Google收购的GIPS公司的音频处理技术是很牛的，现在开源了，这么好的技术应该拿来用的，这里就简单的介绍一下怎样使用VoiceEngine，欢迎大家拍砖指导。
WebRTC相关的VideoEngine和VoiceEngine的API详细说明文档：http://www.webrtc.org/system/app/pages/subPages?path=/reference/webrtc-internals
WebRTC的VideoEngine和VoiceEngine源码在：http://code.google.com/p/webrtc/source/browse/#svn%2Fbranches
iSAC(Internet Speech Audio Codec 互联网语音音频编解码器)相关编码的参数：
取样频率16kHz、24kHz或32kHz，自适应速率为10kbit/s至52kbit/s，自适应包大小为30至60ms，由于算法复杂度和自适应可变速率，相比于G.722.2每帧延时3ms左右。
关于如何配置iSAC的参数，可以参看这里文章的介绍。
当前的版本VideoEngine是：ViE3.1.0
VoiceEngine是：VoE4.1.0
- WebRTC音频引擎版本VoE4.1.0
- ***/
- //初始化VoiceEngine以及Sub_APIS
- VoiceEngine* _voiceEngine;
- VoEBase* _veBase;
- VoENetwork* _veNetwork;
- VoECodec* _veCodec;
- VoERTP_RTCP* _veRTCP;
- _voiceEngine = VoiceEngine::Create();
- _veBase = VoEBase::GetInterface(_voiceEngine);
- _veNetwork = VoENetwork::GetInterface(_voiceEngine);
- _veCodec = VoECodec::GetInterface(_voiceEngine);
- _veRTCP = VoERTP_RTCP::GetInterface(_voiceEngine);
- _vieBase->SetVoiceEngine(_voiceEngine);
- //编码器选择，编码的配置参数可以配置CodecInst：
- // Each codec supported can be described by this structure.
- /********
- struct CodecInst
- {
- int pltype;
- char plname[32];
- int plfreq;
- int pacsize;
- int channels;
- int rate;
- };********/
- CodecInst voiceCodec;
- // define iSAC codec parameters
- strcpy(voiceCodec.plname, "ISAC");
- voiceCodec.plfreq = 16000; // iSAC宽带模式
- voiceCodec.pltype = 103; // 默认动态负载类型
- voiceCodec.pacsize = 480; // 480kbps，即使用30ms的packet size
- voiceCodec.channels = 1; // 单声道
- voiceCodec.rate = -1; // 信道自适应模式，单位bps
- int numOfVeCodecs = _veCodec->NumOfCodecs();
- for(int i=0; i<numOfVeCodecs;++i)
- {
- if(_veCodec->GetCodec(i,voiceCodec)!=-1)
- {
- if(strncmp(voiceCodec.plname,"ISAC",4)==0)
- break;
- }
- }
- //网络传输应用
- _audioChannel = _veBase->CreateChannel();
- _veRTCP->SetRTCPStatus(_audioChannel, true);
- _veCodec->SetSendCodec(_audioChannel, voiceCodec);
- _veBase->StartPlayout(_audioChannel);
- //音频和视频绑定
- _vieBase->ConnectAudioChannel(_channelId,_audioChannel);
- //网络发送接收配置，远程端口：remotePort 目的IP：IP
- _veBase->SetSendDestination(_audioChannel, remotePort,IP);
- //本地接收
- int res=_veBase->SetLocalReceiver(_audioChannel,localPort);
- _veBase->StartSend(_audioChannel);
- _veBase->StartReceive(_audioChannel);
- _veBase->StopReceive(_audioChannel);
- _veBase->StopSend(_audioChannel);
- //结束，释放资源
- if (_voiceEngine)
- {
- _veBase->DeleteChannel(_audioChannel);
- _veBase->Release();
- _veNetwork->Release();
- _veCodec->Release();
- _veRTCP->Release();
- VoiceEngine::Delete(_voiceEngine);
- }