Refactor FakeAudioDevice to have separate methods for starting recording and playout.

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 <memory>
 #include <string>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.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, float speed);
+  // Creates a new FakeAudioDevice. |speed| controls how much faster or slower
+  // time elapse compared to the system clock. It can be used to simulate
+  // clock drift. 1.0 means that the system clock will be used.
+  FakeAudioDevice(Clock* clock, float speed, int sampling_frequency_in_hz);
+  ~FakeAudioDevice() override;
 
-  virtual ~FakeAudioDevice();
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
 
+  // Generates a signal where every second frame is zero and every second frame
+  // is evenly distributed random noise with max amplitude |max_amplitude|.
+  void StartRecordingPulsedNoise(int16_t max_amplitude);
+  int32_t StopRecording() override;
+
+ private:
   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();
+  static bool Run(void* obj);
+  void ProcessAudio();
 
- private:
-  static bool Run(void* obj);
-  void CaptureAudio();
+  const int sampling_frequency_in_hz_;
 
-  static const uint32_t kFrequencyHz = 16000;
-  static const size_t kBufferSizeBytes = 2 * kFrequencyHz;
+  rtc::CriticalSection lock_;
+  AudioTransport* audio_callback_ GUARDED_BY(lock_);
+  bool rendering_ GUARDED_BY(lock_);
 
-  AudioTransport* audio_callback_;
-  bool capturing_;
-  int8_t captured_audio_[kBufferSizeBytes];
-  int8_t playout_buffer_[kBufferSizeBytes];
+  class PulsedNoiseCapturer;
+  std::unique_ptr<PulsedNoiseCapturer> capturer_ GUARDED_BY(lock_);
+
+  // Used for playout.
+  rtc::BufferT<int16_t> playout_buffer_;

[peah-webrtc, 2017/01/26 09:24:00]

Since the size of playout_buffer_ is static, I think it is better to use
std::vector instead. That allows you to set the length in the initializer list
as well.

   const float speed_;
   int64_t last_playout_ms_;
 
   DriftingClock clock_;
   std::unique_ptr<EventTimerWrapper> tick_;
-  rtc::CriticalSection lock_;
   rtc::PlatformThread thread_;
-  std::unique_ptr<ModuleFileUtility> file_utility_;
-  std::unique_ptr<FileWrapper> input_stream_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_