Merge branch '2.0release' into develop

@@ -542,6 +542,11 @@ bool SrsHlsMuxer::is_segment_overflow()
 {
 {
     srs_assert(current);
     srs_assert(current);
     
     
+    // to prevent very small segment.
+    if (current->duration * 1000 < 2 * SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
+        return false;
+    }
+    
     // use N% deviation, to smoother.
     // use N% deviation, to smoother.
     double deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0.0;
     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",
     srs_info("hls: dur=%.2f, tar=%.2f, dev=%.2fms/%dp, frag=%.2f",
@@ -559,7 +564,18 @@ bool SrsHlsMuxer::is_segment_absolutely_overflow()
 {
 {
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-83553950
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-83553950
     srs_assert(current);
     srs_assert(current);
-    return current->duration >= hls_aof_ratio * hls_fragment;
+    
+    // to prevent very small segment.
+    if (current->duration * 1000 < 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",
+             current->duration, hls_fragment + deviation, deviation, deviation_ts, hls_fragment);
+    
+    return current->duration >= hls_aof_ratio * hls_fragment + deviation;
 }
 }
 
 
 int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
 int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
@@ -968,7 +984,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     // we use absolutely overflow of segment to make jwplayer/ffplay happy
     // we use absolutely overflow of segment to make jwplayer/ffplay happy
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
     if (cache->audio && muxer->is_segment_absolutely_overflow()) {
     if (cache->audio && muxer->is_segment_absolutely_overflow()) {
-        srs_warn("hls: absolute audio reap segment.");
+        srs_info("hls: absolute audio reap segment.");
         if ((ret = reap_segment("audio", muxer, cache->audio->pts)) != ERROR_SUCCESS) {
         if ((ret = reap_segment("audio", muxer, cache->audio->pts)) != ERROR_SUCCESS) {
             return ret;
             return ret;
         }
         }
@@ -991,7 +1007,7 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
         // do reap ts if any of:
         // do reap ts if any of:
         //      a. wait keyframe and got keyframe.
         //      a. wait keyframe and got keyframe.
         //      b. always reap when not wait keyframe.
         //      b. always reap when not wait keyframe.
-        if (!muxer->wait_keyframe() || sample->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
+        if (!muxer->wait_keyframe()|| sample->frame_type == SrsCodecVideoAVCFrameKeyFrame) {
             // when wait keyframe, there must exists idr frame in sample.
             // when wait keyframe, there must exists idr frame in sample.
             if (!sample->has_idr && muxer->wait_keyframe()) {
             if (!sample->has_idr && muxer->wait_keyframe()) {
                 srs_warn("hls: ts starts without IDR, first nalu=%d, idr=%d", sample->first_nalu_type, sample->has_idr);
                 srs_warn("hls: ts starts without IDR, first nalu=%d, idr=%d", sample->first_nalu_type, sample->has_idr);

@@ -78,6 +78,7 @@ SrsTsChannel::SrsTsChannel()
     stream = SrsTsStreamReserved;
     stream = SrsTsStreamReserved;
     msg = NULL;
     msg = NULL;
     continuity_counter = 0;
     continuity_counter = 0;
+    context = NULL;
 }
 }
 
 
 SrsTsChannel::~SrsTsChannel()
 SrsTsChannel::~SrsTsChannel()
@@ -196,6 +197,7 @@ ISrsTsHandler::~ISrsTsHandler()
 
 
 SrsTsContext::SrsTsContext()
 SrsTsContext::SrsTsContext()
 {
 {
+    pure_audio = false;
     vcodec = SrsCodecVideoReserved;
     vcodec = SrsCodecVideoReserved;
     acodec = SrsCodecAudioReserved1;
     acodec = SrsCodecAudioReserved1;
 }
 }
@@ -210,6 +212,24 @@ SrsTsContext::~SrsTsContext()
     pids.clear();
     pids.clear();
 }
 }
 
 
+bool SrsTsContext::is_pure_audio()
+{
+    return pure_audio;
+}
+
+void SrsTsContext::on_pmt_parsed()
+{
+    pure_audio = true;
+    
+    std::map<int, SrsTsChannel*>::iterator it;
+    for (it = pids.begin(); it != pids.end(); ++it) {
+        SrsTsChannel* channel = it->second;
+        if (channel->apply == SrsTsPidApplyVideo) {
+            pure_audio = false;
+        }
+    }
+}
+
 void SrsTsContext::reset()
 void SrsTsContext::reset()
 {
 {
     vcodec = SrsCodecVideoReserved;
     vcodec = SrsCodecVideoReserved;
@@ -230,6 +250,7 @@ void SrsTsContext::set(int pid, SrsTsPidApply apply_pid, SrsTsStream stream)
 
 
     if (pids.find(pid) == pids.end()) {
     if (pids.find(pid) == pids.end()) {
         channel = new SrsTsChannel();
         channel = new SrsTsChannel();
+        channel->context = this;
         pids[pid] = channel;
         pids[pid] = channel;
     } else {
     } else {
         channel = pids[pid];
         channel = pids[pid];
@@ -2302,6 +2323,7 @@ int SrsTsPayloadPAT::psi_decode(SrsStream* stream)
 
 
     // update the apply pid table.
     // update the apply pid table.
     packet->context->set(packet->pid, SrsTsPidApplyPAT);
     packet->context->set(packet->pid, SrsTsPidApplyPAT);
+    packet->context->on_pmt_parsed();
 
 
     return ret;
     return ret;
 }
 }

@@ -172,6 +172,7 @@ struct SrsTsChannel
     SrsTsPidApply apply;
     SrsTsPidApply apply;
     SrsTsStream stream;
     SrsTsStream stream;
     SrsTsMessage* msg;
     SrsTsMessage* msg;
+    SrsTsContext* context;
     // for encoder.
     // for encoder.
     u_int8_t continuity_counter;
     u_int8_t continuity_counter;
 
