* [PATCH] Get TSC2101 and TSC2102 drivers in sync with new ALSA struct naming
@ 2007-05-04 2:44 andrzej zaborowski
2007-05-04 17:09 ` Tony Lindgren
0 siblings, 1 reply; 2+ messages in thread
From: andrzej zaborowski @ 2007-05-04 2:44 UTC (permalink / raw)
To: linux-omap-open-source
[-- Attachment #1: Type: text/plain, Size: 246 bytes --]
Update the TSC2101/2102 drivers to use the new ALSA type names. This
makes the TSC2102 driver build again - TSC2101 is still broken though.
Contains also minor formatting fix in TSC2101 code.
Signed-off-by: Andrzej Zaborowski <balrog@zabor.org>
[-- Attachment #2: 0004-Update-TSC2101-and-TSC2102-drivers-with-new-ALSA-structure-names.txt --]
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From 0800f9eaf98b9c525303a44e5582124a34187560 Mon Sep 17 00:00:00 2001
From: Andrzej Zaborowski <balrog@zabor.org>
Date: Fri, 4 May 2007 05:30:40 +0200
Subject: [PATCH] Update TSC2101 and TSC2102 drivers with new ALSA structure names.
---
sound/arm/omap/omap-alsa-tsc2101-mixer.c | 83 ++++++++++++++++++++----------
sound/arm/omap/omap-alsa-tsc2101-mixer.h | 2 +-
sound/arm/omap/omap-alsa-tsc2101.c | 6 +-
sound/arm/omap/omap-alsa-tsc2101.h | 2 +-
sound/arm/omap/omap-alsa-tsc2102-mixer.c | 54 ++++++++++----------
sound/arm/omap/omap-alsa-tsc2102.c | 4 +-
6 files changed, 89 insertions(+), 62 deletions(-)
diff --git a/sound/arm/omap/omap-alsa-tsc2101-mixer.c b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
index 0f92266..8faba95 100644
--- a/sound/arm/omap/omap-alsa-tsc2101-mixer.c
+++ b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
@@ -699,7 +699,8 @@ void snd_omap_init_mixer(void)
FN_OUT(0);
}
-static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __pcm_playback_target_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
static char *texts[PLAYBACK_TARGET_COUNT] = {
"Loudspeaker", "Headphone", "Cellphone"
@@ -716,13 +717,15 @@ static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_inf
return 0;
}
-static int __pcm_playback_target_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_target_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = current_playback_target;
return 0;
}
-static int __pcm_playback_target_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_target_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
int retVal;
int curVal;
@@ -749,7 +752,8 @@ static int __pcm_playback_target_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_valu
return retVal;
}
-static int __pcm_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
@@ -762,7 +766,8 @@ static int __pcm_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_inf
* Alsa mixer interface function for getting the volume read from the DGC in a
* 0 -100 alsa mixer format.
*/
-static int __pcm_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 volL;
u16 volR;
@@ -786,13 +791,15 @@ static int __pcm_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_valu
return 0;
}
-static int __pcm_playback_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
return set_mixer_volume_as_dac_gain_control_volume(ucontrol->value.integer.value[0],
ucontrol->value.integer.value[1]);
}
-static int __pcm_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
@@ -806,7 +813,8 @@ static int __pcm_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_inf
* When DGC_DALMU is 0, left channel is not muted.
* Same logic apply also for the right channel.
*/
-static int __pcm_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
@@ -815,13 +823,15 @@ static int __pcm_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_valu
return 0;
}
-static int __pcm_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
return dac_gain_control_unmute(ucontrol->value.integer.value[0],
ucontrol->value.integer.value[1]);
}
-static int __headset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __headset_playback_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
@@ -830,7 +840,8 @@ static int __headset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
return 0;
}
-static int __headset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __headset_playback_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 val;
u16 vol;
@@ -847,12 +858,14 @@ static int __headset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_
return 0;
}
-static int __headset_playback_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __headset_playback_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
return set_mixer_volume_as_headset_gain_control_volume(ucontrol->value.integer.value[0]);
}
-static int __headset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __headset_playback_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
@@ -864,14 +877,16 @@ static int __headset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
/* When HGC_ADMUT_HED (bit 15) is 1, the headset is muted.
* When HGC_ADMUT_HED is 0, headset is not muted.
