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July 2008
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08 Jul '08
DAPM contains debug output controlled by a DAPM_DEBUG macro. Change this
to be controlled by the standard DEBUG, dropping the custom dbg() macro
as we go.
Also fix the error printed when configuring an unknown pin to be an
unconditionally displayed error rather than debug output.
Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>
---
sound/soc/soc-dapm.c | 22 ++++++++++------------
1 files changed, 10 insertions(+), 12 deletions(-)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 94296b5..d18ebc6 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -45,13 +45,10 @@
#include <sound/initval.h>
/* debug */
-#define DAPM_DEBUG 0
-#if DAPM_DEBUG
+#ifdef DEBUG
#define dump_dapm(codec, action) dbg_dump_dapm(codec, action)
-#define dbg(format, arg...) printk(format, ## arg)
#else
#define dump_dapm(codec, action)
-#define dbg(format, arg...)
#endif
/* dapm power sequences - make this per codec in the future */
@@ -233,7 +230,8 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
snd_soc_write(codec, widget->reg, new);
pop_wait();
}
- dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change);
+ pr_debug("reg %x old %x new %x change %d\n", widget->reg,
+ old, new, change);
return change;
}
@@ -591,8 +589,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
/* call any power change event handlers */
if (power_change) {
if (w->event) {
- dbg("power %s event for %s flags %x\n",
- w->power ? "on" : "off", w->name, w->event_flags);
+ pr_debug("power %s event for %s flags %x\n",
+ w->power ? "on" : "off", w->name, w->event_flags);
if (power) {
/* power up event */
if (w->event_flags & SND_SOC_DAPM_PRE_PMU) {
@@ -634,7 +632,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
return ret;
}
-#if DAPM_DEBUG
+#ifdef DEBUG
static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
{
struct snd_soc_dapm_widget *w;
@@ -887,13 +885,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
list_for_each_entry(w, &codec->dapm_widgets, list) {
if (!strcmp(w->name, pin)) {
- dbg("dapm: %s: pin %s\n", codec->name, pin);
+ pr_debug("dapm: %s: pin %s\n", codec->name, pin);
w->connected = status;
return 0;
}
}
- dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+ pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
return -EINVAL;
}
@@ -1397,8 +1395,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
{
if (!w->sname)
continue;
- dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname,
- stream, event);
+ pr_debug("widget %s\n %s stream %s event %d\n",
+ w->name, w->sname, stream, event);
if (strstr(w->sname, stream)) {
switch(event) {
case SND_SOC_DAPM_STREAM_START:
--
1.5.6
2
2
An gpio_mask value was defined twice needlessly.
---
Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com>
diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c
index c4f3489..a6d1388 100644
--- a/pci/hda/patch_sigmatel.c
+++ b/pci/hda/patch_sigmatel.c
@@ -3669,7 +3669,6 @@ again:
/* GPIO0 High = EAPD */
spec->gpio_mask = 0x01;
spec->gpio_dir = 0x01;
- spec->gpio_mask = 0x01;
spec->gpio_data = 0x01;
spec->mux_nids = stac92hd71bxx_mux_nids;
1
0
[alsa-devel] [PATCH 01/10] asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08
by Liam Girdwood 07 Jul '08
07 Jul '08
This patch series merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai in preparation for further
ASoC v2 patches.
This merger removes duplication in both DAI structures and simplifies
the API for other users.
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
include/sound/soc.h | 71 ++++++++++++++++---------------------------------
sound/soc/soc-core.c | 50 +++++++++++++++++-----------------
2 files changed, 48 insertions(+), 73 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 340223a..778e57e 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -221,8 +221,7 @@ struct snd_soc_pcm_stream;
struct snd_soc_ops;
struct snd_soc_dai_mode;
struct snd_soc_pcm_runtime;
-struct snd_soc_codec_dai;
-struct snd_soc_cpu_dai;
+struct snd_soc_dai;
struct snd_soc_codec;
struct snd_soc_machine_config;
struct soc_enum;
@@ -317,50 +316,24 @@ struct snd_soc_ops {
/* ASoC DAI ops */
struct snd_soc_dai_ops {
/* DAI clocking configuration */
- int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai,
+ int (*set_sysclk)(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir);
- int (*set_pll)(struct snd_soc_codec_dai *codec_dai,
+ int (*set_pll)(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out);
- int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai,
- int div_id, int div);
+ int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/* DAI format configuration */
- int (*set_fmt)(struct snd_soc_codec_dai *codec_dai,
- unsigned int fmt);
- int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai,
+ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+ int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int mask, int slots);
- int (*set_tristate)(struct snd_soc_codec_dai *, int tristate);
+ int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/* digital mute */
- int (*digital_mute)(struct snd_soc_codec_dai *, int mute);
+ int (*digital_mute)(struct snd_soc_dai *dai, int mute);
};
-/* SoC Codec DAI */
-struct snd_soc_codec_dai {
- char *name;
- int id;
- unsigned char type;
-
- /* DAI capabilities */
- struct snd_soc_pcm_stream playback;
- struct snd_soc_pcm_stream capture;
-
- /* DAI runtime info */
- struct snd_soc_codec *codec;
- unsigned int active;
- unsigned char pop_wait:1;
-
- /* ops */
- struct snd_soc_ops ops;
- struct snd_soc_dai_ops dai_ops;
-
- /* DAI private data */
- void *private_data;
-};
-
-/* SoC CPU DAI */
-struct snd_soc_cpu_dai {
-
+/* SoC DAI (Digital Audio Interface) */
+struct snd_soc_dai {
/* DAI description */
char *name;
unsigned int id;
@@ -368,13 +341,13 @@ struct snd_soc_cpu_dai {
/* DAI callbacks */
int (*probe)(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai);
+ struct snd_soc_dai *dai);
void (*remove)(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai);
+ struct snd_soc_dai *dai);
int (*suspend)(struct platform_device *pdev,
- struct snd_soc_cpu_dai *cpu_dai);
+ struct snd_soc_dai *dai);
int (*resume)(struct platform_device *pdev,
- struct snd_soc_cpu_dai *cpu_dai);
+ struct snd_soc_dai *dai);
/* ops */
struct snd_soc_ops ops;
@@ -386,7 +359,9 @@ struct snd_soc_cpu_dai {
/* DAI runtime info */
struct snd_pcm_runtime *runtime;
- unsigned char active:1;
+ struct snd_soc_codec *codec;
+ unsigned int active;
+ unsigned char pop_wait:1;
void *dma_data;
/* DAI private data */
@@ -428,7 +403,7 @@ struct snd_soc_codec {
struct delayed_work delayed_work;
/* codec DAI's */
- struct snd_soc_codec_dai *dai;
+ struct snd_soc_dai *dai;
unsigned int num_dai;
};
@@ -447,12 +422,12 @@ struct snd_soc_platform {
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
int (*suspend)(struct platform_device *pdev,
- struct snd_soc_cpu_dai *cpu_dai);
+ struct snd_soc_dai *dai);
int (*resume)(struct platform_device *pdev,
- struct snd_soc_cpu_dai *cpu_dai);
+ struct snd_soc_dai *dai);
/* pcm creation and destruction */
- int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *,
+ int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
struct snd_pcm *);
