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/*
* Copyright 2012 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.
*/
// This class implements an AudioCaptureModule that can be used to detect if
// audio is being received properly if it is fed by another AudioCaptureModule
// in some arbitrary audio pipeline where they are connected. It does not play
// out or record any audio so it does not need access to any hardware and can
// therefore be used in the gtest testing framework.
// Note P postfix of a function indicates that it should only be called by the
// processing thread.
#ifndef PC_TEST_FAKE_AUDIO_CAPTURE_MODULE_H_
#define PC_TEST_FAKE_AUDIO_CAPTURE_MODULE_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include <optional>
#include "api/audio/audio_device.h"
#include "api/audio/audio_device_defines.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
class FakeAudioCaptureModule : public webrtc::AudioDeviceModule {
public:
typedef uint16_t Sample;
// The value for the following constants have been derived by running VoE
// using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
static const size_t kNumberSamples = 440;
static const size_t kNumberBytesPerSample = sizeof(Sample);
// Creates a FakeAudioCaptureModule or returns NULL on failure.
static webrtc::scoped_refptr<FakeAudioCaptureModule> Create();
// Returns the number of frames that have been successfully pulled by the
// instance. Note that correctly detecting success can only be done if the
// pulled frame was generated/pushed from a FakeAudioCaptureModule.
int frames_received() const RTC_LOCKS_EXCLUDED(mutex_);
int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
// Note: Calling this method from a callback may result in deadlock.
int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback) override
RTC_LOCKS_EXCLUDED(mutex_);
int32_t Init() override;
int32_t Terminate() override;
bool Initialized() const override;
int16_t PlayoutDevices() override;
int16_t RecordingDevices() override;
int32_t PlayoutDeviceName(uint16_t index,
char name[webrtc::kAdmMaxDeviceNameSize],
char guid[webrtc::kAdmMaxGuidSize]) override;
int32_t RecordingDeviceName(uint16_t index,
char name[webrtc::kAdmMaxDeviceNameSize],
char guid[webrtc::kAdmMaxGuidSize]) override;
int32_t SetPlayoutDevice(uint16_t index) override;
int32_t SetPlayoutDevice(WindowsDeviceType device) override;
int32_t SetRecordingDevice(uint16_t index) override;
int32_t SetRecordingDevice(WindowsDeviceType device) override;
int32_t PlayoutIsAvailable(bool* available) override;
int32_t InitPlayout() override;
bool PlayoutIsInitialized() const override;
int32_t RecordingIsAvailable(bool* available) override;
int32_t InitRecording() override;
bool RecordingIsInitialized() const override;
int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
bool Playing() const RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
bool Recording() const RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() override;
bool MicrophoneIsInitialized() const override;
int32_t SpeakerVolumeIsAvailable(bool* available) override;
int32_t SetSpeakerVolume(uint32_t volume) override;
int32_t SpeakerVolume(uint32_t* volume) const override;
int32_t MaxSpeakerVolume(uint32_t* max_volume) const override;
int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
int32_t MicrophoneVolumeIsAvailable(bool* available) override;
int32_t SetMicrophoneVolume(uint32_t volume)
RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t MicrophoneVolume(uint32_t* volume) const
RTC_LOCKS_EXCLUDED(mutex_) override;
int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
int32_t SpeakerMuteIsAvailable(bool* available) override;
int32_t SetSpeakerMute(bool enable) override;
int32_t SpeakerMute(bool* enabled) const override;
int32_t MicrophoneMuteIsAvailable(bool* available) override;
int32_t SetMicrophoneMute(bool enable) override;
int32_t MicrophoneMute(bool* enabled) const override;
int32_t StereoPlayoutIsAvailable(bool* available) const override;
int32_t SetStereoPlayout(bool enable) override;
int32_t StereoPlayout(bool* enabled) const override;
int32_t StereoRecordingIsAvailable(bool* available) const override;
int32_t SetStereoRecording(bool enable) override;
int32_t StereoRecording(bool* enabled) const override;
int32_t PlayoutDelay(uint16_t* delay_ms) const override;
bool BuiltInAECIsAvailable() const override { return false; }
int32_t EnableBuiltInAEC(bool enable) override { return -1; }
bool BuiltInAGCIsAvailable() const override { return false; }
int32_t EnableBuiltInAGC(bool enable) override { return -1; }
bool BuiltInNSIsAvailable() const override { return false; }
int32_t EnableBuiltInNS(bool enable) override { return -1; }
int32_t GetPlayoutUnderrunCount() const override { return -1; }
std::optional<webrtc::AudioDeviceModule::Stats> GetStats() const override {
return webrtc::AudioDeviceModule::Stats();
}
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(
webrtc::AudioParameters* params) const override {
return -1;
}
int GetRecordAudioParameters(webrtc::AudioParameters* params) const override {
return -1;
}
#endif // WEBRTC_IOS
// End of functions inherited from webrtc::AudioDeviceModule.