 
@@ -343,6 +344,7 @@ class SrsTsContext
 // codec
 // codec
 private:
 private:
     std::map<int, SrsTsChannel*> pids;
     std::map<int, SrsTsChannel*> pids;
+    bool pure_audio;
 // encoder
 // encoder
 private:
 private:
     // when any codec changed, write the PAT/PMT.
     // when any codec changed, write the PAT/PMT.
@@ -353,6 +355,14 @@ public:
     virtual ~SrsTsContext();
     virtual ~SrsTsContext();
 public:
 public:
     /**
     /**
+     * whether the hls stream is pure audio stream.
+     */
+    virtual bool is_pure_audio();
+    /**
+     * when PMT table parsed, we know some info about stream.
+     */
+    virtual void on_pmt_parsed();
+    /**
      * reset the context for a new ts segment start.
      * reset the context for a new ts segment start.
      */
      */
     virtual void reset();
     virtual void reset();

@@ -561,7 +561,7 @@ void SrsIngestSrsInput::fetch_all_ts(bool fresh_m3u8)
         }
         }
         
         
         // only wait for a duration of last piece.
         // only wait for a duration of last piece.
-        if (i == pieces.size() - 1) {
+        if (i == (int)pieces.size() - 1) {
             next_connect_time = srs_update_system_time_ms() + (int)tp->duration * 1000;
             next_connect_time = srs_update_system_time_ms() + (int)tp->duration * 1000;
         }
         }
     }
     }
@@ -657,7 +657,7 @@ public:
     SrsIngestSrsOutput(SrsHttpUri* rtmp) {
     SrsIngestSrsOutput(SrsHttpUri* rtmp) {
         out_rtmp = rtmp;
         out_rtmp = rtmp;
         disconnected = false;
         disconnected = false;
-        raw_aac_dts = 0;
+        raw_aac_dts = srs_update_system_time_ms();
         
         
         req = NULL;
         req = NULL;
         io = NULL;
         io = NULL;
@@ -807,12 +807,14 @@ int SrsIngestSrsOutput::do_on_aac_frame(SrsStream* avs, double duration)
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
     
     
+    u_int32_t duration_ms = (u_int32_t)(duration * 1000);
+    
     // ts tbn to flv tbn.
     // ts tbn to flv tbn.
     u_int32_t dts = (u_int32_t)raw_aac_dts;
     u_int32_t dts = (u_int32_t)raw_aac_dts;
-    raw_aac_dts += (int64_t)(duration * 1000);
+    raw_aac_dts += duration_ms;
     
     
     // got the next msg to calc the delta duration for each audio.
     // got the next msg to calc the delta duration for each audio.
-    u_int32_t max_dts = dts + (u_int32_t)(duration * 1000);
+    u_int32_t max_dts = dts + duration_ms;
     
     
     // send each frame.
     // send each frame.
     while (!avs->empty()) {
     while (!avs->empty()) {
@@ -852,7 +854,7 @@ int SrsIngestSrsOutput::do_on_aac_frame(SrsStream* avs, double duration)
         }
         }
         
         
         // calc the delta of dts, when previous frame output.
         // calc the delta of dts, when previous frame output.
-        u_int32_t delta = (duration * 1000) / (avs->size() / frame_size);
+        u_int32_t delta = duration_ms / (avs->size() / frame_size);
         dts = (u_int32_t)(srs_min(max_dts, dts + delta));
         dts = (u_int32_t)(srs_min(max_dts, dts + delta));
     }
     }
     
     
@@ -863,28 +865,36 @@ int SrsIngestSrsOutput::parse_message_queue()
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
     
     
+    if (queue.empty()) {
+        return ret;
+    }
+    
+    SrsTsMessage* first_ts_msg = queue.begin()->second;
+    SrsTsContext* context = first_ts_msg->channel->context;
+    bool cpa = context->is_pure_audio();
+    
     int nb_videos = 0;
     int nb_videos = 0;
-    int nb_audios = 0;
-    std::multimap<int64_t, SrsTsMessage*>::iterator it;
-    for (it = queue.begin(); it != queue.end(); ++it) {
-        SrsTsMessage* msg = it->second;
+    if (!cpa) {
+        std::multimap<int64_t, SrsTsMessage*>::iterator it;
+        for (it = queue.begin(); it != queue.end(); ++it) {
+            SrsTsMessage* msg = it->second;
+            
+            // publish audio or video.
+            if (msg->channel->stream == SrsTsStreamVideoH264) {
+                nb_videos++;
+            }
+        }
         
         
-        // publish audio or video.
-        if (msg->channel->stream == SrsTsStreamVideoH264) {
-            nb_videos++;
-        } else {
-            nb_audios++;
+        // always wait 2+ videos, to left one video in the queue.
+        // TODO: FIXME: support pure audio hls.
+        if (nb_videos <= 1) {
+            return ret;
         }
         }
     }
     }
     
     
-    // always wait 2+ videos, to left one video in the queue.
-    // TODO: FIXME: support pure audio hls.
-    if (nb_videos <= 1) {
-        return ret;
-    }
-    
     // parse messages util the last video.
     // parse messages util the last video.
-    while (nb_videos > 1 && queue.size() > 0) {
+    while ((cpa && queue.size() > 1) || nb_videos > 1) {
+        srs_assert(!queue.empty());
         std::multimap<int64_t, SrsTsMessage*>::iterator it = queue.begin();
         std::multimap<int64_t, SrsTsMessage*>::iterator it = queue.begin();
         
         
         SrsTsMessage* msg = it->second;
         SrsTsMessage* msg = it->second;