Remove old methods in AudioTransport, make it pass a gn build

webrtc/voice_engine/voe_base_impl.h

 /*
  *  Copyright (c) 2012 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 60 matching lines @@
                                   const bool keyPressed,
                                   uint32_t& newMicLevel) override;
   int32_t NeedMorePlayData(const size_t nSamples,
                            const size_t nBytesPerSample,
                            const size_t nChannels,
                            const uint32_t samplesPerSec,
                            void* audioSamples,
                            size_t& nSamplesOut,
                            int64_t* elapsed_time_ms,
                            int64_t* ntp_time_ms) override;
-  int OnDataAvailable(const int voe_channels[],
-                      size_t number_of_voe_channels,
-                      const int16_t* audio_data,
-                      int sample_rate,
-                      size_t number_of_channels,
-                      size_t number_of_frames,
-                      int audio_delay_milliseconds,
-                      int current_volume,
-                      bool key_pressed,
-                      bool need_audio_processing) override;
-  void OnData(int voe_channel,
-              const void* audio_data,
-              int bits_per_sample,
-              int sample_rate,
-              size_t number_of_channels,
-              size_t number_of_frames) override;
   void PushCaptureData(int voe_channel,
                        const void* audio_data,
                        int bits_per_sample,
                        int sample_rate,
                        size_t number_of_channels,
                        size_t number_of_frames) override;
   void PullRenderData(int bits_per_sample,
                       int sample_rate,
                       size_t number_of_channels,
                       size_t number_of_frames,
@@ skipping 40 matching lines @@
   rtc::CriticalSection callbackCritSect_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 
   AudioFrame audioFrame_;
   voe::SharedData* shared_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H