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authorTakashi Iwai <tiwai@suse.de>2011-09-16 08:29:04 +0200
committerTakashi Iwai <tiwai@suse.de>2011-09-16 08:29:04 +0200
commit0308110615acb4b6409fe042caf22bb9fb075d85 (patch)
treed129f11ebdbe49b21246a72e6ba52c50e81e8e7e
parentc731bc96ad641a5fa3d50a87b474652505507282 (diff)
parent763437a9e7737535b2fc72175ad4974048769be6 (diff)
downloadop-kernel-dev-0308110615acb4b6409fe042caf22bb9fb075d85.zip
op-kernel-dev-0308110615acb4b6409fe042caf22bb9fb075d85.tar.gz
Merge branch 'fix/misc' into topic/misc
-rw-r--r--MAINTAINERS5
-rw-r--r--sound/aoa/fabrics/layout.c2
-rw-r--r--sound/core/pcm_lib.c33
-rw-r--r--sound/pci/hda/hda_codec.c6
-rw-r--r--sound/pci/hda/hda_eld.c31
-rw-r--r--sound/pci/hda/patch_cirrus.c10
-rw-r--r--sound/pci/hda/patch_conexant.c57
-rw-r--r--sound/pci/hda/patch_realtek.c28
-rw-r--r--sound/soc/blackfin/bf5xx-ad193x.c6
-rw-r--r--sound/soc/codecs/ad193x.c11
-rw-r--r--sound/soc/codecs/ad193x.h5
-rw-r--r--sound/soc/codecs/sta32x.c1
-rw-r--r--sound/soc/codecs/wm8962.c12
-rw-r--r--sound/soc/codecs/wm8996.c28
-rw-r--r--sound/soc/ep93xx/ep93xx-i2s.c5
-rw-r--r--sound/soc/fsl/fsl_dma.c2
-rw-r--r--sound/soc/fsl/mpc5200_dma.c6
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c18
-rw-r--r--sound/soc/fsl/p1022_ds.c4
-rw-r--r--sound/soc/imx/imx-pcm-fiq.c1
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c2
-rw-r--r--sound/soc/omap/ams-delta.c6
-rw-r--r--sound/soc/samsung/Kconfig1
-rw-r--r--sound/soc/samsung/h1940_uda1380.c1
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c1
-rw-r--r--sound/soc/samsung/speyside_wm8962.c6
-rw-r--r--sound/soc/soc-cache.c12
-rw-r--r--sound/soc/soc-core.c6
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/soc/soc-io.c23
-rw-r--r--sound/soc/soc-jack.c4
-rw-r--r--sound/soc/soc-pcm.c3
-rw-r--r--sound/soc/tegra/tegra_wm8903.c4
33 files changed, 222 insertions, 120 deletions
diff --git a/MAINTAINERS b/MAINTAINERS
index 46e3e6b..0b4ccdd 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5532,6 +5532,7 @@ F: include/media/*7146*
SAMSUNG AUDIO (ASoC) DRIVERS
M: Jassi Brar <jassisinghbrar@gmail.com>
+M: Sangbeom Kim <sbkim73@samsung.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Supported
F: sound/soc/samsung
@@ -7199,6 +7200,9 @@ W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
S: Supported
F: Documentation/hwmon/wm83??
F: drivers/leds/leds-wm83*.c
+F: drivers/input/misc/wm831x-on.c
+F: drivers/input/touchscreen/wm831x-ts.c
+F: drivers/input/touchscreen/wm97*.c
F: drivers/mfd/wm8*.c
F: drivers/power/wm83*.c
F: drivers/rtc/rtc-wm83*.c
@@ -7208,6 +7212,7 @@ F: drivers/watchdog/wm83*_wdt.c
F: include/linux/mfd/wm831x/
F: include/linux/mfd/wm8350/
F: include/linux/mfd/wm8400*
+F: include/linux/wm97xx.h
F: include/sound/wm????.h
F: sound/soc/codecs/wm*
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 3fd1a7e..552b97a 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
sdev->pcmid = -1;
list_del(&ldev->list);
layouts_list_items--;
+ kfree(ldev);
outnodev:
of_node_put(sound);
layout_device = NULL;
- kfree(ldev);
return -ENODEV;
}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 86d0caf..62e90b8 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
snd_pcm_uframes_t avail = 0;
long wait_time, tout;
+ init_waitqueue_entry(&wait, current);
+ set_current_state(TASK_INTERRUPTIBLE);
+ add_wait_queue(&runtime->tsleep, &wait);
+
if (runtime->no_period_wakeup)
wait_time = MAX_SCHEDULE_TIMEOUT;
else {
@@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
}
wait_time = msecs_to_jiffies(wait_time * 1000);
}
- init_waitqueue_entry(&wait, current);
- add_wait_queue(&runtime->tsleep, &wait);
+
for (;;) {
if (signal_pending(current)) {
err = -ERESTARTSYS;
break;
}
+
+ /*
+ * We need to check if space became available already
+ * (and thus the wakeup happened already) first to close
+ * the race of space already having become available.
