we ne allow audioOutput plugins to set the final outAudioFormat that will be used. we now use alsa's _near functions to detect what to use.

git-svn-id: https://svn.musicpd.org/mpd/trunk@3038 09075e82-0dd4-0310-85a5-a0d7c8717e4f

3 files changed, 27 insertions, 13 deletions

diff --git a/src/audioOutput.c b/src/audioOutput.c
index c973e1beb..1186dbe0d 100644
--- a/src/audioOutput.c
+++ b/src/audioOutput.c
@@ -73,13 +73,14 @@ AudioOutput * newAudioOutput(ConfigParam * param) {
 
 		memset(&ret->inAudioFormat, 0, sizeof(AudioFormat));
 		memset(&ret->outAudioFormat, 0, sizeof(AudioFormat));
+		memset(&ret->reqAudioFormat, 0, sizeof(AudioFormat));
 
 		getBlockParam(AUDIO_OUTPUT_FORMAT, format, 0);
 
 		if(format) {
 			ret->convertAudioFormat = 1;
 
-			if(0 != parseAudioConfig(&ret->outAudioFormat, format))
+			if(0 != parseAudioConfig(&ret->reqAudioFormat, format))
 			{
 		        	ERROR("error parsing format at line %i\n", 
 						bp->line);
@@ -102,6 +103,8 @@ AudioOutput * newAudioOutput(ConfigParam * param) {
 }
 
 int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat) {
+	int ret;
+	
 	if(audioOutput->open) {
 		if(cmpAudioFormat(audioFormat, &audioOutput->inAudioFormat) 
 				== 0) 
@@ -114,20 +117,24 @@ int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat) {
 	copyAudioFormat(&audioOutput->inAudioFormat, audioFormat);
 
 	if(audioOutput->convertAudioFormat) {
-		if(cmpAudioFormat(&audioOutput->inAudioFormat, 
-				&audioOutput->outAudioFormat) == 0) 
-		{
-			audioOutput->sameInAndOutFormats = 1;
-		}
-		else audioOutput->sameInAndOutFormats = 0;
+		copyAudioFormat(&audioOutput->outAudioFormat,
+				&audioOutput->reqAudioFormat);
 	}
 	else {
-		audioOutput->sameInAndOutFormats = 1;
 		copyAudioFormat(&audioOutput->outAudioFormat, 
 				&audioOutput->inAudioFormat);
 	}
 
-	return audioOutput->openDeviceFunc(audioOutput);
+	ret = audioOutput->openDeviceFunc(audioOutput);
+
+	if(cmpAudioFormat(&audioOutput->inAudioFormat, 
+			  &audioOutput->outAudioFormat) == 0)
+	{
+		audioOutput->sameInAndOutFormats = 1;
+	}
+	else audioOutput->sameInAndOutFormats = 0;
+
+	return ret;
 }
 
 static void convertAudioFormat(AudioOutput * audioOutput, char ** chunkArgPtr,
diff --git a/src/audioOutput.h b/src/audioOutput.h
index 13c549555..85868eaeb 100644
--- a/src/audioOutput.h
+++ b/src/audioOutput.h
@@ -60,6 +60,7 @@ struct _AudioOutput {
 	int convertAudioFormat;
 	AudioFormat inAudioFormat;
 	AudioFormat outAudioFormat;
+	AudioFormat reqAudioFormat;
 	char * convBuffer;
 	int convBufferLen;
 	int sameInAndOutFormats;
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
index 0cf453ffe..d804720b0 100644
--- a/src/audioOutputs/audioOutput_alsa.c
+++ b/src/audioOutputs/audioOutput_alsa.c
@@ -93,6 +93,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 	snd_pcm_hw_params_t * hwparams;
 	snd_pcm_sw_params_t * swparams;
 	unsigned int sampleRate = audioFormat->sampleRate;
+	unsigned int channels = audioFormat->channels;
 	snd_pcm_uframes_t alsa_buffer_size;
 	snd_pcm_uframes_t alsa_period_size;
 	unsigned int alsa_buffer_time = MPD_ALSA_BUFFER_TIME;
@@ -107,10 +108,10 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 		bitformat = SND_PCM_FORMAT_S16;
 		break;
 	case 24:
-		bitformat = SND_PCM_FORMAT_S16;
+		bitformat = SND_PCM_FORMAT_S24;
 		break;
 	case 32:
-		bitformat = SND_PCM_FORMAT_S16;
+		bitformat = SND_PCM_FORMAT_S32;
 		break;
 	default:
 		ERROR("Alsa device \"%s\" doesn't support %i bit audio\n",
@@ -162,14 +163,15 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 		goto fail;
 	}
 
-	err = snd_pcm_hw_params_set_channels(ad->pcmHandle, hwparams, 
-			audioFormat->channels);
+	err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams, 
+			&channels);
 	if(err < 0) {
 		ERROR("Alsa device \"%s\" does not support %i channels: "
 				"%s\n", ad->device, (int)audioFormat->channels, 
 				snd_strerror(-err));
 		goto fail;
 	}
+	audioFormat->channels = channels;
 
 	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, 
 			&sampleRate, 0);
@@ -178,6 +180,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 				ad->device, (int)audioFormat->sampleRate);
 		goto fail;
 	}
+	audioFormat->sampleRate = sampleRate;
 
 	err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
 			&alsa_buffer_time, 0);
@@ -214,6 +217,9 @@ static int alsa_openDevice(AudioOutput * audioOutput)
 
 	audioOutput->open = 1;
 
+	DEBUG("alsa device \"%s\" will be playing %i channel audio at %i Hz\n",
+			ad->device, channels, sampleRate);
+
 	return 0;
 
 error: