merge from 2.0release

README.md

@@ -1,13 +1,13 @@
 #Simple-RTMP-Server
 
-SRS/3.0，开发代号：[OuXuli](https://github.com/simple-rtmp-server/srs/wiki/v1_CN_Product#release30)
+SRS/3.0, [OuXuli](https://github.com/simple-rtmp-server/srs/wiki/v1_CN_Product#release30)
 
 SRS定位是运营级的互联网直播服务器集群，追求更好的概念完整性和最简单实现的代码。<br/>
 SRS is industrial-strength live streaming cluster, for the best conceptual integrity and the simplest implementation. 
 
 Download from github.io: [Centos6-x86_64][centos0], [more...][more0]<br/>
 Download from ossrs.net: [Centos6-x86_64][centos1], [more...][more1]<br/>
-Contact by QQ or Skype, read [Contact][contact]
+Website for SRS/2.0, read SRS 2.0 [Chinese][srs2_CN] or [English][srs2_EN].
 
 ## Why SRS?
 
@@ -1025,5 +1025,6 @@ Winlin
 [centos0]: http://winlinvip.github.io/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.32.zip
 [centos1]: http://www.ossrs.net/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.32.zip
 
-
+[srs2_CN]: https://github.com/simple-rtmp-server/srs/wiki/v2_CN_Home
+[srs2_EN]: https://github.com/simple-rtmp-server/srs/wiki/v2_EN_Home
 

trunk/research/librtmp/srs_ingest_flv.c

@@ -38,6 +38,7 @@ int proxy(srs_flv_t flv, srs_rtmp_t ortmp);
 int connect_oc(srs_rtmp_t ortmp);
 
 #define RE_PULSE_MS 300
+#define RE_PULSE_JITTER_MS 3000
 int64_t re_create();
 void re_update(int64_t re, int32_t starttime, u_int32_t time);
 void re_cleanup(int64_t re, int32_t starttime, u_int32_t time);
@@ -256,7 +257,7 @@ void re_update(int64_t re, int32_t starttime, u_int32_t time)
     // send by pulse algorithm.
     int64_t now = srs_utils_time_ms();
     int64_t diff = time - starttime - (now -re);
-    if (diff > RE_PULSE_MS) {
+    if (diff > RE_PULSE_MS && diff < RE_PULSE_JITTER_MS) {
         usleep((useconds_t)(diff * 1000));
     }
 }

trunk/src/app/srs_app_config.cpp

@@ -853,6 +853,7 @@ int SrsConfig::reload_vhost(SrsConfDirective* old_root)
                 }
             }
 
+            // TODO: reload new http_remux in on_vhost_add
             // http_remux, only one per vhost.
             if (get_vhost_http_remux_enabled(vhost)) {
                 for (it = subscribes.begin(); it != subscribes.end(); ++it) {
@@ -4417,3 +4418,18 @@ bool srs_config_dvr_is_plan_append(string plan)
 {
     return plan == SRS_CONF_DEFAULT_DVR_PLAN_APPEND;
 }
+
+bool srs_stream_caster_is_udp(string caster)
+{
+    return caster == SRS_CONF_DEFAULT_STREAM_CASTER_MPEGTS_OVER_UDP;
+}
+
+bool srs_stream_caster_is_rtsp(string caster)
+{
+    return caster == SRS_CONF_DEFAULT_STREAM_CASTER_RTSP;
+}
+
+bool srs_stream_caster_is_flv(string caster)
+{
+    return caster == SRS_CONF_DEFAULT_STREAM_CASTER_FLV;
+}

trunk/src/app/srs_app_config.hpp

@@ -1143,6 +1143,9 @@ extern bool srs_config_ingest_is_stream(std::string type);
 extern bool srs_config_dvr_is_plan_segment(std::string plan);
 extern bool srs_config_dvr_is_plan_session(std::string plan);
 extern bool srs_config_dvr_is_plan_append(std::string plan);
+extern bool srs_stream_caster_is_udp(std::string caster);
+extern bool srs_stream_caster_is_rtsp(std::string caster);
+extern bool srs_stream_caster_is_flv(std::string caster);
 
