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merge from 2.0release

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pull/528/head
winlin 9 years ago
parent fe27e76492 d1979c760f
commit 1d57e53910
6 changed files with 62 additions and 49 deletions
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9
README.md
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@ -814,7 +814,9 @@ About the HLS overhead of SRS, we compare the overhead to FLV by remux the HLS t
| 5147kbps | 370s | 195040 | 200280 | 2.68% |
| 5158kbps | 1327s | 835664 | 858092 | 2.68% |
The HLS overhead is calc by: (HLS - FLV) / FLV * 100%
The HLS overhead is calc by: (HLS - FLV) / FLV * 100%.
The overhead is larger than this benchmark(48kbps audio is best overhead), for we fix the [#512][bug#512].
## Architecture
@ -1265,6 +1267,7 @@ Winlin
[bug #59]: https://github.com/simple-rtmp-server/srs/issues/59
[bug #50]: https://github.com/simple-rtmp-server/srs/issues/50
[bug #34]: https://github.com/simple-rtmp-server/srs/issues/34
<<<<<<< HEAD
[bug #367]: https://github.com/simple-rtmp-server/srs/issues/367
[bug #319]: https://github.com/simple-rtmp-server/srs/issues/319
[bug #367]: https://github.com/simple-rtmp-server/srs/issues/367
@ -1282,6 +1285,10 @@ Winlin
[bug #502]: https://github.com/simple-rtmp-server/srs/issues/502
[bug #467]: https://github.com/simple-rtmp-server/srs/issues/467
[bug #xxxxxxx]: https://github.com/simple-rtmp-server/srs/issues/xxxxxxx
=======
[bug #512]: https://github.com/simple-rtmp-server/srs/issues/512
[bug #xxxxxxxxxx]: https://github.com/simple-rtmp-server/srs/issues/xxxxxxxxxx
>>>>>>> 2.0release
[r2.0a2]: https://github.com/simple-rtmp-server/srs/releases/tag/v2.0-a2
[r2.0a1]: https://github.com/simple-rtmp-server/srs/releases/tag/2.0a1

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;
}
}

4
trunk/src/app/srs_app_hls.hpp
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@ -309,6 +309,10 @@ public:
virtual bool is_segment_absolutely_overflow();
public:
virtual int update_acodec(SrsCodecAudio ac);
/**
* whether current hls muxer is pure audio mode.
*/
virtual bool pure_audio();
virtual int flush_audio(SrsTsCache* cache);
virtual int flush_video(SrsTsCache* cache);
/**

6
trunk/src/kernel/srs_kernel_codec.hpp
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@ -246,12 +246,6 @@ extern int aac_sample_rates[];
#define SRS_SRS_MAX_CODEC_SAMPLE 128
#define SRS_AAC_SAMPLE_RATE_UNSET 15
// in ms, for HLS aac flush the audio
#define SRS_CONF_DEFAULT_AAC_DELAY 60
// max PES packets size to flush the video.
#define SRS_AUTO_HLS_AUDIO_CACHE_SIZE 128 * 1024
/**
* the FLV/RTMP supported audio sample size.
* Size of each audio sample. This parameter only pertains to

36
trunk/src/kernel/srs_kernel_ts.cpp
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@ -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)

9
trunk/src/kernel/srs_kernel_ts.hpp
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@ -54,6 +54,9 @@ class SrsTsContext;
// Transport Stream packets are 188 bytes in length.
#define SRS_TS_PACKET_SIZE 188
// the aggregate pure audio for hls, in ts tbn(ms * 90).
#define SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE 720 * 90
/**
* the pid of ts packet,
* Table 2-3 - PID table, hls-mpeg-ts-iso13818-1.pdf, page 37
@ -360,6 +363,7 @@ public:
/**
* whether the hls stream is pure audio stream.
*/
// TODO: FIXME: merge with muxer codec detect.
virtual bool is_pure_audio();
/**
* when PMT table parsed, we know some info about stream.
@ -1594,6 +1598,11 @@ public:
* close the writer.
*/
virtual void close();
public:
/**
* get the video codec of ts muxer.
*/
virtual SrsCodecVideo video_codec();
};
/**

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