aboutsummaryrefslogtreecommitdiff
path: root/sound
diff options
context:
space:
mode:
Diffstat (limited to 'sound')
-rw-r--r--sound/arm/aaci.c2
-rw-r--r--sound/arm/pxa2xx-ac97-lib.c2
-rw-r--r--sound/core/pcm_lib.c17
-rw-r--r--sound/core/pcm_native.c6
-rw-r--r--sound/drivers/pcsp/pcsp_mixer.c4
-rw-r--r--sound/drivers/serial-u16550.c11
-rw-r--r--sound/isa/msnd/msnd.c6
-rw-r--r--sound/pci/ac97/ac97_patch.c7
-rw-r--r--sound/pci/bt87x.c6
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c2
-rw-r--r--sound/pci/echoaudio/indigodjx.c1
-rw-r--r--sound/pci/echoaudio/indigoiox.c1
-rw-r--r--sound/pci/hda/patch_realtek.c1
-rw-r--r--sound/pci/hda/patch_sigmatel.c7
-rw-r--r--sound/pci/korg1212/korg1212.c6
-rw-r--r--sound/pci/riptide/riptide.c10
-rw-r--r--sound/pci/via82xx.c2
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c3
-rw-r--r--sound/soc/au1x/dbdma2.c2
-rw-r--r--sound/soc/codecs/twl4030.c8
-rw-r--r--sound/soc/codecs/wm8350.c2
-rw-r--r--sound/soc/codecs/wm8990.c40
-rw-r--r--sound/soc/davinci/Kconfig7
-rw-r--r--sound/soc/davinci/davinci-evm.c63
-rw-r--r--sound/soc/davinci/davinci-i2s.c26
-rw-r--r--sound/soc/davinci/davinci-pcm.c71
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c3
-rw-r--r--sound/soc/sh/dma-sh7760.c3
-rw-r--r--sound/soc/soc-core.c3
-rw-r--r--sound/sparc/dbri.c3
-rw-r--r--sound/usb/usx2y/usbusx2yaudio.c3
31 files changed, 225 insertions, 103 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 7fbd68fab94..5c48e36038f 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -1074,7 +1074,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
return i;
}
-static int __devinit aaci_probe(struct amba_device *dev, void *id)
+static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
{
struct aaci *aaci;
int ret, i;
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index a2c12d105c9..6fdca97186e 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action)
switch (resetgpio_action) {
case RESETGPIO_NORMAL_ALTFUNC:
if (reset_gpio == 113)
- mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
+ mode = 113 | GPIO_ALT_FN_2_OUT;
if (reset_gpio == 95)
mode = 95 | GPIO_ALT_FN_1_OUT;
break;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 63d088f2265..d659995ac3a 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,17 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
new_hw_ptr = hw_base + pos;
}
}
+
+ /* Do jiffies check only in xrun_debug mode */
+ if (!xrun_debug(substream))
+ goto no_jiffies_check;
+
+ /* Skip the jiffies check for hardwares with BATCH flag.
+ * Such hardware usually just increases the position at each IRQ,
+ * thus it can't give any strange position.
+ */
+ if (runtime->hw.info & SNDRV_PCM_INFO_BATCH)
+ goto no_jiffies_check;
hdelta = new_hw_ptr - old_hw_ptr;
jdelta = jiffies - runtime->hw_ptr_jiffies;
if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
@@ -272,6 +283,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
hw_base -= hw_base % runtime->buffer_size;
delta = 0;
}
+ no_jiffies_check:
if (delta > runtime->period_size + runtime->period_size / 2) {
hw_ptr_error(substream,
"Lost interrupts? "
@@ -329,7 +341,9 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
hw_base = 0;
new_hw_ptr = hw_base + pos;
}
- if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
+ /* Do jiffies check only in xrun_debug mode */
+ if (xrun_debug(substream) &&
+ ((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
hw_ptr_error(substream,
"hw_ptr skipping! "
"(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n",
@@ -1471,7 +1485,6 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream,
runtime->status->hw_ptr %= runtime->buffer_size;
else
runtime->status->hw_ptr = 0;
- runtime->hw_ptr_jiffies = jiffies;
snd_pcm_stream_unlock_irqrestore(substream, flags);
return 0;
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index cb769d415db..45dc53fcfa2 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -849,6 +849,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
{
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_trigger_tstamp(substream);
+ runtime->hw_ptr_jiffies = jiffies;
runtime->status->state = state;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
@@ -962,6 +963,11 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
{
if (substream->runtime->trigger_master != substream)
return 0;
+ /* The jiffies check in snd_pcm_update_hw_ptr*() is done by
+ * a delta betwen the current jiffies, this gives a large enough
+ * delta, effectively to skip the check once.
+ */
+ substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000;
return substream->ops->trigger(substream,
push ? SNDRV_PCM_TRIGGER_PAUSE_PUSH :
SNDRV_PCM_TRIGGER_PAUSE_RELEASE);
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index caeb0f57fcc..199b0337714 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,8 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
uinfo->value.enumerated.items = chip->max_treble + 1;
if (uinfo->value.enumerated.item > chip->max_treble)
uinfo->value.enumerated.item = chip->max_treble;
- sprintf(uinfo->value.enumerated.name, "%d",
- PCSP_CALC_RATE(uinfo->value.enumerated.item));
+ sprintf(uinfo->value.enumerated.name, "%lu",
+ (unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item));
return 0;
}
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index b2b6d50c942..a25fb7b1f44 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -963,16 +963,11 @@ static int __devinit snd_serial_probe(struct platform_device *devptr)
if (err < 0)
goto _err;
- sprintf(card->longname, "%s at 0x%lx, irq %d speed %d div %d outs %d ins %d adaptor %s droponfull %d",
+ sprintf(card->longname, "%s [%s] at %#lx, irq %d",
card->shortname,
- uart->base,
- uart->irq,
- uart->speed,
- (int)uart->divisor,
- outs[dev],
- ins[dev],
adaptor_names[uart->adaptor],
- uart->drop_on_full);
+ uart->base,
+ uart->irq);
snd_card_set_dev(card, &devptr->dev);
diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
index 906454413ed..3a1526ae172 100644
--- a/sound/isa/msnd/msnd.c
+++ b/sound/isa/msnd/msnd.c
@@ -438,7 +438,8 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip,
static struct snd_pcm_hardware snd_msnd_playback = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
@@ -456,7 +457,8 @@ static struct snd_pcm_hardware snd_msnd_playback = {
static struct snd_pcm_hardware snd_msnd_capture = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 81bc93e5f1e..7337abdbe4e 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97)
}
static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker =
-AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch",
+ AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+/* "Sigmatel " removed due to excessive name length: */
static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert =
-AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
+AC97_SINGLE("Surround Phase Inversion Playback Switch",
+ AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = {
AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0),
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index a299340519d..ce3f2e90f4d 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -349,7 +349,8 @@ static struct snd_pcm_hardware snd_bt87x_digital_hw = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = 0, /* set at runtime */
.channels_min = 2,
@@ -365,7 +366,8 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
.rates = SNDRV_PCM_RATE_KNOT,
.rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX,
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index ad2888705d2..c111efe61c3 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
"Capture Volume",
"External Amplifier",
"Sigmatel 4-Speaker Stereo Playback Switch",
- "Sigmatel Surround Phase Inversion Playback ",
+ "Surround Phase Inversion Playback Switch",
NULL
};
static char *ca0106_rename_ctls[] = {
diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
index 3482ef69f49..2e44316530a 100644
--- a/sound/pci/echoaudio/indigodjx.c
+++ b/sound/pci/echoaudio/indigodjx.c
@@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
.rates = SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000,
.rate_min = 32000,
diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c
index aebee27a40f..eb3819f9654 100644
--- a/sound/pci/echoaudio/indigoiox.c
+++ b/sound/pci/echoaudio/indigoiox.c
@@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = {
.rates = SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000,
.rate_min = 32000,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8a0d3e7927..bcbb736f94f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12058,6 +12058,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 917bc5d3ac2..03b3646018a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4079,7 +4079,12 @@ static int stac92xx_init(struct hda_codec *codec)
pinctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
/* if PINCTL already set then skip */
- if (!(pinctl & AC_PINCTL_IN_EN)) {
+ /* Also, if both INPUT and OUTPUT are set,
+ * it must be a BIOS bug; need to override, too
+ */
+ if (!(pinctl & AC_PINCTL_IN_EN) ||
+ (pinctl & AC_PINCTL_OUT_EN)) {
+ pinctl &= ~AC_PINCTL_OUT_EN;
pinctl |= AC_PINCTL_IN_EN;
stac92xx_auto_set_pinctl(codec, nid,
pinctl);
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8b79969034b..7cc38a11e99 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1238,7 +1238,8 @@ static struct snd_pcm_hardware snd_korg1212_playback_info =
{
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED),
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = (SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000),
@@ -1258,7 +1259,8 @@ static struct snd_pcm_hardware snd_korg1212_capture_info =
{
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED),
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = (SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000),
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 6f1034417a0..e51a5ef1954 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -889,7 +889,7 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
spin_lock_irqsave(&cif->lock, irqflags);
while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport))
udelay(10);
- if (i >= CMDIF_TIMEOUT) {
+ if (i > CMDIF_TIMEOUT) {
err = -EBUSY;
goto errout;
}
@@ -907,8 +907,10 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
WRITE_PORT_ULONG(cmdport->data1, cmd); /* write cmd */
if ((flags & RESP) && ret) {
while (!IS_DATF(cmdport) &&
- time++ < CMDIF_TIMEOUT)
+ time < CMDIF_TIMEOUT) {
udelay(10);
+ time++;
+ }
if (time < CMDIF_TIMEOUT) { /* read response */
ret->retlongs[0] =
READ_PORT_ULONG(cmdport->data1);
@@ -1454,7 +1456,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd)
SEND_GPOS(cif, 0, data->id, &rptr);
udelay(1);
} while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY);
- if (j >= MAX_WRITE_RETRY)
+ if (j > MAX_WRITE_RETRY)
snd_printk(KERN_ERR "Riptide: Could not stop stream!");
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -1783,7 +1785,7 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg,
SEND_SACR(cif, val, reg);
SEND_RACR(cif, reg, &rptr);
} while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY);
- if (i == MAX_WRITE_RETRY)
+ if (i > MAX_WRITE_RETRY)
snd_printdd("Write AC97 reg failed\n");
}
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 809b233dd4a..1ef58c51c21 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1687,7 +1687,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol,
return change;
}
-static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1);
static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = {
.name = "PCM Playback Volume",
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 01066c95580..d057e648964 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -240,7 +240,8 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
static struct snd_pcm_hardware pdacf_pcm_capture_hw = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_MMAP_VALID),
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a25914..594c6c5b783 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
/* PCM hardware DMA capabilities - platform specific */
static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED,
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
.formats = AU1XPSC_PCM_FMTS,
.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
.period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28..df7c8c281d2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
/*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
* Capture gain after the ADCs
* from 0 dB to 31 dB in 1 dB steps
*/
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
4, 3, 0, output_tvl),
SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
- TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+ TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
/* Common capture gain controls */
SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed..0275321ff8a 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
* required for LRC in master mode. The DACs or ADCs need a
* valid audio path i.e. pin -> ADC or DAC -> pin before
* the LRC will be enabled in master mode. */
- if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+ if (!master || cmd != SNDRV_PCM_TRIGGER_START)
return 0;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3..40cd274eb1e 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
