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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_
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