 // global config
 extern SrsConfig* _srs_config;

trunk/src/app/srs_app_forward.cpp

@@ -164,7 +164,7 @@ int SrsForwarder::on_meta_data(SrsSharedPtrMessage* shared_metadata)
     SrsSharedPtrMessage* metadata = shared_metadata->copy();
     
     // TODO: FIXME: config the jitter of Forwarder.
-    if ((ret = jitter->correct(metadata, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    if ((ret = jitter->correct(metadata, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
         srs_freep(metadata);
         return ret;
     }
@@ -183,7 +183,7 @@ int SrsForwarder::on_audio(SrsSharedPtrMessage* shared_audio)
     SrsSharedPtrMessage* msg = shared_audio->copy();
     
     // TODO: FIXME: config the jitter of Forwarder.
-    if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
         srs_freep(msg);
         return ret;
     }
@@ -207,7 +207,7 @@ int SrsForwarder::on_video(SrsSharedPtrMessage* shared_video)
     SrsSharedPtrMessage* msg = shared_video->copy();
     
     // TODO: FIXME: config the jitter of Forwarder.
-    if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
         srs_freep(msg);
         return ret;
     }

trunk/src/app/srs_app_hls.cpp

@@ -53,6 +53,8 @@ using namespace std;
 
 // drop the segment when duration of ts too small.
 #define SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS 100
+// when hls timestamp jump, reset it.
+#define SRS_AUTO_HLS_SEGMENT_TIMESTAMP_JUMP_MS 300
 
 // fragment plus the deviation percent.
 #define SRS_HLS_FLOOR_REAP_PERCENT 0.3
@@ -161,6 +163,11 @@ void SrsHlsSegment::update_duration(int64_t current_frame_dts)
     // update the segment duration, which is nagetive,
     // just ignore it.
     if (current_frame_dts < segment_start_dts) {
+        // for atc and timestamp jump, reset the start dts.
+        if (current_frame_dts < segment_start_dts - SRS_AUTO_HLS_SEGMENT_TIMESTAMP_JUMP_MS * 90) {
+            srs_warn("hls timestamp jump %"PRId64"=>%"PRId64, segment_start_dts, current_frame_dts);
+            segment_start_dts = current_frame_dts;
+        }
         return;
     }
     
@@ -280,7 +287,7 @@ SrsHlsMuxer::SrsHlsMuxer()
     previous_floor_ts = 0;
     accept_floor_ts = 0;
     hls_ts_floor = false;
-    target_duration = 0;
+    max_td = 0;
     _sequence_no = 0;
     current = NULL;
     acodec = SrsCodecAudioReserved1;
@@ -400,10 +407,8 @@ int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix,
     m3u8_url = srs_path_build_stream(m3u8_file, req->vhost, req->app, req->stream);
     m3u8 = path + "/" + m3u8_url;
 
-    // we always keep the target duration increasing.
+    // when update config, reset the history target duration.
-    int max_td = srs_max(target_duration, (int)(fragment * _srs_config->get_hls_td_ratio(r->vhost)));
+    max_td = (int)(fragment * _srs_config->get_hls_td_ratio(r->vhost));
-    srs_info("hls update target duration %d=>%d, aof=%.2f", target_duration, max_td, aof_ratio);
-    target_duration = max_td;
 
     std::string storage = _srs_config->get_hls_storage(r->vhost);
     if (storage == "ram") {
@@ -699,7 +704,9 @@ int SrsHlsMuxer::segment_close(string log_desc)
     srs_assert(it == segments.end());
 
     // valid, add to segments if segment duration is ok
-    if (current->duration * 1000 >= SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
+    // when too small, it maybe not enough data to play.
+    // when too large, it maybe timestamp corrupt.
+    if (current->duration * 1000 >= SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS && (int)current->duration <= max_td) {
         segments.push_back(current);
         
         // use async to call the http hooks, for it will cause thread switch.
@@ -750,7 +757,6 @@ int SrsHlsMuxer::segment_close(string log_desc)
         // rename from tmp to real path
         std::string tmp_file = current->full_path + ".tmp";
         if (should_write_file) {
-            unlink(tmp_file.c_str());
             if (unlink(tmp_file.c_str()) < 0) {
                 srs_warn("ignore unlink path failed, file=%s.", tmp_file.c_str());
             }
@@ -817,6 +823,11 @@ int SrsHlsMuxer::refresh_m3u8()
 {
     int ret = ERROR_SUCCESS;
     
+    // no segments, also no m3u8, return.
+    if (segments.size() == 0) {
+        return ret;
+    }
+    
     std::string temp_m3u8 = m3u8 + ".temp";
     if ((ret = _refresh_m3u8(temp_m3u8)) == ERROR_SUCCESS) {
         if (should_write_file && rename(temp_m3u8.c_str(), m3u8.c_str()) < 0) {
@@ -826,7 +837,11 @@ int SrsHlsMuxer::refresh_m3u8()
     }
     
     // remove the temp file.
-    unlink(temp_m3u8.c_str());
+    if (srs_path_exists(temp_m3u8)) {
+        if (unlink(temp_m3u8.c_str()) < 0) {
+            srs_warn("ignore remove m3u8 failed, %s", temp_m3u8.c_str());
+        }
+    }
     
     return ret;
 }
@@ -861,6 +876,9 @@ int SrsHlsMuxer::_refresh_m3u8(string m3u8_file)
     ss << "#EXT-X-MEDIA-SEQUENCE:" << first->sequence_no << SRS_CONSTS_LF;
     srs_verbose("write m3u8 sequence success.");
     
+    // iterator shared for td generation and segemnts wrote.
+    std::vector<SrsHlsSegment*>::iterator it;
+    
     // #EXT-X-TARGETDURATION:4294967295\n
     /**
     * @see hls-m3u8-draft-pantos-http-live-streaming-12.pdf, page 25
@@ -871,11 +889,13 @@ int SrsHlsMuxer::_refresh_m3u8(string m3u8_file)
     * typical target duration is 10 seconds.
     */
     // @see https://github.com/simple-rtmp-server/srs/issues/304#issuecomment-74000081
-    std::vector<SrsHlsSegment*>::iterator it;
+    int target_duration = 0;
     for (it = segments.begin(); it != segments.end(); ++it) {
         SrsHlsSegment* segment = *it;
         target_duration = srs_max(target_duration, (int)ceil(segment->duration));
     }
+    target_duration = srs_max(target_duration, max_td);
+    
     ss << "#EXT-X-TARGETDURATION:" << target_duration << SRS_CONSTS_LF;
     srs_verbose("write m3u8 duration success.");
     
@@ -1335,7 +1355,7 @@ int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio)
     }
     
     // TODO: FIXME: config the jitter of HLS.
-    if ((ret = jitter->correct(audio, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    if ((ret = jitter->correct(audio, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
         srs_error("rtmp jitter correct audio failed. ret=%d", ret);
         return ret;
     }
@@ -1393,7 +1413,7 @@ int SrsHls::on_video(SrsSharedPtrMessage* shared_video)
     }
     
     // TODO: FIXME: config the jitter of HLS.
-    if ((ret = jitter->correct(video, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
+    if ((ret = jitter->correct(video, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
         srs_error("rtmp jitter correct video failed. ret=%d", ret);
         return ret;
     }

trunk/src/app/srs_app_hls.hpp

@@ -233,7 +233,7 @@ private:
     int64_t previous_floor_ts;
 private:
     int _sequence_no;
-    int target_duration;
+    int max_td;
     std::string m3u8;
     std::string m3u8_url;
 private:

trunk/src/app/srs_app_http_stream.cpp

@@ -805,7 +805,14 @@ int SrsHttpStreamServer::http_mount(SrsSource* s, SrsRequest* r)
         entry->cache = new SrsStreamCache(s, r);
         entry->stream = new SrsLiveStream(s, r, entry->cache);
 
-        srs_assert(!tmpl->req);
+        // TODO: FIXME: maybe refine the logic of http remux service.
+        // if user push streams followed:
+        //     rtmp://test.com/live/stream1
+        //     rtmp://test.com/live/stream2
+        // and they will using the same template, such as: [vhost]/[app]/[stream].flv
+        // so, need to free last request object, otherwise, it will cause memory leak.
+        srs_freep(tmpl->req);
+
         tmpl->source = s;
         tmpl->req = r->copy();
         
@@ -1170,8 +1177,8 @@ int SrsHttpStreamServer::hijack(ISrsHttpMessage* request, ISrsHttpHandler** ph)
     std::string sid = r->get_stream_url();
     // check if the stream is enabled.
     if (sflvs.find(sid) != sflvs.end()) {
-        SrsLiveEntry* entry = sflvs[sid];
+        SrsLiveEntry* s_entry = sflvs[sid];
-        if (!entry->stream->entry->enabled) {
+        if (!s_entry->stream->entry->enabled) {
             srs_error("stream is disabled, hijack failed. ret=%d", ret);
             return ret;
         }
@@ -1228,7 +1235,9 @@ int SrsHttpStreamServer::initialize_flv_streaming()
             continue;
         }
 
-        initialize_flv_entry(conf->arg0());
+        if ((ret = initialize_flv_entry(conf->arg0())) != ERROR_SUCCESS) {
+            return ret;
+        }
     }
     return ret;
 }

trunk/src/app/srs_app_server.cpp

@@ -1120,11 +1120,11 @@ int SrsServer::listen_stream_caster()
         SrsListener* listener = NULL;
 
         std::string caster = _srs_config->get_stream_caster_engine(stream_caster);
-        if (caster == SRS_CONF_DEFAULT_STREAM_CASTER_MPEGTS_OVER_UDP) {
+        if (srs_stream_caster_is_udp(caster)) {
             listener = new SrsUdpCasterListener(this, SrsListenerMpegTsOverUdp, stream_caster);
-        } else if (caster == SRS_CONF_DEFAULT_STREAM_CASTER_RTSP) {
+        } else if (srs_stream_caster_is_rtsp(caster)) {
             listener = new SrsRtspListener(this, SrsListenerRtsp, stream_caster);
-        } else if (caster == SRS_CONF_DEFAULT_STREAM_CASTER_FLV) {
+        } else if (srs_stream_caster_is_flv(caster)) {
             listener = new SrsHttpFlvListener(this, SrsListenerFlv, stream_caster);
         } else {
             ret = ERROR_STREAM_CASTER_ENGINE;

trunk/src/app/srs_app_source.cpp

@@ -55,8 +55,8 @@ using namespace std;
 // 115 packets is 3s.
 #define SRS_PURE_AUDIO_GUESS_COUNT 115
 
-// when got these videos or audios, mix ok.
+// when got these videos or audios, pure audio or video, mix ok.
-#define SRS_MIX_CORRECT_MIX_AV 10
+#define SRS_MIX_CORRECT_PURE_AV 10
 
 int _srs_time_jitter_string2int(std::string time_jitter)
 {
@@ -849,12 +849,25 @@ void SrsMixQueue::push(SrsSharedPtrMessage* msg)
 
 SrsSharedPtrMessage* SrsMixQueue::pop()
 {
-    // when got 10+ videos or audios, mix ok.
+    bool mix_ok = false;
-    // when got 1 video and 1 audio, mix ok.
+    
-    if (nb_videos < SRS_MIX_CORRECT_MIX_AV && nb_audios < SRS_MIX_CORRECT_MIX_AV) {
+    // pure video
-        if (nb_videos < 1 || nb_audios < 1) {
+    if (nb_videos >= SRS_MIX_CORRECT_PURE_AV && nb_audios == 0) {
-            return NULL;
+        mix_ok = true;
-        }
+    }
+    
+    // pure audio
+    if (nb_audios >= SRS_MIX_CORRECT_PURE_AV && nb_videos == 0) {
+        mix_ok = true;
+    }
+    
+    // got 1 video and 1 audio, mix ok.
+    if (nb_videos >= 1 && nb_audios >= 1) {
+        mix_ok = true;
+    }
+    
+    if (!mix_ok) {
+        return NULL;
     }
     
     // pop the first msg.