Remove usage of VoEBase::AssociateSendChannel() from WVoMC.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 389 matching lines @@
   event_log_->LogAudioSendStreamConfig(config);
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
       bitrate_allocator_.get(), event_log_);
   {
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
   }
+  {
+    ReadLockScoped read_lock(*receive_crit_);
+    for (const auto& kv : audio_receive_ssrcs_) {
+      if (kv.second->config().rtp.local_ssrc == config.rtp.ssrc) {
+        kv.second->AssociateSendStream(send_stream);
+      }
+    }
+  }
   send_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
 
   webrtc::internal::AudioSendStream* audio_send_stream =
       static_cast<webrtc::internal::AudioSendStream*>(send_stream);
+  uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
   {
     WriteLockScoped write_lock(*send_crit_);
-    size_t num_deleted = audio_send_ssrcs_.erase(
-        audio_send_stream->config().rtp.ssrc);
-    RTC_DCHECK(num_deleted == 1);
+    size_t num_deleted = audio_send_ssrcs_.erase(ssrc);
+    RTC_DCHECK_EQ(1, num_deleted);
+  }
+  {
+    ReadLockScoped read_lock(*receive_crit_);
+    for (const auto& kv : audio_receive_ssrcs_) {
+      if (kv.second->config().rtp.local_ssrc == ssrc) {
+        kv.second->AssociateSendStream(nullptr);
+      }
+    }
   }
   UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   event_log_->LogAudioReceiveStreamConfig(config);
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       congestion_controller_.get(), config, config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                audio_receive_ssrcs_.end());
     audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     ConfigureSync(config.sync_group);
   }
+  {
+    ReadLockScoped read_lock(*send_crit_);
+    auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
+    if (it != audio_send_ssrcs_.end()) {
+      receive_stream->AssociateSendStream(it->second);
+    }
+  }
   receive_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return receive_stream;
 }
 
 void Call::DestroyAudioReceiveStream(
     webrtc::AudioReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
@@ skipping 606 matching lines @@
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   ReadLockScoped read_lock(*receive_crit_);
   auto it = video_receive_ssrcs_.find(ssrc);
   if (it == video_receive_ssrcs_.end())
     return false;
   return it->second->OnRecoveredPacket(packet, length);
 }
 
 }  // namespace internal
 }  // namespace webrtc