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/*
* 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 <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/event.h"
#include "webrtc/rtc_base/platform_thread.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class EventTimerWrapper;
namespace test {
// FakeAudioDevice implements an AudioDevice module that can act both as a
// capturer and a renderer. It will use 10ms audio frames.
class FakeAudioDevice : public FakeAudioDeviceModule {
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;
// Replaces the contents of |buffer| with 10ms of captured audio data
// (see FakeAudioDevice::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;
// 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;
};
// Creates a new FakeAudioDevice. 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.
FakeAudioDevice(std::unique_ptr<Capturer> capturer,
std::unique_ptr<Renderer> renderer,
float speed = 1);
~FakeAudioDevice() override;
// Returns a Capturer instance that generates a signal where every second
// frame is zero and every second frame is evenly distributed random noise
// with max amplitude |max_amplitude|.
static std::unique_ptr<Capturer> CreatePulsedNoiseCapturer(
int16_t max_amplitude, int sampling_frequency_in_hz);
// Returns a Capturer instance that gets its data from a file.
static std::unique_ptr<Capturer> CreateWavFileReader(
std::string filename, int sampling_frequency_in_hz);
// Returns a Capturer instance that gets its data from a file.
// Automatically detects sample rate.
static std::unique_ptr<Capturer> CreateWavFileReader(std::string filename);
// Returns a Renderer instance that writes its data to a file.
static std::unique_ptr<Renderer> CreateWavFileWriter(
std::string filename, int sampling_frequency_in_hz);
// 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(
std::string filename, int sampling_frequency_in_hz);
// Returns a Renderer instance that does nothing with the audio data.
static std::unique_ptr<Renderer> CreateDiscardRenderer(
int sampling_frequency_in_hz);
int32_t Init() override;
int32_t RegisterAudioCallback(AudioTransport* callback) override;
int32_t StartPlayout() override;
int32_t StopPlayout() override;
int32_t StartRecording() override;
int32_t StopRecording() override;
bool Playing() const override;
bool Recording() const override;
// Blocks until the Renderer refuses to receive data.
// Returns false if |timeout_ms| passes before that happens.
bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever);
// Blocks until the Recorder stops producing data.
// Returns false if |timeout_ms| passes before that happens.
bool WaitForRecordingEnd(int timeout_ms = rtc::Event::kForever);
private:
static bool Run(void* obj);
void ProcessAudio();
const std::unique_ptr<Capturer> capturer_ GUARDED_BY(lock_);
const std::unique_ptr<Renderer> renderer_ GUARDED_BY(lock_);
const float speed_;
rtc::CriticalSection lock_;
AudioTransport* audio_callback_ GUARDED_BY(lock_);
bool rendering_ GUARDED_BY(lock_);
bool capturing_ GUARDED_BY(lock_);
rtc::Event done_rendering_;
rtc::Event done_capturing_;
std::vector<int16_t> playout_buffer_ GUARDED_BY(lock_);
rtc::BufferT<int16_t> recording_buffer_ GUARDED_BY(lock_);
std::unique_ptr<EventTimerWrapper> tick_;
rtc::PlatformThread thread_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_