Remove dead code

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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_G722_AUDIO_ENCODER_G722_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
 #include <memory>
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/rtc_base/constructormagic.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 class AudioEncoderG722Impl final : public AudioEncoder {
  public:
-  static rtc::Optional<AudioEncoderG722Config> SdpToConfig(
-      const SdpAudioFormat& format);
-
   AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
   explicit AudioEncoderG722Impl(const CodecInst& codec_inst);
-  AudioEncoderG722Impl(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderG722Impl() override;
 
-  static constexpr const char* GetPayloadName() { return "G722"; }
-  static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
-      const SdpAudioFormat& format);
-
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   void Reset() override;
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
@@ skipping 17 matching lines @@
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   const std::unique_ptr<EncoderState[]> encoders_;
   rtc::Buffer interleave_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722Impl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_