&wm8990_dapm_ainlmux_controls, inmixer_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
&wm8990_dapm_ainrmux_controls, inmixer_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"LIN12 PGA", "LIN2 Switch", "LIN2"},
/* LIN34 PGA */
{"LIN34 PGA", "LIN3 Switch", "LIN3"},
- {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+ {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
/* INMIXL */
{"INMIXL", "Record Left Volume", "LOMIX"},
{"INMIXL", "LIN2 Volume", "LIN2"},
{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
- /* AILNMUX */
- {"AILNMUX", "INMIXL Mix", "INMIXL"},
- {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
- {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
- {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
- {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* AINLMUX */
+ {"AINLMUX", "INMIXL Mix", "INMIXL"},
+ {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+ {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+ {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
/* ADC */
- {"Left ADC", NULL, "AILNMUX"},
+ {"Left ADC", NULL, "AINLMUX"},
/* RIN12 PGA */
{"RIN12 PGA", "RIN1 Switch", "RIN1"},
{"RIN12 PGA", "RIN2 Switch", "RIN2"},
/* RIN34 PGA */
{"RIN34 PGA", "RIN3 Switch", "RIN3"},
- {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+ {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
/* INMIXL */
{"INMIXR", "Record Right Volume", "ROMIX"},
{"INMIXR", "RIN2 Volume", "RIN2"},
{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
- /* AIRNMUX */
- {"AIRNMUX", "INMIXR Mix", "INMIXR"},
- {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
- {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
- {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
- {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* AINRMUX */
+ {"AINRMUX", "INMIXR Mix", "INMIXR"},
+ {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+ {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+ {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
/* ADC */
- {"Right ADC", NULL, "AIRNMUX"},
+ {"Right ADC", NULL, "AINRMUX"},
/* LOMIX */
{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
/* OUT3MIX */
- {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+ {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
/* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
/* Output Pins */
{"LON", NULL, "LONMIX"},
{"LOP", NULL, "LOPMIX"},
- {"OUT", NULL, "OUT3MIX"},
+ {"OUT3", NULL, "OUT3MIX"},
{"LOUT", NULL, "LOUT PGA"},
{"SPKN", NULL, "SPKMIX"},
{"ROUT", NULL, "ROUT PGA"},
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c9657..411a710be66 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
tristate
config SND_DAVINCI_SOC_EVM
- tristate "SoC Audio support for DaVinci EVM"
- depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+ tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+ depends on SND_DAVINCI_SOC
+ depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
select SND_DAVINCI_SOC_I2S
select SND_SOC_TLV320AIC3X
help
Say Y if you want to add support for SoC audio on TI
- DaVinci EVM platform.
+ DaVinci DM6446 or DM355 EVM platforms.
config SND_DAVINCI_SOC_SFFSDR
tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b347007..58fd1cbedd8 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
#include <sound/soc-dapm.h>
#include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
#include "../codecs/tlv320aic3x.h"
#include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
/* evm audio private data */
static struct aic3x_setup_data evm_aic3x_setup = {
- .i2c_bus = 0,
+ .i2c_bus = 1,
.i2c_address = 0x1b,
};
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
.codec_data = &evm_aic3x_setup,
};
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
static struct resource evm_snd_resources[] = {
{
- .start = DAVINCI_MCBSP_BASE,
- .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+ .start = DAVINCI_ASP0_BASE,
+ .end = DAVINCI_ASP0_BASE + SZ_8K - 1,
.flags = IORESOURCE_MEM,
},
};
static struct evm_snd_platform_data evm_snd_data = {
- .tx_dma_ch = DM644X_DMACH_MCBSP_TX,
- .rx_dma_ch = DM644X_DMACH_MCBSP_RX,
+ .tx_dma_ch = DAVINCI_DMA_ASP0_TX,
+ .rx_dma_ch = DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+ {
+ .start = DAVINCI_ASP1_BASE,
+ .end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+ .flags = IORESOURCE_MEM,