protected:
// The constructor is protected because the class needs to be created as a
// reference counted object (for memory managment reasons). It could be
// exposed in which case the burden of proper instantiation would be put on
// the creator of a FakeAudioCaptureModule instance. To create an instance of
// this class use the Create(..) API.
FakeAudioCaptureModule();
// The destructor is protected because it is reference counted and should not
// be deleted directly.
virtual ~FakeAudioCaptureModule();
private:
// Initializes the state of the FakeAudioCaptureModule. This API is called on
// creation by the Create() API.
bool Initialize();
// SetBuffer() sets all samples in send_buffer_ to `value`.
void SetSendBuffer(int value);
// Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0.
void ResetRecBuffer();
// Returns true if rec_buffer_ contains one or more sample greater than or
// equal to `value`.
bool CheckRecBuffer(int value);
// Returns true/false depending on if recording or playback has been
// enabled/started.
bool ShouldStartProcessing() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Starts or stops the pushing and pulling of audio frames.
void UpdateProcessing(bool start) RTC_LOCKS_EXCLUDED(mutex_);
// Starts the periodic calling of ProcessFrame() in a thread safe way.
void StartProcessP();
// Periodcally called function that ensures that frames are pulled and pushed
// periodically if enabled/started.
void ProcessFrameP() RTC_LOCKS_EXCLUDED(mutex_);
// Pulls frames from the registered webrtc::AudioTransport.
void ReceiveFrameP() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Pushes frames to the registered webrtc::AudioTransport.
void SendFrameP() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Callback for playout and recording.
webrtc::AudioTransport* audio_callback_ RTC_GUARDED_BY(mutex_);
bool recording_ RTC_GUARDED_BY(
mutex_); // True when audio is being pushed from the instance.
bool playing_ RTC_GUARDED_BY(
mutex_); // True when audio is being pulled by the instance.
bool play_is_initialized_; // True when the instance is ready to pull audio.
bool rec_is_initialized_; // True when the instance is ready to push audio.
// Input to and output from RecordedDataIsAvailable(..) makes it possible to
// modify the current mic level. The implementation does not care about the
// mic level so it just feeds back what it receives.
uint32_t current_mic_level_ RTC_GUARDED_BY(mutex_);
// next_frame_time_ is updated in a non-drifting manner to indicate the next
// wall clock time the next frame should be generated and received. started_
// ensures that next_frame_time_ can be initialized properly on first call.
bool started_ RTC_GUARDED_BY(mutex_);
int64_t next_frame_time_ RTC_GUARDED_BY(process_thread_checker_);
std::unique_ptr<webrtc::Thread> process_thread_;
// Buffer for storing samples received from the webrtc::AudioTransport.
char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
// Buffer for samples to send to the webrtc::AudioTransport.
char send_buffer_[kNumberSamples * kNumberBytesPerSample];
// Counter of frames received that have samples of high enough amplitude to
// indicate that the frames are not faked somewhere in the audio pipeline
// (e.g. by a jitter buffer).
int frames_received_;
// Protects variables that are accessed from process_thread_ and
// the main thread.
mutable webrtc::Mutex mutex_;
webrtc::SequenceChecker process_thread_checker_{
webrtc::SequenceChecker::kDetached};
};
#endif // PC_TEST_FAKE_AUDIO_CAPTURE_MODULE_H_
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