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28 // This class implements an AudioCaptureModule that can be used to detect if
29 // audio is being received properly if it is fed by another AudioCaptureModule
30 // in some arbitrary audio pipeline where they are connected. It does not play
31 // out or record any audio so it does not need access to any hardware and can
32 // therefore be used in the gtest testing framework.
34 // Note P postfix of a function indicates that it should only be called by the
37 #ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
38 #define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
40 #include "webrtc/base/basictypes.h"
41 #include "webrtc/base/criticalsection.h"
42 #include "webrtc/base/messagehandler.h"
43 #include "webrtc/base/scoped_ref_ptr.h"
44 #include "webrtc/common_types.h"
45 #include "webrtc/modules/audio_device/include/audio_device.h"
53 class FakeAudioCaptureModule
54 : public webrtc::AudioDeviceModule,
55 public rtc::MessageHandler {
57 typedef uint16 Sample;
59 // The value for the following constants have been derived by running VoE
60 // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
61 enum{kNumberSamples = 440};
62 enum{kNumberBytesPerSample = sizeof(Sample)};
64 // Creates a FakeAudioCaptureModule or returns NULL on failure.
65 // |process_thread| is used to push and pull audio frames to and from the
66 // returned instance. Note: ownership of |process_thread| is not handed over.
67 static rtc::scoped_refptr<FakeAudioCaptureModule> Create(
68 rtc::Thread* process_thread);
70 // Returns the number of frames that have been successfully pulled by the
71 // instance. Note that correctly detecting success can only be done if the
72 // pulled frame was generated/pushed from a FakeAudioCaptureModule.
73 int frames_received() const;
75 // Following functions are inherited from webrtc::AudioDeviceModule.
76 // Only functions called by PeerConnection are implemented, the rest do
77 // nothing and return success. If a function is not expected to be called by
78 // PeerConnection an assertion is triggered if it is in fact called.
79 virtual int32_t Version(char* version,
80 uint32_t& remaining_buffer_in_bytes,
81 uint32_t& position) const;
82 virtual int32_t TimeUntilNextProcess();
83 virtual int32_t Process();
84 virtual int32_t ChangeUniqueId(const int32_t id);
86 virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const;
88 virtual ErrorCode LastError() const;
89 virtual int32_t RegisterEventObserver(
90 webrtc::AudioDeviceObserver* event_callback);
92 // Note: Calling this method from a callback may result in deadlock.
93 virtual int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback);
95 virtual int32_t Init();
96 virtual int32_t Terminate();
97 virtual bool Initialized() const;
99 virtual int16_t PlayoutDevices();
100 virtual int16_t RecordingDevices();
101 virtual int32_t PlayoutDeviceName(uint16_t index,
102 char name[webrtc::kAdmMaxDeviceNameSize],
103 char guid[webrtc::kAdmMaxGuidSize]);
104 virtual int32_t RecordingDeviceName(uint16_t index,
105 char name[webrtc::kAdmMaxDeviceNameSize],
106 char guid[webrtc::kAdmMaxGuidSize]);
108 virtual int32_t SetPlayoutDevice(uint16_t index);
109 virtual int32_t SetPlayoutDevice(WindowsDeviceType device);
110 virtual int32_t SetRecordingDevice(uint16_t index);
111 virtual int32_t SetRecordingDevice(WindowsDeviceType device);
113 virtual int32_t PlayoutIsAvailable(bool* available);
114 virtual int32_t InitPlayout();
115 virtual bool PlayoutIsInitialized() const;
116 virtual int32_t RecordingIsAvailable(bool* available);
117 virtual int32_t InitRecording();
118 virtual bool RecordingIsInitialized() const;
120 virtual int32_t StartPlayout();
121 virtual int32_t StopPlayout();
122 virtual bool Playing() const;
123 virtual int32_t StartRecording();
124 virtual int32_t StopRecording();
125 virtual bool Recording() const;
127 virtual int32_t SetAGC(bool enable);
128 virtual bool AGC() const;
130 virtual int32_t SetWaveOutVolume(uint16_t volume_left,
131 uint16_t volume_right);
132 virtual int32_t WaveOutVolume(uint16_t* volume_left,
133 uint16_t* volume_right) const;
135 virtual int32_t SpeakerIsAvailable(bool* available);
136 virtual int32_t InitSpeaker();
137 virtual bool SpeakerIsInitialized() const;
138 virtual int32_t MicrophoneIsAvailable(bool* available);
139 virtual int32_t InitMicrophone();
140 virtual bool MicrophoneIsInitialized() const;
142 virtual int32_t SpeakerVolumeIsAvailable(bool* available);
143 virtual int32_t SetSpeakerVolume(uint32_t volume);
144 virtual int32_t SpeakerVolume(uint32_t* volume) const;
145 virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const;
146 virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const;
147 virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const;
149 virtual int32_t MicrophoneVolumeIsAvailable(bool* available);
150 virtual int32_t SetMicrophoneVolume(uint32_t volume);
151 virtual int32_t MicrophoneVolume(uint32_t* volume) const;
152 virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const;
154 virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const;
155 virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const;
157 virtual int32_t SpeakerMuteIsAvailable(bool* available);
158 virtual int32_t SetSpeakerMute(bool enable);
159 virtual int32_t SpeakerMute(bool* enabled) const;
161 virtual int32_t MicrophoneMuteIsAvailable(bool* available);
162 virtual int32_t SetMicrophoneMute(bool enable);
