/* * 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 #include #include #include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_device/include/audio_device_defines.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" namespace rtc { class Thread; } // namespace rtc 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 rtc::scoped_refptr 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; } #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 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_; }; #endif // PC_TEST_FAKE_AUDIO_CAPTURE_MODULE_H_