+ },
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+ .tx_dma_ch = DAVINCI_DMA_ASP1_TX,
+ .rx_dma_ch = DAVINCI_DMA_ASP1_RX,
};
static struct platform_device *evm_snd_device;
static int __init evm_init(void)
{
+ struct resource *resources;
+ unsigned num_resources;
+ struct evm_snd_platform_data *data;
+ int index;
int ret;
- evm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (machine_is_davinci_evm()) {
+ davinci_cfg_reg(DM644X_MCBSP);
+
+ resources = evm_snd_resources;
+ num_resources = ARRAY_SIZE(evm_snd_resources);
+ data = &evm_snd_data;
+ index = 0;
+ } else if (machine_is_davinci_dm355_evm()) {
+ /* we don't use ASP1 IRQs, or we'd need to mux them ... */
+ davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+ davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+ resources = dm335evm_snd_resources;
+ num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+ data = &dm335evm_snd_data;
+ index = 1;
+ } else
+ return -EINVAL;
+
+ evm_snd_device = platform_device_alloc("soc-audio", index);
if (!evm_snd_device)
return -ENOMEM;
platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
evm_snd_devdata.dev = &evm_snd_device->dev;
- platform_device_add_data(evm_snd_device, &evm_snd_data,
- sizeof(evm_snd_data));
+ platform_device_add_data(evm_snd_device, data, sizeof(*data));
- ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
- ARRAY_SIZE(evm_snd_resources));
+ ret = platform_device_add_resources(evm_snd_device, resources,
+ num_resources);
if (ret) {
platform_device_put(evm_snd_device);
return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb9439d3d..b1ea52fc83c 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
#include "davinci-pcm.h"
+
+/*
+ * NOTE: terminology here is confusing.
+ *
+ * - This driver supports the "Audio Serial Port" (ASP),
+ * found on dm6446, dm355, and other DaVinci chips.
+ *
+ * - But it labels it a "Multi-channel Buffered Serial Port"
+ * (McBSP) as on older chips like the dm642 ... which was
+ * backward-compatible, possibly explaining that confusion.
+ *
+ * - OMAP chips have a controller called McBSP, which is
+ * incompatible with the DaVinci flavor of McBSP.
+ *
+ * - Newer DaVinci chips have a controller called McASP,
+ * incompatible with ASP and with either McBSP.
+ *
+ * In short: this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
#define DAVINCI_MCBSP_DRR_REG 0x00
#define DAVINCI_MCBSP_DXR_REG 0x04
#define DAVINCI_MCBSP_SPCR_REG 0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
struct davinci_mcbsp_dev *dev;
struct resource *mem, *ioarea;
struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
cpu_dai->private_data = dev;
- dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+ dev->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(dev->clk)) {
ret = -ENODEV;
goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+ struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
struct resource *mem;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b3a53..a0599658848 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
#include <sound/soc.h>
#include <asm/dma.h>
+#include <mach/edma.h>
#include "davinci-pcm.h"
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
spinlock_t lock;
int period; /* current DMA period */
int master_lch; /* Master DMA channel */
- int slave_lch; /* Slave DMA channel */
+ int slave_lch; /* linked parameter RAM reload slot */
struct davinci_pcm_dma_params *params; /* DMA params */
};
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
dst_bidx = data_type;
}
- davinci_set_dma_src_params(lch, src, INCR, W8BIT);
- davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
- davinci_set_dma_src_index(lch, src_bidx, 0);
- davinci_set_dma_dest_index(lch, dst_bidx, 0);
- davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+ edma_set_src(lch, src, INCR, W8BIT);
+ edma_set_dest(lch, dst, INCR, W8BIT);
+ edma_set_src_index(lch, src_bidx, 0);
+ edma_set_dest_index(lch, dst_bidx, 0);
+ edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
prtd->period++;
if (unlikely(prtd->period >= runtime->periods))
prtd->period = 0;
}
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
{
struct snd_pcm_substream *substream = data;
struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
struct davinci_runtime_data *prtd = substream->runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
- int tcc = TCC_ANY;
+ struct edmacc_param p_ram;
int ret;