+ * This check must happen after been added to the waitqueue
+ * and having current state be INTERRUPTIBLE.
+ */
+ if (is_playback)
+ avail = snd_pcm_playback_avail(runtime);
+ else
+ avail = snd_pcm_capture_avail(runtime);
+ if (avail >= runtime->twake)
+ break;
snd_pcm_stream_unlock_irq(substream);
- tout = schedule_timeout_interruptible(wait_time);
+
+ tout = schedule_timeout(wait_time);
+
snd_pcm_stream_lock_irq(substream);
+ set_current_state(TASK_INTERRUPTIBLE);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_SUSPENDED:
err = -ESTRPIPE;
@@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
err = -EIO;
break;
}
- if (is_playback)
- avail = snd_pcm_playback_avail(runtime);
- else
- avail = snd_pcm_capture_avail(runtime);
- if (avail >= runtime->twake)
- break;
}
_endloop:
+ set_current_state(TASK_RUNNING);
remove_wait_queue(&runtime->tsleep, &wait);
*availp = avail;
return err;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c..f3aefef 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
return -1;
}
recursive++;
- for (i = 0; i < nums; i++)
+ for (i = 0; i < nums; i++) {
+ unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+ if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+ continue;
if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
return i;
+ }
return -1;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d..c34f730 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
SNDRV_PCM_RATE_192000, /* 7: 192000Hz */
};
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
int byte_index)
{
unsigned int val;
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_HDMI_ELDD, byte_index);
-
#ifdef BE_PARANOID
printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
#endif
-
- if ((val & AC_ELDD_ELD_VALID) == 0) {
- snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
- byte_index);
- val = 0;
- }
-
- return val & AC_ELDD_ELD_DATA;
+ return val;
}
#define GRAB_BITS(buf, byte, lowbit, bits) \
@@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
if (!buf)
return -ENOMEM;
- for (i = 0; i < size; i++)
- buf[i] = hdmi_get_eld_byte(codec, nid, i);
+ for (i = 0; i < size; i++) {
+ unsigned int val = hdmi_get_eld_data(codec, nid, i);
+ if (!(val & AC_ELDD_ELD_VALID)) {
+ if (!i) {
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ snd_printd(KERN_INFO
+ "HDMI: invalid ELD data byte %d\n", i);
+ val = 0;
+ } else
+ val &= AC_ELDD_ELD_DATA;
+ buf[i] = val;
+ }
ret = hdmi_update_eld(eld, buf, size);
+error:
kfree(buf);
return ret;
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc..c45f3e6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
unsigned int *idxp)
{
- int i;
+ int i, idx;
hda_nid_t nid;
nid = codec->start_nid;
@@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
- *idxp = snd_hda_get_conn_index(codec, nid, pin, false);
- if (*idxp >= 0)
+ idx = snd_hda_get_conn_index(codec, nid, pin, false);
+ if (idx >= 0) {
+ *idxp = idx;
return nid;
+ }
}
return 0;
}
@@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
int index, unsigned int pval, int dir,
struct snd_kcontrol **kctlp)
{
- char tmp[32];
+ char tmp[44];
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc94..7696d05 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
#define MAX_AUTO_DACS 5
+#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */
+
/* fill analog DAC list from the widget tree */
static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
{
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
/* fill pin_dac_pair list from the pin and dac list */
static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
int num_pins, hda_nid_t *dacs, int *rest,
- struct pin_dac_pair *filled, int type)
+ struct pin_dac_pair *filled, int nums,
+ int type)
{
- int i, nums;
+ int i, start = nums;
- nums = 0;
- for (i = 0; i < num_pins; i++) {
+ for (i = 0; i < num_pins; i++, nums++) {
filled[nums].pin = pins[i];
filled[nums].type = type;
filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
- nums++;
+ if (filled[nums].dac)
+ continue;
+ if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+ filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+ filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
}
return nums;
}
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
rest = fill_cx_auto_dacs(codec, dacs);
/* parse all analog output pins */
nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
- dacs, &rest, spec->dac_info,
- AUTO_PIN_LINE_OUT);
- nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_HP_OUT);
- nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_SPEAKER_OUT);
+ dacs, &rest, spec->dac_info, 0,
+ AUTO_PIN_LINE_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_HP_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_SPEAKER_OUT);
spec->dac_info_filled = nums;
/* fill multiout struct */
for (i = 0; i < nums; i++) {
hda_nid_t dac = spec->dac_info[i].dac;
- if (!dac)
+ if (!dac || (dac & DAC_SLAVE_FLAG))
continue;
switch (spec->dac_info[i].type) {
case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
}
if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
cx_auto_check_auto_mic(codec);
- if (imux->num_items > 1 && !spec->auto_mic) {
+ if (imux->num_items > 1) {
for (i = 1; i < imux->num_items; i++) {
if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
nid = spec->dac_info[i].dac;
if (!nid)
nid = spec->multiout.dac_nids[0];
+ else if (nid & DAC_SLAVE_FLAG)
+ nid &= ~DAC_SLAVE_FLAG;
select_connection(codec, spec->dac_info[i].pin, nid);
}
if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
hda_nid_t pin, const char *name, int idx)
{
unsigned int caps;
- caps = query_amp_caps(codec, dac, HDA_OUTPUT);
- if (caps & AC_AMPCAP_NUM_STEPS)
- return cx_auto_add_pb_volume(codec, dac, name, idx);
+ if (dac && !(dac & DAC_SLAVE_FLAG)) {
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ if (caps & AC_AMPCAP_NUM_STEPS)
+ return cx_auto_add_pb_volume(codec, dac, name, idx);
+ }
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (caps & AC_AMPCAP_NUM_STEPS)
return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
- if (!spec->dac_info[i].dac)
- continue;
+ hda_nid_t dac = spec->dac_info[i].dac;
type = spec->dac_info[i].type;
if (type == AUTO_PIN_LINE_OUT)
type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
idx = num_spk++;
break;
}
- err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+ err = try_add_pb_volume(codec, dac,
spec->dac_info[i].pin,
label, idx);
if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fcb11af..7cabd73 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute)
- return;
spec->jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins);
+ if (!spec->automute)
+ return;
update_speakers(codec);
}
@@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute || !spec->detect_line)
- return;
spec->line_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins);
+ if (!spec->automute || !spec->detect_line)
+ return;
update_speakers(codec);
}
@@ -3083,16 +3083,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
+ hda_nid_t pin, dac;
pin = spec->autocfg.hp_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
- spec->multiout.hp_nid);
+ if (pin) {
+ dac = spec->multiout.hp_nid;
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+ }
pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
- spec->multiout.extra_out_nid[0]);
+ if (pin) {
+ dac = spec->multiout.extra_out_nid[0];
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+ }
}
/*
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c0..5956584 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
switch (params_rate(params)) {
case 48000:
- clk = 12288000;
+ clk = 24576000;
break;
}
@@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
.cpu_dai_name = "bfin-tdm.0",
.codec_dai_name ="ad193x-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad193x.5",
+ .codec_name = "spi0.5",
.ops = &bf5xx_ad193x_ops,
},
{
@@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
.cpu_dai_name = "bfin-tdm.1",
.codec_dai_name ="ad193x-hifi",
.platform_name = "bfin-tdm-pcm-audio",
- .codec_name = "ad193x.5",
+ .codec_name = "spi0.5",
.ops = &bf5xx_ad193x_ops,
},
};
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5..eedb6f5 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -27,11 +27,6 @@ struct ad193x_priv {
int sysclk;
};
-/* ad193x register cache & default register settings */
-static const u8 ad193x_reg[AD193X_NUM_REGS] = {
- 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
/*
* AD193X volume/mute/de-emphasis etc. controls
*/
@@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
- reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+ reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+ | (word_len << AD193X_DAC_WORD_LEN_SHFT);
snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
@@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
.probe = ad193x_probe,
- .reg_cache_default = ad193x_reg,
- .reg_cache_size = AD193X_NUM_REGS,
- .reg_word_size = sizeof(u16),
};
#if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b54..cccc2e8 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
#define AD193X_DAC_LEFT_HIGH (1 << 3)
#define AD193X_DAC_BCLK_INV (1 << 7)
#define AD193X_DAC_CTRL2 0x804
-#define AD193X_DAC_WORD_LEN_MASK 0xC
+#define AD193X_DAC_WORD_LEN_SHFT 3
+#define AD193X_DAC_WORD_LEN_MASK 0x18
#define AD193X_DAC_MASTER_MUTE 1
#define AD193X_DAC_CHNL_MUTE 0x805
#define AD193X_DACL1_MUTE 0
@@ -63,7 +64,7 @@
#define AD193X_ADC_CTRL1 0x80f
#define AD193X_ADC_SERFMT_MASK 0x60
#define AD193X_ADC_SERFMT_STEREO (0 << 5)
-#define AD193X_ADC_SERFMT_TDM (1 << 2)
+#define AD193X_ADC_SERFMT_TDM (1 << 5)
#define AD193X_ADC_SERFMT_AUX (2 << 5)
#define AD193X_ADC_WORD_LEN_MASK 0x3
#define AD193X_ADC_CTRL2 0x810
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 409d89d..fbd7eb9 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+ kfree(sta32x);
return ret;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 60d740e..1725550 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (fll) {
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_ENA, WM8962_FLL_ENA);
if (wm8962->irq) {
@@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
WM8962_BIAS_ENA | 0x180);
msleep(5);
-
- snd_soc_update_bits(codec, WM8962_CLOCKING2,
- WM8962_CLKREG_OVD,
- WM8962_CLKREG_OVD);
}
/* VMID 2*250k */
@@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
WM8962_FLL_ENA, fll1);
@@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec)
*/
snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
+ /* Ensure we have soft control over all registers */
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
if (pdata) {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index ab8e9d1..0cdb9d1 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = {
};
static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text);
+ SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
static const char *hpf_mode_text[] = {
"HiFi", "Custom", "Voice"
@@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
-SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-
-SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux),
+SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux),
SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0),
@@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
{ "AIF2RX0", NULL, "AIFCLK" },
{ "AIF2RX1", NULL, "AIFCLK" },
+ { "AIF1TX0", NULL, "AIFCLK" },
+ { "AIF1TX1", NULL, "AIFCLK" },
+ { "AIF1TX2", NULL, "AIFCLK" },
+ { "AIF1TX3", NULL, "AIFCLK" },
+ { "AIF1TX4", NULL, "AIFCLK" },
+ { "AIF1TX5", NULL, "AIFCLK" },
+
+ { "AIF2TX0", NULL, "AIFCLK" },
+ { "AIF2TX1", NULL, "AIFCLK" },
+
{ "DSP1RXL", NULL, "SYSDSPCLK" },
{ "DSP1RXR", NULL, "SYSDSPCLK" },
{ "DSP2RXL", NULL, "SYSDSPCLK" },
@@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
+ /* Clear any pending completions (eg, from failed startups) */
+ try_wait_for_completion(&wm8996->fll_lock);
+
snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
WM8996_FLL_ENA, WM8996_FLL_ENA);
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 56efa0c..099614e 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
err = -ENODEV;
- goto fail;
+ goto fail_free_info;
}
info->mem = request_mem_region(res->start, resource_size(res),
pdev->name);
if (!info->mem) {
err = -EBUSY;
- goto fail;
+ goto fail_free_info;
}
info->regs = ioremap(info->mem->start, resource_size(info->mem));
@@ -435,6 +435,7 @@ fail_unmap_mem:
iounmap(info->regs);
fail_release_mem:
release_mem_region(info->mem->start, resource_size(info->mem));
+fail_free_info:
kfree(info);
fail:
return err;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 732208c..cb50598 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
* assume that device_node pointers are a valid comparison.
