From 9a70c4d06d51cca24dfb93847d1f82187d454e31 Mon Sep 17 00:00:00 2001
From: Viliam Mateicka <viliam.mateicka@gmail.com>
Date: Wed, 31 Dec 2008 16:46:41 +0100
Subject: Moving mixers to audio outputs

---
 src/output/alsa_plugin.c | 22 +++++++++++++++++++++-
 src/output/oss_plugin.c  | 19 +++++++++++++++++++
 2 files changed, 40 insertions(+), 1 deletion(-)

(limited to 'src/output')

diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 57d102c5..5f53729a 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 4727823c..1f582b2b 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,
 };
-- 
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