Added A/V sync tests with drifting clocks.

webrtc/test/fake_audio_device.h

 /*
  *  Copyright (c) 2013 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.
  */
 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
 #define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
 
 #include <string>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/test/drifting_clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class Clock;
 class EventTimerWrapper;
 class FileWrapper;
 class ModuleFileUtility;
 
 namespace test {
 
 class FakeAudioDevice : public FakeAudioDeviceModule {
  public:
-  FakeAudioDevice(Clock* clock, const std::string& filename);
+  FakeAudioDevice(Clock* clock, const std::string& filename, float speed);
 
   virtual ~FakeAudioDevice();
 
   int32_t Init() override;
   int32_t RegisterAudioCallback(AudioTransport* callback) override;
 
   bool Playing() const override;
   int32_t PlayoutDelay(uint16_t* delay_ms) const override;
   bool Recording() const override;
 
   void Start();
   void Stop();
 
  private:
   static bool Run(void* obj);
   void CaptureAudio();
 
   static const uint32_t kFrequencyHz = 16000;
   static const size_t kBufferSizeBytes = 2 * kFrequencyHz;
 
   AudioTransport* audio_callback_;
   bool capturing_;
   int8_t captured_audio_[kBufferSizeBytes];
   int8_t playout_buffer_[kBufferSizeBytes];
+  const float speed_;
   int64_t last_playout_ms_;
 
-  Clock* clock_;
+  DriftingClock clock_;
   rtc::scoped_ptr<EventTimerWrapper> tick_;
   rtc::CriticalSection lock_;
   rtc::PlatformThread thread_;
   rtc::scoped_ptr<ModuleFileUtility> file_utility_;
   rtc::scoped_ptr<FileWrapper> input_stream_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_