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authorWarren Dukes <warren.dukes@gmail.com>2005-03-08 01:53:13 +0000
committerWarren Dukes <warren.dukes@gmail.com>2005-03-08 01:53:13 +0000
commitb5e55a648f12cea229f49c00730ef11a8583b66b (patch)
tree2ca0130ddee4b343c3aea396f0539136010ff185
parentfa08a7e3ff0d0f60415c29c080cc4130cf4032c9 (diff)
downloadmpd-b5e55a648f12cea229f49c00730ef11a8583b66b.tar.gz
mpd-b5e55a648f12cea229f49c00730ef11a8583b66b.tar.xz
mpd-b5e55a648f12cea229f49c00730ef11a8583b66b.zip
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
Diffstat (limited to '')
-rw-r--r--src/audioOutput.c25
-rw-r--r--src/audioOutput.h1
-rw-r--r--src/audioOutputs/audioOutput_alsa.c14
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: