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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 TimeUntilNextProcess() OVERRIDE;
80 virtual int32_t Process() OVERRIDE;
81 virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
83 virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const OVERRIDE;
85 virtual ErrorCode LastError() const OVERRIDE;
86 virtual int32_t RegisterEventObserver(
87 webrtc::AudioDeviceObserver* event_callback) OVERRIDE;
89 // Note: Calling this method from a callback may result in deadlock.
90 virtual int32_t RegisterAudioCallback(
91 webrtc::AudioTransport* audio_callback) OVERRIDE;
93 virtual int32_t Init() OVERRIDE;
94 virtual int32_t Terminate() OVERRIDE;
95 virtual bool Initialized() const OVERRIDE;
97 virtual int16_t PlayoutDevices() OVERRIDE;
98 virtual int16_t RecordingDevices() OVERRIDE;
99 virtual int32_t PlayoutDeviceName(
101 char name[webrtc::kAdmMaxDeviceNameSize],
102 char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE;
103 virtual int32_t RecordingDeviceName(
105 char name[webrtc::kAdmMaxDeviceNameSize],
106 char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE;
108 virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
109 virtual int32_t SetPlayoutDevice(WindowsDeviceType device) OVERRIDE;
110 virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
111 virtual int32_t SetRecordingDevice(WindowsDeviceType device) OVERRIDE;
113 virtual int32_t PlayoutIsAvailable(bool* available) OVERRIDE;
114 virtual int32_t InitPlayout() OVERRIDE;
115 virtual bool PlayoutIsInitialized() const OVERRIDE;
116 virtual int32_t RecordingIsAvailable(bool* available) OVERRIDE;
117 virtual int32_t InitRecording() OVERRIDE;
118 virtual bool RecordingIsInitialized() const OVERRIDE;
120 virtual int32_t StartPlayout() OVERRIDE;
121 virtual int32_t StopPlayout() OVERRIDE;
122 virtual bool Playing() const OVERRIDE;
123 virtual int32_t StartRecording() OVERRIDE;
124 virtual int32_t StopRecording() OVERRIDE;
125 virtual bool Recording() const OVERRIDE;
127 virtual int32_t SetAGC(bool enable) OVERRIDE;
128 virtual bool AGC() const OVERRIDE;
130 virtual int32_t SetWaveOutVolume(uint16_t volume_left,
131 uint16_t volume_right) OVERRIDE;
132 virtual int32_t WaveOutVolume(uint16_t* volume_left,
133 uint16_t* volume_right) const OVERRIDE;
135 virtual int32_t InitSpeaker() OVERRIDE;
136 virtual bool SpeakerIsInitialized() const OVERRIDE;
137 virtual int32_t InitMicrophone() OVERRIDE;
138 virtual bool MicrophoneIsInitialized() const OVERRIDE;
140 virtual int32_t SpeakerVolumeIsAvailable(bool* available) OVERRIDE;
141 virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
142 virtual int32_t SpeakerVolume(uint32_t* volume) const OVERRIDE;
143 virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const OVERRIDE;
144 virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const OVERRIDE;
145 virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const OVERRIDE;
147 virtual int32_t MicrophoneVolumeIsAvailable(bool* available) OVERRIDE;
148 virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
149 virtual int32_t MicrophoneVolume(uint32_t* volume) const OVERRIDE;
150 virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const OVERRIDE;
152 virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const OVERRIDE;
153 virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const OVERRIDE;
155 virtual int32_t SpeakerMuteIsAvailable(bool* available) OVERRIDE;
156 virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
157 virtual int32_t SpeakerMute(bool* enabled) const OVERRIDE;
159 virtual int32_t MicrophoneMuteIsAvailable(bool* available) OVERRIDE;
160 virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
161 virtual int32_t MicrophoneMute(bool* enabled) const OVERRIDE;
163 virtual int32_t MicrophoneBoostIsAvailable(bool* available) OVERRIDE;
164 virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
165 virtual int32_t MicrophoneBoost(bool* enabled) const OVERRIDE;
167 virtual int32_t StereoPlayoutIsAvailable(bool* available) const OVERRIDE;
168 virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
169 virtual int32_t StereoPlayout(bool* enabled) const OVERRIDE;
170 virtual int32_t StereoRecordingIsAvailable(bool* available) const OVERRIDE;
171 virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
172 virtual int32_t StereoRecording(bool* enabled) const OVERRIDE;
173 virtual int32_t SetRecordingChannel(const ChannelType channel) OVERRIDE;
174 virtual int32_t RecordingChannel(ChannelType* channel) const OVERRIDE;
176 virtual int32_t SetPlayoutBuffer(const BufferType type,
177 uint16_t size_ms = 0) OVERRIDE;
178 virtual int32_t PlayoutBuffer(BufferType* type,
179 uint16_t* size_ms) const OVERRIDE;
180 virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE;
181 virtual int32_t RecordingDelay(uint16_t* delay_ms) const OVERRIDE;
183 virtual int32_t CPULoad(uint16_t* load) const OVERRIDE;
185 virtual int32_t StartRawOutputFileRecording(
186 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) OVERRIDE;
187 virtual int32_t StopRawOutputFileRecording() OVERRIDE;
188 virtual int32_t StartRawInputFileRecording(
189 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) OVERRIDE;
190 virtual int32_t StopRawInputFileRecording() OVERRIDE;
192 virtual int32_t SetRecordingSampleRate(
193 const uint32_t samples_per_sec) OVERRIDE;
194 virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
195 virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) OVERRIDE;
196 virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
198 virtual int32_t ResetAudioDevice() OVERRIDE;
199 virtual int32_t SetLoudspeakerStatus(bool enable) OVERRIDE;
200 virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
201 // End of functions inherited from webrtc::AudioDeviceModule.
203 // The following function is inherited from rtc::MessageHandler.
204 virtual void OnMessage(rtc::Message* msg) OVERRIDE;
207 // The constructor is protected because the class needs to be created as a
208 // reference counted object (for memory managment reasons). It could be
209 // exposed in which case the burden of proper instantiation would be put on
210 // the creator of a FakeAudioCaptureModule instance. To create an instance of
211 // this class use the Create(..) API.
212 explicit FakeAudioCaptureModule(rtc::Thread* process_thread);
213 // The destructor is protected because it is reference counted and should not
214 // be deleted directly.
215 virtual ~FakeAudioCaptureModule();
218 // Initializes the state of the FakeAudioCaptureModule. This API is called on
219 // creation by the Create() API.
221 // SetBuffer() sets all samples in send_buffer_ to |value|.
222 void SetSendBuffer(int value);
223 // Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0.
224 void ResetRecBuffer();
225 // Returns true if rec_buffer_ contains one or more sample greater than or
227 bool CheckRecBuffer(int value);
229 // Returns true/false depending on if recording or playback has been
231 bool ShouldStartProcessing();
233 // Starts or stops the pushing and pulling of audio frames.
234 void UpdateProcessing(bool start);
236 // Starts the periodic calling of ProcessFrame() in a thread safe way.
237 void StartProcessP();
238 // Periodcally called function that ensures that frames are pulled and pushed
239 // periodically if enabled/started.
240 void ProcessFrameP();
241 // Pulls frames from the registered webrtc::AudioTransport.
242 void ReceiveFrameP();
243 // Pushes frames to the registered webrtc::AudioTransport.
245 // Stops the periodic calling of ProcessFrame() in a thread safe way.
248 // The time in milliseconds when Process() was last called or 0 if no call
250 uint32 last_process_time_ms_;
252 // Callback for playout and recording.
253 webrtc::AudioTransport* audio_callback_;
255 bool recording_; // True when audio is being pushed from the instance.
256 bool playing_; // True when audio is being pulled by the instance.
258 bool play_is_initialized_; // True when the instance is ready to pull audio.
259 bool rec_is_initialized_; // True when the instance is ready to push audio.
261 // Input to and output from RecordedDataIsAvailable(..) makes it possible to
262 // modify the current mic level. The implementation does not care about the
263 // mic level so it just feeds back what it receives.
264 uint32_t current_mic_level_;
266 // next_frame_time_ is updated in a non-drifting manner to indicate the next
267 // wall clock time the next frame should be generated and received. started_
268 // ensures that next_frame_time_ can be initialized properly on first call.
270 uint32 next_frame_time_;
272 // User provided thread context.
273 rtc::Thread* process_thread_;
275 // Buffer for storing samples received from the webrtc::AudioTransport.
276 char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
277 // Buffer for samples to send to the webrtc::AudioTransport.
278 char send_buffer_[kNumberSamples * kNumberBytesPerSample];
280 // Counter of frames received that have samples of high enough amplitude to
281 // indicate that the frames are not faked somewhere in the audio pipeline
282 // (e.g. by a jitter buffer).
283 int frames_received_;
285 // Protects variables that are accessed from process_thread_ and
287 mutable rtc::CriticalSection crit_;
288 // Protects |audio_callback_| that is accessed from process_thread_ and
290 rtc::CriticalSection crit_callback_;
293 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_