Remove dead code

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

 /*
  *  Copyright (c) 2015 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 163 matching lines @@
   }
 
   static rtc::ArrayView<const CodecInst> Database();
 
   static rtc::Optional<bool> IsSupportedNumChannels(CodecId codec_id,
                                                     int num_channels);
 
   static rtc::Optional<NetEqDecoder> NetEqDecoderFromCodecId(CodecId codec_id,
                                                              int num_channels);
 
+  // Parse codec_inst and extract payload types. If the given CodecInst was for
+  // the wrong sort of codec, return kSkip; otherwise, if the rate was illegal,
+  // return kBadFreq; otherwise, update the given RTP timestamp rate (Hz) ->
+  // payload type map and return kOk.
+  enum class RegistrationResult { kOk, kSkip, kBadFreq };
+  static RegistrationResult RegisterCngPayloadType(std::map<int, int>* pt_map,
+                                                   const CodecInst& codec_inst);
+  static RegistrationResult RegisterRedPayloadType(std::map<int, int>* pt_map,
+                                                   const CodecInst& codec_inst);
+
   RentACodec();
   ~RentACodec();
 
   // Creates and returns an audio encoder built to the given specification.
   // Returns null in case of error. The returned encoder is live until the next
   // successful call to this function, or until the Rent-A-Codec is destroyed.
   AudioEncoder* RentEncoder(const CodecInst& codec_inst);
 
   // Creates and returns an audio encoder stack where the given speech encoder
   // is augmented with the specified CNG/VAD and RED encoders. Leave either
@@ skipping 26 matching lines @@
   AudioEncoder* encoder_stack_ = nullptr;
   LockedIsacBandwidthInfo isac_bandwidth_info_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RentACodec);
 };
 
 }  // namespace acm2
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_