void (*pcm_free)(struct snd_pcm *);
@@ -466,8 +441,8 @@ struct snd_soc_dai_link {
char *stream_name; /* Stream name */
/* DAI */
- struct snd_soc_codec_dai *codec_dai;
- struct snd_soc_cpu_dai *cpu_dai;
+ struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *cpu_dai;
/* machine stream operations */
struct snd_soc_ops *ops;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bdbbc6a..4d626b4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -134,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
- struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
- struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret = 0;
mutex_lock(&pcm_mutex);
@@ -272,7 +272,7 @@ static void close_delayed_work(struct work_struct *work)
struct snd_soc_device *socdev =
container_of(work, struct snd_soc_device, delayed_work.work);
struct snd_soc_codec *codec = socdev->codec;
- struct snd_soc_codec_dai *codec_dai;
+ struct snd_soc_dai *codec_dai;
int i;
mutex_lock(&pcm_mutex);
@@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
- struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
- struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
mutex_lock(&pcm_mutex);
@@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
- struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
- struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
int ret = 0;
@@ -489,8 +489,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
- struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
- struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret = 0;
mutex_lock(&pcm_mutex);
@@ -559,8 +559,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
- struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
- struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
struct snd_soc_codec *codec = socdev->codec;
mutex_lock(&pcm_mutex);
@@ -594,8 +594,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_dai_link *machine = rtd->dai;
struct snd_soc_platform *platform = socdev->platform;
- struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
- struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
int ret;
if (codec_dai->ops.trigger) {
@@ -651,7 +651,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
/* mute any active DAC's */
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+ struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
if (dai->dai_ops.digital_mute && dai->playback.active)
dai->dai_ops.digital_mute(dai, 1);
}
@@ -664,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
machine->suspend_pre(pdev, state);
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
cpu_dai->suspend(pdev, cpu_dai);
if (platform->suspend)
@@ -690,7 +690,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
codec_dev->suspend(pdev, state);
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
cpu_dai->suspend(pdev, cpu_dai);
}
@@ -726,7 +726,7 @@ static void soc_resume_deferred(struct work_struct *work)
machine->resume_pre(pdev);
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
cpu_dai->resume(pdev, cpu_dai);
}
@@ -747,13 +747,13 @@ static void soc_resume_deferred(struct work_struct *work)
/* unmute any active DACs */
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+ struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
if (dai->dai_ops.digital_mute && dai->playback.active)
dai->dai_ops.digital_mute(dai, 0);
}
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
cpu_dai->resume(pdev, cpu_dai);
if (platform->resume)
@@ -803,7 +803,7 @@ static int soc_probe(struct platform_device *pdev)
}
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->probe) {
ret = cpu_dai->probe(pdev, cpu_dai);
if (ret < 0)
@@ -838,7 +838,7 @@ platform_err:
cpu_dai_err:
for (i--; i >= 0; i--) {
- struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
@@ -867,7 +867,7 @@ static int soc_remove(struct platform_device *pdev)
codec_dev->remove(pdev);
for (i = 0; i < machine->num_links; i++) {
- struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
if (cpu_dai->remove)
cpu_dai->remove(pdev, cpu_dai);
}
@@ -895,8 +895,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
struct snd_soc_dai_link *dai_link, int num)
{
struct snd_soc_codec *codec = socdev->codec;
- struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai;
- struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai;
+ struct snd_soc_dai *codec_dai = dai_link->codec_dai;
+ struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
struct snd_soc_pcm_runtime *rtd;
struct snd_pcm *pcm;
char new_name[64];
@@ -1211,7 +1211,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
#ifdef CONFIG_SND_SOC_AC97_BUS
- struct snd_soc_codec_dai *codec_dai;
+ struct snd_soc_dai *codec_dai;
int i;
#endif
--
1.5.4.2
2
1
07 Jul '08
Refactored snd_soc_dapm_set_endpoint() to snd_soc_dapm_enable_pin() and
snd_soc_dapm_disable_pin().
Renamed snd_soc_dapm_sync_endpoints() to snd_soc_dapm_sync().
Renamed snd_soc_dapm_get_endpoint_status() to
snd_soc_dapm_get_pin_status().
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
include/sound/soc-dapm.h | 11 ++--
sound/soc/at91/eti_b1_wm8731.c | 10 ++--
sound/soc/codecs/tlv320aic3x.c | 4 +-
sound/soc/davinci/davinci-evm.c | 16 +++---
sound/soc/omap/n810.c | 21 ++++--
sound/soc/pxa/corgi.c | 42 +++++++------
sound/soc/pxa/poodle.c | 24 ++++----
sound/soc/pxa/spitz.c | 62 ++++++++++----------
sound/soc/pxa/tosa.c | 30 +++++----
sound/soc/s3c24xx/neo1973_wm8753.c | 116 ++++++++++++++++++------------------
sound/soc/sh/exmmb-ac97.c | 2 +-
sound/soc/sh/exmmb-i2s.c | 2 +-
sound/soc/sh/sh7760-ac97.c | 2 +-
sound/soc/soc-dapm.c | 81 ++++++++++++++++---------
14 files changed, 227 insertions(+), 196 deletions(-)
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index f8223fa..64adbe5 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -227,12 +227,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
-/* dapm audio endpoint control */
-int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
- char *pin, int status);
-int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec,
- char *pin);
-int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec);
+/* dapm audio pin control and status */
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_sync(struct snd_soc_codec *codec);
/* dapm widget types */
enum snd_soc_dapm_type {
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index 83614dd..c8b6bc2 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -216,14 +216,14 @@ static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
/* not connected */
- snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
- snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+ snd_soc_dapm_disable_pin(codec, "RLINEIN");
+ snd_soc_dapm_disable_pin(codec, "LLINEIN");
/* always connected */
- snd_soc_dapm_set_endpoint(codec, "Int Mic", 1);
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+ snd_soc_dapm_enable_pin(codec, "Int Mic");
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 528c26a..b735ddb 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -29,7 +29,7 @@
* ---------------------------------------
*
* Hence the machine layer should disable unsupported inputs/outputs by
- * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc.
+ * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc.