*/
np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
}
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index fd0dc46..5c6c245 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
.pcm_free = &psc_dma_free,
};
-static int mpc5200_hpcd_probe(struct of_device *op)
+static int mpc5200_hpcd_probe(struct platform_device *op)
{
phys_addr_t fifo;
struct psc_dma *psc_dma;
@@ -487,7 +487,7 @@ out_unmap:
return ret;
}
-static int mpc5200_hpcd_remove(struct of_device *op)
+static int mpc5200_hpcd_remove(struct platform_device *op)
{
struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
@@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match);
static struct platform_driver mpc5200_hpcd_of_driver = {
.probe = mpc5200_hpcd_probe,
.remove = mpc5200_hpcd_remove,
- .dev = {
+ .driver = {
.owner = THIS_MODULE,
.name = "mpc5200-pcm-audio",
.of_match_table = mpc5200_hpcd_match,
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a192979..358f0ba 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
- if (!machine_data)
- return -ENOMEM;
+ if (!machine_data) {
+ ret = -ENOMEM;
+ goto error_alloc;
+ }
machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
machine_data->dai[0].ops = &mpc8610_hpcd_ops;
@@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
ret = platform_device_add(sound_device);
if (ret) {
dev_err(&pdev->dev, "platform device add failed\n");
- goto error;
+ goto error_sound;
}
dev_set_drvdata(&pdev->dev, sound_device);
@@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
return 0;
+error_sound:
+ platform_device_unregister(sound_device);
error:
- of_node_put(codec_np);
-
- if (sound_device)
- platform_device_unregister(sound_device);
-
kfree(machine_data);
-
+error_alloc:
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 8fa4d5f..fcb862e 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np,
* dai->platform name should already point to an allocated buffer.
*/
ret = of_address_to_resource(dma_channel_np, 0, &res);
- if (ret)
+ if (ret) {
+ of_node_put(dma_channel_np);
return ret;
+ }
snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
(unsigned long long) res.start, dma_channel_np->name);
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 309c59e..7945625 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -240,7 +240,6 @@ static int ssi_irq = 0;
static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_card *card = rtd->card->snd_card;
struct snd_soc_dai *dai = rtd->cpu_dai;
struct snd_pcm *pcm = rtd->pcm;
int ret;
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index a33fc51..d0bcf3f 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
if (!priv->mem) {
dev_err(&pdev->dev, "request_mem_region failed\n");
err = -EBUSY;
- goto error;
+ goto err_alloc;
}
priv->io = ioremap(priv->mem->start, SZ_16K);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 30fe0d0..0aa475f 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
}
/* Set codec bias level */
- ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
/* Add hook switch - can be used to control the codec from userspace
* even if line discipline fails */
@@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void)
ams_delta_hook_switch_gpios);
/* Keep modem power on */
- ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(&ams_delta_audio_card,
+ &ams_delta_audio_card.rtd[0].codec->dapm,
+ SND_SOC_BIAS_STANDBY);
platform_device_unregister(cx20442_platform_device);
platform_device_unregister(ams_delta_audio_platform_device);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index b99091f..65f980e 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
+ select SND_SOC_WM1250_EV1
config SND_SOC_SPEYSIDE_WM8962
tristate "Audio support for Wolfson Speyside with WM8962"
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 241f55d..c6c6589 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,6 +13,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 1e574a5..bc8c167 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,6 +17,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 0b9eb5f..72535f2 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
@@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_STANDBY:
ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index d9f8ade..20b7f3b 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -203,14 +203,14 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec)
rbnode = rb_entry(node, struct snd_soc_rbtree_node, node);
for (i = 0; i < rbnode->blklen; ++i) {
regtmp = rbnode->base_reg + i;
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, regtmp));
val = snd_soc_rbtree_get_register(rbnode, i);
def = snd_soc_get_cache_val(codec->reg_def_copy, i,
rbnode->word_size);
if (val == def)
continue;
+ WARN_ON(!snd_soc_codec_writable_register(codec, regtmp));
+
codec->cache_bypass = 1;
ret = snd_soc_write(codec, regtmp, val);