if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
prtd->params = dma_data;
/* Request master DMA channel */
- ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+ ret = edma_alloc_channel(prtd->params->channel,
davinci_pcm_dma_irq, substream,
- &prtd->master_lch, &tcc, EVENTQ_0);
- if (ret)
+ EVENTQ_0);
+ if (ret < 0)
return ret;
+ prtd->master_lch = ret;
- /* Request slave DMA channel */
- ret = davinci_request_dma(PARAM_ANY, "Link",
- NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
- if (ret) {
- davinci_free_dma(prtd->master_lch);
+ /* Request parameter RAM reload slot */
+ ret = edma_alloc_slot(EDMA_SLOT_ANY);
+ if (ret < 0) {
+ edma_free_channel(prtd->master_lch);
return ret;
}
-
- /* Link slave DMA channel in loopback */
- davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+ prtd->slave_lch = ret;
+
+ /* Issue transfer completion IRQ when the channel completes a
+ * transfer, then always reload from the same slot (by a kind
+ * of loopback link). The completion IRQ handler will update
+ * the reload slot with a new buffer.
+ *
+ * REVISIT save p_ram here after setting up everything except
+ * the buffer and its length (ccnt) ... use it as a template
+ * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+ */
+ edma_read_slot(prtd->slave_lch, &p_ram);
+ p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+ p_ram.link_bcntrld = prtd->slave_lch << 5;
+ edma_write_slot(prtd->slave_lch, &p_ram);
return 0;
}
@@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- davinci_start_dma(prtd->master_lch);
+ edma_start(prtd->master_lch);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- davinci_stop_dma(prtd->master_lch);
+ edma_stop(prtd->master_lch);
break;
default:
ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
- struct paramentry_descriptor temp;
+ struct edmacc_param temp;
prtd->period = 0;
davinci_pcm_enqueue_dma(substream);
- /* Get slave channel dma params for master channel startup */
- davinci_get_dma_params(prtd->slave_lch, &temp);
- davinci_set_dma_params(prtd->master_lch, &temp);
+ /* Copy self-linked parameter RAM entry into master channel */
+ edma_read_slot(prtd->slave_lch, &temp);
+ edma_write_slot(prtd->master_lch, &temp);
return 0;
}
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
spin_lock(&prtd->lock);
- davinci_dma_getposition(prtd->master_lch, &src, &dst);
+ edma_get_position(prtd->master_lch, &src, &dst);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
count = src - runtime->dma_addr;
else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct davinci_runtime_data *prtd = runtime->private_data;
- davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+ edma_unlink(prtd->slave_lch);
- davinci_free_dma(prtd->slave_lch);
- davinci_free_dma(prtd->master_lch);
+ edma_free_slot(prtd->slave_lch);
+ edma_free_channel(prtd->master_lch);
kfree(prtd);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b..1111c710118 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
.rate_min = 8000,
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb92..baddb1242c7 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID),
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH),
.formats = DMABRG_FMTS,
.rates = DMABRG_RATES,
.rate_min = 8000,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f652d0..1cd149b9ce6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev)
struct snd_soc_platform *platform = card->platform;
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ if (!card->instantiated)
+ return 0;
+
run_delayed_work(&card->delayed_work);
if (platform->remove)
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index af95ff1e126..1d2e51b3f91 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -1975,7 +1975,8 @@ static struct snd_pcm_hardware snd_dbri_pcm_hw = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID,
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH,
.formats = SNDRV_PCM_FMTBIT_MU_LAW |
SNDRV_PCM_FMTBIT_A_LAW |
SNDRV_PCM_FMTBIT_U8 |
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 9a608fa8515..dd1ab617784 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -870,7 +870,8 @@ static struct snd_pcm_hardware snd_usX2Y_2c =
{
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID),
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH),
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE,
.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.rate_min = 44100,