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

webrtc/modules/audio_device/test/audio_device_test_api.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 36 matching lines @@
 using namespace webrtc;
 
 class AudioEventObserverAPI: public AudioDeviceObserver {
  public:
   AudioEventObserverAPI(
       const rtc::scoped_refptr<AudioDeviceModule>& audioDevice)
       : error_(kRecordingError),
         warning_(kRecordingWarning),
         audio_device_(audioDevice) {}
 
-  ~AudioEventObserverAPI() {}
+  ~AudioEventObserverAPI() override {}
 
-  virtual void OnErrorIsReported(const ErrorCode error) {
+  void OnErrorIsReported(const ErrorCode error) override {
     TEST_LOG("\n[*** ERROR ***] => OnErrorIsReported(%d)\n\n", error);
     error_ = error;
   }
 
-  virtual void OnWarningIsReported(const WarningCode warning) {
+  void OnWarningIsReported(const WarningCode warning) override {
     TEST_LOG("\n[*** WARNING ***] => OnWarningIsReported(%d)\n\n", warning);
     warning_ = warning;
     EXPECT_EQ(0, audio_device_->StopRecording());
     EXPECT_EQ(0, audio_device_->StopPlayout());
   }
 
  public:
   ErrorCode error_;
   WarningCode warning_;
  private:
   rtc::scoped_refptr<AudioDeviceModule> audio_device_;
 };
 
 class AudioTransportAPI: public AudioTransport {
  public:
   AudioTransportAPI(const rtc::scoped_refptr<AudioDeviceModule>& audioDevice)
       : rec_count_(0),
         play_count_(0) {
   }
 
-  ~AudioTransportAPI() {}
+  ~AudioTransportAPI() override {}
 
   int32_t RecordedDataIsAvailable(const void* audioSamples,
                                   const size_t nSamples,
                                   const size_t nBytesPerSample,
                                   const size_t nChannels,
                                   const uint32_t sampleRate,
                                   const uint32_t totalDelay,
                                   const int32_t clockSkew,
                                   const uint32_t currentMicLevel,
                                   const bool keyPressed,
@@ skipping 27 matching lines @@
       if (nChannels == 1) {
         TEST_LOG("+");
       } else {
         TEST_LOG("++");
       }
     }
     nSamplesOut = 480;
     return 0;
   }
 
+  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 {}
+
  private:
   uint32_t rec_count_;
   uint32_t play_count_;
 };
 
 class AudioDeviceAPITest: public testing::Test {
  protected:
   AudioDeviceAPITest() {}
 
-  virtual ~AudioDeviceAPITest() {}
+  ~AudioDeviceAPITest() override {}
 
   static void SetUpTestCase() {
     process_thread_ = ProcessThread::Create("ProcessThread");
     process_thread_->Start();
 
     // Windows:
     //      if (WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
     //          user can select between default (Core) or Wave
     //      else
     //          user can select between default (Wave) or Wave
@@ skipping 98 matching lines @@
     }
     if (audio_transport_) {
       delete audio_transport_;
       audio_transport_ = NULL;
     }
     if (audio_device_)
       EXPECT_EQ(0, audio_device_.release()->Release());
     PRINT_TEST_RESULTS;
   }
 
-  void SetUp() {
+  void SetUp() override {
     if (linux_alsa_) {
       FAIL() << "API Test is not available on ALSA on Linux!";
     }
     EXPECT_EQ(0, audio_device_->Init());
     EXPECT_TRUE(audio_device_->Initialized());
   }
 
-  void TearDown() {
-    EXPECT_EQ(0, audio_device_->Terminate());
-  }
+  void TearDown() override { EXPECT_EQ(0, audio_device_->Terminate()); }
 
   void CheckVolume(uint32_t expected, uint32_t actual) {
     // Mac and Windows have lower resolution on the volume settings.
 #if defined(WEBRTC_MAC) || defined(_WIN32)
     int diff = abs(static_cast<int>(expected - actual));
     EXPECT_LE(diff, 5);
 #else
     EXPECT_TRUE((actual == expected) || (actual == expected-1));
 #endif
   }
@@ skipping 1516 matching lines @@
   // TODO(kjellander): Fix so these tests pass on Mac.
 #if !defined(WEBRTC_MAC)
   EXPECT_EQ(0, audio_device_->InitPlayout());
   EXPECT_EQ(0, audio_device_->StartPlayout());
 #endif
 
   EXPECT_EQ(-1, audio_device_->GetLoudspeakerStatus(&loudspeakerOn));
 #endif
   EXPECT_EQ(0, audio_device_->StopPlayout());
 }