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/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);

[peah-webrtc, 2017/01/25 10:45:28]

It seems to me very strange that the sample rate is not part of the constructor
call. Afaics, the rate is not hardcoded to be 16 kHz, which means that it will
produce 16 kHz of data. Is that desired and sufficient for the usage of this?

For instance, the audio processing differs quite a lot if you use 16 or 48 kHz.

+  // 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);
+  ~FakeAudioDevice() override;
 
-  virtual ~FakeAudioDevice();
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
 
+  // Generates a sine tone with |frequency_in_hz| and |peak_to_peak|.
+  void StartRecordingSine(int frequency_in_hz, uint16_t peak_to_peak);
+  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();
-
- private:
   static bool Run(void* obj);
-  void CaptureAudio();
+  void ProcessAudio();
 
   static const uint32_t kFrequencyHz = 16000;
   static const size_t kBufferSizeBytes = 2 * kFrequencyHz;
 
-  AudioTransport* audio_callback_;
-  bool capturing_;
-  int8_t captured_audio_[kBufferSizeBytes];
+  rtc::CriticalSection lock_;
+  AudioTransport* audio_callback_ GUARDED_BY(lock_);
+  bool rendering_ GUARDED_BY(lock_);
+
+  class SinusCapturer;
+  std::unique_ptr<SinusCapturer> capturer_ GUARDED_BY(lock_);
+
+  // Used for playout.
   int8_t playout_buffer_[kBufferSizeBytes];
   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_