1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H 12 #define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H 13 14 #include "webrtc/modules/audio_device/audio_device_utility.h" 15 16 #include <list> 17 #include <string> 18 19 #include "webrtc/common_audio/resampler/include/resampler.h" 20 #include "webrtc/modules/audio_device/include/audio_device.h" 21 #include "webrtc/modules/audio_device/test/audio_device_test_defines.h" 22 #include "webrtc/system_wrappers/interface/file_wrapper.h" 23 #include "webrtc/typedefs.h" 24 25 #if defined(WEBRTC_IOS) || defined(ANDROID) 26 #define USE_SLEEP_AS_PAUSE 27 #else 28 //#define USE_SLEEP_AS_PAUSE 29 #endif 30 31 // Sets the default pause time if using sleep as pause 32 #define DEFAULT_PAUSE_TIME 5000 33 34 #if defined(USE_SLEEP_AS_PAUSE) 35 #define PAUSE(a) SleepMs(a); 36 #else 37 #define PAUSE(a) AudioDeviceUtility::WaitForKey(); 38 #endif 39 40 #define ADM_AUDIO_LAYER AudioDeviceModule::kPlatformDefaultAudio 41 //#define ADM_AUDIO_LAYER AudioDeviceModule::kLinuxPulseAudio 42 43 enum TestType 44 { 45 TTInvalid = -1, 46 TTAll = 0, 47 TTAudioLayerSelection = 1, 48 TTDeviceEnumeration = 2, 49 TTDeviceSelection = 3, 50 TTAudioTransport = 4, 51 TTSpeakerVolume = 5, 52 TTMicrophoneVolume = 6, 53 TTSpeakerMute = 7, 54 TTMicrophoneMute = 8, 55 TTMicrophoneBoost = 9, 56 TTMicrophoneAGC = 10, 57 TTLoopback = 11, 58 TTDeviceRemoval = 13, 59 TTMobileAPI = 14, 60 TTTest = 66, 61 }; 62 63 struct AudioPacket 64 { 65 uint8_t dataBuffer[4 * 960]; 66 uint16_t nSamples; 67 uint16_t nBytesPerSample; 68 uint8_t nChannels; 69 uint32_t samplesPerSec; 70 }; 71 72 class ProcessThread; 73 74 namespace webrtc 75 { 76 77 class AudioDeviceModule; 78 class AudioEventObserver; 79 class AudioTransport; 80 81 // ---------------------------------------------------------------------------- 82 // AudioEventObserver 83 // ---------------------------------------------------------------------------- 84 85 class AudioEventObserver: public AudioDeviceObserver 86 { 87 public: 88 virtual void OnErrorIsReported(const ErrorCode error); 89 virtual void OnWarningIsReported(const WarningCode warning); 90 AudioEventObserver(AudioDeviceModule* audioDevice); 91 ~AudioEventObserver(); 92 public: 93 ErrorCode _error; 94 WarningCode _warning; 95 }; 96 97 // ---------------------------------------------------------------------------- 98 // AudioTransport 99 // ---------------------------------------------------------------------------- 100 101 class AudioTransportImpl: public AudioTransport 102 { 103 public: 104 virtual int32_t 105 RecordedDataIsAvailable(const void* audioSamples, 106 const uint32_t nSamples, 107 const uint8_t nBytesPerSample, 108 const uint8_t nChannels, 109 const uint32_t samplesPerSec, 110 const uint32_t totalDelayMS, 111 const int32_t clockDrift, 112 const uint32_t currentMicLevel, 113 const bool keyPressed, 114 uint32_t& newMicLevel); 115 116 virtual int32_t NeedMorePlayData(const uint32_t nSamples, 117 const uint8_t nBytesPerSample, 118 const uint8_t nChannels, 119 const uint32_t samplesPerSec, 120 void* audioSamples, 121 uint32_t& nSamplesOut, 122 int64_t* elapsed_time_ms, 123 int64_t* ntp_time_ms); 124 125 virtual int OnDataAvailable(const int voe_channels[], 126 int number_of_voe_channels, 127 const int16_t* audio_data, 128 int sample_rate, 129 int number_of_channels, 130 int number_of_frames, 131 int audio_delay_milliseconds, 132 