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fix #151, always reap ts whatever audio or video packet. 0.9.223.
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6 changed files with 22 additions and 20 deletions
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@ -208,6 +208,7 @@ Supported operating systems and hardware:
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* 2013-10-17, Created.<br/>
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## History
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* v1.0, 2014-10-08, fix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), always reap ts whatever audio or video packet. 0.9.223.
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* v1.0, 2014-10-08, fix [#162](https://github.com/winlinvip/simple-rtmp-server/issues/162), failed if no epoll. 0.9.222.
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* v1.0, 2014-09-30, fix [#180](https://github.com/winlinvip/simple-rtmp-server/issues/180), crash for multiple edge publishing the same stream. 0.9.220.
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* v1.0, 2014-09-26, fix hls bug, refine config and log, according to clion of jetbrains. 0.9.216.
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@ -135,6 +135,8 @@ u_int8_t mpegts_header[] = {
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/* PMT */
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0xe1, 0x00,
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0xf0, 0x00,
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// must generate header with/without video, @see:
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// https://github.com/winlinvip/simple-rtmp-server/issues/40
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0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
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0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
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/*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
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@ -967,8 +969,6 @@ SrsHlsCache::SrsHlsCache()
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af = new SrsMpegtsFrame();
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vf = new SrsMpegtsFrame();
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video_count = 0;
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}
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SrsHlsCache::~SrsHlsCache()
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@ -999,9 +999,6 @@ int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment
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// get the hls path config
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std::string hls_path = _srs_config->get_hls_path(vhost);
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// reset video count for new publish session.
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video_count = 0;
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// TODO: FIXME: support load exists m3u8, to continue publish stream.
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// for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase.
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@ -1082,9 +1079,13 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
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}
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}
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// for pure audio
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// start new segment when duration overflow.
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if (video_count == 0 && muxer->is_segment_overflow()) {
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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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// pure audio again for audio disabled.
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// so we reap event when the audio incoming when segment overflow.
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/151
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if (muxer->is_segment_overflow()) {
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if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
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return ret;
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}
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@ -1098,8 +1099,6 @@ int SrsHlsCache::write_video(
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{
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int ret = ERROR_SUCCESS;
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video_count++;
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// write video to cache.
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if ((ret = cache_video(codec, sample)) != ERROR_SUCCESS) {
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return ret;
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@ -239,14 +239,6 @@ private:
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int64_t audio_buffer_start_pts;
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// time jitter for aac
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SrsHlsAacJitter* aac_jitter;
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private:
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/**
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* for pure audio HLS application,
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* the video count used to count the video,
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* if zero and audio buffer overflow, reap the ts,
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* just like we got a keyframe.
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*/
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u_int32_t video_count;
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public:
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SrsHlsCache();
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virtual ~SrsHlsCache();
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@ -400,7 +400,7 @@ int SrsGopCache::cache(SrsSharedPtrMessage* msg)
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}
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// no acceptable video or pure audio, disable the cache.
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if (cached_video_count == 0) {
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if (pure_audio()) {
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srs_verbose("ignore any frame util got a h264 video frame.");
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return ret;
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}
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@ -482,6 +482,11 @@ int64_t SrsGopCache::start_time()
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return msg->header.timestamp;
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}
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bool SrsGopCache::pure_audio()
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{
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return cached_video_count == 0;
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}
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std::map<std::string, SrsSource*> SrsSource::pool;
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int SrsSource::find(SrsRequest* req, SrsSource** ppsource)
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@ -263,6 +263,11 @@ public:
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* @return 0 if no packets.
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*/
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virtual int64_t start_time();
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/**
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* whether current stream is pure audio,
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* when no video in gop cache, the stream is pure audio right now.
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*/
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virtual bool pure_audio();
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};
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/**
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@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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// current release version
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#define VERSION_MAJOR "0"
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#define VERSION_MINOR "9"
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#define VERSION_REVISION "222"
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#define VERSION_REVISION "223"
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#define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
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// server info.
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#define RTMP_SIG_SRS_KEY "SRS"
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