webrtc中音频3A处理开关配置

XHW___0012026-02-05 10:48

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

c++ 复制代码

WebRtcVoiceEngine::Init
media/engine/webrtc_voice_engine.h

WebRtcVoiceEngine::ApplyOptions
media/engine/webrtc_voice_engine.h

modules/audio_processing/audio_processing_impl.h
AudioProcessingImpl::ApplyConfig

2 创建audio source时设置3A参数

c++ 复制代码

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_refptr<webrtc::AudioSourceInterface> source = g_factory->CreateAudioSource(options);
rtc::scoped_refptr<webrtc::AudioTrackInterface> trackPtr = g_factory->CreateAudioTrack(label, source);

PeerConnection::AddTransceiver
pc/peer_connection.h
关键参数：
cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track

PeerConnection::CreateSender
pc/peer_connection.h
关键参数：
rtc::scoped_refptr<MediaStreamTrackInterface> 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