| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
| #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
| |
| #include "common_types.h" |
| #include "voe_base.h" |
| #include "file_player.h" |
| #include "file_recorder.h" |
| #include "level_indicator.h" |
| #include "module_common_types.h" |
| #include "monitor_module.h" |
| #include "resampler.h" |
| #include "voice_engine_defines.h" |
| |
| |
| namespace webrtc { |
| |
| class AudioProcessing; |
| class ProcessThread; |
| class VoEExternalMedia; |
| class VoEMediaProcess; |
| |
| namespace voe { |
| |
| class ChannelManager; |
| class MixedAudio; |
| class Statistics; |
| |
| class TransmitMixer : public MonitorObserver, |
| public FileCallback |
| |
| { |
| public: |
| static WebRtc_Word32 Create(TransmitMixer*& mixer, |
| const WebRtc_UWord32 instanceId); |
| |
| static void Destroy(TransmitMixer*& mixer); |
| |
| WebRtc_Word32 SetEngineInformation(ProcessThread& processThread, |
| Statistics& engineStatistics, |
| ChannelManager& channelManager); |
| |
| WebRtc_Word32 SetAudioProcessingModule( |
| AudioProcessing* audioProcessingModule); |
| |
| WebRtc_Word32 PrepareDemux(const WebRtc_Word8* audioSamples, |
| const WebRtc_UWord32 nSamples, |
| const WebRtc_UWord8 nChannels, |
| const WebRtc_UWord32 samplesPerSec, |
| const WebRtc_UWord16 totalDelayMS, |
| const WebRtc_Word32 clockDrift, |
| const WebRtc_UWord16 currentMicLevel); |
| |
| |
| WebRtc_Word32 DemuxAndMix(); |
| |
| WebRtc_Word32 EncodeAndSend(); |
| |
| WebRtc_UWord32 CaptureLevel() const; |
| |
| WebRtc_Word32 StopSend(); |
| |
| // VoEDtmf |
| void UpdateMuteMicrophoneTime(const WebRtc_UWord32 lengthMs); |
| |
| // VoEExternalMedia |
| int RegisterExternalMediaProcessing(VoEMediaProcess& proccess_object); |
| |
| int DeRegisterExternalMediaProcessing(); |
| |
| int GetMixingFrequency(); |
| |
| // VoEVolumeControl |
| int SetMute(const bool enable); |
| |
| bool Mute() const; |
| |
| WebRtc_Word8 AudioLevel() const; |
| |
| WebRtc_Word16 AudioLevelFullRange() const; |
| |
| bool IsRecordingCall(); |
| |
| bool IsRecordingMic(); |
| |
| int StartPlayingFileAsMicrophone(const char* fileName, |
| const bool loop, |
| const FileFormats format, |
| const int startPosition, |
| const float volumeScaling, |
| const int stopPosition, |
| const CodecInst* codecInst); |
| |
| int StartPlayingFileAsMicrophone(InStream* stream, |
| const FileFormats format, |
| const int startPosition, |
| const float volumeScaling, |
| const int stopPosition, |
| const CodecInst* codecInst); |
| |
| int StopPlayingFileAsMicrophone(); |
| |
| int IsPlayingFileAsMicrophone() const; |
| |
| int ScaleFileAsMicrophonePlayout(const float scale); |
| |
| int StartRecordingMicrophone(const char* fileName, |
| const CodecInst* codecInst); |
| |
| int StartRecordingMicrophone(OutStream* stream, |
| const CodecInst* codecInst); |
| |
| int StopRecordingMicrophone(); |
| |
| int StartRecordingCall(const char* fileName, const CodecInst* codecInst); |
| |
| int StartRecordingCall(OutStream* stream, const CodecInst* codecInst); |
| |
| int StopRecordingCall(); |
| |
| void SetMixWithMicStatus(bool mix); |
| |
| WebRtc_Word32 RegisterVoiceEngineObserver(VoiceEngineObserver& observer); |
| |
| virtual ~TransmitMixer(); |
| |
| public: // MonitorObserver |
| void OnPeriodicProcess(); |
| |
| |
| public: // FileCallback |
| void PlayNotification(const WebRtc_Word32 id, |
| const WebRtc_UWord32 durationMs); |
| |
| void RecordNotification(const WebRtc_Word32 id, |
| const WebRtc_UWord32 durationMs); |
| |
| void PlayFileEnded(const WebRtc_Word32 id); |
| |
| void RecordFileEnded(const WebRtc_Word32 id); |
| |
| private: |
| TransmitMixer(const WebRtc_UWord32 instanceId); |
| |
| private: |
| WebRtc_Word32 GenerateAudioFrame(const WebRtc_Word16 audioSamples[], |
| const WebRtc_UWord32 nSamples, |
| const WebRtc_UWord8 nChannels, |
| const WebRtc_UWord32 samplesPerSec, |
| const int mixingFrequency); |
| WebRtc_Word32 RecordAudioToFile(const WebRtc_UWord32 mixingFrequency); |
| |
| WebRtc_Word32 MixOrReplaceAudioWithFile( |
| const int mixingFrequency); |
| |
| WebRtc_Word32 APMProcessStream(const WebRtc_UWord16 totalDelayMS, |
| const WebRtc_Word32 clockDrift, |
| const WebRtc_UWord16 currentMicLevel); |
| |
| #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| int TypingDetection(); |
| #endif |
| |
| private: // uses |
| Statistics* _engineStatisticsPtr; |
| ChannelManager* _channelManagerPtr; |
| AudioProcessing* _audioProcessingModulePtr; |
| VoiceEngineObserver* _voiceEngineObserverPtr; |
| ProcessThread* _processThreadPtr; |
| |
| private: // owns |
| MonitorModule _monitorModule; |
| AudioFrame _audioFrame; |
| Resampler _audioResampler; // ADM sample rate -> mixing rate |
| FilePlayer* _filePlayerPtr; |
| FileRecorder* _fileRecorderPtr; |
| FileRecorder* _fileCallRecorderPtr; |
| int _filePlayerId; |
| int _fileRecorderId; |
| int _fileCallRecorderId; |
| bool _filePlaying; |
| bool _fileRecording; |
| bool _fileCallRecording; |
| voe::AudioLevel _audioLevel; |
| // protect file instances and their variables in MixedParticipants() |
| CriticalSectionWrapper& _critSect; |
| CriticalSectionWrapper& _callbackCritSect; |
| |
| #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| WebRtc_Word32 _timeActive; |
| WebRtc_Word32 _penaltyCounter; |
| WebRtc_UWord32 _typingNoiseWarning; |
| #endif |
| WebRtc_UWord32 _saturationWarning; |
| WebRtc_UWord32 _noiseWarning; |
| |
| private: |
| int _instanceId; |
| bool _mixFileWithMicrophone; |
| WebRtc_UWord32 _captureLevel; |
| bool _externalMedia; |
| VoEMediaProcess* _externalMediaCallbackPtr; |
| bool _mute; |
| WebRtc_Word32 _remainingMuteMicTimeMs; |
| int _mixingFrequency; |
| bool _includeAudioLevelIndication; |
| }; |
| |
| #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
| |
| } // namespace voe |
| |
| } // namespace webrtc |