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

webrtc/api/rtpsender.h

 /*
  *  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.
  */
 
 // This file contains classes that implement RtpSenderInterface.
 // An RtpSender associates a MediaStreamTrackInterface with an underlying
 // transport (provided by AudioProviderInterface/VideoProviderInterface)
 
 #ifndef WEBRTC_API_RTPSENDER_H_
 #define WEBRTC_API_RTPSENDER_H_
 
 #include <string>
 
 #include "webrtc/api/mediastreamprovider.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/statscollector.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/media/base/audiorenderer.h"
+#include "webrtc/media/base/audiosource.h"
 
 namespace webrtc {
 
 // LocalAudioSinkAdapter receives data callback as a sink to the local
-// AudioTrack, and passes the data to the sink of AudioRenderer.
+// AudioTrack, and passes the data to the sink of AudioSource.
 class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
-                              public cricket::AudioRenderer {
+                              public cricket::AudioSource {
  public:
   LocalAudioSinkAdapter();
   virtual ~LocalAudioSinkAdapter();
 
  private:
   // AudioSinkInterface implementation.
   void OnData(const void* audio_data,
               int bits_per_sample,
               int sample_rate,
               size_t number_of_channels,
               size_t number_of_frames) override;
 
-  // cricket::AudioRenderer implementation.
-  void SetSink(cricket::AudioRenderer::Sink* sink) override;
+  // cricket::AudioSource implementation.
+  void SetSink(cricket::AudioSource::Sink* sink) override;
 
-  cricket::AudioRenderer::Sink* sink_;
+  cricket::AudioSource::Sink* sink_;
   // Critical section protecting |sink_|.
   rtc::CriticalSection lock_;
 };
 
 class AudioRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
   // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
   // at the appropriate times.
   AudioRtpSender(AudioTrackInterface* track,
@@ skipping 46 matching lines @@
   std::string id_;
   std::string stream_id_;
   AudioProviderInterface* provider_;
   StatsCollector* stats_;
   rtc::scoped_refptr<AudioTrackInterface> track_;
   uint32_t ssrc_ = 0;
   bool cached_track_enabled_ = false;
   bool stopped_ = false;
 
   // Used to pass the data callback from the |track_| to the other end of
-  // cricket::AudioRenderer.
+  // cricket::AudioSource.
   rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
 };
 
 class VideoRtpSender : public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInterface> {
  public:
   VideoRtpSender(VideoTrackInterface* track,
                  const std::string& stream_id,
                  VideoProviderInterface* provider);
 
@@ skipping 42 matching lines @@
   VideoProviderInterface* provider_;
   rtc::scoped_refptr<VideoTrackInterface> track_;
   uint32_t ssrc_ = 0;
   bool cached_track_enabled_ = false;
   bool stopped_ = false;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_RTPSENDER_H_