163 virtual int32_t MicrophoneMute(bool* enabled) const;
165 virtual int32_t MicrophoneBoostIsAvailable(bool* available);
166 virtual int32_t SetMicrophoneBoost(bool enable);
167 virtual int32_t MicrophoneBoost(bool* enabled) const;
169 virtual int32_t StereoPlayoutIsAvailable(bool* available) const;
170 virtual int32_t SetStereoPlayout(bool enable);
171 virtual int32_t StereoPlayout(bool* enabled) const;
172 virtual int32_t StereoRecordingIsAvailable(bool* available) const;
173 virtual int32_t SetStereoRecording(bool enable);
174 virtual int32_t StereoRecording(bool* enabled) const;
175 virtual int32_t SetRecordingChannel(const ChannelType channel);
176 virtual int32_t RecordingChannel(ChannelType* channel) const;
178 virtual int32_t SetPlayoutBuffer(const BufferType type,
179 uint16_t size_ms = 0);
180 virtual int32_t PlayoutBuffer(BufferType* type,
181 uint16_t* size_ms) const;
182 virtual int32_t PlayoutDelay(uint16_t* delay_ms) const;
183 virtual int32_t RecordingDelay(uint16_t* delay_ms) const;
185 virtual int32_t CPULoad(uint16_t* load) const;
187 virtual int32_t StartRawOutputFileRecording(
188 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
189 virtual int32_t StopRawOutputFileRecording();
190 virtual int32_t StartRawInputFileRecording(
191 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
192 virtual int32_t StopRawInputFileRecording();
194 virtual int32_t SetRecordingSampleRate(const uint32_t samples_per_sec);
195 virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const;
196 virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec);
197 virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const;
199 virtual int32_t ResetAudioDevice();
200 virtual int32_t SetLoudspeakerStatus(bool enable);
201 virtual int32_t GetLoudspeakerStatus(bool* enabled) const;
202 // End of functions inherited from webrtc::AudioDeviceModule.
204 // The following function is inherited from rtc::MessageHandler.
205 virtual void OnMessage(rtc::Message* msg);
208 // The constructor is protected because the class needs to be created as a
209 // reference counted object (for memory managment reasons). It could be
210 // exposed in which case the burden of proper instantiation would be put on
211 // the creator of a FakeAudioCaptureModule instance. To create an instance of
212 // this class use the Create(..) API.
213 explicit FakeAudioCaptureModule(rtc::Thread* process_thread);
214 // The destructor is protected because it is reference counted and should not
215 // be deleted directly.
216 virtual ~FakeAudioCaptureModule();
219 // Initializes the state of the FakeAudioCaptureModule. This API is called on
220 // creation by the Create() API.
222 // SetBuffer() sets all samples in send_buffer_ to |value|.
223 void SetSendBuffer(int value);
224 // Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0.
225 void ResetRecBuffer();
226 // Returns true if rec_buffer_ contains one or more sample greater than or
228 bool CheckRecBuffer(int value);
230 // Returns true/false depending on if recording or playback has been
232 bool ShouldStartProcessing();
234 // Starts or stops the pushing and pulling of audio frames.
235 void UpdateProcessing(bool start);
237 // Starts the periodic calling of ProcessFrame() in a thread safe way.
238 void StartProcessP();
239 // Periodcally called function that ensures that frames are pulled and pushed
240 // periodically if enabled/started.
241 void ProcessFrameP();
242 // Pulls frames from the registered webrtc::AudioTransport.
243 void ReceiveFrameP();
244 // Pushes frames to the registered webrtc::AudioTransport.
246 // Stops the periodic calling of ProcessFrame() in a thread safe way.
249 // The time in milliseconds when Process() was last called or 0 if no call
251 uint32 last_process_time_ms_;
253 // Callback for playout and recording.
254 webrtc::AudioTransport* audio_callback_;
256 bool recording_; // True when audio is being pushed from the instance.
257 bool playing_; // True when audio is being pulled by the instance.
259 bool play_is_initialized_; // True when the instance is ready to pull audio.
260 bool rec_is_initialized_; // True when the instance is ready to push audio.
262 // Input to and output from RecordedDataIsAvailable(..) makes it possible to
263 // modify the current mic level. The implementation does not care about the
264 // mic level so it just feeds back what it receives.
265 uint32_t current_mic_level_;
267 // next_frame_time_ is updated in a non-drifting manner to indicate the next
268 // wall clock time the next frame should be generated and received. started_
269 // ensures that next_frame_time_ can be initialized properly on first call.
271 uint32 next_frame_time_;
273 // User provided thread context.
274 rtc::Thread* process_thread_;
276 // Buffer for storing samples received from the webrtc::AudioTransport.
277 char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
278 // Buffer for samples to send to the webrtc::AudioTransport.
279 char send_buffer_[kNumberSamples * kNumberBytesPerSample];
281 // Counter of frames received that have samples of high enough amplitude to
282 // indicate that the frames are not faked somewhere in the audio pipeline
283 // (e.g. by a jitter buffer).
284 int frames_received_;
286 // Protects variables that are accessed from process_thread_ and
288 mutable rtc::CriticalSection crit_;
289 // Protects |audio_callback_| that is accessed from process_thread_ and
291 rtc::CriticalSection crit_callback_;
294 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_