Variable audio bitrate.

webrtc/audio_send_stream.h

 /*
  *  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 70 matching lines @@
     // components.
     // TODO(solenberg): Remove when VoiceEngine channels are created outside
     // of Call.
     int voe_channel_id = -1;
 
     // Ownership of the encoder object is transferred to Call when the config is
     // passed to Call::CreateAudioSendStream().
     // TODO(solenberg): Implement, once we configure codecs through the new API.
     // std::unique_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
+
+    // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
+    // disable audio bitrate adaptation.
+    // Note: This is still an experimental feature and not ready for real usage.
+    int min_bitrate_kbps = -1;
+    int max_bitrate_kbps = -1;
   };
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
 
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_