webrtc m76 simulcast代码片段

simucast同一个视频使用不同码率，帧率，宽高发送多个视频流，接收方可以根据选择接受那一个视频

sdp表现：  
//simucast  直播多个流，可以使用不同码率，宽度高度帧率等参数，webrtc内部有一固定算法进行转化
a=ssrc-group:SIM 2267484833 2714558033 2714558034

如何设置：

planb模式下：  
   RTCOfferAnswerOptions.num_simulcast_layer =3;   
   PeerConnection::CreateOffer();

unified plan模式下:
   使用rid可以代替他的.
   
   PeerConnection.GetSenders();
   RtpParameters parma=RtpSender.GetParameters();
   //设置param中编码参数的rid
   ...
   RtpSender.SetParameters();
   
   struct RtpParameters{
       //多组编码参数 ，用于支持svc编码  
       std::vector encodings;
   }
   
   struct RtpEncodingParameters{
      // Value to use for RID RTP header extension.
      // Called "encodingId" in ORTC.
      std::string rid;
   }

webrtc m76中创建simulcast相关：

	media_session.cc
	
	static bool AddStreamParams(){
	...
	  //这是对应的流的信息 
	  StreamParams stream_param =
	          sender.rids.empty()
	              ?
	              // Signal SSRCs and legacy simulcast (if requested).
	              //老版本planb. rids为空，使用num_simulcast_layer来创建，内部调用GenerateSsrcs		              
	              CreateStreamParamsForNewSenderWithSsrcs(
	                  sender, rtcp_cname, include_rtx_streams,
	                  include_flexfec_stream, ssrc_generator)
	              :
	              // Signal RIDs and spec-compliant simulcast (if requested).
	              CreateStreamParamsForNewSenderWithRids(sender, rtcp_cname);
	
	...
	}

planb 使用的函数：

	//num_layers>1表示simulcast流
	void StreamParams::GenerateSsrcs(int num_layers,
	                                 bool generate_fid,
	                                 bool generate_fec_fr,
	                                 rtc::UniqueRandomIdGenerator* ssrc_generator) {
	  RTC_DCHECK_GE(num_layers, 0);
	  RTC_DCHECK(ssrc_generator);
	  std::vector primary_ssrcs;
	  for (int i = 0; i < num_layers; ++i) {
	    uint32_t ssrc = ssrc_generator->GenerateId();
	    primary_ssrcs.push_back(ssrc);
	    add_ssrc(ssrc);
	  }
	
	  if (num_layers > 1) {
	    SsrcGroup simulcast(kSimSsrcGroupSemantics, primary_ssrcs);
	    ssrc_groups.push_back(simulcast);
	  }
	
	  if (generate_fid) {
	    for (uint32_t ssrc : primary_ssrcs) {
	      AddFidSsrc(ssrc, ssrc_generator->GenerateId());
	    }
	  }
	
	  if (generate_fec_fr) {
	    for (uint32_t ssrc : primary_ssrcs) {
	      AddFecFrSsrc(ssrc, ssrc_generator->GenerateId());
	    }
	  }
	}




...
验证ssrc和rid一系列参数

unimlucast plan 模式下:

	 BaseChannel::UpdateLocalStreams_w(){ 
	 ssrc应该是空的，用rid数量填充simulcast
	    // At this point we use the legacy simulcast group in StreamParams to
	    // indicate that we want multiple layers to the media channel.
	    if (!new_stream.has_ssrcs()) {
	      // TODO(bugs.webrtc.org/10250): Indicate if flex is desired here.
	      new_stream.GenerateSsrcs(new_stream.rids().size(), /* rtx = */ true,
	                               /* flex_fec = */ false, ssrc_generator_);
	    }
	
	    if (media_channel()->AddSendStream(new_stream)){}
	
	
	}

创建编码器配置，进行校验：

	WebRtcVideoChannel ::WebRtcVideoSendStream::CreateVideoEncoderConfig(){
	// By default, the stream count for the codec configuration should match the
	  // number of negotiated ssrcs. But if the codec is blacklisted for simulcast
	  // or a screencast (and not in simulcast screenshare experiment), only
	  // configure a single stream.
	  encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();
	  if (IsCodecBlacklistedForSimulcast(codec.name) ||
	      (is_screencast && (!ScreenshareSimulcastFieldTrialEnabled() ||
	                         !parameters_.conference_mode))) {
     //如果编码器器在黑名单中，则强制设置单个流，不进行广播

	    encoder_config.number_of_streams = 1;
	
	}