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

webrtc/modules/audio_device/test/func_test_manager.cc

 /*
  *  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 545 matching lines @@
         {
             TEST_LOG("++");
         }
     }  // if (_playCount % 100 == 0)
 
     nSamplesOut = nSamples;
 
     return 0;
 }
 
+void AudioTransportImpl::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) {}
+
+void AudioTransportImpl::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) {}
+
 FuncTestManager::FuncTestManager() :
     _audioDevice(NULL),
     _audioEventObserver(NULL),
     _audioTransport(NULL)
 {
   _playoutFile48 = webrtc::test::ResourcePath("audio_device\\audio_short48",
                                               "pcm");
   _playoutFile44 = webrtc::test::ResourcePath("audio_device\\audio_short44",
                                               "pcm");
   _playoutFile16 = webrtc::test::ResourcePath("audio_device\\audio_short16",
@@ skipping 2151 matching lines @@
 
     TEST_LOG("\n");
     PRINT_TEST_RESULTS;
 
     return 0;
 }
 
 }  // namespace webrtc
 
 // EOF