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 "webrtc/base/basictypes.h" 41 #include "webrtc/base/criticalsection.h" 42 #include "webrtc/base/messagehandler.h" 43 #include "webrtc/base/scoped_ptr.h" 44 #include "webrtc/base/scoped_ref_ptr.h" 45 #include "webrtc/common_types.h" 46 #include "webrtc/modules/audio_device/include/audio_device.h" 47 48 namespace rtc { 49 class Thread; 50 } // namespace rtc 51 52 class FakeAudioCaptureModule 53 : public webrtc::AudioDeviceModule, 54 public rtc::MessageHandler { 55 public: 56 typedef uint16_t Sample; 57 58 // The value for the following constants have been derived by running VoE 59 // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz. 60 static const size_t kNumberSamples = 440; 61 static const size_t kNumberBytesPerSample = sizeof(Sample); 62 63 // Creates a FakeAudioCaptureModule or returns NULL on failure. 64 static rtc::scoped_refptr<FakeAudioCaptureModule> Create(); 65 66 // Returns the number of frames that have been successfully pulled by the 67 // instance. Note that correctly detecting success can only be done if the 68 // pulled frame was generated/pushed from a FakeAudioCaptureModule. 69 int frames_received() const; 70 71 // Following functions are inherited from webrtc::AudioDeviceModule. 72 // Only functions called by PeerConnection are implemented, the rest do 73 // nothing and return success. If a function is not expected to be called by 74 // PeerConnection an assertion is triggered if it is in fact called. 75 int64_t TimeUntilNextProcess() override; 76 int32_t Process() override; 77 78 int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override; 79 80 ErrorCode LastError() const override; 81 int32_t RegisterEventObserver( 82 webrtc::AudioDeviceObserver* event_callback) override; 83 84 // Note: Calling this method from a callback may result in deadlock. 85 int32_t RegisterAudioCallback( 86 webrtc::AudioTransport* audio_callback) override; 87 88 int32_t Init() override; 89 int32_t Terminate() override; 90 bool Initialized() const override; 91 92 int16_t PlayoutDevices() override; 93 int16_t RecordingDevices() override; 94 int32_t PlayoutDeviceName(uint16_t index, 95 char name[webrtc::kAdmMaxDeviceNameSize], 96 char guid[webrtc::kAdmMaxGuidSize]) override; 97 int32_t RecordingDeviceName(uint16_t index, 98 char name[webrtc::kAdmMaxDeviceNameSize], 99 char guid[webrtc::kAdmMaxGuidSize]) override; 100 101 int32_t SetPlayoutDevice(uint16_t index) override; 102 int32_t SetPlayoutDevice(WindowsDeviceType device) override; 103 int32_t SetRecordingDevice(uint16_t index) override; 104 int32_t SetRecordingDevice(WindowsDeviceType device) override; 105 106 int32_t PlayoutIsAvailable(bool* available) override; 107 int32_t InitPlayout() override; 108 bool PlayoutIsInitialized() const override; 109 int32_t RecordingIsAvailable(bool* available) override; 110 int32_t InitRecording() override; 111 bool RecordingIsInitialized() const override; 112 113 int32_t StartPlayout() override; 114 int32_t StopPlayout() override; 115 bool Playing() const override; 116 int32_t StartRecording() override; 117 int32_t StopRecording() override; 118 bool Recording() const override; 119 120 int32_t SetAGC(bool enable) override; 121 bool AGC() const override; 122 123 int32_t SetWaveOutVolume(uint16_t volume_left, 124 uint16_t volume_right) override; 125 int32_t WaveOutVolume(uint16_t* volume_left, 126 uint16_t* volume_right) const override; 127 128 int32_t InitSpeaker() override; 129 bool SpeakerIsInitialized() const override; 130 int32_t InitMicrophone() override; 131 bool MicrophoneIsInitialized() const override; 132 133 int32_t SpeakerVolumeIsAvailable(bool* available) override; 134 int32_t SetSpeakerVolume(uint32_t volume) override; 135 int32_t SpeakerVolume(uint32_t* volume) const override; 136 int32_t MaxSpeakerVolume(uint32_t* max_volume) const override; 137 int32_t MinSpeakerVolume(uint32_t* min_volume) const override; 138 int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override; 139 140 int32_t MicrophoneVolumeIsAvailable(bool* available) override; 141 int32_t SetMicrophoneVolume(uint32_t volume) override; 142 int32_t MicrophoneVolume(uint32_t* volume) const override; 143 int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override; 144 145 int32_t MinMicrophoneVolume(uint32_t* min_volume) const override; 146 int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override; 147 148 int32_t SpeakerMuteIsAvailable(bool* available) override; 149 int32_t SetSpeakerMute(bool enable) override; 150 int32_t SpeakerMute(bool* enabled) const override; 151 152 int32_t MicrophoneMuteIsAvailable(bool* available) override; 153 int32_t SetMicrophoneMute(bool enable) override; 154 int32_t MicrophoneMute(bool* enabled) const override; 155 156 int32_t MicrophoneBoostIsAvailable(bool* available) override; 157 int32_t SetMicrophoneBoost(bool enable) override; 158 int32_t MicrophoneBoost(bool* enabled) const override; 159 160 int32_t StereoPlayoutIsAvailable(bool* available) const override; 161 int32_t SetStereoPlayout(bool enable) override; 162 int32_t StereoPlayout(bool* enabled) const override; 163 int32_t StereoRecordingIsAvailable(bool* available) const override; 164 int32_t SetStereoRecording(bool enable) override; 165 int32_t StereoRecording(bool* enabled) const override; 166 int32_t SetRecordingChannel(const ChannelType channel) override; 167 int32_t RecordingChannel(ChannelType* channel) const override; 168 169 int32_t SetPlayoutBuffer(const BufferType type, 170 uint16_t size_ms = 0) override; 171 int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override; 172 int32_t PlayoutDelay(uint16_t* delay_ms) const override; 173 int32_t RecordingDelay(uint16_t* delay_ms) const override; 174 175 int32_t CPULoad(uint16_t* load) const override; 176 177 int32_t StartRawOutputFileRecording( 178 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override; 179 int32_t StopRawOutputFileRecording() override; 180 int32_t StartRawInputFileRecording( 181 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override; 182 int32_t StopRawInputFileRecording() override; 183 184 int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override; 185 int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override; 186 int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override; 187 int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override; 188 189 int32_t ResetAudioDevice() override; 190 int32_t SetLoudspeakerStatus(bool enable) override; 191 int32_t GetLoudspeakerStatus(bool* enabled) const override; 192 virtual bool BuiltInAECIsAvailable() const { return false; } 193 virtual int32_t EnableBuiltInAEC(bool enable) { return -1; } 194 virtual bool BuiltInAGCIsAvailable() const { return false; } 195 virtual int32_t EnableBuiltInAGC(bool enable) { return -1; } 196 virtual bool BuiltInNSIsAvailable() const { return false; } 197 virtual int32_t EnableBuiltInNS(bool enable) { return -1; } 198 // End of functions inherited from webrtc::AudioDeviceModule. 199 200 // The following function is inherited from rtc::MessageHandler. 201 void OnMessage(rtc::Message* msg) override; 202 203 protected: 204 // The constructor is protected because the class needs to be created as a 205 // reference counted object (for memory managment reasons). It could be 206 // exposed in which case the burden of proper instantiation would be put on 207 // the creator of a FakeAudioCaptureModule instance. To create an instance of 208 // this class use the Create(..) API. 209 explicit FakeAudioCaptureModule(); 210 // The destructor is protected because it is reference counted and should not 211 // be deleted directly. 212 virtual ~FakeAudioCaptureModule(); 213 214 private: 215 // Initializes the state of the FakeAudioCaptureModule. This API is called on 216 // creation by the Create() API. 217 bool Initialize(); 218 // SetBuffer() sets all samples in send_buffer_ to |value|. 219 void SetSendBuffer(int value); 220 // Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0. 221 void ResetRecBuffer(); 222 // Returns true if rec_buffer_ contains one or more sample greater than or 223 // equal to |value|. 224 bool CheckRecBuffer(int value); 225 226 // Returns true/false depending on if recording or playback has been 227 // enabled/started. 228 bool ShouldStartProcessing(); 229 230 // Starts or stops the pushing and pulling of audio frames. 231 void UpdateProcessing(bool start); 232 233 // Starts the periodic calling of ProcessFrame() in a thread safe way. 234 void StartProcessP(); 235 // Periodcally called function that ensures that frames are pulled and pushed 236 // periodically if enabled/started. 237 void ProcessFrameP(); 238 // Pulls frames from the registered webrtc::AudioTransport. 239 void ReceiveFrameP(); 240 // Pushes frames to the registered webrtc::AudioTransport. 241 void SendFrameP(); 242 243 // The time in milliseconds when Process() was last called or 0 if no call 244 // has been made. 245 uint32_t last_process_time_ms_; 246 247 // Callback for playout and recording. 248 webrtc::AudioTransport* audio_callback_; 249 250 bool recording_; // True when audio is being pushed from the instance. 251 bool playing_; // True when audio is being pulled by the instance. 252 253 bool play_is_initialized_; // True when the instance is ready to pull audio. 254 bool rec_is_initialized_; // True when the instance is ready to push audio. 255 256 // Input to and output from RecordedDataIsAvailable(..) makes it possible to 257 // modify the current mic level. The implementation does not care about the 258 // mic level so it just feeds back what it receives. 259 uint32_t current_mic_level_; 260 261 // next_frame_time_ is updated in a non-drifting manner to indicate the next 262 // wall clock time the next frame should be generated and received. started_ 263 // ensures that next_frame_time_ can be initialized properly on first call. 264 bool started_; 265 uint32_t next_frame_time_; 266 267 rtc::scoped_ptr<rtc::Thread> process_thread_; 268 269 // Buffer for storing samples received from the webrtc::AudioTransport. 270 char rec_buffer_[kNumberSamples * kNumberBytesPerSample]; 271 // Buffer for samples to send to the webrtc::AudioTransport. 272 char send_buffer_[kNumberSamples * kNumberBytesPerSample]; 273 274 // Counter of frames received that have samples of high enough amplitude to 275 // indicate that the frames are not faked somewhere in the audio pipeline 276 // (e.g. by a jitter buffer). 277 int frames_received_; 278 279 // Protects variables that are accessed from process_thread_ and 280 // the main thread. 281 mutable rtc::CriticalSection crit_; 282 // Protects |audio_callback_| that is accessed from process_thread_ and 283 // the main thread. 284 rtc::CriticalSection crit_callback_; 285 }; 286 287 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ 288
