Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.

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 1406 matching lines @@
   void SetPlayout(bool playout) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     if (playout) {
       LOG(LS_INFO) << "Starting playout for channel #" << channel();
       stream_->Start();
     } else {
       LOG(LS_INFO) << "Stopping playout for channel #" << channel();
       stream_->Stop();
     }
+    playout_ = playout;
   }
 
  private:
   void RecreateAudioReceiveStream(
       uint32_t local_ssrc,
       bool use_transport_cc,
       bool use_nack,
       const std::vector<webrtc::RtpExtension>& extensions) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioReceiveStream(stream_);
       stream_ = nullptr;
     }
     config_.rtp.local_ssrc = local_ssrc;
     config_.rtp.transport_cc = use_transport_cc;
     config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
     config_.rtp.extensions = extensions;
     RTC_DCHECK(!stream_);
     stream_ = call_->CreateAudioReceiveStream(config_);
     RTC_CHECK(stream_);
+    SetPlayout(playout_);
   }
 
   rtc::ThreadChecker worker_thread_checker_;
   webrtc::Call* call_ = nullptr;
   webrtc::AudioReceiveStream::Config config_;
   // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
   // configuration changes.
   webrtc::AudioReceiveStream* stream_ = nullptr;
+  bool playout_ = false;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
 };
 
 WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                                                  const MediaConfig& config,
                                                  const AudioOptions& options,
                                                  webrtc::Call* call)
     : VoiceMediaChannel(config), engine_(engine), call_(call) {
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
@@ skipping 1196 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