void uninit_audio_chain(struct MPContext *mpctx)
{
if (mpctx->d_audio) {
mixer_uninit_audio(mpctx->mixer);
audio_uninit(mpctx->d_audio);
mpctx->d_audio = NULL;
talloc_free(mpctx->ao_buffer);
mpctx->ao_buffer = NULL;
mpctx->audio_status = STATUS_EOF;
reselect_demux_streams(mpctx);
}
}
//******************************************************************************************
// open & setup audio device
// return: 1=success 0=fail
static int audio_init(int rate, int channels, int format, int flags)
{
CSingleLock lock(m_critAudio);
char strFourCC[10];
char strAudioCodec[128];
long lBitRate;
long lSampleRate;
int iChannels;
BOOL bVBR;
bool bAC3PassThru = false;
audio_uninit(1); //Make sure nothing else was uninted first. mplayer sometimes forgets.
mplayer_GetAudioInfo(strFourCC, strAudioCodec, &lBitRate, &lSampleRate, &iChannels, &bVBR);
int ao_format_bits;
//Make sure we only accept formats that we can handle
switch (format)
{
case AFMT_AC3:
ao_format_bits = 16;
break;
case AFMT_S16_LE:
case AFMT_S8:
case AFMT_U8: //Not sure about this one, added as we have handling for it later
ao_format_bits = audio_out_format_bits(format);
break;
default:
CLog::Log(LOGDEBUG, "ao_win32: format %d not supported defaulting to Signed 16-bit Little-Endian\n", format);
format = AFMT_S16_LE;
ao_format_bits = audio_out_format_bits(format);
}
pao_data = GetAOData();
channels = pao_data->channels;
//In the case of forced audio filter, channel number for the GetAudioInfo, and from pao_data
//is not the same, so the follwing two lines are not correct
//Make sure we only output as many channels as we how. don't try to create any virtual.
//if (channels > iChannels) channels = iChannels;
// Check whether we are passing digital output direct through.
// Anything with 48kHz 2 channel audio can be passed direct.
if (g_guiSettings.GetInt("audiooutput.mode") == AUDIO_DIGITAL)
{
// Check that we are allowed to pass through DD or DTS
if (strstr(strAudioCodec, "SPDIF"))
bAC3PassThru = true;
}
if (bAC3PassThru)
{
channels = 2;
// ac3 passthru
m_pAudioDecoder = new CAc97DirectSound(m_pAudioCallback, channels, rate, ao_format_bits, bAC3PassThru);
}
else
{ // check if we should resample this audio
// currently we don't do this for videos for fear of CPU issues
bool bResample = false;
if( !mplayer_HasVideo() && channels <= 2 && rate != 48000 )
bResample = true;
if ( channels == 3 || channels == 5 || channels > 6 )
return 1; // this is an ugly hack due to our code use mplayer_open_file for both playing file, and format detecttion
if(bResample)
m_pAudioDecoder = new CResampleDirectSound(m_pAudioCallback, channels, rate, ao_format_bits, strAudioCodec, !mplayer_HasVideo());
else
m_pAudioDecoder = new CASyncDirectSound(m_pAudioCallback, channels, rate, ao_format_bits, strAudioCodec, !mplayer_HasVideo());
}
pao_data->channels = channels;
pao_data->samplerate = rate;
pao_data->format = format;
pao_data->bps = channels * rate;
if (format != AFMT_U8 && format != AFMT_S8)
{
pao_data->bps *= (ao_format_bits / 8);
}
pao_data->outburst = m_pAudioDecoder->GetChunkLen();
if( mplayer_HasVideo() )
{
m_waitvideo = 5; /* allow 5 empty returns after filling audio buffer */
m_pAudioDecoder->Pause();
}
return 1;
}
