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+/*
+ *  Copyright (c) 2018 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 MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+// This is test API and is in development, so it can be changed/removed without
+// notice.
+
+// This class exists for historical reasons. For now it only contains static
+// methods to create test AudioDeviceModule. Implementation details of that
+// module are considered private. This class isn't intended to be instantiated.
+class TestAudioDeviceModule {
+ public:
+  // Returns the number of samples that Capturers and Renderers with this
+  // sampling frequency will work with every time Capture or Render is called.
+  static size_t SamplesPerFrame(int sampling_frequency_in_hz);
+
+  class Capturer {
+   public:
+    virtual ~Capturer() {}
+    // Returns the sampling frequency in Hz of the audio data that this
+    // capturer produces.
+    virtual int SamplingFrequency() const = 0;
+    // Returns the number of channels of captured audio data.
+    virtual int NumChannels() const = 0;
+    // Replaces the contents of `buffer` with 10ms of captured audio data
+    // (see TestAudioDeviceModule::SamplesPerFrame). Returns true if the
+    // capturer can keep producing data, or false when the capture finishes.
+    virtual bool Capture(rtc::BufferT<int16_t>* buffer) = 0;
+  };
+
+  class Renderer {
+   public:
+    virtual ~Renderer() {}
+    // Returns the sampling frequency in Hz of the audio data that this
+    // renderer receives.
+    virtual int SamplingFrequency() const = 0;
+    // Returns the number of channels of audio data to be required.
+    virtual int NumChannels() const = 0;
+    // Renders the passed audio data and returns true if the renderer wants
+    // to keep receiving data, or false otherwise.
+    virtual bool Render(rtc::ArrayView<const int16_t> data) = 0;
+  };
+
+  // A fake capturer that generates pulses with random samples between
+  // -max_amplitude and +max_amplitude.
+  class PulsedNoiseCapturer : public Capturer {
+   public:
+    ~PulsedNoiseCapturer() override {}
+
+    virtual void SetMaxAmplitude(int16_t amplitude) = 0;
+  };
+
+  // Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
+  // frames will be processed every 10ms / `speed`.
+  // `capturer` is an object that produces audio data. Can be nullptr if this
+  // device is never used for recording.
+  // `renderer` is an object that receives audio data that would have been
+  // played out. Can be nullptr if this device is never used for playing.
+  // Use one of the Create... functions to get these instances.
+  static rtc::scoped_refptr<AudioDeviceModule> Create(
+      TaskQueueFactory* task_queue_factory,
+      std::unique_ptr<Capturer> capturer,
+      std::unique_ptr<Renderer> renderer,
+      float speed = 1);
+
+  // Returns a Capturer instance that generates a signal of `num_channels`
+  // channels where every second frame is zero and every second frame is evenly
+  // distributed random noise with max amplitude `max_amplitude`.
+  static std::unique_ptr<PulsedNoiseCapturer> CreatePulsedNoiseCapturer(
+      int16_t max_amplitude,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Renderer instance that does nothing with the audio data.
+  static std::unique_ptr<Renderer> CreateDiscardRenderer(
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // WavReader and WavWriter creation based on file name.
+
+  // Returns a Capturer instance that gets its data from a WAV file. The sample
+  // rate and channels will be checked against the Wav file.
+  static std::unique_ptr<Capturer> CreateWavFileReader(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Capturer instance that gets its data from a file.
+  // Automatically detects sample rate and num of channels.
+  // `repeat` - if true, the file will be replayed from the start when we reach
+  // the end of file.
+  static std::unique_ptr<Capturer> CreateWavFileReader(
+      absl::string_view filename,
+      bool repeat = false);
+
+  // Returns a Renderer instance that writes its data to a file.
+  static std::unique_ptr<Renderer> CreateWavFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Renderer instance that writes its data to a WAV file, cutting
+  // off silence at the beginning (not necessarily perfect silence, see
+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
+  static std::unique_ptr<Renderer> CreateBoundedWavFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Capturer instance that gets its data from a raw file (*.raw).
+  static std::unique_ptr<Capturer> CreateRawFileReader(
+      absl::string_view filename,
+      int sampling_frequency_in_hz = 48000,
+      int num_channels = 2,
+      bool repeat = true);
+
+  // Returns a Renderer instance that writes its data to a raw file (*.raw),
+  // cutting off silence at the beginning (not necessarily perfect silence, see
+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
+  static std::unique_ptr<Renderer> CreateRawFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz = 48000,
+      int num_channels = 2);
+
+ private:
+  TestAudioDeviceModule() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_