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fix #151, always reap ts whatever audio or video packet. 0.9.223.

This commit is contained in:
winlin 2014-10-08 15:15:57 +08:00
parent 13b092704d
commit 9789335d0b
6 changed files with 22 additions and 20 deletions

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@ -208,6 +208,7 @@ Supported operating systems and hardware:
* 2013-10-17, Created.<br/>
## History
* 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.
* v1.0, 2014-10-08, fix [#162](https://github.com/winlinvip/simple-rtmp-server/issues/162), failed if no epoll. 0.9.222.
* 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.
* 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[] = {
/* PMT */
0xe1, 0x00,
0xf0, 0x00,
// must generate header with/without video, @see:
// https://github.com/winlinvip/simple-rtmp-server/issues/40
0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
/*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
@ -967,8 +969,6 @@ SrsHlsCache::SrsHlsCache()
af = new SrsMpegtsFrame();
vf = new SrsMpegtsFrame();
video_count = 0;
}
SrsHlsCache::~SrsHlsCache()
@ -999,9 +999,6 @@ int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment
// get the hls path config
std::string hls_path = _srs_config->get_hls_path(vhost);
// reset video count for new publish session.
video_count = 0;
// TODO: FIXME: support load exists m3u8, to continue publish stream.
// for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase.
@ -1082,9 +1079,13 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
}
}
// for pure audio
// start new segment when duration overflow.
if (video_count == 0 && muxer->is_segment_overflow()) {
// reap when current source is pure audio.
// it maybe changed when stream info changed,
// for example, pure audio when start, audio/video when publishing,
// pure audio again for audio disabled.
// so we reap event when the audio incoming when segment overflow.
// @see https://github.com/winlinvip/simple-rtmp-server/issues/151
if (muxer->is_segment_overflow()) {
if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
return ret;
}
@ -1098,8 +1099,6 @@ int SrsHlsCache::write_video(
{
int ret = ERROR_SUCCESS;
video_count++;
// write video to cache.
if ((ret = cache_video(codec, sample)) != ERROR_SUCCESS) {
return ret;

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@ -239,14 +239,6 @@ private:
int64_t audio_buffer_start_pts;
// time jitter for aac
SrsHlsAacJitter* aac_jitter;
private:
/**
* for pure audio HLS application,
* the video count used to count the video,
* if zero and audio buffer overflow, reap the ts,
* just like we got a keyframe.
*/
u_int32_t video_count;
public:
SrsHlsCache();
virtual ~SrsHlsCache();

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@ -400,7 +400,7 @@ int SrsGopCache::cache(SrsSharedPtrMessage* msg)
}
// no acceptable video or pure audio, disable the cache.
if (cached_video_count == 0) {
if (pure_audio()) {
srs_verbose("ignore any frame util got a h264 video frame.");
return ret;
}
@ -482,6 +482,11 @@ int64_t SrsGopCache::start_time()
return msg->header.timestamp;
}
bool SrsGopCache::pure_audio()
{
return cached_video_count == 0;
}
std::map<std::string, SrsSource*> SrsSource::pool;
int SrsSource::find(SrsRequest* req, SrsSource** ppsource)

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@ -263,6 +263,11 @@ public:
* @return 0 if no packets.
*/
virtual int64_t start_time();
/**
* whether current stream is pure audio,
* when no video in gop cache, the stream is pure audio right now.
*/
virtual bool pure_audio();
};
/**

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@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
// current release version
#define VERSION_MAJOR "0"
#define VERSION_MINOR "9"
#define VERSION_REVISION "222"
#define VERSION_REVISION "223"
#define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
// server info.
#define RTMP_SIG_SRS_KEY "SRS"