Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

 /*
  *  Copyright (c) 2012 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 167 matching lines @@
   // Get 10 milliseconds of raw audio data to play out, and
   // automatic resample to the requested frequency if > 0.
   int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
 
   /////////////////////////////////////////
   //   Statistics
   //
 
   int GetNetworkStatistics(NetworkStatistics* statistics) override;
 
-  int SetISACMaxRate(int max_bit_per_sec) override;
-
-  int SetISACMaxPayloadSize(int max_size_bytes) override;
-
   int SetOpusApplication(OpusApplicationMode application) override;
 
   // If current send codec is Opus, informs it about the maximum playback rate
   // the receiver will render.
   int SetOpusMaxPlaybackRate(int frequency_hz) override;
 
   int EnableOpusDtx() override;
 
   int DisableOpusDtx() override;
 
@@ skipping 169 matching lines @@
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_