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for #512, partical hotfix the hls pure audio. 2.0.196
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5 changed files with 25 additions and 6 deletions
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@ -336,6 +336,7 @@ Remark:
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## History
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* v2.0, 2015-10-27, for [#512][bug #512] partical hotfix the hls pure audio. 2.0.196
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* <strong>v2.0, 2015-10-08, [2.0 alpha2(2.0.195)][r2.0a2] released. 89358 lines.</strong>
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* v2.0, 2015-10-04, for [#448][bug #448] fix the bug of response of http hooks. 2.0.195
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* v2.0, 2015-10-01, for [#497][bug #497] response error when client not found to kickoff. 2.0.194
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@ -578,6 +578,7 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
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current->full_path.c_str(), tmp_file.c_str());
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// set the segment muxer audio codec.
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// TODO: FIXME: refine code, use event instead.
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if (acodec != SrsCodecAudioReserved1) {
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current->muxer->update_acodec(acodec);
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}
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@ -1069,6 +1070,13 @@ 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, aggregate some frame to one.
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if (muxer->pure_audio() && cache->audio) {
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if (pts - cache->audio->start_pts < SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE) {
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return ret;
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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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@ -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 2
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#define VERSION_MINOR 0
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#define VERSION_REVISION 195
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#define VERSION_REVISION 196
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// server info.
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#define RTMP_SIG_SRS_KEY "SRS"
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@ -459,8 +459,11 @@ int SrsTsContext::encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t p
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while (p < end) {
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SrsTsPacket* pkt = NULL;
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if (p == start) {
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// for pure audio stream, always write pcr.
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// write pcr according to message.
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bool write_pcr = msg->write_pcr;
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// for pure audio, always write pcr.
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// TODO: FIXME: maybe only need to write at begin and end of ts.
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if (pure_audio && msg->is_audio()) {
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write_pcr = true;
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}
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@ -2785,11 +2788,12 @@ int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample*
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if (!audio) {
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audio = new SrsTsMessage();
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audio->write_pcr = false;
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audio->start_pts = dts;
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audio->dts = audio->pts = audio->start_pts = dts;
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}
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audio->dts = dts;
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audio->pts = audio->dts;
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// TODO: FIXME: refine code.
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//audio->dts = dts;
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//audio->pts = audio->dts;
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audio->sid = SrsTsPESStreamIdAudioCommon;
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// must be aac or mp3
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@ -3139,6 +3143,8 @@ 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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// TODO: FIXME: for pure audio, aggregate some frame to one.
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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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@ -54,6 +54,9 @@ class SrsTsContext;
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// Transport Stream packets are 188 bytes in length.
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#define SRS_TS_PACKET_SIZE 188
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// the aggregate pure audio for hls, in ts tbn(ms * 90).
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#define SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE 720 * 90
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/**
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* the pid of ts packet,
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* Table 2-3 - PID table, hls-mpeg-ts-iso13818-1.pdf, page 37
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@ -359,6 +362,7 @@ public:
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/**
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* whether the hls stream is pure audio stream.
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*/
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// TODO: FIXME: merge with muxer codec detect.
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virtual bool is_pure_audio();
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/**
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* when PMT table parsed, we know some info about stream.
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