merge from 2.0release

@@ -814,7 +814,9 @@ About the HLS overhead of SRS, we compare the overhead to FLV by remux the HLS t
 | 5147kbps  |   370s        |   195040      |   200280      |   2.68%       |
 | 5147kbps  |   370s        |   195040      |   200280      |   2.68%       |
 | 5158kbps  |   1327s       |   835664      |   858092      |   2.68%       |
 | 5158kbps  |   1327s       |   835664      |   858092      |   2.68%       |
 
 
-The HLS overhead is calc by: (HLS - FLV) / FLV * 100%
+The HLS overhead is calc by: (HLS - FLV) / FLV * 100%.
+
+The overhead is larger than this benchmark(48kbps audio is best overhead), for we fix the [#512][bug#512].
 
 
 ## Architecture
 ## Architecture
 
 
@@ -1265,6 +1267,7 @@ Winlin
 [bug #59]: https://github.com/simple-rtmp-server/srs/issues/59
 [bug #59]: https://github.com/simple-rtmp-server/srs/issues/59
 [bug #50]: https://github.com/simple-rtmp-server/srs/issues/50
 [bug #50]: https://github.com/simple-rtmp-server/srs/issues/50
 [bug #34]: https://github.com/simple-rtmp-server/srs/issues/34
 [bug #34]: https://github.com/simple-rtmp-server/srs/issues/34
+<<<<<<< HEAD
 [bug #367]: https://github.com/simple-rtmp-server/srs/issues/367
 [bug #367]: https://github.com/simple-rtmp-server/srs/issues/367
 [bug #319]: https://github.com/simple-rtmp-server/srs/issues/319
 [bug #319]: https://github.com/simple-rtmp-server/srs/issues/319
 [bug #367]: https://github.com/simple-rtmp-server/srs/issues/367
 [bug #367]: https://github.com/simple-rtmp-server/srs/issues/367
@@ -1282,6 +1285,10 @@ Winlin
 [bug #502]: https://github.com/simple-rtmp-server/srs/issues/502
 [bug #502]: https://github.com/simple-rtmp-server/srs/issues/502
 [bug #467]: https://github.com/simple-rtmp-server/srs/issues/467
 [bug #467]: https://github.com/simple-rtmp-server/srs/issues/467
 [bug #xxxxxxx]: https://github.com/simple-rtmp-server/srs/issues/xxxxxxx
 [bug #xxxxxxx]: https://github.com/simple-rtmp-server/srs/issues/xxxxxxx
+=======
+[bug #512]: https://github.com/simple-rtmp-server/srs/issues/512
+[bug #xxxxxxxxxx]: https://github.com/simple-rtmp-server/srs/issues/xxxxxxxxxx
+>>>>>>> 2.0release
 
 
 [r2.0a2]: https://github.com/simple-rtmp-server/srs/releases/tag/v2.0-a2
 [r2.0a2]: https://github.com/simple-rtmp-server/srs/releases/tag/v2.0-a2
 [r2.0a1]: https://github.com/simple-rtmp-server/srs/releases/tag/2.0a1
 [r2.0a1]: https://github.com/simple-rtmp-server/srs/releases/tag/2.0a1

@@ -578,6 +578,7 @@ int SrsHlsMuxer::segment_open(int64_t segment_start_dts)
         current->full_path.c_str(), tmp_file.c_str());
         current->full_path.c_str(), tmp_file.c_str());
 
 
     // set the segment muxer audio codec.
     // set the segment muxer audio codec.
+    // TODO: FIXME: refine code, use event instead.
     if (acodec != SrsCodecAudioReserved1) {
     if (acodec != SrsCodecAudioReserved1) {
         current->muxer->update_acodec(acodec);
         current->muxer->update_acodec(acodec);
     }
     }
@@ -646,6 +647,11 @@ int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
     return current->muxer->update_acodec(ac);
     return current->muxer->update_acodec(ac);
 }
 }
 
 
+bool SrsHlsMuxer::pure_audio()
+{
+    return current && current->muxer && current->muxer->video_codec() == SrsCodecVideoDisabled;
+}
+
 int SrsHlsMuxer::flush_audio(SrsTsCache* cache)
 int SrsHlsMuxer::flush_audio(SrsTsCache* cache)
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
@@ -1049,25 +1055,6 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
         return ret;
         return ret;
     }
     }
     
     
-    // flush if buffer exceed max size.
-    if (cache->audio->payload->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
-        if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
-            return ret;
-        }
-    }
-
-    // TODO: config it.
-    // in ms, audio delay to flush the audios.
-    int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
-    // flush if audio delay exceed
-    // cache->audio will be free in flush_audio
-    // so we must check whether it's null ptr.
-    if (cache->audio && pts - cache->audio->start_pts > audio_delay * 90) {
-        if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
-            return ret;
-        }
-    }
-    
     // reap when current source is pure audio.
     // reap when current source is pure audio.
     // it maybe changed when stream info changed,
     // it maybe changed when stream info changed,
     // for example, pure audio when start, audio/video when publishing,
     // for example, pure audio when start, audio/video when publishing,
@@ -1083,6 +1070,21 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
         }
         }
     }
     }
     
     
+    // for pure audio, aggregate some frame to one.
+    if (muxer->pure_audio() && cache->audio) {
+        if (pts - cache->audio->start_pts < SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE) {
+            return ret;
+        }
+    }
+    
+    // directly write the audio frame by frame to ts,
+    // it's ok for the hls overload, or maybe cause the audio corrupt,
+    // which introduced by aggregate the audios to a big one.
+    // @see https://github.com/simple-rtmp-server/srs/issues/512
+    if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
     return ret;
     return ret;
 }
 }
     
     
@@ -1100,7 +1102,7 @@ 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);
@@ -1110,9 +1112,6 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
             if ((ret = reap_segment("video", muxer, cache->video->dts)) != ERROR_SUCCESS) {
             if ((ret = reap_segment("video", muxer, cache->video->dts)) != ERROR_SUCCESS) {
                 return ret;
                 return ret;
             }
             }
-            
-            // the video must be flushed, just return.
-            return ret;
         }
         }
     }
     }
     
     

@@ -309,6 +309,10 @@ public:
     virtual bool is_segment_absolutely_overflow();
     virtual bool is_segment_absolutely_overflow();
 public:
 public:
     virtual int update_acodec(SrsCodecAudio ac);
     virtual int update_acodec(SrsCodecAudio ac);
+    /**
+     * whether current hls muxer is pure audio mode.
+     */
+    virtual bool pure_audio();
     virtual int flush_audio(SrsTsCache* cache);
     virtual int flush_audio(SrsTsCache* cache);
     virtual int flush_video(SrsTsCache* cache);
     virtual int flush_video(SrsTsCache* cache);
     /**
     /**

@@ -246,12 +246,6 @@ extern int aac_sample_rates[];
 #define SRS_SRS_MAX_CODEC_SAMPLE 128
 #define SRS_SRS_MAX_CODEC_SAMPLE 128
 #define SRS_AAC_SAMPLE_RATE_UNSET 15
 #define SRS_AAC_SAMPLE_RATE_UNSET 15
 
 
-// in ms, for HLS aac flush the audio
-#define SRS_CONF_DEFAULT_AAC_DELAY 60
-
-// max PES packets size to flush the video.
-#define SRS_AUTO_HLS_AUDIO_CACHE_SIZE 128 * 1024
-
 /**
 /**
 * the FLV/RTMP supported audio sample size.
 * the FLV/RTMP supported audio sample size.
 * Size of each audio sample. This parameter only pertains to
 * Size of each audio sample. This parameter only pertains to

