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refine the overflow algorithm, prevent smaller piece.
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parent
d19dfa528e
commit
1d973e2178
2 changed files with 14 additions and 2 deletions
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@ -542,6 +542,11 @@ bool SrsHlsMuxer::is_segment_overflow()
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{
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srs_assert(current);
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// to prevent very small segment.
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if (current->duration < 2 * SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
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return false;
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}
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// use N% deviation, to smoother.
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double deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0.0;
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srs_info("hls: dur=%.2f, tar=%.2f, dev=%.2fms/%dp, frag=%.2f",
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@ -560,6 +565,11 @@ bool SrsHlsMuxer::is_segment_absolutely_overflow()
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-83553950
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srs_assert(current);
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// to prevent very small segment.
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if (current->duration < 2 * SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
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return false;
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}
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// use N% deviation, to smoother.
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double deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0.0;
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srs_info("hls: dur=%.2f, tar=%.2f, dev=%.2fms/%dp, frag=%.2f",
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@ -871,9 +871,10 @@ int SrsIngestSrsOutput::parse_message_queue()
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SrsTsMessage* first_ts_msg = queue.begin()->second;
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SrsTsContext* context = first_ts_msg->channel->context;
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bool cpa = context->is_pure_audio();
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int nb_videos = 0;
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if (!context->is_pure_audio()) {
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if (!cpa) {
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std::multimap<int64_t, SrsTsMessage*>::iterator it;
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for (it = queue.begin(); it != queue.end(); ++it) {
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SrsTsMessage* msg = it->second;
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@ -892,7 +893,8 @@ int SrsIngestSrsOutput::parse_message_queue()
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}
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// parse messages util the last video.
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while ((nb_videos > 1 || context->is_pure_audio()) && queue.size() > 0) {
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while ((cpa && queue.size() > 1) || nb_videos > 1) {
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srs_assert(!queue.empty());
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std::multimap<int64_t, SrsTsMessage*>::iterator it = queue.begin();
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SrsTsMessage* msg = it->second;
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