Merge branch 'develop' into feature/rtc

trunk/src/app/srs_app_gb28181.cpp

@@ -412,7 +412,7 @@ srs_error_t SrsGb28181PsRtpProcessor::rtmpmuxer_enqueue_data(SrsGb28181RtmpMuxer
                 muxer->get_channel_id().c_str(), ssrc, muxer->channel_peer_port(), peer_port);
         }else {
             //muxer->ps_packet_enqueue(pkt);
-             muxer->insert_jitterbuffer(pkt);
+            muxer->insert_jitterbuffer(pkt);
         }//end if (muxer->channel_peer_port() != peer_port)
     }//end  if (muxer)
 
@@ -593,19 +593,6 @@ int64_t  SrsPsStreamDemixer::parse_ps_timestamp(const uint8_t* p)
 	return val;
 }
 
-bool SrsPsStreamDemixer::is_aac(){
-    // SrsBuffer  *avs = new SrsBuffer(stream->bytes(), stream->length());
-    // SrsAutoFree(SrsBuffer, avs);
-    // if (!avs->empty()) {
-    //     char* frame = NULL;
-    //     int frame_size = 0;
-    //     SrsRawAacStreamCodec codec;
-    //     if ((err = aac->adts_demux(avs, &frame, &frame_size, codec)) != srs_success) {
-    //         return srs_error_wrap(err, "demux adts");
-    //     }
-    return true;
-}
-
 srs_error_t SrsPsStreamDemixer::on_ps_stream(char* ps_data, int ps_size, uint32_t timestamp, uint32_t ssrc)
 {
     srs_error_t err = srs_success;
@@ -976,8 +963,13 @@ SrsGb28181RtmpMuxer::SrsGb28181RtmpMuxer(SrsGb28181Manger* c, std::string id, bo
     source_publish = true;
 
     jitter_buffer = new SrsPsJitterBuffer(id);
+    jitter_buffer_audio = new SrsPsJitterBuffer(id);
+
     ps_buflen = 0;
     ps_buffer = NULL;
+
+    ps_buflen_auido = 0;
+    ps_buffer_audio = NULL;
 }
 
 SrsGb28181RtmpMuxer::~SrsGb28181RtmpMuxer()
@@ -988,7 +980,10 @@ SrsGb28181RtmpMuxer::~SrsGb28181RtmpMuxer()
     srs_cond_destroy(wait_ps_queue);
  
     srs_freep(jitter_buffer);
+    srs_freep(jitter_buffer_audio);
     srs_freepa(ps_buffer);
+    srs_freepa(ps_buffer_audio);
+
     srs_freep(channel);
     srs_freep(ps_demixer);
     srs_freep(trd);
@@ -1094,6 +1089,14 @@ srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
     jitter_buffer->SetNackMode(kNack, -1, -1);
     jitter_buffer->SetNackSettings(250, 450, 0);
 
+    if (!jitter_buffer_audio) {
+        jitter_buffer_audio = new SrsPsJitterBuffer(channel_id);
+    }
+
+    jitter_buffer_audio->SetDecodeErrorMode(kSelectiveErrors);
+    jitter_buffer_audio->SetNackMode(kNack, -1, -1);
+    jitter_buffer_audio->SetNackSettings(250, 450, 0);
+
     if (!source_publish) return err;
 
     req = r;
@@ -1146,6 +1149,17 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
                     };
                 }
             }
+
+            if(jitter_buffer_audio->FoundFrame(cur_timestamp)){
+                jitter_buffer_audio->GetPsFrame(&ps_buffer_audio, ps_buflen_auido, buffer_size, cur_timestamp);
+            
+                if (buffer_size > 0){
+                    if ((err = ps_demixer->on_ps_stream(ps_buffer_audio, buffer_size, cur_timestamp, 0)) != srs_success){
+                        srs_warn("gb28181: demix ps stream error:%s",  srs_error_desc(err).c_str());
+                        srs_freep(err);
+                    };
+                }
+            }
         }else {
             //demix ps to h264/aac, to rtmp
             while(!ps_queue.empty()){
@@ -1224,8 +1238,39 @@ void SrsGb28181RtmpMuxer::stop()
 
 void SrsGb28181RtmpMuxer::insert_jitterbuffer(SrsPsRtpPacket *pkt)
 {
+    if (!pkt){
+        return;
+    }
+        
     recv_rtp_stream_time = srs_get_system_time();
-    jitter_buffer->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
+
+    char *payload = pkt->payload->bytes();
+
+    uint8_t p1 = (uint8_t)(payload[0]);
+    uint8_t p2 = (uint8_t)(payload[1]);
+    uint8_t p3 = (uint8_t)(payload[2]);
+    uint8_t p4 = (uint8_t)(payload[3]);
+
+   
+    //check for rtp ps audio streaming
+    bool av_same_ts = true;
+
+    if (p1 == 0x00 && p2 == 0x00 && p3 == 0x01 && p4 == 0xC0 &&
+        ps_rtp_video_ts != pkt->timestamp) {
+        av_same_ts = false;
+    }
+
+    //if audio and video are the same clock, 
+    //if both audio and video use jitter_buffer, 
+    //otherwise audio uses jitter_buffer_audio, and video uses jitter_buffer
+    if (av_same_ts){
+        pkt->marker = false;
+        jitter_buffer->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
+        ps_rtp_video_ts = pkt->timestamp;
+    }else {
+        jitter_buffer_audio->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
+    }
+ 
     //srs_cond_signal(wait_ps_queue);
 }
 

trunk/src/app/srs_app_gb28181.hpp

@@ -260,7 +260,6 @@ private:
 public:
     int64_t parse_ps_timestamp(const uint8_t* p);
     std::string get_ps_map_type_str(uint8_t);
-    bool is_aac();
     virtual srs_error_t on_ps_stream(char* ps_data, int ps_size, uint32_t timestamp, uint32_t ssrc);
 };
 
@@ -303,8 +302,16 @@ private:
     SrsServer* server;
 
     SrsPsJitterBuffer *jitter_buffer;
+    SrsPsJitterBuffer *jitter_buffer_audio;
+
     char *ps_buffer;
+    char *ps_buffer_audio;
+
     int ps_buflen;
+    int ps_buflen_auido;
+
+    uint32_t ps_rtp_video_ts;
+    uint32_t ps_rtp_audio_ts;
 
     bool source_publish; 
 

trunk/src/app/srs_app_gb28181_jitbuffer.cpp

@@ -306,11 +306,11 @@ PsFrameBufferEnum SrsPsFrameBuffer::InsertPacket(const VCMPacket& packet, const
     }
     
     //TODO: not check marker, check a complete frame with timestamp
-    // if (packet.markerBit &&
-    //     (last_packet_seq_num_ == -1 ||
-    //     IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_))) {
-    //     last_packet_seq_num_ = packet.seqNum;
-    // }
+    if (packet.markerBit &&
+        (last_packet_seq_num_ == -1 ||
+        IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_))) {
+        last_packet_seq_num_ = packet.seqNum;
+    }
 
     // The insert operation invalidates the iterator |rit|.
     PacketIterator packet_list_it = packets_.insert(rit.base(), packet);

trunk/src/app/srs_app_gb28181_sip.cpp

@@ -163,7 +163,8 @@ srs_error_t SrsGb28181SipSession::do_cycle()
         if (_register_status == SrsGb28181SipSessionRegisterOk &&
             _alive_status == SrsGb28181SipSessionAliveOk)
         {
-               
+            std::list<std::string> auto_play_list;
+           
             std::map<std::string, SrsGb28181Device*>::iterator it;
             for (it = _device_list.begin(); it != _device_list.end(); it++) {
                 SrsGb28181Device *device = it->second;
@@ -191,8 +192,15 @@ srs_error_t SrsGb28181SipSession::do_cycle()
                 //offline or already invite device does not need to send invite
                 if (device->device_status != "ON" || 
                     device->invite_status != SrsGb28181SipSessionUnkonw) continue;
+                
+                auto_play_list.push_back(chid);
+            }//end for (it) 
 
-
+            //auto send sip invite and create stream chennal
+            while(auto_play_list.size() > 0){
+                std::string chid = auto_play_list.front();
+                auto_play_list.pop_front();
+        
                 SrsGb28181StreamChannel ch;
             
                 ch.set_channel_id(_session_id + "@" + chid);
@@ -226,7 +234,7 @@ srs_error_t SrsGb28181SipSession::do_cycle()
             
                 srs_trace("gb28181: %s clients device=%s send invite code=%d", 
                     _session_id.c_str(), chid.c_str(), code);
-            }//end for (it)
+            }//end while (auto_play_list.size())
         }//end if (config)
 
         if (_register_status == SrsGb28181SipSessionRegisterOk &&
@@ -911,7 +919,7 @@ srs_error_t SrsGb28181SipService::fetch_or_create_sip_session(SrsSipRequest *req
     if ((sess = fetch(req->sip_auth_id)) != NULL) {
         *sip_session = sess;
         return err;
-    }
+    }
     
     sess = new SrsGb28181SipSession(this, req);;
     if ((err = sess->serve()) != srs_success) {