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merge from 2.0release

This commit is contained in:
winlin 2015-10-27 17:45:14 +08:00
commit 1d57e53910
6 changed files with 62 additions and 49 deletions

View file

@ -469,8 +469,11 @@ int SrsTsContext::encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t p
while (p < end) {
SrsTsPacket* pkt = NULL;
if (p == start) {
// for pure audio stream, always write pcr.
// write pcr according to message.
bool write_pcr = msg->write_pcr;
// for pure audio, always write pcr.
// TODO: FIXME: maybe only need to write at begin and end of ts.
if (pure_audio && msg->is_audio()) {
write_pcr = true;
}
@ -2772,6 +2775,11 @@ void SrsTSMuxer::close()
writer->close();
}
SrsCodecVideo SrsTSMuxer::video_codec()
{
return vcodec;
}
SrsTsCache::SrsTsCache()
{
audio = NULL;
@ -2792,11 +2800,12 @@ int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample*
if (!audio) {
audio = new SrsTsMessage();
audio->write_pcr = false;
audio->start_pts = dts;
audio->dts = audio->pts = audio->start_pts = dts;
}
audio->dts = dts;
audio->pts = audio->dts;
// TODO: FIXME: refine code.
//audio->dts = dts;
//audio->pts = audio->dts;
audio->sid = SrsTsPESStreamIdAudioCommon;
// must be aac or mp3
@ -3146,20 +3155,11 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
return ret;
}
// flush if buffer exceed max size.
if (cache->audio->payload->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
return flush_video();
}
// TODO: config it.
// in ms, audio delay to flush the audios.
int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
// flush if audio delay exceed
if (dts - cache->audio->start_pts > audio_delay * 90) {
return flush_audio();
}
return ret;
// TODO: FIXME: for pure audio, aggregate some frame to one.
// always flush audio frame by frame.
// @see https://github.com/simple-rtmp-server/srs/issues/512
return flush_audio();
}
int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)