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:
1 -(BOOL)initWebrtcObjects 2 {
//转载请说明出处: RTC_Blacker http://www.cnblogs.com/lingyunhu 3 if ((voE = webrtc::VoiceEngine::Create()) == NULL) { 4 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 5 return FALSE; 6 } 7 if ((voeBase = webrtc::VoEBase::GetInterface(voE)) == NULL) { 8 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 9 return FALSE; 10 } 11 if ((voeCodec = webrtc::VoECodec::GetInterface(voE)) == NULL) { 12 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 13 return FALSE; 14 } 15 if ((voeFile=webrtc::VoEFile::GetInterface(voE))==NULL) { 16 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 17 } 18 if ((voeHardware = webrtc::VoEHardware::GetInterface(voE)) == NULL) { 19 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 20 return FALSE; 21 } 22 if ((voeNetwork = webrtc::VoENetwork::GetInterface(voE)) == NULL) { 23 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 24 return FALSE; 25 } 26 if ((voeAudioProccessing = webrtc::VoEAudioProcessing::GetInterface(voE)) == NULL) { 27 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 28 return FALSE; 29 } 30 if ((voeRtpRtcp = webrtc::VoERTP_RTCP::GetInterface(voE)) == NULL) { 31 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 32 return FALSE; 33 } 34 35 if(voeBase->Init()!=0){ 36 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 37 } 38 39 if ((viE = webrtc::VideoEngine::Create()) == NULL) { 40 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 41 return FALSE; 42 } 43 if ((vieBase = webrtc::ViEBase::GetInterface(viE)) == NULL) { 44 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 45 return FALSE; 46 } 47 if ((vieCapture = webrtc::ViECapture::GetInterface(viE)) == NULL) { 48 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 49 return FALSE; 50 } 51 if ((vieRender = webrtc::ViERender::GetInterface(viE)) == NULL) { 52 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 53 return FALSE; 54 } 55 if ((vieCodec = webrtc::ViECodec::GetInterface(viE)) == NULL) { 56 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 57 return FALSE; 58 } 59 if ((vieNetwork = webrtc::ViENetwork::GetInterface(viE)) == NULL) { 60 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 61 return FALSE; 62 } 63 if ((vieRtpRtcp = webrtc::ViERTP_RTCP::GetInterface(viE)) == NULL) { 64 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 65 return FALSE; 66 } 67 68 if (vieBase->Init() != 0) { 69 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 70 return FALSE; 71 } 72 73 [self initAudioCodec]; 74 [self initVideoCodec]; 75 76 captureID = 0; 77 videoChannel = -1; 78 79 return TRUE; 80 } 81 82 -(void)initAudioCodec 83 { 84 memset(&voeCodecInst, 0, sizeof(webrtc::CodecInst)); 85 86 if (voeCodec != NULL) { 87 for (int index=0; index < voeCodec->NumOfCodecs(); index++) { 88 webrtc::CodecInst ci; 89 voeCodec->GetCodec(index, ci); 90 if (strncmp(ci.plname, "ISAC", 4) == 0) { 91 memcpy(&voeCodecInst, &ci, sizeof(webrtc::CodecInst)); 92 break; 93 } 94 } 95 //voeCodecInst.channels = 1; 96 //voeCodecInst.rate = -1; 97 } 98 } 99 100 -(BOOL)start 101 { 102 f ((audioChannel = voeBase->CreateChannel())!=0) { 103 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 104 return FALSE; 105 } 106 if (vieBase->CreateChannel(videoChannel) != 0) { 107 DebugLog(@"AVErr: %d %s at line %d", vieBase->LastError(),__FUNCTION__, __LINE__); 108 return FALSE; 109 } 110 DebugLog(@"AVInfo: CreateChannel success! %d, %d",videoChannel,audioChannel); 111 112 //vieCodec->SetReceiveCodec(videoChannel,videoCodec); 113 114 if(voeAudioProccessing->SetAecmMode()!=0){ 115 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 116 return FALSE; 117 } 118 voeAudioProccessing->SetAgcStatus(TRUE, webrtc::kAgcDefault); 119 voeAudioProccessing->SetNsStatus(TRUE, webrtc::kNsHighSuppression); 120 _voice_capture_device_index = -1; 121 voeHardware->SetRecordingDevice(_voice_capture_device_index); 122 voeHardware->SetPlayoutDevice(_voice_playback_device_index); 123 if(voeHardware->SetLoudspeakerStatus(true)!=0){ 124 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 125 } 126 voeCodec->SetSendCodec(audioChannel, voeCodecInst); 127 128 RtpRtcpStreamStruct streamStruct=[self createRtpStreamStruct]; 129 voeChannelTransport=new webrtc::test::VoiceChannelTransport(voeNetwork, audioChannel); 130 voeChannelTransport->SetLocalReceiver2(localARtpPort.rtp,streamStruct ); 131 voeChannelTransport->SetSendDestination2([remoteIPAddress UTF8String], remoteARtpPort.rtp, remoteARtpPort.rtcp); 132 133 if(vieCodec->SetSendCodec(videoChannel, videoCodec) != 0) 134 { 135 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 136 return FALSE; 137 } 138 vieRtpRtcp->SetNACKStatus(videoChannel, TRUE); 139 vieRtpRtcp->SetRTCPStatus(videoChannel, webrtc::kRtcpNonCompound_RFC5506); 140 vieRtpRtcp->SetKeyFrameRequestMethod(videoChannel, webrtc::kViEKeyFrameRequestPliRtcp); 141 142 vieBase->SetVoiceEngine(voE); 143 if (vieBase->ConnectAudioChannel(videoChannel, audioChannel)) { 144 DebugLog(@"AVErr:%s at line %d",__FUNCTION__,__LINE__); 