refine the overflow algorithm, prevent smaller piece.

trunk/src/app/srs_app_hls.cpp

@@ -542,6 +542,11 @@ bool SrsHlsMuxer::is_segment_overflow()
 {
     srs_assert(current);
     
+    // to prevent very small segment.
+    if (current->duration < 2 * SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
+        return false;
+    }
+    
     // use N% deviation, to smoother.
     double deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0.0;
     srs_info("hls: dur=%.2f, tar=%.2f, dev=%.2fms/%dp, frag=%.2f",
@@ -560,6 +565,11 @@ bool SrsHlsMuxer::is_segment_absolutely_overflow()
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-83553950
     srs_assert(current);
     
+    // to prevent very small segment.
+    if (current->duration < 2 * SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
+        return false;
+    }
+    
     // use N% deviation, to smoother.
     double deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0.0;
     srs_info("hls: dur=%.2f, tar=%.2f, dev=%.2fms/%dp, frag=%.2f",

trunk/src/main/srs_main_ingest_hls.cpp

@@ -871,9 +871,10 @@ int SrsIngestSrsOutput::parse_message_queue()
     
     SrsTsMessage* first_ts_msg = queue.begin()->second;
     SrsTsContext* context = first_ts_msg->channel->context;
+    bool cpa = context->is_pure_audio();
     
     int nb_videos = 0;
-    if (!context->is_pure_audio()) {
+    if (!cpa) {
         std::multimap<int64_t, SrsTsMessage*>::iterator it;
         for (it = queue.begin(); it != queue.end(); ++it) {
             SrsTsMessage* msg = it->second;
@@ -892,7 +893,8 @@ int SrsIngestSrsOutput::parse_message_queue()
     }
     
     // parse messages util the last video.
-    while ((nb_videos > 1 || context->is_pure_audio()) && queue.size() > 0) {
+    while ((cpa && queue.size() > 1) || nb_videos > 1) {
+        srs_assert(!queue.empty());
         std::multimap<int64_t, SrsTsMessage*>::iterator it = queue.begin();
         
         SrsTsMessage* msg = it->second;