*/
-static int __headset_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __headset_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
ucontrol->value.integer.value[0] = IS_UNMUTED(15, val);
return 0;
}
-static int __headset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __headset_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
// mute/unmute headset
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
@@ -879,7 +894,8 @@ static int __headset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_
15);
}
-static int __handset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __handset_playback_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
@@ -888,7 +904,8 @@ static int __handset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
return 0;
}
-static int __handset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __handset_playback_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 val;
u16 vol;
@@ -904,12 +921,14 @@ static int __handset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_
return 0;
}
-static int __handset_playback_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __handset_playback_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
return set_mixer_volume_as_handset_gain_control_volume(ucontrol->value.integer.value[0]);
}
-static int __handset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __handset_playback_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
@@ -921,14 +940,16 @@ static int __handset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
/* When HNGC_ADMUT_HND (bit 15) is 1, the handset is muted.
* When HNGC_ADMUT_HND is 0, handset is not muted.
*/
-static int __handset_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __handset_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
ucontrol->value.integer.value[0] = IS_UNMUTED(15, val);
return 0;
}
-static int __handset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __handset_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
// handset mute/unmute
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
@@ -936,7 +957,8 @@ static int __handset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_
15);
}
-static int __cellphone_input_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __cellphone_input_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
@@ -948,21 +970,24 @@ static int __cellphone_input_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_
/* When BGC_MUT_CP (bit 15) = 1, power down cellphone input pga.
* When BGC_MUT_CP = 0, power up cellphone input pga.
*/
-static int __cellphone_input_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __cellphone_input_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
ucontrol->value.integer.value[0] = IS_UNMUTED(15, val);
return 0;
}
-static int __cellphone_input_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __cellphone_input_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_BUZZER_GAIN_CTRL,
15);
}
-static int __buzzer_input_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __buzzer_input_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
@@ -974,21 +999,23 @@ static int __buzzer_input_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_inf
/* When BGC_MUT_BU (bit 6) = 1, power down cellphone input pga.
* When BGC_MUT_BU = 0, power up cellphone input pga.
*/
-static int __buzzer_input_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __buzzer_input_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
ucontrol->value.integer.value[0] = IS_UNMUTED(6, val);
return 0;
}
-static int __buzzer_input_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __buzzer_input_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_BUZZER_GAIN_CTRL,
6);
}
-static snd_kcontrol_new_t tsc2101_control[] __devinitdata = {
+static struct snd_kcontrol_new tsc2101_control[] __devinitdata = {
{
.name = "Target Playback Route",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/arm/omap/omap-alsa-tsc2101-mixer.h b/sound/arm/omap/omap-alsa-tsc2101-mixer.h
index 3f27e27..26fabfb 100644
--- a/sound/arm/omap/omap-alsa-tsc2101-mixer.h
+++ b/sound/arm/omap/omap-alsa-tsc2101-mixer.h
@@ -1,5 +1,5 @@
/*
- * sound/arm/omap/omap-alsa-tsc2101-mixer.c
+ * sound/arm/omap/omap-alsa-tsc2101-mixer.h
*
* Alsa Driver for TSC2101 codec for OMAP platform boards.
*
diff --git a/sound/arm/omap/omap-alsa-tsc2101.c b/sound/arm/omap/omap-alsa-tsc2101.c
index 9bbf0c7..24b0cc6 100644
--- a/sound/arm/omap/omap-alsa-tsc2101.c
+++ b/sound/arm/omap/omap-alsa-tsc2101.c
@@ -1,5 +1,5 @@
/*
- * arch/arm/mach-omap1/omap-alsa-tsc2101.c
+ * sound/arm/omap/omap-alsa-tsc2101.c
*
* Alsa codec Driver for TSC2101 chip for OMAP platform boards.
* Code obtained from oss omap drivers
@@ -59,7 +59,7 @@ static unsigned int rates[] = {
29400, 32000, 44100, 48000,
};
-static snd_pcm_hw_constraint_list_t tsc2101_hw_constraints_rates = {
+static struct snd_pcm_hw_constraint_list tsc2101_hw_constraints_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
@@ -93,7 +93,7 @@ static const struct tsc2101_samplerate_reg_info
{48000, 0, 0},
};
-static snd_pcm_hardware_t tsc2101_snd_omap_alsa_playback = {
+static struct snd_pcm_hardware tsc2101_snd_omap_alsa_playback = {
.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
#ifdef CONFIG_MACH_OMAP_H6300
diff --git a/sound/arm/omap/omap-alsa-tsc2101.h b/sound/arm/omap/omap-alsa-tsc2101.h
index 803d215..6cb2db3 100644
--- a/sound/arm/omap/omap-alsa-tsc2101.h
+++ b/sound/arm/omap/omap-alsa-tsc2101.h
@@ -1,5 +1,5 @@
/*
- * arch/arc/mach-omap1/omap-alsa-tsc2101.h
+ * sound/arm/omap/omap-alsa-tsc2101.h
*
* Alsa Driver for TSC2101 codec for OMAP platform boards.