int current_volume, 133 bool key_pressed, 134 bool need_audio_processing); 135 136 virtual void PushCaptureData(int voe_channel, const void* audio_data, 137 int bits_per_sample, int sample_rate, 138 int number_of_channels, 139 int number_of_frames); 140 141 virtual void PullRenderData(int bits_per_sample, int sample_rate, 142 int number_of_channels, int number_of_frames, 143 void* audio_data, 144 int64_t* elapsed_time_ms, 145 int64_t* ntp_time_ms); 146 147 AudioTransportImpl(AudioDeviceModule* audioDevice); 148 ~AudioTransportImpl(); 149 150 public: 151 int32_t SetFilePlayout(bool enable, const char* fileName = NULL); 152 void SetFullDuplex(bool enable); SetSpeakerVolume(bool enable)153 void SetSpeakerVolume(bool enable) 154 { 155 _speakerVolume = enable; 156 } 157 ; SetSpeakerMute(bool enable)158 void SetSpeakerMute(bool enable) 159 { 160 _speakerMute = enable; 161 } 162 ; SetMicrophoneMute(bool enable)163 void SetMicrophoneMute(bool enable) 164 { 165 _microphoneMute = enable; 166 } 167 ; SetMicrophoneVolume(bool enable)168 void SetMicrophoneVolume(bool enable) 169 { 170 _microphoneVolume = enable; 171 } 172 ; SetMicrophoneBoost(bool enable)173 void SetMicrophoneBoost(bool enable) 174 { 175 _microphoneBoost = enable; 176 } 177 ; SetLoopbackMeasurements(bool enable)178 void SetLoopbackMeasurements(bool enable) 179 { 180 _loopBackMeasurements = enable; 181 } 182 ; SetMicrophoneAGC(bool enable)183 void SetMicrophoneAGC(bool enable) 184 { 185 _microphoneAGC = enable; 186 } 187 ; 188 189 private: 190 typedef std::list<AudioPacket*> AudioPacketList; 191 AudioDeviceModule* _audioDevice; 192 193 bool _playFromFile; 194 bool _fullDuplex; 195 bool _speakerVolume; 196 bool _speakerMute; 197 bool _microphoneVolume; 198 bool _microphoneMute; 199 bool _microphoneBoost; 200 bool _microphoneAGC; 201 bool _loopBackMeasurements; 202 203 FileWrapper& _playFile; 204 205 uint32_t _recCount; 206 uint32_t _playCount; 207 AudioPacketList _audioList; 208 209 Resampler _resampler; 210 }; 211 212 // ---------------------------------------------------------------------------- 213 // FuncTestManager 214 // ---------------------------------------------------------------------------- 215 216 class FuncTestManager 217 { 218 public: 219 FuncTestManager(); 220 ~FuncTestManager(); 221 int32_t Init(); 222 int32_t Close(); 223 int32_t DoTest(const TestType testType); 224 private: 225 int32_t TestAudioLayerSelection(); 226 int32_t TestDeviceEnumeration(); 227 int32_t TestDeviceSelection(); 228 int32_t TestAudioTransport(); 229 int32_t TestSpeakerVolume(); 230 int32_t TestMicrophoneVolume(); 231 int32_t TestSpeakerMute(); 232 int32_t TestMicrophoneMute(); 233 int32_t TestMicrophoneBoost(); 234 int32_t TestLoopback(); 235 int32_t TestDeviceRemoval(); 236 int32_t TestExtra(); 237 int32_t TestMicrophoneAGC(); 238 int32_t SelectPlayoutDevice(); 239 int32_t SelectRecordingDevice(); 240 int32_t TestAdvancedMBAPI(); 241 private: 242 // Paths to where the resource files to be used for this test are located. 243 std::string _playoutFile48; 244 std::string _playoutFile44; 245 std::string _playoutFile16; 246 std::string _playoutFile8; 247 248 ProcessThread* _processThread; 249 AudioDeviceModule* _audioDevice; 250 AudioEventObserver* _audioEventObserver; 251 AudioTransportImpl* _audioTransport; 252 }; 253 254 } // namespace webrtc 255 256 #endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H 257