145 return FALSE; 146 } 147 148 if (deviceUniqueID == nil) { 149 DebugLog(@"AVInfo NumberOfCaptureDevices is %d", vieCapture->NumberOfCaptureDevices()); 150 int list_count=vieCapture->NumberOfCaptureDevices(); 151 if ( list_count> 0) { 152 int list_number=0; 153 if (list_count>1) { 154 list_number=1;//[[AVShareData instance] isUseFrontCamera]?0:1; 155 } 156 char device_name[KMaxDeviceNameLength]; 157 char unique_id[KMaxUniqueIdLength]; 158 memset(unique_id, 0, KMaxUniqueIdLength); 159 vieCapture->GetCaptureDevice(list_number, device_name, KMaxDeviceNameLength, unique_id, KMaxUniqueIdLength); 160 deviceUniqueID = [NSString stringWithFormat:@"%s", unique_id]; 161 } 162 } 163 DebugLog(@"AVInfo deviceUniqueID is %@", deviceUniqueID); 164 165 if ((vieCapture->AllocateCaptureDevice([deviceUniqueID UTF8String], deviceUniqueID.length, captureID)) != 0) { 166 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 167 return FALSE; 168 } 169 170 DebugLog(@"AVInfo captureID is %d", captureID); 171 172 if (vieCapture->ConnectCaptureDevice(captureID, videoChannel) != 0) { 173 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 174 return FALSE; 175 } 176 177 webrtc::CaptureCapability captureCapability; 178 captureCapability.width=352; 179 captureCapability.height=288; 180 captureCapability.codecType=webrtc::kVideoCodecVP8; 181 captureCapability.maxFPS=DEFAULT_VIDEO_CODEC_MAX_FRAMERATE; 182 //vieCapture->SetRotateCapturedFrames(captureID, <#const webrtc::RotateCapturedFrame rotation#>) 183 if (vieCapture->StartCapture(captureID,captureCapability) != 0) { 184 //if (vieCapture->StartCapture(captureID) != 0) { 185 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 186 return FALSE; 187 } 188 if((vieRender->AddRenderer(captureID, [self localRenderView], 0, 0.0, 0.0, 1.0, 1.0)) != 0){ 189 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 190 return FALSE; 191 } 192 /* 193 if((vieRender->AddRenderer(captureID, [self localRenderView2], 0, 0.0, 0.0, 1.0, 1.0)) != 0){ 194 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 195 return FALSE; 196 } 197 */ 198 199 if (vieRender->StartRender(captureID) != 0) { 200 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 201 return FALSE; 202 } 203 if(vieRender->AddRenderer(videoChannel, [self remoteRenderView], 1, 0.0f, 0.0f, 1.0f, 1.0f)!=0){ 204 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 205 return FALSE; 206 } 207 if(vieRender->StartRender(videoChannel)!=0){ 208 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 209 return FALSE; 210 } 211 212 if (vieBase->StartReceive(videoChannel)!=0) { 213 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 214 return FALSE; 215 } 216 if (vieBase->StartSend(videoChannel)!=0) { 217 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 218 return FALSE; 219 } 220 if(voeBase->StartReceive(audioChannel) != 0) 221 { 222 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 223 return FALSE; 224 } 225 if(voeBase->StartPlayout(audioChannel) != 0) 226 { 227 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 228 return FALSE; 229 } 230 if(voeBase->StartSend(audioChannel) != 0) 231 { 232 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 233 return FALSE; 234 } 235 236 //webrtc::CodecInst ci; 237 //voeFile->StartRecordingMicrophone(@"a.avi",ci,1000); 238 239 DebugLog(@"AVInfo: %s at line %d success!", __FUNCTION__, __LINE__); 240 return TRUE; 241 } 242 243 -(BOOL)stop 244 { 245 if(voeBase->StopSend(audioChannel)!=0){ 246 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 247 return FALSE; 248 } 249 if(voeBase->StopReceive(audioChannel)!=0){ 250 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 251 return FALSE; 252 } 253 if(voeBase->StopPlayout(audioChannel)!=0){ 254 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 255 return FALSE; 256 } 257 258 if(vieBase->StopSend(videoChannel)!=0){ 259 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 260 return FALSE; 261 } 262 if(vieBase->StopReceive(videoChannel)!=0){ 263 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 264 return FALSE; 265 } 266 if(vieCapture->StopCapture(captureID)!=0){ 267 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 268 return FALSE; 269 } 270 if(vieCapture->ReleaseCaptureDevice(captureID)!=0){ 271 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 272 return FALSE; 273 } 274 if(vieRender->StopRender(videoChannel)!=0){ 275 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 276 return FALSE; 277 } 278 if(vieRender->RemoveRenderer(videoChannel)!=0){ 279 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 280 return FALSE; 281 } 282 283 if(voeBase->DeleteChannel(audioChannel)!=0){ 284 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 285 return FALSE; 286 } 287 if(vieBase->DeleteChannel(videoChannel)!=0){ 288 DebugLog(@"AVErr: %s at line %d", __FUNCTION__, __LINE__); 289 return FALSE; 290 } 291 292 DebugLog(@"AVInfo: %s at line %d success", __FUNCTION__, __LINE__); 293 294 return TRUE; 295 }
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.
ChitGPT提问
1752
被折叠的 条评论
为什么被折叠?
到【灌水乐园】发言
