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merge from 2.0release

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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
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47
trunk/src/app/srs_app_hls.cpp
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@ -578,6 +578,7 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
current->full_path.c_str(), tmp_file.c_str());
// set the segment muxer audio codec.
// TODO: FIXME: refine code, use event instead.
if (acodec != SrsCodecAudioReserved1) {
current->muxer->update_acodec(acodec);
}
@ -646,6 +647,11 @@ int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
return current->muxer->update_acodec(ac);
}
bool SrsHlsMuxer::pure_audio()
{
return current && current->muxer && current->muxer->video_codec() == SrsCodecVideoDisabled;
}
int SrsHlsMuxer::flush_audio(SrsTsCache* cache)
{
int ret = ERROR_SUCCESS;
@ -1039,7 +1045,7 @@ int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
// when the sequence header changed, the stream is not republish.
return muxer->on_sequence_header();
}
int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
@ -1049,25 +1055,6 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
return ret;
}
// flush if buffer exceed max size.
if (cache->audio->payload->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
return ret;
}
}
// 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
// cache->audio will be free in flush_audio
// so we must check whether it's null ptr.
if (cache->audio && pts - cache->audio->start_pts > audio_delay * 90) {
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
return ret;
}
}
// reap when current source is pure audio.
// it maybe changed when stream info changed,
// for example, pure audio when start, audio/video when publishing,
@ -1083,6 +1070,21 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
}
}
// for pure audio, aggregate some frame to one.
if (muxer->pure_audio() && cache->audio) {
if (pts - cache->audio->start_pts < SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE) {
return ret;
}
}
// directly write the audio frame by frame to ts,
// it's ok for the hls overload, or maybe cause the audio corrupt,
// which introduced by aggregate the audios to a big one.
// @see https://github.com/simple-rtmp-server/srs/issues/512
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
@ -1100,7 +1102,7 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
// do reap ts if any of:
// a. wait keyframe and got keyframe.
// b. always reap when not wait keyframe.
if (!muxer->wait_keyframe()|| sample->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
if (!muxer->wait_keyframe() || sample->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
// when wait keyframe, there must exists idr frame in sample.
if (!sample->has_idr && muxer->wait_keyframe()) {
srs_warn("hls: ts starts without IDR, first nalu=%d, idr=%d", sample->first_nalu_type, sample->has_idr);
@ -1110,9 +1112,6 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
if ((ret = reap_segment("video", muxer, cache->video->dts)) != ERROR_SUCCESS) {
return ret;
}
// the video must be flushed, just return.
return ret;
}
}