Android IOS WebRTC 音视频开发总结（八）-- ios上移植webRTCDemo

这篇文章主要介绍ios webrtcdemo的实现及相关注意事项，转载请说明出处（博客园RTC.Blacker）

前面很多人问webrtc android下有webrtcdemo，ios上怎么找不到，放在哪里呢?

答案:webrtcdemo在ios上没有实现，如果要实现也很简单，既然安卓都有了，依葫芦画瓢即可移植到ios上，不过可能要求您熟悉android语法，这里给出ios上的参考代码:

 -(BOOL)initWebrtcObjects
 {
　　　　　//转载请说明出处:　RTC_Blacker http://www.cnblogs.com/lingyunhu
     if ((voE = webrtc::VoiceEngine::Create()) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((voeBase = webrtc::VoEBase::GetInterface(voE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((voeCodec = webrtc::VoECodec::GetInterface(voE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((voeFile=webrtc::VoEFile::GetInterface(voE))==NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
     }
     if ((voeHardware = webrtc::VoEHardware::GetInterface(voE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((voeNetwork = webrtc::VoENetwork::GetInterface(voE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((voeAudioProccessing = webrtc::VoEAudioProcessing::GetInterface(voE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((voeRtpRtcp = webrtc::VoERTP_RTCP::GetInterface(voE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     if(voeBase->Init()!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
     }

     if ((viE = webrtc::VideoEngine::Create()) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((vieBase = webrtc::ViEBase::GetInterface(viE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((vieCapture = webrtc::ViECapture::GetInterface(viE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((vieRender = webrtc::ViERender::GetInterface(viE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((vieCodec = webrtc::ViECodec::GetInterface(viE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((vieNetwork = webrtc::ViENetwork::GetInterface(viE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if ((vieRtpRtcp = webrtc::ViERTP_RTCP::GetInterface(viE)) == NULL) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     if (vieBase->Init() != ) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     [self initAudioCodec];
     [self initVideoCodec];

     captureID = ;
     videoChannel = -;

     return TRUE;
 }

 -(void)initAudioCodec
 {
     memset(&voeCodecInst, , sizeof(webrtc::CodecInst));

     if (voeCodec != NULL) {
         for (int index=; index < voeCodec->NumOfCodecs(); index++) {
             webrtc::CodecInst ci;
             voeCodec->GetCodec(index, ci);
             if (strncmp(ci.plname, "ISAC", ) == ) {
                 memcpy(&voeCodecInst, &ci, sizeof(webrtc::CodecInst));
                 break;
             }
         }
         //voeCodecInst.channels = 1;
         //voeCodecInst.rate = -1;
     }
 }

 -(BOOL)start
 {
    f ((audioChannel = voeBase->CreateChannel())!=) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if (vieBase->CreateChannel(videoChannel) != ) {
         DebugLog(@"AVErr: %d %s at line %d", vieBase->LastError(),__FUNCTION__, __LINE__);
         return FALSE;
     }
     DebugLog(@"AVInfo: CreateChannel success! %d, %d",videoChannel,audioChannel);

     //vieCodec->SetReceiveCodec(videoChannel,videoCodec);

     if(voeAudioProccessing->SetAecmMode()!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     voeAudioProccessing->SetAgcStatus(TRUE, webrtc::kAgcDefault);
     voeAudioProccessing->SetNsStatus(TRUE, webrtc::kNsHighSuppression);
     _voice_capture_device_index = -;
     voeHardware->SetRecordingDevice(_voice_capture_device_index);
     voeHardware->SetPlayoutDevice(_voice_playback_device_index);
     if(voeHardware->SetLoudspeakerStatus(true)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
     }
     voeCodec->SetSendCodec(audioChannel, voeCodecInst);

     RtpRtcpStreamStruct streamStruct=[self createRtpStreamStruct];
     voeChannelTransport=new webrtc::test::VoiceChannelTransport(voeNetwork, audioChannel);
     voeChannelTransport->SetLocalReceiver2(localARtpPort.rtp,streamStruct );
     voeChannelTransport->SetSendDestination2([remoteIPAddress UTF8String], remoteARtpPort.rtp, remoteARtpPort.rtcp);

     if(vieCodec->SetSendCodec(videoChannel, videoCodec) != )
     {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     vieRtpRtcp->SetNACKStatus(videoChannel, TRUE);
     vieRtpRtcp->SetRTCPStatus(videoChannel, webrtc::kRtcpNonCompound_RFC5506);
     vieRtpRtcp->SetKeyFrameRequestMethod(videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);

     vieBase->SetVoiceEngine(voE);
     if (vieBase->ConnectAudioChannel(videoChannel, audioChannel)) {
         DebugLog(@"AVErr:%s at line %d",__FUNCTION__,__LINE__);
         return FALSE;
     }

