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