/* * libjingle * Copyright 2012, Google Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: * * 1. Redistributions of source code must retain the above copyright notice, * this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright notice, * this list of conditions and the following disclaimer in the documentation * and/or other materials provided with the distribution. * 3. The name of the author may not be used to endorse or promote products * derived from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ // 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 TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ #define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ #include "talk/base/basictypes.h" #include "talk/base/criticalsection.h" #include "talk/base/messagehandler.h" #include "talk/base/scoped_ref_ptr.h" #include "webrtc/common_types.h" #include "webrtc/modules/audio_device/include/audio_device.h" namespace talk_base { class Thread; } // namespace talk_base class FakeAudioCaptureModule : public webrtc::AudioDeviceModule, public talk_base::MessageHandler { public: typedef uint16 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. enum{kNumberSamples = 440}; enum{kNumberBytesPerSample = sizeof(Sample)}; // Creates a FakeAudioCaptureModule or returns NULL on failure. // |process_thread| is used to push and pull audio frames to and from the // returned instance. Note: ownership of |process_thread| is not handed over. static talk_base::scoped_refptr Create( talk_base::Thread* process_thread); // 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; // Following functions are inherited from webrtc::AudioDeviceModule. // Only functions called by PeerConnection are implemented, the rest do // nothing and return success. If a function is not expected to be called by // PeerConnection an assertion is triggered if it is in fact called. virtual int32_t Version(char* version, uint32_t& remaining_buffer_in_bytes, uint32_t& position) const; virtual int32_t TimeUntilNextProcess(); virtual int32_t Process(); virtual int32_t ChangeUniqueId(const int32_t id); virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const; virtual ErrorCode LastError() const; virtual int32_t RegisterEventObserver( webrtc::AudioDeviceObserver* event_callback); // Note: Calling this method from a callback may result in deadlock. virtual int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback); virtual int32_t Init(); virtual int32_t Terminate(); virtual bool Initialized() const; virtual int16_t PlayoutDevices(); virtual int16_t RecordingDevices(); virtual int32_t PlayoutDeviceName(uint16_t index, char name[webrtc::kAdmMaxDeviceNameSize], char guid[webrtc::kAdmMaxGuidSize]); virtual int32_t RecordingDeviceName(uint16_t index, char name[webrtc::kAdmMaxDeviceNameSize], char guid[webrtc::kAdmMaxGuidSize]); virtual int32_t SetPlayoutDevice(uint16_t index); virtual int32_t SetPlayoutDevice(WindowsDeviceType device); virtual int32_t SetRecordingDevice(uint16_t index); virtual int32_t SetRecordingDevice(WindowsDeviceType device); virtual int32_t PlayoutIsAvailable(bool* available); virtual int32_t InitPlayout(); virtual bool PlayoutIsInitialized() const; virtual int32_t RecordingIsAvailable(bool* available); virtual int32_t InitRecording(); virtual bool RecordingIsInitialized() const; virtual int32_t StartPlayout(); virtual int32_t StopPlayout(); virtual bool Playing() const; virtual int32_t StartRecording(); virtual int32_t StopRecording(); virtual bool Recording() const; virtual int32_t SetAGC(bool enable); virtual bool AGC() const; virtual int32_t SetWaveOutVolume(uint16_t volume_left, uint16_t volume_right); virtual int32_t WaveOutVolume(uint16_t* volume_left, uint16_t* volume_right) const; virtual int32_t SpeakerIsAvailable(bool* available); virtual int32_t InitSpeaker(); virtual bool SpeakerIsInitialized() const; virtual int32_t MicrophoneIsAvailable(bool* available); virtual int32_t InitMicrophone(); virtual bool MicrophoneIsInitialized() const; virtual int32_t SpeakerVolumeIsAvailable(bool* available); virtual int32_t SetSpeakerVolume(uint32_t volume); virtual int32_t SpeakerVolume(uint32_t* volume) const; virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const; virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const; virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const; virtual int32_t MicrophoneVolumeIsAvailable(bool* available); virtual int32_t SetMicrophoneVolume(uint32_t volume); virtual int32_t MicrophoneVolume(uint32_t* volume) const; virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const; virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const; virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const; virtual int32_t SpeakerMuteIsAvailable(bool* available); virtual int32_t SetSpeakerMute(bool enable); virtual int32_t SpeakerMute(bool* enabled) const; virtual int32_t MicrophoneMuteIsAvailable(bool* available); virtual int32_t SetMicrophoneMute(bool enable); virtual int32_t MicrophoneMute(bool* enabled) const; virtual int32_t MicrophoneBoostIsAvailable(bool* available); virtual int32_t SetMicrophoneBoost(bool enable); virtual int32_t MicrophoneBoost(bool* enabled) const; virtual int32_t StereoPlayoutIsAvailable(bool* available) const; virtual int32_t SetStereoPlayout(bool enable); virtual int32_t StereoPlayout(bool* enabled) const; virtual int32_t StereoRecordingIsAvailable(bool* available) const; virtual int32_t SetStereoRecording(bool enable); virtual int32_t StereoRecording(bool* enabled) const; virtual int32_t SetRecordingChannel(const ChannelType channel); virtual int32_t RecordingChannel(ChannelType* channel) const; virtual int32_t SetPlayoutBuffer(const BufferType type, uint16_t size_ms = 0); virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const; virtual int32_t PlayoutDelay(uint16_t* delay_ms) const; virtual int32_t RecordingDelay(uint16_t* delay_ms) const; virtual int32_t CPULoad(uint16_t* load) const; virtual int32_t StartRawOutputFileRecording( const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]); virtual int32_t StopRawOutputFileRecording(); virtual int32_t StartRawInputFileRecording( const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]); virtual int32_t StopRawInputFileRecording(); virtual int32_t SetRecordingSampleRate(const uint32_t samples_per_sec); virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const; virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec); virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const; virtual int32_t ResetAudioDevice(); virtual int32_t SetLoudspeakerStatus(bool enable); virtual int32_t GetLoudspeakerStatus(bool* enabled) const; // End of functions inherited from webrtc::AudioDeviceModule. // The following function is inherited from talk_base::MessageHandler. virtual void OnMessage(talk_base::Message* msg); 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. explicit FakeAudioCaptureModule(talk_base::Thread* process_thread); // 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(); // Starts or stops the pushing and pulling of audio frames. void UpdateProcessing(bool start); // 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(); // Pulls frames from the registered webrtc::AudioTransport. void ReceiveFrameP(); // Pushes frames to the registered webrtc::AudioTransport. void SendFrameP(); // Stops the periodic calling of ProcessFrame() in a thread safe way. void StopProcessP(); // The time in milliseconds when Process() was last called or 0 if no call // has been made. uint32 last_process_time_ms_; // Callback for playout and recording. webrtc::AudioTransport* audio_callback_; bool recording_; // True when audio is being pushed from the instance. bool playing_; // 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_; // 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_; uint32 next_frame_time_; // User provided thread context. talk_base::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 talk_base::CriticalSection crit_; // Protects |audio_callback_| that is accessed from process_thread_ and // the main thread. talk_base::CriticalSection crit_callback_; }; #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_