     if (deviceUniqueID == nil) {
         DebugLog(@"AVInfo NumberOfCaptureDevices is %d", vieCapture->NumberOfCaptureDevices());
         int list_count=vieCapture->NumberOfCaptureDevices();
         if ( list_count> ) {
             int list_number=;
             if (list_count>) {
                 list_number=;//[[AVShareData instance] isUseFrontCamera]?0:1;
             }
             char device_name[KMaxDeviceNameLength];
             char unique_id[KMaxUniqueIdLength];
             memset(unique_id, , KMaxUniqueIdLength);
             vieCapture->GetCaptureDevice(list_number, device_name, KMaxDeviceNameLength, unique_id, KMaxUniqueIdLength);
             deviceUniqueID = [NSString stringWithFormat:@"%s", unique_id];
         }
     }
     DebugLog(@"AVInfo deviceUniqueID is %@", deviceUniqueID);

     if ((vieCapture->AllocateCaptureDevice([deviceUniqueID UTF8String], deviceUniqueID.length, captureID)) != ) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     DebugLog(@"AVInfo captureID is %d", captureID);

     if (vieCapture->ConnectCaptureDevice(captureID, videoChannel) != ) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     webrtc::CaptureCapability captureCapability;
     captureCapability.width=;
     captureCapability.height=;
     captureCapability.codecType=webrtc::kVideoCodecVP8;
     captureCapability.maxFPS=DEFAULT_VIDEO_CODEC_MAX_FRAMERATE;
     //vieCapture->SetRotateCapturedFrames(captureID, <#const webrtc::RotateCapturedFrame rotation#>)
     if (vieCapture->StartCapture(captureID,captureCapability) != ) {
         //if (vieCapture->StartCapture(captureID) != 0) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if((vieRender->AddRenderer(captureID, [self localRenderView], , 0.0, 0.0, 1.0, 1.0)) != ){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     /*
     if((vieRender->AddRenderer(captureID, [self localRenderView2], 0, 0.0, 0.0, 1.0, 1.0)) != 0){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     */

     if (vieRender->StartRender(captureID) != ) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieRender->AddRenderer(videoChannel, [self remoteRenderView], , 0.0f, 0.0f, 1.0f, 1.0f)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieRender->StartRender(videoChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     if (vieBase->StartReceive(videoChannel)!=) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if (vieBase->StartSend(videoChannel)!=) {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(voeBase->StartReceive(audioChannel) != )
     {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(voeBase->StartPlayout(audioChannel) != )
     {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(voeBase->StartSend(audioChannel) != )
     {
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     //webrtc::CodecInst ci;
     //voeFile->StartRecordingMicrophone(@"a.avi",ci,1000);

     DebugLog(@"AVInfo: %s at line %d success!", __FUNCTION__, __LINE__);
     return TRUE;
 }

 -(BOOL)stop
 {
     if(voeBase->StopSend(audioChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(voeBase->StopReceive(audioChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(voeBase->StopPlayout(audioChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     if(vieBase->StopSend(videoChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieBase->StopReceive(videoChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieCapture->StopCapture(captureID)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieCapture->ReleaseCaptureDevice(captureID)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieRender->StopRender(videoChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieRender->RemoveRenderer(videoChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     if(voeBase->DeleteChannel(audioChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }
     if(vieBase->DeleteChannel(videoChannel)!=){
         DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__);
         return FALSE;
     }

     DebugLog(@"AVInfo: %s at line %d success", __FUNCTION__, __LINE__);

     return TRUE;
 }

相关说明:

1，声音处理:

1.1. webrtc支持很多种音频编码，ilbc. isac. G711. G722. opus等等，不能编码适用不同场景，可根据自己需求调整.

1.2. 声音处理最大的难题就是噪声，回声，抖动，自动增益的处理，这也是最有价值的部分，webrtc和系统里面都有相应的处理，不过因为安卓机型众多，加上厂商DIY所以不同机器问题不一样，有些问题还得自己去处理，如webrtc团队基本上就不会用小米，酷派啥的测试.

1.3. AECM目前在安卓上都是通过软件在处理，看资料说后面一些厂商会直接集成到硬件上，具体效果拭目以待.

2，视频处理:

2.1. webrtc默认使用vp8编码，这也是Google力推的一种编码格式，后面会推VP9.

2.2. 如果需兼容H264，则需要自己去集成，实际上有人已经这么做了，不过WebRTC后面也会支持H264.

2.3. vp8与和h264孰优孰劣，最好自己去比较测试，不要道听途书，我相信Google力推的东西不会差到哪去.

2.4. NACK，字面解释就是协商确认包，实际就是起到丢包重传的作用，网络不好时因为丢包造成花屏，通过这个可解决，但会带来一定的延迟.

2.5. FEC，字面解释就是向前纠错编码，与NACK不同，包里面已经携带了纠错码，即时前一包未正确接收，也可根据他的信息正确计算出来.