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);
+  RTC_DCHECK_EQ(0, error);

[the sun, 2016/03/07 21:24:53]

I don't think a DCHECK is appropriate here. There are many reasons why
VoEBase::StartSend() could fail - it enables the ADM for recording! So it should
be treated as a legitimate run-time condition, not a logic error. Thus, to
maintain the current logic, add a conditional log line here, just like the one
in pc/channel.cc which will never be triggered anyway, following this CL.

(same in Stop())

[Taylor Brandstetter, 2016/03/08 00:00:31]

Done.

 }
 
 void AudioSendStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  int error = base->StopSend(config_.voe_channel_id);
+  RTC_DCHECK_EQ(0, 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