Remove dead code

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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
 
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 template <typename T>
 class AudioEncoderIsacT final : public AudioEncoder {
@@ skipping 21 matching lines @@
 
     // In adaptive mode, prevent adaptive changes to the frame size. (Not used
     // in nonadaptive mode.)
     bool enforce_frame_size = false;
   };
 
   explicit AudioEncoderIsacT(const Config& config);
   explicit AudioEncoderIsacT(
       const CodecInst& codec_inst,
       const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
-  AudioEncoderIsacT(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderIsacT() override;
 
-  static constexpr const char* GetPayloadName() { return "ISAC"; }
-  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;
 
@@ skipping 19 matching lines @@
 
   // Timestamp of the previously encoded packet.
   uint32_t last_encoded_timestamp_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_