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

webrtc/modules/audio_device/test/func_test_manager.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 88 matching lines @@
 
     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;
 
+    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,
+                        void* audio_data,
+                        int64_t* elapsed_time_ms,
+                        int64_t* ntp_time_ms) override;
+
     AudioTransportImpl(AudioDeviceModule* audioDevice);
     ~AudioTransportImpl();
 
 public:
     int32_t SetFilePlayout(bool enable, const char* fileName = NULL);
     void SetFullDuplex(bool enable);
     void SetSpeakerVolume(bool enable)
     {
         _speakerVolume = enable;
     }
@@ skipping 90 matching lines @@
 
     std::unique_ptr<ProcessThread> _processThread;
     AudioDeviceModule* _audioDevice;
     AudioEventObserver* _audioEventObserver;
     AudioTransportImpl* _audioTransport;
 };
 
 }  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H