Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.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 49 matching lines @@
   int SampleRateHz() const override;
   int NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
-  void SetMaxPayloadSize(int max_payload_size_bytes) override;
-  void SetMaxBitrate(int max_rate_bps) override;
 
  private:
   // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
   // STREAM_MAXW16_60MS for iSAC fix (60 ms).
   static const size_t kSufficientEncodeBufferSizeBytes = 400;
 
   static const int kDefaultBitRate = 32000;
 
   // Recreate the iSAC encoder instance with the given settings, and save them.
   void RecreateEncoderInstance(const Config& config);
@@ skipping 40 matching lines @@
  private:
   typename T::instance_type* isac_state_;
   LockedIsacBandwidthInfo* bwinfo_;
   int decoder_sample_rate_hz_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_