codec->cache_bypass = 0;
@@ -563,8 +563,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
lzo_blocks = codec->reg_cache;
for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) {
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, i));
+ WARN_ON(!snd_soc_codec_writable_register(codec, i));
ret = snd_soc_cache_read(codec, i, &val);
if (ret)
return ret;
@@ -823,8 +822,6 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
codec_drv = codec->driver;
for (i = 0; i < codec_drv->reg_cache_size; ++i) {
- WARN_ON(codec->writable_register &&
- codec->writable_register(codec, i));
ret = snd_soc_cache_read(codec, i, &val);
if (ret)
return ret;
@@ -832,6 +829,9 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
if (snd_soc_get_cache_val(codec->reg_def_copy,
i, codec_drv->reg_word_size) == val)
continue;
+
+ WARN_ON(!snd_soc_codec_writable_register(codec, i));
+
ret = snd_soc_write(codec, i, val);
if (ret)
return ret;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca..d2ef014 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1633,7 +1633,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec,
if (codec->readable_register)
return codec->readable_register(codec, reg);
else
- return 0;
+ return 1;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register);
@@ -1651,7 +1651,7 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec,
if (codec->writable_register)
return codec->writable_register(codec, reg);
else
- return 0;
+ return 1;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register);
@@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
if (prefix) {
name_len = strlen(long_name) + strlen(prefix) + 2;
- name = kmalloc(name_len, GFP_ATOMIC);
+ name = kmalloc(name_len, GFP_KERNEL);
if (!name)
return NULL;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7e15914..d67c637 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2763,7 +2763,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
/**
* snd_soc_dapm_free - free dapm resources
- * @card: SoC device
+ * @dapm: DAPM context
*
* Free all dapm widgets and resources.
*/
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c..a62f7dd 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
#define snd_soc_16_8_read_i2c NULL
#endif
+#if defined(CONFIG_SPI_MASTER)
+static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ struct spi_device *spi = codec->control_data;
+
+ const u16 reg = cpu_to_be16(r | 0x100);
+ u8 data;
+ int ret;
+
+ ret = spi_write_then_read(spi, &reg, 2, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_16_8_read_spi NULL
+#endif
+
static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
@@ -295,6 +314,7 @@ static struct {
int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+ unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
} io_types[] = {
{
.addr_bits = 4, .data_bits = 12,
@@ -318,6 +338,7 @@ static struct {
.addr_bits = 16, .data_bits = 8,
.write = snd_soc_16_8_write,
.i2c_read = snd_soc_16_8_read_i2c,
+ .spi_read = snd_soc_16_8_read_spi,
},
{
.addr_bits = 16, .data_bits = 16,
@@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
#ifdef CONFIG_SPI_MASTER
codec->hw_write = do_spi_write;
#endif
+ if (io_types[i].spi_read)
+ codec->hw_read = io_types[i].spi_read;
codec->control_data = container_of(codec->dev,
struct spi_device,
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98..fa31d9c 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
snd_soc_dapm_sync(dapm);
- snd_jack_report(jack->jack, status);
+ snd_jack_report(jack->jack, jack->status);
out:
mutex_unlock(&codec->mutex);
@@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
IRQF_TRIGGER_FALLING,
gpios[i].name,
&gpios[i]);
- if (ret)
+ if (ret < 0)
goto err;
if (gpios[i].wake) {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b575939..2879c88 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
codec_dai->active--;
codec->active--;
+ if (!cpu_dai->active && !codec_dai->active)
+ rtd->rate = 0;
+
/* Muting the DAC suppresses artifacts caused during digital
* shutdown, for example from stopping clocks.
*/
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 661373c..be27f1d 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
/* FIXME: Calculate automatically based on DAPM routes? */
- if (!machine_is_harmony() && !machine_is_ventana())
+ if (!machine_is_harmony())
snd_soc_dapm_nc_pin(dapm, "IN1L");
if (!machine_is_seaboard() && !machine_is_aebl())
snd_soc_dapm_nc_pin(dapm, "IN1R");
@@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, card);
snd_soc_card_set_drvdata(card, machine);
- if (machine_is_harmony() || machine_is_ventana()) {
+ if (machine_is_harmony()) {
card->dapm_routes = harmony_audio_map;
card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
} else if (machine_is_seaboard()) {
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