Mercurial > mplayer.hg
changeset 12281:e6f6dac5d77b
needed for a/v sync with compressed audio (e.g. raw .mp2 or .ac3 file)
| author | rfelker |
|---|---|
| date | Mon, 26 Apr 2004 03:11:08 +0000 |
| parents | f5cb1afdff53 |
| children | 76dc8fd54dec |
| files | libmpdemux/demux_rawaudio.c |
| diffstat | 1 files changed, 8 insertions(+), 1 deletions(-) [+] |
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--- a/libmpdemux/demux_rawaudio.c Sun Apr 25 22:43:03 2004 +0000 +++ b/libmpdemux/demux_rawaudio.c Mon Apr 26 03:11:08 2004 +0000 @@ -17,6 +17,7 @@ static int channels = 2; static int samplerate = 44100; static int samplesize = 2; +static int bitrate = 0; static int format = 0x1; // Raw PCM m_option_t demux_rawaudio_opts[] = { @@ -24,6 +25,7 @@ { "channels", &channels, CONF_TYPE_INT,CONF_RANGE,1,8, NULL }, { "rate", &samplerate, CONF_TYPE_INT,CONF_RANGE,1000,8*48000, NULL }, { "samplesize", &samplesize, CONF_TYPE_INT,CONF_RANGE,1,8, NULL }, + { "bitrate", &bitrate, CONF_TYPE_INT,CONF_MIN,0,0, NULL }, { "format", &format, CONF_TYPE_INT, CONF_MIN, 0 , 0, NULL }, {NULL, NULL, 0, 0, 0, 0, NULL} }; @@ -40,7 +42,12 @@ w->wFormatTag = sh_audio->format = format; w->nChannels = sh_audio->channels = channels; w->nSamplesPerSec = sh_audio->samplerate = samplerate; - w->nAvgBytesPerSec = samplerate*samplesize*channels; + if (bitrate > 999) + w->nAvgBytesPerSec = bitrate/8; + else if (bitrate > 0) + w->nAvgBytesPerSec = bitrate*125; + else + w->nAvgBytesPerSec = samplerate*samplesize*channels; w->nBlockAlign = channels*samplesize; sh_audio->samplesize = samplesize; w->wBitsPerSample = 8*samplesize;