andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 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_VOICE_ENGINE_TRANSMIT_MIXER_H |
| 12 | #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
| 13 | |
| 14 | #include "common_types.h" |
| 15 | #include "voe_base.h" |
| 16 | #include "file_player.h" |
| 17 | #include "file_recorder.h" |
| 18 | #include "level_indicator.h" |
| 19 | #include "module_common_types.h" |
| 20 | #include "monitor_module.h" |
| 21 | #include "resampler.h" |
| 22 | #include "voice_engine_defines.h" |
| 23 | |
| 24 | |
| 25 | namespace webrtc { |
| 26 | |
| 27 | class AudioProcessing; |
| 28 | class ProcessThread; |
| 29 | class VoEExternalMedia; |
| 30 | class VoEMediaProcess; |
| 31 | |
| 32 | namespace voe { |
| 33 | |
| 34 | class ChannelManager; |
| 35 | class MixedAudio; |
| 36 | class Statistics; |
| 37 | |
| 38 | class TransmitMixer : public MonitorObserver, |
| 39 | public FileCallback |
| 40 | |
| 41 | { |
| 42 | public: |
| 43 | static WebRtc_Word32 Create(TransmitMixer*& mixer, |
| 44 | const WebRtc_UWord32 instanceId); |
| 45 | |
| 46 | static void Destroy(TransmitMixer*& mixer); |
| 47 | |
| 48 | WebRtc_Word32 SetEngineInformation(ProcessThread& processThread, |
| 49 | Statistics& engineStatistics, |
| 50 | ChannelManager& channelManager); |
| 51 | |
| 52 | WebRtc_Word32 SetAudioProcessingModule( |
| 53 | AudioProcessing* audioProcessingModule); |
| 54 | |
| 55 | WebRtc_Word32 PrepareDemux(const void* audioSamples, |
| 56 | const WebRtc_UWord32 nSamples, |
| 57 | const WebRtc_UWord8 nChannels, |
| 58 | const WebRtc_UWord32 samplesPerSec, |
| 59 | const WebRtc_UWord16 totalDelayMS, |
| 60 | const WebRtc_Word32 clockDrift, |
| 61 | const WebRtc_UWord16 currentMicLevel); |
| 62 | |
| 63 | |
| 64 | WebRtc_Word32 DemuxAndMix(); |
| 65 | |
| 66 | WebRtc_Word32 EncodeAndSend(); |
| 67 | |
| 68 | WebRtc_UWord32 CaptureLevel() const; |
| 69 | |
| 70 | WebRtc_Word32 StopSend(); |
| 71 | |
| 72 | // VoEDtmf |
| 73 | void UpdateMuteMicrophoneTime(const WebRtc_UWord32 lengthMs); |
| 74 | |
| 75 | // VoEExternalMedia |
| 76 | int RegisterExternalMediaProcessing(VoEMediaProcess* object, |
| 77 | ProcessingTypes type); |
| 78 | int DeRegisterExternalMediaProcessing(ProcessingTypes type); |
| 79 | |
| 80 | int GetMixingFrequency(); |
| 81 | |
| 82 | // VoEVolumeControl |
| 83 | int SetMute(const bool enable); |
| 84 | |
| 85 | bool Mute() const; |
| 86 | |
| 87 | WebRtc_Word8 AudioLevel() const; |
| 88 | |
| 89 | WebRtc_Word16 AudioLevelFullRange() const; |
| 90 | |
| 91 | bool IsRecordingCall(); |
| 92 | |
| 93 | bool IsRecordingMic(); |
| 94 | |
| 95 | int StartPlayingFileAsMicrophone(const char* fileName, |
| 96 | const bool loop, |
| 97 | const FileFormats format, |
| 98 | const int startPosition, |
| 99 | const float volumeScaling, |
| 100 | const int stopPosition, |
| 101 | const CodecInst* codecInst); |
| 102 | |
| 103 | int StartPlayingFileAsMicrophone(InStream* stream, |
| 104 | const FileFormats format, |
| 105 | const int startPosition, |
| 106 | const float volumeScaling, |
| 107 | const int stopPosition, |
| 108 | const CodecInst* codecInst); |
| 109 | |
| 110 | int StopPlayingFileAsMicrophone(); |
| 111 | |
| 112 | int IsPlayingFileAsMicrophone() const; |
| 113 | |
| 114 | int ScaleFileAsMicrophonePlayout(const float scale); |
| 115 | |
| 116 | int StartRecordingMicrophone(const char* fileName, |
| 117 | const CodecInst* codecInst); |
| 118 | |
| 119 | int StartRecordingMicrophone(OutStream* stream, |
| 120 | const CodecInst* codecInst); |
| 121 | |
| 122 | int StopRecordingMicrophone(); |
| 123 | |
| 124 | int StartRecordingCall(const char* fileName, const CodecInst* codecInst); |
| 125 | |
| 126 | int StartRecordingCall(OutStream* stream, const CodecInst* codecInst); |
| 127 | |
| 128 | int StopRecordingCall(); |
| 129 | |
| 130 | void SetMixWithMicStatus(bool mix); |
| 131 | |
| 132 | WebRtc_Word32 RegisterVoiceEngineObserver(VoiceEngineObserver& observer); |
| 133 | |