*/
#include <linux/module.h>
@@ -206,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
}
if (found)
- snd_soc_dapm_sync_endpoints(widget->codec);
+ snd_soc_dapm_sync(widget->codec);
}
ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 4c70a0e..091eae3 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -103,17 +103,17 @@ static int evm_aic3x_init(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* not connected */
- snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
- snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
- snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+ snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
+ snd_soc_dapm_disable_pin(codec, "HPLCOM");
+ snd_soc_dapm_disable_pin(codec, "HPRCOM");
/* always connected */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
- snd_soc_dapm_set_endpoint(codec, "Line Out", 1);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
- snd_soc_dapm_set_endpoint(codec, "Line In", 1);
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Out");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index c168a64..182d690 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -49,10 +49,17 @@ static int n810_jack_func;
static void n810_ext_control(struct snd_soc_codec *codec)
{
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func);
+ if (n810_spk_func)
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
+
+ if (n810_jack_func)
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int n810_startup(struct snd_pcm_substream *substream)
@@ -204,9 +211,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
int i, err;
/* Not connected */
- snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
- snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
- snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+ snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
+ snd_soc_dapm_disable_pin(codec, "HPLCOM");
+ snd_soc_dapm_disable_pin(codec, "HPRCOM");
/* Add N810 specific controls */
for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
@@ -223,7 +230,7 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
/* Set up N810 specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index edeea63..db18ef6 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -50,47 +50,51 @@ static int corgi_spk_func;
static void corgi_ext_control(struct snd_soc_codec *codec)
{
- int spk = 0, mic = 0, line = 0, hp = 0, hs = 0;
-
/* set up jack connection */
switch (corgi_jack_func) {
case CORGI_HP:
- hp = 1;
/* set = unmute headphone */
set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_MIC:
- mic = 1;
/* reset = mute headphone */
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_LINE:
- line = 1;
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case CORGI_HEADSET:
- hs = 1;
- mic = 1;
reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
break;
}
if (corgi_spk_func == CORGI_SPK_ON)
- spk = 1;
-
- /* set the enpoints to their new connetion states */
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", line);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int corgi_startup(struct snd_pcm_substream *substream)
@@ -285,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
{
int i, err;
- snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
- snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+ snd_soc_dapm_disable_pin(codec, "LLINEIN");
+ snd_soc_dapm_disable_pin(codec, "RLINEIN");
/* Add corgi specific controls */
for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
@@ -303,7 +307,7 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
/* Set up corgi specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 810f1fe..36cbf69 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -48,8 +48,6 @@ static int poodle_spk_func;
static void poodle_ext_control(struct snd_soc_codec *codec)
{
- int spk = 0;
-
/* set up jack connection */
if (poodle_jack_func == POODLE_HP) {
/* set = unmute headphone */
@@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
POODLE_LOCOMO_GPIO_MUTE_L, 1);
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_R, 1);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
} else {
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_L, 0);
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_R, 0);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
}
- if (poodle_spk_func == POODLE_SPK_ON)
- spk = 1;
-
/* set the enpoints to their new connetion states */
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
+ if (poodle_spk_func == POODLE_SPK_ON)
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int poodle_startup(struct snd_pcm_substream *substream)
@@ -248,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
{
int i, err;
- snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
- snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
- snd_soc_dapm_set_endpoint(codec, "MICIN", 1);
+ snd_soc_dapm_disable_pin(codec, "LLINEIN");
+ snd_soc_dapm_disable_pin(codec, "RLINEIN");
+ snd_soc_dapm_enable_pin(codec, "MICIN");
/* Add poodle specific controls */
for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
@@ -267,7 +265,7 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
/* Set up poodle specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 092b5c7..ec18163 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -51,60 +51,60 @@ static int spitz_spk_func;
static void spitz_ext_control(struct snd_soc_codec *codec)
{
if (spitz_spk_func == SPITZ_SPK_ON)
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
else
- snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0);
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
/* set up jack connection */
switch (spitz_jack_func) {
case SPITZ_HP:
/* enable and unmute hp jack, disable mic bias */
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
case SPITZ_HP_OFF:
/* jack removed, everything off */
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
- snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
break;
}
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int spitz_startup(struct snd_pcm_substream *substream)
@@ -291,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
int i, err;
/* NC codec pins */
- snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0);
- snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0);
- snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0);
- snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "MONO", 0);
+ snd_soc_dapm_disable_pin(codec, "RINPUT1");
+ snd_soc_dapm_disable_pin(codec, "LINPUT2");
+ snd_soc_dapm_disable_pin(codec, "RINPUT2");
+ snd_soc_dapm_disable_pin(codec, "LINPUT3");
+ snd_soc_dapm_disable_pin(codec, "RINPUT3");
+ snd_soc_dapm_disable_pin(codec, "OUT3");
+ snd_soc_dapm_disable_pin(codec, "MONO");
/* Add spitz specific controls */
for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
@@ -314,7 +314,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
/* Set up spitz specific audio paths */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 465ff0f..dba7689 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -52,29 +52,31 @@ static int tosa_spk_func;
static void tosa_ext_control(struct snd_soc_codec *codec)
{
- int spk = 0, mic_int = 0, hp = 0, hs = 0;
-
/* set up jack connection */
switch (tosa_jack_func) {
case TOSA_HP:
- hp = 1;
+ snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case TOSA_MIC_INT:
- mic_int = 1;
+ snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
break;
case TOSA_HEADSET:
- hs = 1;
+ snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
break;
}
if (tosa_spk_func == TOSA_SPK_ON)
- spk = 1;
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
- snd_soc_dapm_set_endpoint(codec, "Speaker", spk);
- snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int);
- snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
- snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
}
static int tosa_startup(struct snd_pcm_substream *substream)
@@ -191,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
{
int i, err;
- snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0);
+ snd_soc_dapm_disable_pin(codec, "OUT3");
+ snd_soc_dapm_disable_pin(codec, "MONOOUT");
/* add tosa specific controls */
for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
@@ -209,7 +211,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
/* set up tosa specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 3485123..f053e85 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -250,77 +250,77 @@ static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
switch (neo1973_scenario) {
case NEO_AUDIO_OFF:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_HANDSET:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_HEADSET:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line In");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_BLUETOOTH:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_STEREO_TO_SPEAKERS:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_STEREO_TO_HEADPHONES:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_CAPTURE_HANDSET:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Call Mic");
break;
case NEO_CAPTURE_HEADSET:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_CAPTURE_BLUETOOTH:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
default:
- snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
- snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
- snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
- snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
}
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
@@ -511,12 +511,12 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
DBG("Entered %s\n", __func__);
/* set up NC codec pins */
- snd_soc_dapm_set_endpoint(codec, "LOUT2", 0);
- snd_soc_dapm_set_endpoint(codec, "ROUT2", 0);
- snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
- snd_soc_dapm_set_endpoint(codec, "OUT4", 0);
- snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
- snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
+ snd_soc_dapm_disable_pin(codec, "LOUT2");
+ snd_soc_dapm_disable_pin(codec, "ROUT2");
+ snd_soc_dapm_disable_pin(codec, "OUT3");
+ snd_soc_dapm_disable_pin(codec, "OUT4");
+ snd_soc_dapm_disable_pin(codec, "LINE1");
+ snd_soc_dapm_disable_pin(codec, "LINE2");
/* set endpoints to default mode */
@@ -539,7 +539,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
err = snd_soc_dapm_add_routes(codec, dapm_routes,
ARRAY_SIZE(dapm_routes));
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/sh/exmmb-ac97.c b/sound/soc/sh/exmmb-ac97.c
index e8d710f..b0b4a20 100644
--- a/sound/soc/sh/exmmb-ac97.c
+++ b/sound/soc/sh/exmmb-ac97.c
@@ -46,7 +46,7 @@ static int exm7760_ac97_machine_init(struct snd_soc_codec *codec)
{
MSG("machine_init() enter\n");
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
MSG("machine_init() leave\n");
return 0;
diff --git a/sound/soc/sh/exmmb-i2s.c b/sound/soc/sh/exmmb-i2s.c
index db1341e..f253f94 100644
--- a/sound/soc/sh/exmmb-i2s.c
+++ b/sound/soc/sh/exmmb-i2s.c
@@ -116,7 +116,7 @@ static int exm7760_i2s_machine_init(struct snd_soc_codec *codec)
cs4251x_gpio_mode(codec, CS4251X_RXP_6,
CS4251X_GPIO_MODE_GPOLOW, 0, 0);
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 2f91de8..846d1b3 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -25,7 +25,7 @@ extern struct snd_soc_platform sh7760_soc_platform;
static int machine_init(struct snd_soc_codec *codec)
{
- snd_soc_dapm_sync_endpoints(codec);
+ snd_soc_dapm_sync(codec);
return 0;
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 9e83357..0813c08 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -857,8 +857,25 @@ static void dapm_free_widgets(struct snd_soc_codec *codec)
}
}
+static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
+ char *pin, int status)
+{
+ struct snd_soc_dapm_widget *w;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ if (!strcmp(w->name, pin)) {
+ dbg("dapm: %s: pin %s\n", codec->name, pin);
+ w->connected = status;
+ return 0;
+ }
+ }
+
+ dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+ return -EINVAL;
+}
+
/**
- * snd_soc_dapm_sync_endpoints - scan and power dapm paths
+ * snd_soc_dapm_sync - scan and power dapm paths
* @codec: audio codec
*
* Walks all dapm audio paths and powers widgets according to their
@@ -866,11 +883,11 @@ static void dapm_free_widgets(struct snd_soc_codec *codec)
*
* Returns 0 for success.
*/
-int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec)
+int snd_soc_dapm_sync(struct snd_soc_codec *codec)
{
return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints);
+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
const char *sink, const char *control, const char *source)
@@ -1418,53 +1435,57 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
}
/**
- * snd_soc_dapm_set_endpoint - set audio endpoint status
- * @codec: audio codec
- * @endpoint: audio signal endpoint (or start point)
- * @status: point status
- *
- * Set audio endpoint status - connected or disconnected.
+ * snd_soc_dapm_enable_pin - enable pin.
+ * @snd_soc_codec: SoC codec
+ * @pin: pin name
*
- * Returns 0 for success else error.
+ * Enables input/output pin and it's parents or children widgets iff there is
+ * a valid audio route and active audio stream.
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
*/
-int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
- char *endpoint, int status)
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin)
{
- struct snd_soc_dapm_widget *w;
-
- list_for_each_entry(w, &codec->dapm_widgets, list) {
- if (!strcmp(w->name, endpoint)) {
- w->connected = status;
- return 0;
- }
- }
+ return snd_soc_dapm_set_pin(codec, pin, 1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
- return -ENODEV;
+/**
+ * snd_soc_dapm_disable_pin - disable pin.
+ * @codec: SoC codec
+ * @pin: pin name
+ *
+ * Disables input/output pin and it's parents or children widgets.
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
+{
+ return snd_soc_dapm_set_pin(codec, pin, 0);
}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
+EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
/**
- * snd_soc_dapm_get_endpoint_status - get audio endpoint status
+ * snd_soc_dapm_get_pin_status - get audio pin status
* @codec: audio codec
- * @endpoint: audio signal endpoint (or start point)
+ * @pin: audio signal pin endpoint (or start point)
*
- * Get audio endpoint status - connected or disconnected.
+ * Get audio pin status - connected or disconnected.
*
- * Returns status
+ * Returns 1 for connected otherwise 0.
*/
-int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec,
- char *endpoint)
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin)
{
struct snd_soc_dapm_widget *w;
list_for_each_entry(w, &codec->dapm_widgets, list) {
- if (!strcmp(w->name, endpoint))
+ if (!strcmp(w->name, pin))
return w->connected;
}
return 0;
}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status);
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
/**
* snd_soc_dapm_free - free dapm resources
--
1.5.4.2
2
4
[alsa-devel] [PATCH 10/10] asoc: sh - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08
by Liam Girdwood 07 Jul '08
07 Jul '08
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the SuperH platform.
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
sound/soc/sh/dma-sh7760.c | 2 +-
sound/soc/sh/hac.c | 2 +-
sound/soc/sh/sh7760-ac97.c | 2 +-
sound/soc/sh/ssi.c | 8 ++++----
4 files changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 7a3ce80..9faa126 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -326,7 +326,7 @@ static void camelot_pcm_free(struct snd_pcm *pcm)
}
static int camelot_pcm_new(struct snd_card *card,
- struct snd_soc_codec_dai *dai,
+ struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
/* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index b7b676b..df7bc34 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream,
#define AC97_FMTS \
SNDRV_PCM_FMTBIT_S16_LE
-struct snd_soc_cpu_dai sh4_hac_dai[] = {
+struct snd_soc_dai sh4_hac_dai[] = {
{
.name = "HAC0",
.id = 0,
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 846d1b3..92bfaf4 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -20,7 +20,7 @@
#define IPSEL 0xFE400034
/* platform specific structs can be declared here */
-extern struct snd_soc_cpu_dai sh4_hac_dai[2];
+extern struct snd_soc_dai sh4_hac_dai[2];
extern struct snd_soc_platform sh7760_soc_platform;
static int machine_init(struct snd_soc_codec *codec)
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 3388bc3..55c3464 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -208,7 +208,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id,
+static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
unsigned int freq, int dir)
{
struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id];
@@ -222,7 +222,7 @@ static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id,
* This divider is used to generate the SSI_SCK (I2S bitclock) from the
* clock at the HAC_BIT_CLK ("oversampling clock") pin.
*/
-static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div)
+static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div)
{
struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
unsigned long ssicr;
@@ -245,7 +245,7 @@ static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div)
return 0;
}
-static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt)
+static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
unsigned long ssicr = SSIREG(SSICR);
@@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt)
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
-struct snd_soc_cpu_dai sh4_ssi_dai[] = {
+struct snd_soc_dai sh4_ssi_dai[] = {
{
.name = "SSI0",
.id = 0,
--
1.5.4.2
1
0
[alsa-devel] [PATCH 09/10] asoc: s3c24xx - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08
by Liam Girdwood 07 Jul '08
07 Jul '08
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the S3C24xx platform.
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
sound/soc/s3c24xx/neo1973_wm8753.c | 12 ++++++------
sound/soc/s3c24xx/s3c2412-i2s.c | 14 +++++++-------
sound/soc/s3c24xx/s3c2412-i2s.h | 2 +-
sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++-----
sound/soc/s3c24xx/s3c24xx-ac97.h | 2 +-
sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +++++++-------
sound/soc/s3c24xx/s3c24xx-i2s.h | 2 +-
sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +-
8 files changed, 29 insertions(+), 29 deletions(-)
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index f053e85..51a4ce3 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -66,8 +66,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int pll_out = 0, bclk = 0;
int ret = 0;
unsigned long iis_clkrate;
@@ -156,7 +156,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
DBG("Entered %s\n", __func__);
@@ -176,7 +176,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
unsigned int pcmdiv = 0;
int ret = 0;
unsigned long iis_clkrate;
@@ -222,7 +222,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
DBG("Entered %s\n", __func__);
@@ -546,7 +546,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
/*
* BT Codec DAI
*/
-static struct snd_soc_cpu_dai bt_dai = {
+static struct snd_soc_dai bt_dai = {
.name = "Bluetooth",
.id = 0,
.type = SND_SOC_DAI_PCM,
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index c463a82..ee4676e 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -295,7 +295,7 @@ static inline int s3c2412_snd_is_clkmaster(void)
/*
* Set S3C2412 I2S DAI format
*/
-static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
u32 iismod;
@@ -500,7 +500,7 @@ EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
/*
* Set S3C2412 Clock source
*/
-static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
@@ -528,7 +528,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
/*
* Set S3C2412 Clock dividers
*/
-static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
int div_id, int div)
{
struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
@@ -602,7 +602,7 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
static int s3c2412_i2s_probe(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
DBG("Entered %s\n", __func__);
@@ -648,7 +648,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
#ifdef CONFIG_PM
static int s3c2412_i2s_suspend(struct platform_device *dev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
u32 iismod;
@@ -676,7 +676,7 @@ static int s3c2412_i2s_suspend(struct platform_device *dev,
}
static int s3c2412_i2s_resume(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
@@ -708,7 +708,7 @@ static int s3c2412_i2s_resume(struct platform_device *pdev,
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-struct snd_soc_cpu_dai s3c2412_i2s_dai = {
+struct snd_soc_dai s3c2412_i2s_dai = {
.name = "s3c2412-i2s",
.id = 0,
.type = SND_SOC_DAI_I2S,
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
index 27f48e1..aac08a2 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.h
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -24,7 +24,7 @@
extern struct clk *s3c2412_get_iisclk(void);
-extern struct snd_soc_cpu_dai s3c2412_i2s_dai;
+extern struct snd_soc_dai s3c2412_i2s_dai;
struct s3c2412_rate_calc {
unsigned int clk_div; /* for prescaler */
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 533565b..783349b 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -210,7 +210,7 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = {
};
static int s3c2443_ac97_probe(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
int ret;
u32 ac_glbctrl;
@@ -262,7 +262,7 @@ static int s3c2443_ac97_probe(struct platform_device *pdev,
}
static void s3c2443_ac97_remove(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
free_irq(IRQ_S3C244x_AC97, NULL);
clk_disable(s3c24xx_ac97.ac97_clk);
@@ -274,7 +274,7 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out;
@@ -316,7 +316,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
@@ -352,7 +352,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
-struct snd_soc_cpu_dai s3c2443_ac97_dai[] = {
+struct snd_soc_dai s3c2443_ac97_dai[] = {
{
.name = "s3c2443-ac97",
.id = 0,
diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h
index bf03e8e..a96dcad 100644
--- a/sound/soc/s3c24xx/s3c24xx-ac97.h
+++ b/sound/soc/s3c24xx/s3c24xx-ac97.h
@@ -26,6 +26,6 @@
#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97
#endif
-extern struct snd_soc_cpu_dai s3c2443_ac97_dai[];
+extern struct snd_soc_dai s3c2443_ac97_dai[];
#endif /*S3C24XXAC97_H_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 42e96b5..3975242 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -205,7 +205,7 @@ static inline int s3c24xx_snd_is_clkmaster(void)
/*
* Set S3C24xx I2S DAI format
*/
-static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
u32 iismod;
@@ -313,7 +313,7 @@ exit_err:
/*
* Set S3C24xx Clock source
*/
-static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -339,7 +339,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
/*
* Set S3C24xx Clock dividers
*/
-static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
int div_id, int div)
{
u32 reg;
@@ -378,7 +378,7 @@ u32 s3c24xx_i2s_get_clockrate(void)
EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
static int s3c24xx_i2s_probe(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
DBG("Entered %s\n", __func__);
@@ -411,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
#ifdef CONFIG_PM
static int s3c24xx_i2s_suspend(struct platform_device *pdev,
- struct snd_soc_cpu_dai *cpu_dai)
+ struct snd_soc_dai *cpu_dai)
{
DBG("Entered %s\n", __func__);
@@ -426,7 +426,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev,
}
static int s3c24xx_i2s_resume(struct platform_device *pdev,
- struct snd_soc_cpu_dai *cpu_dai)
+ struct snd_soc_dai *cpu_dai)
{
DBG("Entered %s\n", __func__);
clk_enable(s3c24xx_i2s.iis_clk);
@@ -449,7 +449,7 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev,
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-struct snd_soc_cpu_dai s3c24xx_i2s_dai = {
+struct snd_soc_dai s3c24xx_i2s_dai = {
.name = "s3c24xx-i2s",
.id = 0,
.type = SND_SOC_DAI_I2S,
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h
index 537b4ec..726d91c 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.h
@@ -32,6 +32,6 @@
u32 s3c24xx_i2s_get_clockrate(void);
-extern struct snd_soc_cpu_dai s3c24xx_i2s_dai;
+extern struct snd_soc_dai s3c24xx_i2s_dai;
#endif /*S3C24XXI2S_H_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index ef59974..cef79b3 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -429,7 +429,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK;
static int s3c24xx_pcm_new(struct snd_card *card,
- struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
+ struct snd_soc_dai *dai, struct snd_pcm *pcm)
{
int ret = 0;
--
1.5.4.2
1
0
[alsa-devel] [PATCH 08/10] asoc: pxa - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08
by Liam Girdwood 07 Jul '08
07 Jul '08
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the PXA platform.
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
sound/soc/pxa/corgi.c | 4 ++--
sound/soc/pxa/poodle.c | 4 ++--
sound/soc/pxa/pxa2xx-ac97.c | 14 +++++++-------
sound/soc/pxa/pxa2xx-ac97.h | 2 +-
sound/soc/pxa/pxa2xx-i2s.c | 14 +++++++-------
sound/soc/pxa/pxa2xx-i2s.h | 2 +-
sound/soc/pxa/pxa2xx-pcm.c | 2 +-
sound/soc/pxa/spitz.c | 4 ++--
8 files changed, 23 insertions(+), 23 deletions(-)
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index db18ef6..782afbf 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -123,8 +123,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int clk = 0;
int ret = 0;
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 36cbf69..ce25b6b 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -102,8 +102,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int clk = 0;
int ret = 0;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index cb94795..b458667 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -283,7 +283,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
#ifdef CONFIG_PM
static int pxa2xx_ac97_suspend(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
GCR |= GCR_ACLINK_OFF;
clk_disable(ac97_clk);
@@ -291,7 +291,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev,
}
static int pxa2xx_ac97_resume(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
@@ -311,7 +311,7 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev,
#endif
static int pxa2xx_ac97_probe(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
int ret;
@@ -373,7 +373,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
@@ -387,7 +387,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
@@ -401,7 +401,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
@@ -419,7 +419,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
* There is only 1 physical AC97 interface for pxa2xx, but it
* has extra fifo's that can be used for aux DACs and ADCs.
