Remove dead code

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 #include "webrtc/rtc_base/constructormagic.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 class AudioEncoderIlbcImpl final : public AudioEncoder {
  public:
-  static rtc::Optional<AudioEncoderIlbcConfig> SdpToConfig(
-      const SdpAudioFormat& format);
-
   AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
   explicit AudioEncoderIlbcImpl(const CodecInst& codec_inst);
-  AudioEncoderIlbcImpl(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderIlbcImpl() override;
 
-  static constexpr const char* GetPayloadName() { return "ILBC"; }
-  static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
-      const SdpAudioFormat& format);
-
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
   void Reset() override;
 
  private:
   size_t RequiredOutputSizeBytes() const;
 
   static constexpr size_t kMaxSamplesPerPacket = 480;
   const int frame_size_ms_;
   const int payload_type_;
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   int16_t input_buffer_[kMaxSamplesPerPacket];
   IlbcEncoderInstance* encoder_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbcImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_