| 134 | virtual ~TransmitMixer(); |
| 135 | |
| 136 | // MonitorObserver |
| 137 | void OnPeriodicProcess(); |
| 138 | |
| 139 | |
| 140 | // FileCallback |
| 141 | void PlayNotification(const WebRtc_Word32 id, |
| 142 | const WebRtc_UWord32 durationMs); |
| 143 | |
| 144 | void RecordNotification(const WebRtc_Word32 id, |
| 145 | const WebRtc_UWord32 durationMs); |
| 146 | |
| 147 | void PlayFileEnded(const WebRtc_Word32 id); |
| 148 | |
| 149 | void RecordFileEnded(const WebRtc_Word32 id); |
| 150 | |
| 151 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 152 | // Typing detection |
| 153 | int TimeSinceLastTyping(int &seconds); |
| 154 | int SetTypingDetectionParameters(int timeWindow, |
| 155 | int costPerTyping, |
| 156 | int reportingThreshold, |
| 157 | int penaltyDecay, |
| 158 | int typeEventDelay); |
| 159 | #endif |
| 160 | |
| 161 | void EnableStereoChannelSwapping(bool enable); |
| 162 | bool IsStereoChannelSwappingEnabled(); |
| 163 | |
| 164 | private: |
| 165 | TransmitMixer(const WebRtc_UWord32 instanceId); |
| 166 | |
| 167 | void CheckForSendCodecChanges(); |
| 168 | |
| 169 | int GenerateAudioFrame(const int16_t audioSamples[], |
| 170 | int nSamples, |
| 171 | int nChannels, |
| 172 | int samplesPerSec); |
| 173 | WebRtc_Word32 RecordAudioToFile(const WebRtc_UWord32 mixingFrequency); |
| 174 | |
| 175 | WebRtc_Word32 MixOrReplaceAudioWithFile( |
| 176 | const int mixingFrequency); |
| 177 | |
| 178 | WebRtc_Word32 APMProcessStream(const WebRtc_UWord16 totalDelayMS, |
| 179 | const WebRtc_Word32 clockDrift, |
| 180 | const WebRtc_UWord16 currentMicLevel); |
| 181 | |
| 182 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 183 | int TypingDetection(); |
| 184 | #endif |
| 185 | |
| 186 | // uses |
| 187 | Statistics* _engineStatisticsPtr; |
| 188 | ChannelManager* _channelManagerPtr; |
| 189 | AudioProcessing* _audioProcessingModulePtr; |
| 190 | VoiceEngineObserver* _voiceEngineObserverPtr; |
| 191 | ProcessThread* _processThreadPtr; |
| 192 | |
| 193 | // owns |
| 194 | MonitorModule _monitorModule; |
| 195 | AudioFrame _audioFrame; |
| 196 | Resampler _audioResampler; // ADM sample rate -> mixing rate |
| 197 | FilePlayer* _filePlayerPtr; |
| 198 | FileRecorder* _fileRecorderPtr; |
| 199 | FileRecorder* _fileCallRecorderPtr; |
| 200 | int _filePlayerId; |
| 201 | int _fileRecorderId; |
| 202 | int _fileCallRecorderId; |
| 203 | bool _filePlaying; |
| 204 | bool _fileRecording; |
| 205 | bool _fileCallRecording; |
| 206 | voe::AudioLevel _audioLevel; |
| 207 | // protect file instances and their variables in MixedParticipants() |
| 208 | CriticalSectionWrapper& _critSect; |
| 209 | CriticalSectionWrapper& _callbackCritSect; |
| 210 | |
| 211 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 212 | WebRtc_Word32 _timeActive; |
| 213 | WebRtc_Word32 _timeSinceLastTyping; |
| 214 | WebRtc_Word32 _penaltyCounter; |
| 215 | WebRtc_UWord32 _typingNoiseWarning; |
| 216 | |
| 217 | // Tunable treshold values |
| 218 | int _timeWindow; // nr of10ms slots accepted to count as a hit. |
| 219 | int _costPerTyping; // Penalty added for a typing + activity coincide. |
| 220 | int _reportingThreshold; // Threshold for _penaltyCounter. |
| 221 | int _penaltyDecay; // How much we reduce _penaltyCounter every 10 ms. |
| 222 | int _typeEventDelay; // How old typing events we allow |
| 223 | |
| 224 | #endif |
| 225 | WebRtc_UWord32 _saturationWarning; |
| 226 | WebRtc_UWord32 _noiseWarning; |
| 227 | |
| 228 | int _instanceId; |
| 229 | bool _mixFileWithMicrophone; |
| 230 | WebRtc_UWord32 _captureLevel; |
| 231 | VoEMediaProcess* external_postproc_ptr_; |
| 232 | VoEMediaProcess* external_preproc_ptr_; |
| 233 | bool _mute; |
| 234 | WebRtc_Word32 _remainingMuteMicTimeMs; |
| 235 | int _mixingFrequency; |
| 236 | bool stereo_codec_; |
| 237 | bool swap_stereo_channels_; |
| 238 | }; |
| 239 | |
| 240 | #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
| 241 | |
| 242 | } // namespace voe |
| 243 | |
| 244 | } // namespace webrtc |