d89f6dc0090 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2072,9 +2072,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
}
}
}
- mutex_unlock(&codec->mutex);
dapm_power_widgets(codec, event);
+ mutex_unlock(&codec->mutex);
dump_dapm(codec, __func__);
return 0;
}
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 121af0644fd..86b2c3b92df 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -62,10 +62,14 @@ static void
activate_substream(struct snd_usb_caiaqdev *dev,
struct snd_pcm_substream *sub)
{
+ spin_lock(&dev->spinlock);
+
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
dev->sub_playback[sub->number] = sub;
else
dev->sub_capture[sub->number] = sub;
+
+ spin_unlock(&dev->spinlock);
}
static void
@@ -269,16 +273,22 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
{
int index = sub->number;
struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub);
+ snd_pcm_uframes_t ptr;
+
+ spin_lock(&dev->spinlock);
if (dev->input_panic || dev->output_panic)
- return SNDRV_PCM_POS_XRUN;
+ ptr = SNDRV_PCM_POS_XRUN;
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
- return bytes_to_frames(sub->runtime,
+ ptr = bytes_to_frames(sub->runtime,
dev->audio_out_buf_pos[index]);
else
- return bytes_to_frames(sub->runtime,
+ ptr = bytes_to_frames(sub->runtime,
dev->audio_in_buf_pos[index]);
+
+ spin_unlock(&dev->spinlock);
+ return ptr;
}
/* operators for both playback and capture */
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 83e6c1312d4..a3f02dd9744 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 8e7f78941ba..e9a3a9dca15 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -210,7 +210,7 @@ struct snd_usb_midi_endpoint_info {
/*
*/
-#define combine_word(s) ((*s) | ((unsigned int)(s)[1] << 8))
+#define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8))
#define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16))
#define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24))
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-03 09:45:10 +0000