WebRTC VoiceEngine使用簡單Demo

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

WebRTC技術交流羣：234795279

Google收購的GIPS公司的音頻處理技術是很牛的，現在開源了，這麼好的技術應該拿來用的，這裏就簡單的介紹一下怎樣使用VoiceEngine，歡迎大家拍磚指導。

WebRTC相關的VideoEngine和VoiceEngine的API詳細說明文檔：http://www.webrtc.org/system/app/pages/subPages?path=/reference/webrtc-internals

WebRTC的VideoEngine和VoiceEngine源碼在：http://code.google.com/p/webrtc/source/browse/#svn%2Fbranches

iSAC(Internet Speech Audio Codec 互聯網語音音頻編解碼器)相關編碼的參數：

取樣頻率16kHz、24kHz或32kHz，自適應速率爲10kbit/s至52kbit/s，自適應包大小爲30至60ms，由於算法複雜度和自適應可變速率，相比於G.722.2每幀延時3ms左右。

關於如何配置iSAC的參數，可以參看這裏文章的介紹。

當前的版本VideoEngine是：ViE3.1.0

        VoiceEngine是：VoE4.1.0

<pre name="code" class="cpp">/****
    WebRTC音頻引擎版本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);

//編碼器選擇，編碼的配置參數可以配置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寬帶模式
voiceCodec.pltype   = 103;      // 默認動態負載類型
voiceCodec.pacsize  = 480;      // 480kbps，即使用30ms的packet size
voiceCodec.channels     = 1;        // 單聲道
voiceCodec.rate     = -1;       // 信道自適應模式，單位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;
        }
    }

    //網絡傳輸應用
    _audioChannel = _veBase->CreateChannel();
    _veRTCP->SetRTCPStatus(_audioChannel, true);
    _veCodec->SetSendCodec(_audioChannel, voiceCodec);
    _veBase->StartPlayout(_audioChannel);

//音頻和視頻綁定
_vieBase->ConnectAudioChannel(_channelId,_audioChannel);

//網絡發送接收配置，遠程端口：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);

//結束，釋放資源
    if (_voiceEngine)
    {
        _veBase->DeleteChannel(_audioChannel);
        _veBase->Release();
        _veNetwork->Release();
        _veCodec->Release();
        _veRTCP->Release();

         VoiceEngine::Delete(_voiceEngine);
        }

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