Add AudioEncoder::GetTargetBitrate

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 9025834dfa78704d16f00aed0b5bfe091eda86d7..35fbfad5f7d185b53e6a7e9b30380ad8e3e94bce 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -97,8 +97,9 @@ class AudioEncoder {
   virtual int Max10MsFramesInAPacket() const = 0;
 
   // Changes the target bitrate. The implementation is free to alter this value,
-  // e.g., if the desired value is outside the valid range.
-  virtual void SetTargetBitrate(int bits_per_second) {}
+  // e.g., if the desired value is outside the valid range. Returns the new
+  // target bitrate (which may or may not be equal to the input parameter).
+  virtual int SetTargetBitrate(int bits_per_second) = 0;
 
   // Tells the implementation what the projected packet loss rate is. The rate
   // is in the range [0.0, 1.0]. This rate is typically used to adjust channel