Add the ability to read/write to WAV files in FakeAudioDevice

webrtc/test/file_audio_device.h

Index: webrtc/test/file_audio_device.h
diff --git a/webrtc/test/file_audio_device.h b/webrtc/test/file_audio_device.h
new file mode 100644
index 0000000000000000000000000000000000000000..67e76ef2b0b4ee5620a84fb93b88a149930a0cf6
--- /dev/null
+++ b/webrtc/test/file_audio_device.h
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2017 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_FILE_AUDIO_DEVICE_H_
+#define WEBRTC_TEST_FILE_AUDIO_DEVICE_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class EventTimerWrapper;
+
+namespace test {
+
+// FileReaderAudioDevice implements an AudioDevice module that acts as a
+// capturer (read audio from a WAV file and send it). It uses 10ms audio frames.
+class FileReaderAudioDevice : public FakeAudioDeviceModule {
+ public:
+  // Creates a new FileReaderAudioDevice. When capturing, 10 ms audio frames

[kwiberg-webrtc, 2017/03/07 10:11:57]

"When capturing, one 10 ms audio frame will [...]"

+  // will be processed every 100ms / |speed|.

[kwiberg-webrtc, 2017/03/07 10:11:57]

I'll just note for the record that I find the unit of |speed| very strange, but
keep it like this if you must.

[oprypin_webrtc, 2017/03/09 08:23:24]

Acknowledged.

+  // |sample_rate_hz| can be 8, 16, 32, 44.1 or 48kHz.
+  FileReaderAudioDevice(std::string filename, float speed,
+                        int sample_rate_hz);
+  ~FileReaderAudioDevice() override;
+
+  // Block until the input audio file ends.
+  bool WaitForFileEnd(int timeout_ms = rtc::Event::kForever);
+
+ private:
+  int32_t RegisterAudioCallback(AudioTransport* callback) override;
+
+  // Start reading audio data from the file and sending it.
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+
+  bool Recording() const override;

[kwiberg-webrtc, 2017/03/07 10:11:57]

Are these four public in FakeAudioDeviceModule? If so, they should probably be
so here too.

[oprypin_webrtc, 2017/03/09 08:23:24]

Done.

+
+  static bool Run(void* obj);
+  void ProcessAudio();
+
+  const std::string filename_;
+  const int sample_rate_hz_;
+  const float speed_;
+
+  rtc::CriticalSection lock_;
+  AudioTransport* audio_callback_ GUARDED_BY(lock_);
+  std::unique_ptr<WavReader> wav_reader_ GUARDED_BY(lock_);
+
+  std::vector<int16_t> playout_buffer_ GUARDED_BY(lock_);
+
+  std::unique_ptr<EventTimerWrapper> tick_;

[kwiberg-webrtc, 2017/03/07 10:11:57]

Can this be a const std::unique_ptr<EventTimerWrapper>? You only set it once, in
the constructor.

+  rtc::PlatformThread thread_;
+  // This event is set when the audio input file ends or has not been opened
+  // yet.
+  rtc::Event done_reading_;
+};
+
+// FileWriterAudioDevice implements an AudioDevice module that acts as a
+// renderer (receive audio and write it to a WAV file).
+// It uses 10ms audio frames.
+class FileWriterAudioDevice : public FakeAudioDeviceModule {
+ public:
+  // Creates a new FileWriterAudioDevice. When playing, 10 ms audio frames will
+  // be processed every 100ms / |speed|.
+  // |sample_rate_hz| can be 8, 16, 32, 44.1 or 48kHz.
+  FileWriterAudioDevice(std::string filename, float speed,
+                        int sample_rate_hz);
+  ~FileWriterAudioDevice() override;
+
+ private:
+  int32_t RegisterAudioCallback(AudioTransport* callback) override;
+
+  // Start receiving audio data and writing it to the file.
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+
+  bool Playing() const override;
+
+  static bool Run(void* obj);
+  void ProcessAudio();
+
+  const std::string filename_;
+  const int sample_rate_hz_;
+  const float speed_;
+
+  rtc::CriticalSection lock_;
+  AudioTransport* audio_callback_ GUARDED_BY(lock_);
+  std::unique_ptr<WavWriter> wav_writer_ GUARDED_BY(lock_);
+
+  std::vector<int16_t> playout_buffer_ GUARDED_BY(lock_);
+
+  std::unique_ptr<EventTimerWrapper> tick_;
+  rtc::PlatformThread thread_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_TEST_FILE_AUDIO_DEVICE_H_