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authorViliam Mateicka <viliam.mateicka@gmail.com>2008-12-31 16:46:41 +0100
committerViliam Mateicka <viliam.mateicka@gmail.com>2008-12-31 16:46:41 +0100
commit9a70c4d06d51cca24dfb93847d1f82187d454e31 (patch)
tree444672f0ab76eca96692560201287d6f7a1a73b5 /src/output
parentdd9af72a744d9c93966f5088931d7edbf28e1d1d (diff)
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Moving mixers to audio outputs
Diffstat (limited to 'src/output')
-rw-r--r--src/output/alsa_plugin.c22
-rw-r--r--src/output/oss_plugin.c19
2 files changed, 40 insertions, 1 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 57d102c51..5f53729a7 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -18,6 +18,7 @@
#include "../output_api.h"
#include "../utils.h"
+#include "../mixer.h"
#include <glib.h>
#include <alsa/asoundlib.h>
@@ -51,6 +52,9 @@ typedef struct _AlsaData {
unsigned int period_time;
int sampleSize;
int useMmap;
+
+ struct alsa_mixer *mixer;
+
} AlsaData;
static const char *
@@ -71,12 +75,15 @@ static AlsaData *newAlsaData(void)
ret->buffer_time = MPD_ALSA_BUFFER_TIME_US;
ret->period_time = 0;
+ ret->mixer = alsa_mixer_init();
+
return ret;
}
static void freeAlsaData(AlsaData * ad)
{
g_free(ad->device);
+ alsa_mixer_finish(ad->mixer);
free(ad);
}
@@ -125,8 +132,10 @@ static void *alsa_initDriver(mpd_unused struct audio_output *ao,
free_global_registered = 1;
}
- if (param)
+ if (param) {
alsa_configure(ad, param);
+ alsa_mixer_configure(ad->mixer, param);
+ }
return ad;
}
@@ -181,6 +190,8 @@ static bool alsa_openDevice(void *data, struct audio_format *audioFormat)
unsigned int period_time, period_time_ro;
unsigned int buffer_time;
+ alsa_mixer_open(ad->mixer);
+
if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
g_warning("ALSA device \"%s\" doesn't support %u bit audio\n",
alsa_device(ad), audioFormat->bits);
@@ -403,6 +414,7 @@ static void alsa_closeDevice(void *data)
snd_pcm_close(ad->pcmHandle);
ad->pcmHandle = NULL;
}
+ alsa_mixer_close(ad->mixer);
}
static bool
@@ -436,6 +448,13 @@ alsa_playAudio(void *data, const char *playChunk, size_t size)
return true;
}
+static bool
+alsa_control(void *data, int cmd, void *arg)
+{
+ AlsaData *ad = data;
+ return alsa_mixer_control(ad->mixer, cmd, arg);
+}
+
const struct audio_output_plugin alsaPlugin = {
.name = "alsa",
.test_default_device = alsa_testDefault,
@@ -445,4 +464,5 @@ const struct audio_output_plugin alsaPlugin = {
.play = alsa_playAudio,
.cancel = alsa_dropBufferedAudio,
.close = alsa_closeDevice,
+ .control = alsa_control
};
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
index 4727823c3..1f582b2b4 100644
--- a/src/output/oss_plugin.c
+++ b/src/output/oss_plugin.c
@@ -20,6 +20,7 @@
*/
#include "../output_api.h"
+#include "../mixer.h"
#include <glib.h>
#include <sys/stat.h>
@@ -53,6 +54,7 @@ typedef struct _OssData {
int numSupported[3];
int *unsupported[3];
int numUnsupported[3];
+ struct oss_mixer *mixer;
} OssData;
enum oss_support {
@@ -273,6 +275,8 @@ static OssData *newOssData(void)
supportParam(ret, SNDCTL_DSP_CHANNELS, 2);
supportParam(ret, SNDCTL_DSP_SAMPLESIZE, 16);
+ ret->mixer = oss_mixer_init();
+
return ret;
}
@@ -285,6 +289,8 @@ static void freeOssData(OssData * od)
g_free(od->unsupported[OSS_CHANNELS]);
g_free(od->unsupported[OSS_BITS]);
+ oss_mixer_finish(od->mixer);
+
free(od);
}
@@ -348,6 +354,7 @@ static void *oss_open_default(ConfigParam *param)
if (ret[i] == 0) {
OssData *od = newOssData();
od->device = default_devices[i];
+ oss_mixer_configure(od->mixer, param);
return od;
}
}
@@ -388,6 +395,7 @@ static void *oss_initDriver(mpd_unused struct audio_output *audioOutput,
if (bp) {
OssData *od = newOssData();
od->device = bp->value;
+ oss_mixer_configure(od->mixer, param);
return od;
}
}
@@ -513,6 +521,8 @@ oss_openDevice(void *data, struct audio_format *audioFormat)
od->audio_format.bits, od->audio_format.channels,
od->audio_format.sample_rate);
+ oss_mixer_open(od->mixer);
+
return ret;
}
@@ -521,6 +531,7 @@ static void oss_closeDevice(void *data)
OssData *od = data;
oss_close(od);
+ oss_mixer_close(od->mixer);
}
static void oss_dropBufferedAudio(void *data)
@@ -559,6 +570,13 @@ oss_playAudio(void *data, const char *playChunk, size_t size)
return true;
}
+static bool
+oss_control(void *data, int cmd, void *arg)
+{
+ OssData *od = data;
+ return oss_mixer_control(od->mixer, cmd, arg);
+}
+
const struct audio_output_plugin ossPlugin = {
.name = "oss",
.test_default_device = oss_testDefault,
@@ -568,4 +586,5 @@ const struct audio_output_plugin ossPlugin = {
.play = oss_playAudio,
.cancel = oss_dropBufferedAudio,
.close = oss_closeDevice,
+ .control = oss_control,
};