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);
-
+  FakeAudioDevice(Clock* clock, float speed);

[the sun, 2017/01/24 08:58:12]

What's "speed"?

[perkj_webrtc, 2017/01/24 09:57:03]

Added a comment. Note that this is nothing I want to change.

   virtual ~FakeAudioDevice();

[the sun, 2017/01/24 08:58:12]

override

[perkj_webrtc, 2017/01/24 09:57:03]

Done.

 
+  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_;
+  rtc::CriticalSection lock_;
+  AudioTransport* audio_callback_ GUARDED_BY(lock_);
+  bool recording_ GUARDED_BY(lock_);
+  bool playing_ GUARDED_BY(lock_);
+
+  // Used for recording.
+  double normalized_frequency_ GUARDED_BY(lock_);

[the sun, 2017/01/24 08:58:12]

float

[perkj_webrtc, 2017/01/24 09:57:03]

Done.

+  uint16_t peak_to_peak_ GUARDED_BY(lock_);

[the sun, 2017/01/24 08:58:12]

float

[perkj_webrtc, 2017/01/24 09:57:03]

Done.

+  double n_;

[the sun, 2017/01/24 08:58:12]

float

[perkj_webrtc, 2017/01/24 09:57:03]

Done.

   int8_t captured_audio_[kBufferSizeBytes];
+
+  // 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_;

[the sun, 2017/01/24 08:58:12]

Are there any build file deps you can remove now?

[perkj_webrtc, 2017/01/24 09:57:03]

Done.

-  std::unique_ptr<FileWrapper> input_stream_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_