Support RTMP ATC for HLS/HDS to support backup(failover). change to 0.9.35

@@ -112,13 +112,13 @@ Supported operating systems and hardware:
 26. Support RTMP(play-publish) library: srs-librtmp<br/>
 27. Support ARM([debian armhf, v7cpu](https://github.com/winlinvip/simple-rtmp-server/wiki/SrsLinuxArm)) with rtmp/ssl/hls/librtmp.<br/>
 28. Support [init.d](https://github.com/winlinvip/simple-rtmp-server/wiki/LinuxService) and packge script, log to file. <br/>
-29. [plan] Support system utest<br/>
-30. [plan] Support embeded http server for api and hls(live/vod)<br/>
-31. [plan] Support vod(file to hls stream)<br/>
-32. [plan] Support stream ingester using ffmpeg.<br/>
-33. [plan] Support ingest RTSP(RTP, SDP) stream to RTMP.<br/>
-34. [plan] Support network based cli and json result.<br/>
-35. [plan] Support HLS cluster, use RTMP ATC to generate the TS<br/>
+29. Support RTMP ATC for HLS/HDS to support backup(failover)<br/>
+30. [plan] Support system utest<br/>
+31. [plan] Support embeded http server for api and hls(live/vod)<br/>
+32. [plan] Support vod(file to hls stream)<br/>
+33. [plan] Support stream ingester using ffmpeg.<br/>
+34. [plan] Support ingest RTSP(RTP, SDP) stream to RTMP.<br/>
+35. [plan] Support network based cli and json result.<br/>
 36. [plan] Support RTMP edge server, push/pull stream from any RTMP server<br/>
 37. [plan] Support multiple processes, for both origin and edge<br/>
 38. [no-plan] Support adobe flash refer/token/swf verification.<br/>

@@ -41,6 +41,23 @@ vhost __defaultVhost__ {
     gop_cache       on;
 }
 
+# vhost for atc.
+vhost atc.srs.com {
+    # vhost for atc for hls/hds/rtmp backup.
+    # generally, atc default to off, server delivery rtmp stream to client(flash) timestamp from 0.
+    # when atc is on, server delivery rtmp stream by absolute time.
+    # atc is used, for instance, encoder will copy stream to master and slave server,
+    # server use atc to delivery stream to edge/client, where stream time from master/slave server
+    # is always the same, client/tools can slice RTMP stream to HLS according to the same time,
+    # if the time not the same, the HLS stream cannot slice to support system backup.
+    # 
+    # @see http://www.adobe.com/cn/devnet/adobe-media-server/articles/varnish-sample-for-failover.html
+    # @see http://www.baidu.com/#wd=hds%20hls%20atc
+    #
+    # default: off
+    atc             on;
+}
+
 # vhost for bandwidth check
 # generally, the bandcheck vhost must be: bandcheck.srs.com,
 # or need to modify the vhost of client.

@@ -1340,6 +1340,22 @@ bool SrsConfig::get_gop_cache(string vhost)
     return true;
 }
 
+bool SrsConfig::get_atc(string vhost)
+{
+    SrsConfDirective* conf = get_vhost(vhost);
+
+    if (!conf) {
+        return true;
+    }
+    
+    conf = conf->get("atc");
+    if (conf && conf->arg0() == "on") {
+        return true;
+    }
+    
+    return false;
+}
+
 double SrsConfig::get_queue_length(string vhost)
 {
     SrsConfDirective* conf = get_vhost(vhost);

@@ -153,6 +153,7 @@ public:
     virtual bool                get_deamon();
     virtual int                 get_max_connections();
     virtual bool                get_gop_cache(std::string vhost);
+    virtual bool                get_atc(std::string vhost);
     virtual double              get_queue_length(std::string vhost);
     virtual SrsConfDirective*   get_forward(std::string vhost);
 private:

@@ -268,11 +268,12 @@ int SrsConsumer::enqueue(SrsSharedPtrMessage* msg, int tba, int tbv)
 {
     int ret = ERROR_SUCCESS;
     
-    /* TODO: to support atc */
+    if (!source->is_atc()) {
         if ((ret = jitter->correct(msg, tba, tbv)) != ERROR_SUCCESS) {
             srs_freep(msg);
             return ret;
         }
+    }
     
     if ((ret = queue->enqueue(msg)) != ERROR_SUCCESS) {
         return ret;
@@ -391,6 +392,23 @@ int SrsGopCache::dump(SrsConsumer* consumer, int tba, int tbv)
     return ret;
 }
 
+bool SrsGopCache::empty()
+{
+    return gop_cache.empty();
+}
+
+int64_t SrsGopCache::get_start_time()
+{
+    if (empty()) {
+        return 0;
+    }
+    
+    SrsSharedPtrMessage* msg = gop_cache[0];
+    srs_assert(msg);
+    
+    return msg->header.timestamp;
+}
+
 std::map<std::string, SrsSource*> SrsSource::pool;
 
