WebRTC VoiceEngine使用简单Demo

http://blog.csdn.net/temotemo/article/details/7449525

WebRTC VoiceEngine使用简单Demo

WebRTC技术交流群：234795279

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

 <pre name="code" class="cpp">/****
     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);
         }

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