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

webrtc/media/base/audiosource.h

 /*
  *  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.
  */
 
-#ifndef WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
-#define WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
+#ifndef WEBRTC_MEDIA_BASE_AUDIOSOURCE_H_
+#define WEBRTC_MEDIA_BASE_AUDIOSOURCE_H_
 
 #include <cstddef>
 
 namespace cricket {
 
-// Abstract interface for rendering the audio data.
-class AudioRenderer {
+// Abstract interface for providing the audio data.
+// TODO(deadbeef): Rename this to AudioSourceInterface, and rename
+// webrtc::AudioSourceInterface to AudioTrackSourceInterface.
+class AudioSource {
  public:
   class Sink {
    public:
-    // Callback to receive data from the AudioRenderer.
+    // Callback to receive data from the AudioSource.
     virtual void OnData(const void* audio_data,
                         int bits_per_sample,
                         int sample_rate,
                         size_t number_of_channels,
                         size_t number_of_frames) = 0;
 
-    // Called when the AudioRenderer is going away.
+    // Called when the AudioSource is going away.
     virtual void OnClose() = 0;
 
    protected:
     virtual ~Sink() {}
   };
 
-  // Sets a sink to the AudioRenderer. There can be only one sink connected
-  // to the renderer at a time.
-  virtual void SetSink(Sink* sink) {}
+  // Sets a sink to the AudioSource. There can be only one sink connected
+  // to the source at a time.
+  virtual void SetSink(Sink* sink) = 0;
 
  protected:
-  virtual ~AudioRenderer() {}
+  virtual ~AudioSource() {}
 };
 
 }  // namespace cricket
 
-#endif  // WEBRTC_MEDIA_BASE_AUDIORENDERER_H_
+#endif  // WEBRTC_MEDIA_BASE_AUDIOSOURCE_H_