*/
-struct snd_soc_cpu_dai pxa_ac97_dai[] = {
+struct snd_soc_dai pxa_ac97_dai[] = {
{
.name = "pxa2xx-ac97",
.id = 0,
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
index b8ccfee..e390de8 100644
--- a/sound/soc/pxa/pxa2xx-ac97.h
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -14,7 +14,7 @@
#define PXA2XX_DAI_AC97_AUX 1
#define PXA2XX_DAI_AC97_MIC 2
-extern struct snd_soc_cpu_dai pxa_ac97_dai[3];
+extern struct snd_soc_dai pxa_ac97_dai[3];
/* platform data */
extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 35090c2..9c06553 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -77,7 +77,7 @@ static struct pxa2xx_gpio gpio_bus[] = {
static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
if (!cpu_dai->active) {
SACR0 |= SACR0_RST;
@@ -98,7 +98,7 @@ static int pxa_i2s_wait(void)
return 0;
}
-static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
/* interface format */
@@ -124,7 +124,7 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
return 0;
}
-static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
if (clk_id != PXA2XX_I2S_SYSCLK)
@@ -140,7 +140,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
@@ -237,7 +237,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
#ifdef CONFIG_PM
static int pxa2xx_i2s_suspend(struct platform_device *dev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
if (!dai->active)
return 0;
@@ -255,7 +255,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
}
static int pxa2xx_i2s_resume(struct platform_device *pdev,
- struct snd_soc_cpu_dai *dai)
+ struct snd_soc_dai *dai)
{
if (!dai->active)
return 0;
@@ -280,7 +280,7 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev,
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
-struct snd_soc_cpu_dai pxa_i2s_dai = {
+struct snd_soc_dai pxa_i2s_dai = {
.name = "pxa2xx-i2s",
.id = 0,
.type = SND_SOC_DAI_I2S,
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
index 4435bd9..e2def44 100644
--- a/sound/soc/pxa/pxa2xx-i2s.h
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -15,6 +15,6 @@
/* I2S clock */
#define PXA2XX_I2S_SYSCLK 0
-extern struct snd_soc_cpu_dai pxa_i2s_dai;
+extern struct snd_soc_dai pxa_i2s_dai;
#endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 01ad7bf..2df03ee 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK;
-int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
int ret = 0;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index ec18163..fd1abc7 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -121,8 +121,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int clk = 0;
int ret = 0;
--
1.5.4.2
1
0
[alsa-devel] [PATCH 07/10] asoc: omap - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08
by Liam Girdwood 07 Jul '08
07 Jul '08
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the Omap platform.
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
sound/soc/omap/n810.c | 4 ++--
sound/soc/omap/omap-mcbsp.c | 16 ++++++++--------
sound/soc/omap/omap-mcbsp.h | 2 +-
sound/soc/omap/omap-pcm.c | 2 +-
4 files changed, 12 insertions(+), 12 deletions(-)
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 74f4599..d1233c0 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -86,8 +86,8 @@ static int n810_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int err;
/* Set codec DAI configuration */
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 40d87e6..00b0c9d 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -103,7 +103,7 @@ static const unsigned long omap2420_mcbsp_port[][2] = {};
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
int err = 0;
@@ -116,7 +116,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
if (!cpu_dai->active) {
@@ -128,7 +128,7 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
int err = 0;
@@ -157,7 +157,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
@@ -223,7 +223,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
* This must be called before _set_clkdiv and _set_sysclk since McBSP register
* cache is initialized here
*/
-static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
@@ -292,7 +292,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
return 0;
}
-static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
int div_id, int div)
{
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
@@ -347,7 +347,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
return 0;
}
-static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq,
int dir)
{
@@ -376,7 +376,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
return err;
}
-struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = {
+struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = {
{
.name = "omap-mcbsp-dai",
.id = 0,
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 9965fd4..ed8afb5 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -44,6 +44,6 @@ enum omap_mcbsp_div {
*/
#define NUM_LINKS 1
-extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS];
+extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
#endif
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 6237020..e092f3d 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -316,7 +316,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
}
}
-int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
int ret = 0;
--
1.5.4.2
1
0
[alsa-devel] [PATCH 06/10] asoc: fsl - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08
by Liam Girdwood 07 Jul '08
07 Jul '08
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the Freescale PPC platform.
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
sound/soc/fsl/fsl_dma.c | 2 +-
sound/soc/fsl/fsl_dma.h | 2 +-
sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++------------
sound/soc/fsl/fsl_ssi.h | 4 ++--
sound/soc/fsl/mpc8610_hpcd.c | 4 ++--
5 files changed, 18 insertions(+), 18 deletions(-)
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 78de716..da2bc59 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -282,7 +282,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
* once for each .dai_link in the machine driver's snd_soc_machine
* structure.
*/
-static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
static u64 fsl_dma_dmamask = DMA_BIT_MASK(32);
diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h
index 430a6ce..385d4a4 100644
--- a/sound/soc/fsl/fsl_dma.h
+++ b/sound/soc/fsl/fsl_dma.h
@@ -126,7 +126,7 @@ struct fsl_dma_link_descriptor {
u8 res[4]; /* Reserved */
} __attribute__ ((aligned(32), packed));
-/* DMA information needed to create a snd_soc_cpu_dai object
+/* DMA information needed to create a snd_soc_dai object
*
* ssi_stx_phys: bus address of SSI STX register to use
* ssi_srx_phys: bus address of SSI SRX register to use
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index f588545..71bff33 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -82,7 +82,7 @@ struct fsl_ssi_private {
struct device *dev;
unsigned int playback;
unsigned int capture;
- struct snd_soc_cpu_dai cpu_dai;
+ struct snd_soc_dai cpu_dai;
struct device_attribute dev_attr;
struct {
@@ -479,7 +479,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
* @freq: the frequency of the given clock ID, currently ignored
* @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
*/
-static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
@@ -497,7 +497,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
*
* @format: one of SND_SOC_DAIFMT_xxx
*/
-static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
+static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
{
return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
}
@@ -505,7 +505,7 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
/**
* fsl_ssi_dai_template: template CPU DAI for the SSI
*/
-static struct snd_soc_cpu_dai fsl_ssi_dai_template = {
+static struct snd_soc_dai fsl_ssi_dai_template = {
.playback = {
/* The SSI does not support monaural audio. */
.channels_min = 2,
@@ -569,15 +569,15 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev,
}
/**
- * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure
+ * fsl_ssi_create_dai: create a snd_soc_dai structure
*
- * This function is called by the machine driver to create a snd_soc_cpu_dai
+ * This function is called by the machine driver to create a snd_soc_dai
* structure. The function creates an ssi_private object, which contains
- * the snd_soc_cpu_dai. It also creates the sysfs statistics device.
+ * the snd_soc_dai. It also creates the sysfs statistics device.
*/
-struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
+struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
{
- struct snd_soc_cpu_dai *fsl_ssi_dai;
+ struct snd_soc_dai *fsl_ssi_dai;
struct fsl_ssi_private *ssi_private;
int ret = 0;
struct device_attribute *dev_attr;
@@ -588,7 +588,7 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
return NULL;
}
memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template,
- sizeof(struct snd_soc_cpu_dai));
+ sizeof(struct snd_soc_dai));
fsl_ssi_dai = &ssi_private->cpu_dai;
dev_attr = &ssi_private->dev_attr;
@@ -623,11 +623,11 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
EXPORT_SYMBOL_GPL(fsl_ssi_create_dai);
/**
- * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object
+ * fsl_ssi_destroy_dai: destroy the snd_soc_dai object
*
* This function undoes the operations of fsl_ssi_create_dai()
*/
-void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai)
+void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
{
struct fsl_ssi_private *ssi_private =
container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai);
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index c5ce88e..83b44d7 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -217,8 +217,8 @@ struct fsl_ssi_info {
struct device *dev;
};
-struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
-void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai);
+struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
+void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai);
#endif
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 8820c3f..59d7e49 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -96,8 +96,8 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device)
static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
- struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct mpc8610_hpcd_data *machine_data =
rtd->socdev->dev->platform_data;
int ret = 0;
--
1.5.4.2
1
0
[alsa-devel] [PATCH 05/10] asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08
by Liam Girdwood 07 Jul '08
07 Jul '08
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the codec drivers.
Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>
---
sound/soc/codecs/ac97.c | 2 +-
sound/soc/codecs/ac97.h | 2 +-
sound/soc/codecs/ak4535.c | 8 ++++----
sound/soc/codecs/ak4535.h | 2 +-
sound/soc/codecs/cs4270.c | 8 ++++----
sound/soc/codecs/cs4270.h | 2 +-
sound/soc/codecs/tlv320aic3x.c | 8 ++++----
sound/soc/codecs/tlv320aic3x.h | 2 +-
sound/soc/codecs/uda1380.c | 6 +++---
sound/soc/codecs/uda1380.h | 2 +-
sound/soc/codecs/wm8510.c | 10 +++++-----
sound/soc/codecs/wm8510.h | 2 +-
sound/soc/codecs/wm8731.c | 8 ++++----
sound/soc/codecs/wm8731.h | 2 +-
sound/soc/codecs/wm8750.c | 8 ++++----
sound/soc/codecs/wm8750.h | 2 +-
sound/soc/codecs/wm8753.c | 28 ++++++++++++++--------------
sound/soc/codecs/wm8753.h | 2 +-
sound/soc/codecs/wm8990.c | 12 ++++++------
sound/soc/codecs/wm8990.h | 2 +-
sound/soc/codecs/wm9712.c | 2 +-
sound/soc/codecs/wm9712.h | 2 +-
sound/soc/codecs/wm9713.c | 10 +++++-----
sound/soc/codecs/wm9713.h | 2 +-
24 files changed, 67 insertions(+), 67 deletions(-)
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index e4516f3..61fd96c 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -41,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000)
-struct snd_soc_codec_dai ac97_dai = {
+struct snd_soc_dai ac97_dai = {
.name = "AC97 HiFi",
.type = SND_SOC_DAI_AC97,
.playback = {
diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h
index 2bf6d69..281aa42 100644
--- a/sound/soc/codecs/ac97.h
+++ b/sound/soc/codecs/ac97.h
@@ -14,6 +14,6 @@
#define __LINUX_SND_SOC_AC97_H
extern struct snd_soc_codec_device soc_codec_dev_ac97;
-extern struct snd_soc_codec_dai ac97_dai;
+extern struct snd_soc_dai ac97_dai;
#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 469266e..b26003c 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -329,7 +329,7 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec)
return 0;
}
-static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -369,7 +369,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -394,7 +394,7 @@ static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute)
+static int ak4535_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
@@ -436,7 +436,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
-struct snd_soc_codec_dai ak4535_dai = {
+struct snd_soc_dai ak4535_dai = {
.name = "AK4535",
.playback = {
.stream_name = "Playback",
diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h
index fc686dd..e9fe30e 100644
--- a/sound/soc/codecs/ak4535.h
+++ b/sound/soc/codecs/ak4535.h
@@ -40,7 +40,7 @@ struct ak4535_setup_data {
unsigned short i2c_address;
};
-extern struct snd_soc_codec_dai ak4535_dai;
+extern struct snd_soc_dai ak4535_dai;
extern struct snd_soc_codec_device soc_codec_dev_ak4535;
#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index e73fcfd..9deb8c7 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -201,7 +201,7 @@ static struct {
* driver what the input settings can be. This would need to be implemented
* for stand-alone mode to work.
*/
-static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
* data for playback only, but ASoC currently does not support different
* formats for playback vs. record.
*/
-static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int format)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
* board does not have the MUTEA or MUTEB pins connected to such circuitry,
* then this function will do nothing.
*/
-static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute)
+static int cs4270_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
int reg6;
@@ -667,7 +667,7 @@ error:
#endif /* USE_I2C*/
-struct snd_soc_codec_dai cs4270_dai = {
+struct snd_soc_dai cs4270_dai = {
.name = "CS4270",
.playback = {
.stream_name = "Playback",
diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h
index 0ced49b..adc6cd9 100644
--- a/sound/soc/codecs/cs4270.h
+++ b/sound/soc/codecs/cs4270.h
@@ -16,7 +16,7 @@
* The ASoC codec DAI structure for the CS4270. Assign this structure to
* the .codec_dai field of your machine driver's snd_soc_dai_link structure.
*/
-extern struct snd_soc_codec_dai cs4270_dai;
+extern struct snd_soc_dai cs4270_dai;
/*
* The ASoC codec device structure for the CS4270. Assign this structure
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 954d39b..b1dce5f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -814,7 +814,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
+static int aic3x_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON;
@@ -831,7 +831,7 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
return 0;
}
-static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -841,7 +841,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -990,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
-struct snd_soc_codec_dai aic3x_dai = {
+struct snd_soc_dai aic3x_dai = {
.name = "aic3x",
.playback = {
.stream_name = "Playback",
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index e600946..d76c079 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -228,7 +228,7 @@ struct aic3x_setup_data {
unsigned int gpio_func[2];
};
-extern struct snd_soc_codec_dai aic3x_dai;
+extern struct snd_soc_dai aic3x_dai;
extern struct snd_soc_codec_device soc_codec_dev_aic3x;
#endif /* _AIC3X_H */
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 6d5335b..a52d6d9 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -372,7 +372,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec)
return 0;
}
-static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -499,7 +499,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
uda1380_write(codec, UDA1380_CLK, clk);
}
-static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute)
+static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
@@ -542,7 +542,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
-struct snd_soc_codec_dai uda1380_dai[] = {
+struct snd_soc_dai uda1380_dai[] = {
{
.name = "UDA1380",
.playback = {
diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h
index f9d885c..50c603e 100644
--- a/sound/soc/codecs/uda1380.h
+++ b/sound/soc/codecs/uda1380.h
@@ -83,7 +83,7 @@ struct uda1380_setup_data {
#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */
#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
-extern struct snd_soc_codec_dai uda1380_dai[3];
+extern struct snd_soc_dai uda1380_dai[3];
extern struct snd_soc_codec_device soc_codec_dev_uda1380;
#endif /* _UDA1380_H */
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index b549f67..67325fd 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -332,7 +332,7 @@ static void pll_factors(unsigned int target, unsigned int source)
pll_div.k = K;
}
-static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -368,7 +368,7 @@ static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
/*
* Configure WM8510 clock dividers.
*/
-static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -402,7 +402,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -510,7 +510,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8510_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
@@ -554,7 +554,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-struct snd_soc_codec_dai wm8510_dai = {
+struct snd_soc_dai wm8510_dai = {
.name = "WM8510 HiFi",
.playback = {
.stream_name = "Playback",
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
index c862e7b..f5d2e42 100644
--- a/sound/soc/codecs/wm8510.h
+++ b/sound/soc/codecs/wm8510.h
@@ -97,7 +97,7 @@ struct wm8510_setup_data {
unsigned short i2c_address;
};
-extern struct snd_soc_codec_dai wm8510_dai;
+extern struct snd_soc_dai wm8510_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8510;
#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 3ff42ad..369d39c 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -318,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream)
}
}
-static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8731_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
@@ -330,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
return 0;
}
-static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -349,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
}
-static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -443,7 +443,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
#define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
-struct snd_soc_codec_dai wm8731_dai = {
+struct snd_soc_dai wm8731_dai = {
.name = "WM8731",
.playback = {
.stream_name = "Playback",
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 5bcab6a..99f2e3c 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -38,7 +38,7 @@ struct wm8731_setup_data {
unsigned short i2c_address;
};
-extern struct snd_soc_codec_dai wm8731_dai;
+extern struct snd_soc_dai wm8731_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8731;
#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index eb460c9..e23cb09 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -536,7 +536,7 @@ static inline int get_coeff(int mclk, int rate)
return -EINVAL;
}
-static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -554,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
return -EINVAL;
}
-static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -647,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8750_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
@@ -692,7 +692,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
#define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
-struct snd_soc_codec_dai wm8750_dai = {
+struct snd_soc_dai wm8750_dai = {
.name = "WM8750",
.playback = {
.stream_name = "Playback",
diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h
index a97a54a..8ef30e6 100644
--- a/sound/soc/codecs/wm8750.h
+++ b/sound/soc/codecs/wm8750.h
@@ -61,7 +61,7 @@ struct wm8750_setup_data {
unsigned short i2c_address;
};
-extern struct snd_soc_codec_dai wm8750_dai;
+extern struct snd_soc_dai wm8750_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8750;
#endif
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index be01a73..8604809 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -740,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
pll_div->k = K;
}
-static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
u16 reg, enable;
@@ -863,7 +863,7 @@ static int get_coeff(int mclk, int rate)
/*
* Clock after PLL and dividers
*/
-static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -890,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
/*
* Set's ADC and Voice DAC format.
*/
-static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -960,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
/*
* Set's PCM dai fmt and BCLK.
*/
-static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1026,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1054,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
/*
* Set's HiFi DAC format.
*/
-static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1087,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
/*
* Set's I2S DAI format.
*/
-static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1195,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1210,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
@@ -1218,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1233,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1250,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
}
-static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8753_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
@@ -1316,7 +1316,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
* 3. Voice disabled - HIFI over HIFI
* 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
*/
-static const struct snd_soc_codec_dai wm8753_all_dai[] = {
+static const struct snd_soc_dai wm8753_all_dai[] = {
/* DAI HiFi mode 1 */
{ .name = "WM8753 HiFi",
.id = 1,
@@ -1456,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = {
},
};
-struct snd_soc_codec_dai wm8753_dai[2];
+struct snd_soc_dai wm8753_dai[2];
EXPORT_SYMBOL_GPL(wm8753_dai);
static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index 95e2a1f..44f5f1f 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -120,7 +120,7 @@ struct wm8753_setup_data {
#define WM8753_DAI_HIFI 0
#define WM8753_DAI_VOICE 1
-extern struct snd_soc_codec_dai wm8753_dai[2];
+extern struct snd_soc_dai wm8753_dai[2];
extern struct snd_soc_codec_device soc_codec_dev_wm8753;
#endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index a1371b7..3ecce51 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1029,7 +1029,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
pll_div->k = K;
}
-static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
u16 reg;
@@ -1065,7 +1065,7 @@ static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
/*
* Clock after PLL and dividers
*/
-static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1078,7 +1078,7 @@ static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
/*
* Set's ADC and Voice DAC format.
*/
-static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1131,7 +1131,7 @@ static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1196,7 +1196,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8990_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 val;
@@ -1329,7 +1329,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
* 1. ADC/DAC on Primary Interface
* 2. ADC on Primary Interface/DAC on secondary
*/
-struct snd_soc_codec_dai wm8990_dai = {
+struct snd_soc_dai wm8990_dai = {
/* ADC/DAC on primary */
.name = "WM8990 ADC/DAC Primary",
.id = 1,
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index bf9f882..6bea574 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -825,7 +825,7 @@ struct wm8990_setup_data {
#define WM8990_ADCCLK_DIV 2
#define WM8990_BCLK_DIV 3
-extern struct snd_soc_codec_dai wm8990_dai;
+extern struct snd_soc_dai wm8990_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8990;
#endif /* __WM8990REGISTERDEFS_H__ */
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 4739011..9fc8edd 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -532,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000)
-struct snd_soc_codec_dai wm9712_dai[] = {
+struct snd_soc_dai wm9712_dai[] = {
{
.name = "AC97 HiFi",
.type = SND_SOC_DAI_AC97_BUS,
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
index 719105d..d29e8a1 100644
--- a/sound/soc/codecs/wm9712.h
+++ b/sound/soc/codecs/wm9712.h
@@ -8,7 +8,7 @@
#define WM9712_DAI_AC97_HIFI 0
#define WM9712_DAI_AC97_AUX 1
-extern struct snd_soc_codec_dai wm9712_dai[2];
+extern struct snd_soc_dai wm9712_dai[2];
extern struct snd_soc_codec_device soc_codec_dev_wm9712;
#endif
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index a480618..38d1fe0 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -789,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
return 0;
}
-static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -800,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
* Tristate the PCM DAI lines, tristate can be disabled by calling
* wm9713_set_dai_fmt()
*/
-static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
int tristate)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -816,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
* Configure WM9713 clock dividers.
* Voice DAC needs 256 FS
*/
-static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -858,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
return 0;
}
-static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
@@ -1018,7 +1018,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
SNDRV_PCM_FORMAT_S24_LE)
-struct snd_soc_codec_dai wm9713_dai[] = {
+struct snd_soc_dai wm9713_dai[] = {
{
.name = "AC97 HiFi",
.type = SND_SOC_DAI_AC97_BUS,
diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h
index d357b6c..63b8d81 100644
--- a/sound/soc/codecs/wm9713.h
+++ b/sound/soc/codecs/wm9713.h
@@ -46,7 +46,7 @@
#define WM9713_DAI_PCM_VOICE 2
extern struct snd_soc_codec_device soc_codec_dev_wm9713;
-extern struct snd_soc_codec_dai wm9713_dai[3];
+extern struct snd_soc_dai wm9713_dai[3];
int wm9713_reset(struct snd_soc_codec *codec, int try_warm);
--
1.5.4.2
1
0