@@ -469,8 +469,11 @@ int SrsTsContext::encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t p
     while (p < end) {
     while (p < end) {
         SrsTsPacket* pkt = NULL;
         SrsTsPacket* pkt = NULL;
         if (p == start) {
         if (p == start) {
-            // for pure audio stream, always write pcr.
+            // write pcr according to message.
             bool write_pcr = msg->write_pcr;
             bool write_pcr = msg->write_pcr;
+            
+            // for pure audio, always write pcr.
+            // TODO: FIXME: maybe only need to write at begin and end of ts.
             if (pure_audio && msg->is_audio()) {
             if (pure_audio && msg->is_audio()) {
                 write_pcr = true;
                 write_pcr = true;
             }
             }
@@ -2772,6 +2775,11 @@ void SrsTSMuxer::close()
     writer->close();
     writer->close();
 }
 }
 
 
+SrsCodecVideo SrsTSMuxer::video_codec()
+{
+    return vcodec;
+}
+
 SrsTsCache::SrsTsCache()
 SrsTsCache::SrsTsCache()
 {
 {
     audio = NULL;
     audio = NULL;
@@ -2792,11 +2800,12 @@ int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample*
     if (!audio) {
     if (!audio) {
         audio = new SrsTsMessage();
         audio = new SrsTsMessage();
         audio->write_pcr = false;
         audio->write_pcr = false;
-        audio->start_pts = dts;
+        audio->dts = audio->pts = audio->start_pts = dts;
     }
     }
 
 
-    audio->dts = dts;
-    audio->pts = audio->dts;
+    // TODO: FIXME: refine code.
+    //audio->dts = dts;
+    //audio->pts = audio->dts;
     audio->sid = SrsTsPESStreamIdAudioCommon;
     audio->sid = SrsTsPESStreamIdAudioCommon;
     
     
     // must be aac or mp3
     // must be aac or mp3
@@ -3146,20 +3155,11 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
         return ret;
         return ret;
     }
     }
     
     
-    // flush if buffer exceed max size.
-    if (cache->audio->payload->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
-        return flush_video();
-    }
+    // TODO: FIXME: for pure audio, aggregate some frame to one.
     
     
-    // TODO: config it.
-    // in ms, audio delay to flush the audios.
-    int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
-    // flush if audio delay exceed
-    if (dts - cache->audio->start_pts > audio_delay * 90) {
+    // always flush audio frame by frame.
+    // @see https://github.com/simple-rtmp-server/srs/issues/512
     return flush_audio();
     return flush_audio();
-    }
-
-    return ret;
 }
 }
 
 
 int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)
 int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)

@@ -54,6 +54,9 @@ class SrsTsContext;
 // Transport Stream packets are 188 bytes in length.
 // Transport Stream packets are 188 bytes in length.
 #define SRS_TS_PACKET_SIZE          188
 #define SRS_TS_PACKET_SIZE          188
 
 
+// the aggregate pure audio for hls, in ts tbn(ms * 90).
+#define SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE 720 * 90
+
 /**
 /**
 * the pid of ts packet,
 * the pid of ts packet,
 * Table 2-3 - PID table, hls-mpeg-ts-iso13818-1.pdf, page 37
 * Table 2-3 - PID table, hls-mpeg-ts-iso13818-1.pdf, page 37
@@ -360,6 +363,7 @@ public:
     /**
     /**
      * whether the hls stream is pure audio stream.
      * whether the hls stream is pure audio stream.
      */
      */
+    // TODO: FIXME: merge with muxer codec detect.
     virtual bool is_pure_audio();
     virtual bool is_pure_audio();
     /**
     /**
      * when PMT table parsed, we know some info about stream.
      * when PMT table parsed, we know some info about stream.
@@ -1594,6 +1598,11 @@ public:
     * close the writer.
     * close the writer.
     */
     */
     virtual void close();
     virtual void close();
+public:
+    /**
+     * get the video codec of ts muxer.
+     */
+    virtual SrsCodecVideo video_codec();
 };
 };
 
 
 /**
 /**