WebRTC VoiceEngine使用简单Demo

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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; iGetCodec(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);
        }