webrtc中音频3A处理开关配置1 音频引擎初始化的时对3A处理进行设置WebRtcVoiceEngine::Init media/engine/webrtc_voice_engine.h WebRtcVoiceEngine::ApplyOptions media/engine/webrtc_voice_engine.h modules/audio_processing/audio_processing_impl.h AudioProcessingImpl::ApplyConfig2 创建audio source时设置3A参数cricket::AudioOptions options; options.highpass_filter true; options.echo_cancellation true; options.auto_gain_control true; options.noise_suppression true; options.combined_audio_video_bwe true; options.residual_echo_detector true;//残余回音消除 rtc::scoped_refptrwebrtc::AudioSourceInterface source g_factory-CreateAudioSource(options); rtc::scoped_refptrwebrtc::AudioTrackInterface trackPtr g_factory-CreateAudioTrack(label, source); PeerConnection::AddTransceiver pc/peer_connection.h 关键参数 cricket::MediaType media_type, rtc::scoped_refptrMediaStreamTrackInterface track PeerConnection::CreateSender pc/peer_connection.h 关键参数 rtc::scoped_refptrMediaStreamTrackInterface track RtpSenderBase::SetTrack(MediaStreamTrackInterface* track) pc/rtp_sender.h AudioRtpSender::SetSend pc/rtp_sender.h 备注 1获取track中source的配置3A处理相关选项 2 voice_media_channel()-SetAudioSend(ssrc_, track_enabled, options, sink_adapter_.get()); WebRtcVoiceMediaChannel::SetAudioSend media/engine/webrtc_voice_engine.h WebRtcVoiceMediaChannel::SetOptions media/engine/webrtc_voice_engine.h WebRtcVoiceEngine::ApplyOptions media/engine/webrtc_voice_engine.h modules/audio_processing/audio_processing_impl.h AudioProcessingImpl::ApplyConfig