Add AudioEncoder::GetTargetBitrate

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  Copyright (c) 2014 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 79 matching lines @@
   // the encoder may vary the number of 10 ms frames from packet to packet, but
   // it must decide the length of the next packet no later than when outputting
   // the preceding packet.
   virtual int Num10MsFramesInNextPacket() const = 0;
 
   // Returns the maximum value that can be returned by
   // Num10MsFramesInNextPacket().
   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
   // coding efforts, such as FEC.
   virtual void SetProjectedPacketLossRate(double fraction) {}
 
   // This is the encode function that the inherited classes must implement. It
   // is called from Encode in the base class.
   virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                                      const int16_t* audio,
@@ skipping 26 matching lines @@
   // Sets the maximum rate which the codec may not exceed for any packet.
   virtual void SetMaxRate(int max_rate_bps) = 0;
 
   // Informs the encoder about the maximum sample rate which the decoder will
   // use when decoding the bitstream. The implementation is free to disregard
   // this hint.
   virtual bool SetMaxPlaybackRate(int frequency_hz) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_