Removed MaxEncodedBytes from AudioEncoder.

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 117 matching lines @@
   // encoder is free to adjust or disregard the given bitrate (the default
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
-
- private:
-  // This function is deprecated. It was used to return the maximum number of
-  // bytes that can be produced by the encoder at each Encode() call.  Since the
-  // Encode interface was changed to use rtc::Buffer, this is no longer
-  // applicable. It is only kept in to avoid breaking subclasses that still have
-  // it implemented (with the override attribute). It will be removed as soon
-  // as these subclasses have been given a chance to change.
-  virtual size_t MaxEncodedBytes() const;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_