Prevent a voice channel from sending data before a renderer is set.

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 79 matching lines @@
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->ResetCongestionControlObjects();
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  int error = base->StartSend(config_.voe_channel_id);
+  if (error != 0) {
+    LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error;
+  }
 }
 
 void AudioSendStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  int error = base->StopSend(config_.voe_channel_id);
+  if (error != 0) {
+    LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error;
+  }
 }
 
 void AudioSendStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
@@ skipping 98 matching lines @@
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc