Add SSRC to RtpEncodingParameters for audio.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1199 matching lines @@
         rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
     RTC_DCHECK_GE(ch, 0);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
     config_.rtp.ssrc = ssrc;
     config_.rtp.c_name = c_name;
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
     config_.audio_network_adaptor_config = audio_network_adaptor_config;
+    rtp_parameters_.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
     RecreateAudioSendStream(send_codec_spec);
   }
 
   ~WebRtcAudioSendStream() override {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     ClearSource();
     call_->DestroyAudioSendStream(stream_);
   }
 
   void RecreateAudioSendStream(
@@ skipping 505 matching lines @@
     LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
                     << "with ssrc " << ssrc << " which doesn't exist.";
     return webrtc::RtpParameters();
   }
 
   // TODO(deadbeef): Return stream-specific parameters.
   webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
   for (const AudioCodec& codec : recv_codecs_) {
     rtp_params.codecs.push_back(codec.ToCodecParameters());
   }
+  rtp_params.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
   return rtp_params;
 }
 
 bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (!ValidateRtpParameters(parameters)) {
     return false;
   }
@@ skipping 896 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE