1 /* 2 * libjingle 3 * Copyright 2012, Google Inc. 4 * 5 * Redistribution and use in source and binary forms, with or without 6 * modification, are permitted provided that the following conditions are met: 7 * 8 * 1. Redistributions of source code must retain the above copyright notice, 9 * this list of conditions and the following disclaimer. 10 * 2. Redistributions in binary form must reproduce the above copyright notice, 11 * this list of conditions and the following disclaimer in the documentation 12 * and/or other materials provided with the distribution. 13 * 3. The name of the author may not be used to endorse or promote products 14 * derived from this software without specific prior written permission. 15 * 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 26 */ 27 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. 33 34 // Note P postfix of a function indicates that it should only be called by the 35 // processing thread. 36 37 #ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ 38 #define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ 39 40 #include "talk/base/basictypes.h" 41 #include "talk/base/criticalsection.h" 42 #include "talk/base/messagehandler.h" 43 #include "talk/base/scoped_ref_ptr.h" 44 #include "webrtc/common_types.h" 45 #include "webrtc/modules/audio_device/include/audio_device.h" 46 47 namespace talk_base { 48 49 class Thread; 50 51 } // namespace talk_base 52 53 class FakeAudioCaptureModule 54 : public webrtc::AudioDeviceModule, 55 public talk_base::MessageHandler { 56 public: 57 typedef uint16 Sample; 58 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)}; 63 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 talk_base::scoped_refptr<FakeAudioCaptureModule> Create( 68 talk_base::Thread* process_thread); 69 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; 74 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); 85 86 virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const; 87 88 virtual ErrorCode LastError() const; 89 virtual int32_t RegisterEventObserver( 90 webrtc::AudioDeviceObserver* event_callback); 91 92 // Note: Calling this method from a callback may result in deadlock. 93 virtual int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback); 94 95 virtual int32_t Init(); 96 virtual int32_t Terminate(); 97 virtual bool Initialized() const; 98 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]); 107 108 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); 112 113 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; 119 120 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; 126 127 virtual int32_t SetAGC(bool enable); 128 virtual bool AGC() const; 129 130 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; 134 135 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; 141 142 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; 148 149 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; 153 154 virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const; 155 virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const; 156 157 virtual int32_t SpeakerMuteIsAvailable(bool* available); 158 virtual int32_t SetSpeakerMute(bool enable); 159 virtual int32_t SpeakerMute(bool* enabled) const; 160 161 virtual int32_t MicrophoneMuteIsAvailable(bool* available); 162 virtual int32_t SetMicrophoneMute(bool enable); 163 virtual int32_t MicrophoneMute(bool* enabled) const; 164 165 virtual int32_t MicrophoneBoostIsAvailable(bool* available); 166 virtual int32_t SetMicrophoneBoost(bool enable); 167 virtual int32_t MicrophoneBoost(bool* enabled) const; 168 169 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; 177 178 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; 184 185 virtual int32_t CPULoad(uint16_t* load) const; 186 187 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(); 193 194 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; 198 199 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. 203 204 // The following function is inherited from talk_base::MessageHandler. 205 virtual void OnMessage(talk_base::Message* msg); 206 207 protected: 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(talk_base::Thread* process_thread); 214 // The destructor is protected because it is reference counted and should not 215 // be deleted directly. 216 virtual ~FakeAudioCaptureModule(); 217 218 private: 219 // Initializes the state of the FakeAudioCaptureModule. This API is called on 220 // creation by the Create() API. 221 bool Initialize(); 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 227 // equal to |value|. 228 bool CheckRecBuffer(int value); 229 230 // Returns true/false depending on if recording or playback has been 231 // enabled/started. 232 bool ShouldStartProcessing(); 233 234 // Starts or stops the pushing and pulling of audio frames. 235 void UpdateProcessing(bool start); 236 237 // 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. 245 void SendFrameP(); 246 // Stops the periodic calling of ProcessFrame() in a thread safe way. 247 void StopProcessP(); 248 249 // The time in milliseconds when Process() was last called or 0 if no call 250 // has been made. 251 uint32 last_process_time_ms_; 252 253 // Callback for playout and recording. 254 webrtc::AudioTransport* audio_callback_; 255 256 bool recording_; // True when audio is being pushed from the instance. 257 bool playing_; // True when audio is being pulled by the instance. 258 259 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. 261 262 // 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_; 266 267 // 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. 270 bool started_; 271 uint32 next_frame_time_; 272 273 // User provided thread context. 274 talk_base::Thread* process_thread_; 275 276 // 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]; 280 281 // 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_; 285 286 // Protects variables that are accessed from process_thread_ and 287 // the main thread. 288 mutable talk_base::CriticalSection crit_; 289 // Protects |audio_callback_| that is accessed from process_thread_ and 290 // the main thread. 291 talk_base::CriticalSection crit_callback_; 292 }; 293 294 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ 295