 SrsSource* SrsSource::find(SrsRequest* req)
@@ -425,6 +443,7 @@ SrsSource::SrsSource(SrsRequest* _req)
     gop_cache = new SrsGopCache();
     
     _srs_config->subscribe(this);
+    atc = _srs_config->get_atc(req->vhost);
 }
 
 SrsSource::~SrsSource()
@@ -774,6 +793,16 @@ int SrsSource::on_audio(SrsCommonMessage* audio)
     }
     srs_verbose("cache gop success.");
     
+    // if atc, update the sequence header to abs time.
+    if (atc) {
+        if (cache_sh_audio) {
+            cache_sh_audio->header.timestamp = msg->header.timestamp;
+        }
+        if (cache_metadata) {
+            cache_metadata->header.timestamp = msg->header.timestamp;
+        }
+    }
+    
     return ret;
 }
 
@@ -841,6 +870,16 @@ int SrsSource::on_video(SrsCommonMessage* video)
     }
     srs_verbose("cache gop success.");
     
+    // if atc, update the sequence header to abs time.
+    if (atc) {
+        if (cache_sh_video) {
+            cache_sh_video->header.timestamp = msg->header.timestamp;
+        }
+        if (cache_metadata) {
+            cache_metadata->header.timestamp = msg->header.timestamp;
+        }
+    }
+    
     return ret;
 }
 
@@ -921,6 +960,17 @@ void SrsSource::on_unpublish()
     }
     srs_info("dispatch metadata success");
     
+    // if atc, update the sequence header to gop cache time.
+    if (atc && !gop_cache->empty()) {
+        if (cache_sh_video) {
+            cache_sh_video->header.timestamp = gop_cache->get_start_time();
+        }
+        if (cache_sh_audio) {
+            cache_sh_audio->header.timestamp = gop_cache->get_start_time();
+        }
+    }
+    
+    // copy sequence header
     if (cache_sh_video && (ret = consumer->enqueue(cache_sh_video->copy(), sample_rate, frame_rate)) != ERROR_SUCCESS) {
         srs_error("dispatch video sequence header failed. ret=%d", ret);
         return ret;
@@ -933,6 +983,7 @@ void SrsSource::on_unpublish()
     }
     srs_info("dispatch audio sequence header success");
     
+    // copy gop cache to client.
     if ((ret = gop_cache->dump(consumer, sample_rate, frame_rate)) != ERROR_SUCCESS) {
         return ret;
     }
@@ -957,6 +1008,11 @@ void SrsSource::set_cache(bool enabled)
     gop_cache->set(enabled);
 }
 
+bool SrsSource::is_atc()
+{
+    return atc;
+}
+
 int SrsSource::create_forwarders()
 {
     int ret = ERROR_SUCCESS;

@@ -191,6 +191,12 @@ public:
     virtual int cache(SrsSharedPtrMessage* msg);
     virtual void clear();
     virtual int dump(SrsConsumer* consumer, int tba, int tbv);
+    /**
+    * used for atc to get the time of gop cache,
+    * the atc will adjust the sequence header timestamp to gop cache.
+    */
+    virtual bool empty();
+    virtual int64_t get_start_time();
 };
 
 /**
@@ -238,6 +244,13 @@ private:
     * can publish, true when is not streaming
     */
     bool _can_publish;
+    /**
+    * atc whether atc(use absolute time and donot adjust time),
+    * directly use msg time and donot adjust if atc is true,
+    * otherwise, adjust msg time to start from 0 to make flash happy.
+    */
+    // TODO: FIXME: to support reload atc.
+    bool atc;
 private:
     SrsSharedPtrMessage* cache_metadata;
     // the cached video sequence header.
@@ -279,6 +292,10 @@ public:
     virtual int create_consumer(SrsConsumer*& consumer);
     virtual void on_consumer_destroy(SrsConsumer* consumer);
     virtual void set_cache(bool enabled);
+// internal
+public:
+    // for consumer, atc feature.
+    virtual bool is_atc();
 private:
     virtual int create_forwarders();
     virtual void destroy_forwarders();

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR "0"
 #define VERSION_MINOR "9"
-#define VERSION_REVISION "34"
+#define VERSION_REVISION "35"
 #define RTMP_SIG_SRS_VERSION VERSION_MAJOR"."VERSION_MINOR"."VERSION_REVISION
 // server info.
 #define RTMP_SIG_SRS_KEY "srs"

@@ -1163,7 +1163,7 @@ int SrsComplexHandshake::handshake_with_client(SrsHandshakeBytes* hs_bytes, ISrs
     // never verify c2, for ffmpeg will failed.
     // it's ok for flash.
     
-    srs_trace("comple handshake with client success");
+    srs_trace("complex handshake with client success");
     
     return ret;
 }