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for #512, write audio frame by frame for video+audio hls.
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6 changed files with 36 additions and 44 deletions
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@ -765,7 +765,9 @@ About the HLS overhead of SRS, we compare the overhead to FLV by remux the HLS t
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| 5147kbps | 370s | 195040 | 200280 | 2.68% |
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| 5158kbps | 1327s | 835664 | 858092 | 2.68% |
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The HLS overhead is calc by: (HLS - FLV) / FLV * 100%
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The HLS overhead is calc by: (HLS - FLV) / FLV * 100%.
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The overhead is larger than this benchmark(48kbps audio is best overhead), for we fix the [#512][bug#512].
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## Architecture
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@ -1193,6 +1195,8 @@ Winlin
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[bug #59]: https://github.com/simple-rtmp-server/srs/issues/59
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[bug #50]: https://github.com/simple-rtmp-server/srs/issues/50
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[bug #34]: https://github.com/simple-rtmp-server/srs/issues/34
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[bug #512]: https://github.com/simple-rtmp-server/srs/issues/512
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[bug #xxxxxxxxxx]: https://github.com/simple-rtmp-server/srs/issues/xxxxxxxxxx
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[r2.0a2]: https://github.com/simple-rtmp-server/srs/releases/tag/v2.0-a2
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[r2.0a1]: https://github.com/simple-rtmp-server/srs/releases/tag/2.0a1
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@ -646,6 +646,11 @@ int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
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return current->muxer->update_acodec(ac);
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}
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bool SrsHlsMuxer::pure_audio()
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{
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return current && current->muxer && current->muxer->video_codec() == SrsCodecVideoDisabled;
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}
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int SrsHlsMuxer::flush_audio(SrsTsCache* cache)
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{
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int ret = ERROR_SUCCESS;
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@ -1049,25 +1054,6 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
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return ret;
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}
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// flush if buffer exceed max size.
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if (cache->audio->payload->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
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if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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// TODO: config it.
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// in ms, audio delay to flush the audios.
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int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
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// flush if audio delay exceed
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// cache->audio will be free in flush_audio
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// so we must check whether it's null ptr.
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if (cache->audio && pts - cache->audio->start_pts > audio_delay * 90) {
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if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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// reap when current source is pure audio.
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// it maybe changed when stream info changed,
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// for example, pure audio when start, audio/video when publishing,
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@ -1083,6 +1069,14 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
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}
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}
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// directly write the audio frame by frame to ts,
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// it's ok for the hls overload, or maybe cause the audio corrupt,
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// which introduced by aggregate the audios to a big one.
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// @see https://github.com/simple-rtmp-server/srs/issues/512
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if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
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return ret;
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}
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return ret;
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}
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@ -1100,7 +1094,7 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
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// do reap ts if any of:
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// a. wait keyframe and got keyframe.
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// b. always reap when not wait keyframe.
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if (!muxer->wait_keyframe()|| sample->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
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if (!muxer->wait_keyframe() || sample->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
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// when wait keyframe, there must exists idr frame in sample.
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if (!sample->has_idr && muxer->wait_keyframe()) {
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srs_warn("hls: ts starts without IDR, first nalu=%d, idr=%d", sample->first_nalu_type, sample->has_idr);
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@ -1110,9 +1104,6 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
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if ((ret = reap_segment("video", muxer, cache->video->dts)) != ERROR_SUCCESS) {
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return ret;
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}
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// the video must be flushed, just return.
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return ret;
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}
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}
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@ -309,6 +309,10 @@ public:
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virtual bool is_segment_absolutely_overflow();
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public:
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virtual int update_acodec(SrsCodecAudio ac);
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/**
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* whether current hls muxer is pure audio mode.
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*/
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virtual bool pure_audio();
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virtual int flush_audio(SrsTsCache* cache);
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virtual int flush_video(SrsTsCache* cache);
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/**
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@ -246,12 +246,6 @@ extern int aac_sample_rates[];
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#define SRS_SRS_MAX_CODEC_SAMPLE 128
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#define SRS_AAC_SAMPLE_RATE_UNSET 15
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// in ms, for HLS aac flush the audio
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#define SRS_CONF_DEFAULT_AAC_DELAY 60
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// max PES packets size to flush the video.
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#define SRS_AUTO_HLS_AUDIO_CACHE_SIZE 128 * 1024
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/**
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* the FLV/RTMP supported audio sample size.
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* Size of each audio sample. This parameter only pertains to
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@ -2760,6 +2760,11 @@ void SrsTSMuxer::close()
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writer->close();
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}
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SrsCodecVideo SrsTSMuxer::video_codec()
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{
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return vcodec;
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}
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SrsTsCache::SrsTsCache()
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{
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audio = NULL;
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@ -3134,20 +3139,9 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
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return ret;
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}
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// flush if buffer exceed max size.
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if (cache->audio->payload->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
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return flush_video();
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}
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// TODO: config it.
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// in ms, audio delay to flush the audios.
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int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
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// flush if audio delay exceed
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if (dts - cache->audio->start_pts > audio_delay * 90) {
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return flush_audio();
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}
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return ret;
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// always flush audio frame by frame.
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// @see https://github.com/simple-rtmp-server/srs/issues/512
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return flush_audio();
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}
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int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)
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@ -1586,6 +1586,11 @@ public:
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* close the writer.
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*/
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virtual void close();
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public:
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/**
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* get the video codec of ts muxer.
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*/
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virtual SrsCodecVideo video_codec();
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};
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/**
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