Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize

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 122 matching lines @@
   // Tells the encoder what the projected packet loss rate is. The rate is in
   // the range [0.0, 1.0]. The encoder would typically use this information to
   // adjust channel coding efforts, such as FEC. The default implementation
   // does nothing.
   virtual void SetProjectedPacketLossRate(double fraction);
 
   // Tells the encoder what average bitrate we'd like it to produce. The
   // encoder is free to adjust or disregard the given bitrate (the default
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
-
-  // Sets the maximum bitrate which must not be exceeded for any packet. The
-  // encoder is free to adjust or disregard this value (the default
-  // implementation does the latter).
-  virtual void SetMaxBitrate(int max_bps);
-
-  // Sets an upper limit on the size of packet payloads produced by the
-  // encoder. The encoder is free to adjust or disregard this value (the
-  // default implementation does the latter).
-  virtual void SetMaxPayloadSize(int max_payload_size_bytes);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_