*
diff --git a/sound/arm/omap/omap-alsa-tsc2102-mixer.c b/sound/arm/omap/omap-alsa-tsc2102-mixer.c
index 2e06d70..d1c8b65 100644
--- a/sound/arm/omap/omap-alsa-tsc2102-mixer.c
+++ b/sound/arm/omap/omap-alsa-tsc2102-mixer.c
@@ -99,8 +99,8 @@ void snd_omap_init_mixer(void)
FN_OUT(0);
}
-static int __pcm_playback_volume_info(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
@@ -109,8 +109,8 @@ static int __pcm_playback_volume_info(
return 0;
}
-static int __pcm_playback_volume_get(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = vol[0]; /* L */
ucontrol->value.integer.value[1] = vol[1]; /* R */
@@ -118,8 +118,8 @@ static int __pcm_playback_volume_get(
return 0;
}
-static int __pcm_playback_volume_put(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
set_dac_gain_stereo(
ucontrol->value.integer.value[0], /* L */
@@ -127,8 +127,8 @@ static int __pcm_playback_volume_put(
return 1;
}
-static int __pcm_playback_switch_info(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
@@ -137,8 +137,8 @@ static int __pcm_playback_switch_info(
return 0;
}
-static int __pcm_playback_switch_get(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = !mute[0]; /* L */
ucontrol->value.integer.value[1] = !mute[1]; /* R */
@@ -146,8 +146,8 @@ static int __pcm_playback_switch_get(
return 0;
}
-static int __pcm_playback_switch_put(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
mute[0] = (ucontrol->value.integer.value[0] == 0); /* L */
mute[1] = (ucontrol->value.integer.value[1] == 0); /* R */
@@ -156,8 +156,8 @@ static int __pcm_playback_switch_put(
return 1;
}
-static int __pcm_playback_deemphasis_info(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __pcm_playback_deemphasis_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
@@ -166,15 +166,15 @@ static int __pcm_playback_deemphasis_info(
return 0;
}
-static int __pcm_playback_deemphasis_get(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_deemphasis_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = filter[0];
return 0;
}
-static int __pcm_playback_deemphasis_put(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_deemphasis_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
filter[0] = (ucontrol->value.integer.value[0] > 0);
@@ -182,8 +182,8 @@ static int __pcm_playback_deemphasis_put(
return 1;
}
-static int __pcm_playback_bassboost_info(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int __pcm_playback_bassboost_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
@@ -192,15 +192,15 @@ static int __pcm_playback_bassboost_info(
return 0;
}
-static int __pcm_playback_bassboost_get(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_bassboost_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = filter[1];
return 0;
}
-static int __pcm_playback_bassboost_put(
- snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int __pcm_playback_bassboost_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
filter[1] = (ucontrol->value.integer.value[0] > 0);
@@ -208,7 +208,7 @@ static int __pcm_playback_bassboost_put(
return 1;
}
-static snd_kcontrol_new_t tsc2102_control[] __devinitdata = {
+static struct snd_kcontrol_new tsc2102_controls[] __devinitdata = {
{
.name = "Master Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -271,9 +271,9 @@ int snd_omap_mixer(struct snd_card_omap_codec *tsc2102)
if (!tsc2102)
return -EINVAL;
- for (i = 0; i < ARRAY_SIZE(tsc2102_control); i ++) {
+ for (i = 0; i < ARRAY_SIZE(tsc2102_controls); i ++) {
err = snd_ctl_add(tsc2102->card,
- snd_ctl_new1(&tsc2102_control[i],
+ snd_ctl_new1(&tsc2102_controls[i],
tsc2102->card));
if (err < 0)
diff --git a/sound/arm/omap/omap-alsa-tsc2102.c b/sound/arm/omap/omap-alsa-tsc2102.c
index 41e2d07..2e1643a 100644
--- a/sound/arm/omap/omap-alsa-tsc2102.c
+++ b/sound/arm/omap/omap-alsa-tsc2102.c
@@ -38,13 +38,13 @@ static unsigned int rates[] = {
16000, 22050, 24000, 29400, 32000, 44100, 48000,
};
-static snd_pcm_hw_constraint_list_t tsc2102_hw_constraints_rates = {
+static struct snd_pcm_hw_constraint_list tsc2102_hw_constraints_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
-static snd_pcm_hardware_t tsc2102_snd_omap_alsa_playback = {
+static struct snd_pcm_hardware tsc2102_snd_omap_alsa_playback = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID,
--
1.4.4.3
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2007-05-04 2:44 [PATCH] Get TSC2101 and TSC2102 drivers in sync with new ALSA struct naming andrzej zaborowski
2007-05-04 17:09 ` Tony Lindgren
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