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

webrtc/modules/audio_device/android/audio_device_unittest.cc

 /*
  *  Copyright (c) 2015 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 372 matching lines @@
                 int32_t(const void* audioSamples,
                         const size_t nSamples,
                         const size_t nBytesPerSample,
                         const size_t nChannels,
                         const uint32_t samplesPerSec,
                         const uint32_t totalDelayMS,
                         const int32_t clockDrift,
                         const uint32_t currentMicLevel,
                         const bool keyPressed,
                         uint32_t& newMicLevel));
+
   MOCK_METHOD8(NeedMorePlayData,
                int32_t(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));
 
+  MOCK_METHOD6(PushCaptureData,
+               void(int voe_channel,
+                    const void* audio_data,
+                    int bits_per_sample,
+                    int sample_rate,
+                    size_t number_of_channels,
+                    size_t number_of_frames));
+
+  MOCK_METHOD7(PullRenderData,
+               void(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));
+
   // Set default actions of the mock object. We are delegating to fake
   // implementations (of AudioStreamInterface) here.
   void HandleCallbacks(EventWrapper* test_is_done,
                        AudioStreamInterface* audio_stream,
                        int num_callbacks) {
     test_is_done_ = test_is_done;
     audio_stream_ = audio_stream;
     num_callbacks_ = num_callbacks;
     if (play_mode()) {
       ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
@@ skipping 596 matching lines @@
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc