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_ref_ptr.h" 44#include "webrtc/common_types.h" 45#include "webrtc/modules/audio_device/include/audio_device.h" 46 47namespace rtc { 48 49class Thread; 50 51} // namespace rtc 52 53class FakeAudioCaptureModule 54 : public webrtc::AudioDeviceModule, 55 public rtc::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 rtc::scoped_refptr<FakeAudioCaptureModule> Create( 68 rtc::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 TimeUntilNextProcess() OVERRIDE; 80 virtual int32_t Process() OVERRIDE; 81 virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE; 82 83 virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const OVERRIDE; 84 85 virtual ErrorCode LastError() const OVERRIDE; 86 virtual int32_t RegisterEventObserver( 87 webrtc::AudioDeviceObserver* event_callback) OVERRIDE; 88 89 // Note: Calling this method from a callback may result in deadlock. 90 virtual int32_t RegisterAudioCallback( 91 webrtc::AudioTransport* audio_callback) OVERRIDE; 92 93 virtual int32_t Init() OVERRIDE; 94 virtual int32_t Terminate() OVERRIDE; 95 virtual bool Initialized() const OVERRIDE; 96 97 virtual int16_t PlayoutDevices() OVERRIDE; 98 virtual int16_t RecordingDevices() OVERRIDE; 99 virtual int32_t PlayoutDeviceName( 100 uint16_t index, 101 char name[webrtc::kAdmMaxDeviceNameSize], 102 char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE; 103 virtual int32_t RecordingDeviceName( 104 uint16_t index, 105 char name[webrtc::kAdmMaxDeviceNameSize], 106 char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE; 107 108 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; 112 113 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; 119 120 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; 126 127 virtual int32_t SetAGC(bool enable) OVERRIDE; 128 virtual bool AGC() const OVERRIDE; 129 130 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; 134 135 virtual int32_t InitSpeaker() OVERRIDE; 136 virtual bool SpeakerIsInitialized() const OVERRIDE; 137 virtual int32_t InitMicrophone() OVERRIDE; 138 virtual bool MicrophoneIsInitialized() const OVERRIDE; 139 140 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; 146 147 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; 151 152 virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const OVERRIDE; 153 virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const OVERRIDE; 154 155 virtual int32_t SpeakerMuteIsAvailable(bool* available) OVERRIDE; 156 virtual int32_t SetSpeakerMute(bool enable) OVERRIDE; 157 virtual int32_t SpeakerMute(bool* enabled) const OVERRIDE; 158 159 virtual int32_t MicrophoneMuteIsAvailable(bool* available) OVERRIDE; 160 virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE; 161 virtual int32_t MicrophoneMute(bool* enabled) const OVERRIDE; 162 163 virtual int32_t MicrophoneBoostIsAvailable(bool* available) OVERRIDE; 164 virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE; 165 virtual int32_t MicrophoneBoost(bool* enabled) const OVERRIDE; 166 167 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; 175 176 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; 182 183 virtual int32_t CPULoad(uint16_t* load) const OVERRIDE; 184 185 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; 191 192 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; 197 198 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. 202 203 // The following function is inherited from rtc::MessageHandler. 204 virtual void OnMessage(rtc::Message* msg) OVERRIDE; 205 206 protected: 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(); 216 217 private: 218 // Initializes the state of the FakeAudioCaptureModule. This API is called on 219 // creation by the Create() API. 220 bool Initialize(); 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 226 // equal to |value|. 227 bool CheckRecBuffer(int value); 228 229 // Returns true/false depending on if recording or playback has been 230 // enabled/started. 231 bool ShouldStartProcessing(); 232 233 // Starts or stops the pushing and pulling of audio frames. 234 void UpdateProcessing(bool start); 235 236 // 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. 244 void SendFrameP(); 245 // Stops the periodic calling of ProcessFrame() in a thread safe way. 246 void StopProcessP(); 247 248 // The time in milliseconds when Process() was last called or 0 if no call 249 // has been made. 250 uint32 last_process_time_ms_; 251 252 // Callback for playout and recording. 253 webrtc::AudioTransport* audio_callback_; 254 255 bool recording_; // True when audio is being pushed from the instance. 256 bool playing_; // True when audio is being pulled by the instance. 257 258 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. 260 261 // 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_; 265 266 // 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. 269 bool started_; 270 uint32 next_frame_time_; 271 272 // User provided thread context. 273 rtc::Thread* process_thread_; 274 275 // 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]; 279 280 // 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_; 284 285 // Protects variables that are accessed from process_thread_ and 286 // the main thread. 287 mutable rtc::CriticalSection crit_; 288 // Protects |audio_callback_| that is accessed from process_thread_ and 289 // the main thread. 290 rtc::CriticalSection crit_callback_; 291}; 292 293#endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ 294