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

webrtc/api/rtpsender.cc

 /*
  *  Copyright 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 18 matching lines @@
                                    int sample_rate,
                                    size_t number_of_channels,
                                    size_t number_of_frames) {
   rtc::CritScope lock(&lock_);
   if (sink_) {
     sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
                   number_of_frames);
   }
 }
 
-void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
+void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
   rtc::CritScope lock(&lock_);
   ASSERT(!sink || !sink_);
   sink_ = sink;
 }
 
 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
                                const std::string& stream_id,
                                AudioProviderInterface* provider,
                                StatsCollector* stats)
     : id_(track->id()),
@@ skipping 137 matching lines @@
   // PeerConnection.  This is a bit of a strange way to apply local audio
   // options since it is also applied to all streams/channels, local or remote.
   if (track_->enabled() && track_->GetSource() &&
       !track_->GetSource()->remote()) {
     // TODO(xians): Remove this static_cast since we should be able to connect
     // a remote audio track to a peer connection.
     options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
   }
 #endif
 
-  cricket::AudioRenderer* renderer = sink_adapter_.get();
-  ASSERT(renderer != nullptr);
-  provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
+  cricket::AudioSource* source = sink_adapter_.get();
+  ASSERT(source != nullptr);
+  provider_->SetAudioSend(ssrc_, track_->enabled(), options, source);
 }
 
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
                                const std::string& stream_id,
                                VideoProviderInterface* provider)
     : id_(track->id()),
       stream_id_(stream_id),
       provider_(provider),
       track_(track),
       cached_track_enabled_(track->enabled()) {
@@ skipping 112 matching lines @@
   RTC_DCHECK(!stopped_ && can_send_track());
   const cricket::VideoOptions* options = nullptr;
   VideoTrackSourceInterface* source = track_->GetSource();
   if (track_->enabled() && source) {
     options = source->options();
   }
   provider_->SetVideoSend(ssrc_, track_->enabled(), options);
 }
 
 }  // namespace webrtc