WebRTC VideoEngine综合应用示例（一）——视频通话的基本流程（转）
本系列目前共三篇文章，后续还会更新
WebRTC VideoEngine综合应用示例（一）——视频通话的基本流程
WebRTC VideoEngine综合应用示例（二）——集成OPENH264编解码器
WebRTC VideoEngine综合应用示例（三）——集成X264编码和ffmpeg解码
WebRTC技术的出现改变了传统即时通信的现状，它是一套开源的旨在建立浏览器端对端的通信标准的技术，支持浏览器平台，使用P2P架构。WebRTC所采用的技术都是当前VoIP先进的技术，如内部所采用的音频引擎是Google收购知名GIPS公司获得的核心技术：视频编解码则采用了VP8。
大家都说WebRTC好，是未来的趋势，但是不得不说这个开源项目对新手学习实在是太不友好，光是windows平台下的编译就能耗费整整一天的精力，还未必能成功，关于这个问题在我之前的文章中有所描述。编译成功之后打开一看，整个solution里面有215个项目，绝对让人当时就懵了，而且最重要的是，google方面似乎没给出什么有用的文档供人参考，网络上有关的资料也多是有关于web端开发的，和Native API开发有关的内容少之又少，于是我决定把自己这两天学习VideoEngine的成果分享出来，供大家参考，有什么问题也欢迎大家指出，一起学习一起进步。
首先需要说明的是，webrtc项目的all.sln下有一个vie_auto_test项目，里面包含了一些针对VideoEngine的测试程序，我这里的demo就是基于此修改得到的。
先来看一下VideoEngine的核心API，基本上就在以下几个头文件中了。
具体来说
ViEBase用于
- 创建和销毁 VideoEngine 实例
- 创建和销毁 channels - 将 video channel 和相应的 voice channel 连接到一起并同步 - 发送和接收的开始与停止
ViECapture用于
- 分配capture devices. - 将 capture device 与一个或多个 channels连接起来. - 启动或停止 capture devices. - 获得capture device 的可用性.
ViECodec用于
- 设置发送和接收的编解码器.
- 设置编解码器特性.
- Key frame signaling.
- Stream management settings.
ViEError即一些预定义的错误消息
ViEExternalCodec用于注册除VP8之外的其他编解码器
ViEImageProcess提供以下功能
- Effect filters - 抗闪烁 - 色彩增强
ViENetwork用于
- 配置发送和接收地址. - External transport support. - 端口和地址过滤. - Windows GQoS functions and ToS functions. - Packet timeout notification. - Dead‐or‐Alive connection observations.
ViERender用于
- 为输入视频流、capture device和文件指定渲染目标. - 配置render streams.
ViERTP_RTCP用于
- Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
- SSRC handling. - Transmission of RTCP reports. - Obtaining RTCP data from incoming RTCP sender reports. - RTP and RTCP statistics (jitter, packet loss, RTT etc.). - Forward Error Correction (FEC). - Writing RTP and RTCP packets to binary files for off‐line analysis of the call quality. - Inserting extra RTP packets into active audio stream.
下面将以实现一个视频通话功能为实例详细介绍VideoEngine的使用，在文末将附上相应源码的下载地址
第一步是创建一个VideoEngine实例，如下
webrtc::VideoEngine* ptrViE = NULL; ptrViE = webrtc::VideoEngine::Create(); if (ptrViE == NULL) { printf("ERROR in VideoEngine::Create\n"); return -1; }
然后初始化VideoEngine并创建一个Channel
webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE); if (ptrViEBase == NULL) { printf("ERROR in ViEBase::GetInterface\n"); return -1; } error = ptrViEBase->Init();//这里的Init其实是针对VideoEngine的初始化 if (error == -1) { printf("ERROR in ViEBase::Init\n"); return -1; } webrtc::ViERTP_RTCP* ptrViERtpRtcp = webrtc::ViERTP_RTCP::GetInterface(ptrViE); if (ptrViERtpRtcp == NULL) { printf("ERROR in ViERTP_RTCP::GetInterface\n"); return -1; } int videoChannel = -1; error = ptrViEBase->CreateChannel(videoChannel); if (error == -1) { printf("ERROR in ViEBase::CreateChannel\n"); return -1; }
列出可用的capture devices等待用户进行选择, 然后进行allocate和connect，最后start选中的capture device
webrtc::ViECapture* ptrViECapture = webrtc::ViECapture::GetInterface(ptrViE); if (ptrViEBase == NULL) { printf("ERROR in ViECapture::GetInterface\n"); return -1; } const unsigned int KMaxDeviceNameLength = 128; const unsigned int KMaxUniqueIdLength = 256; char deviceName[KMaxDeviceNameLength]; memset(deviceName, 0, KMaxDeviceNameLength); char uniqueId[KMaxUniqueIdLength]; memset(uniqueId, 0, KMaxUniqueIdLength); printf("Available capture devices:\n"); int captureIdx = 0; for (captureIdx = 0; captureIdx < ptrViECapture->NumberOfCaptureDevices(); captureIdx++) { memset(deviceName, 0, KMaxDeviceNameLength); memset(uniqueId, 0, KMaxUniqueIdLength); error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName, KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength); if (error == -1) { printf("ERROR in ViECapture::GetCaptureDevice\n"); return -1; } printf("\t %d. %s\n", captureIdx + 1, deviceName); } printf("\nChoose capture device: "); if (scanf("%d", &captureIdx) != 1) { printf("Error in scanf()\n"); return -1; } getchar(); captureIdx = captureIdx - 1; // Compensate for idx start at 1. error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName, KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength); if (error == -1) { printf("ERROR in ViECapture::GetCaptureDevice\n"); return -1; } int captureId = 0; error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength, captureId); if (error == -1) { printf("ERROR in ViECapture::AllocateCaptureDevice\n"); return -1; } error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel); if (error == -1) { printf("ERROR in ViECapture::ConnectCaptureDevice\n"); return -1; } error = ptrViECapture->StartCapture(captureId); if (error == -1) { printf("ERROR in ViECapture::StartCapture\n"); return -1; }
设置RTP/RTCP所采用的模式
error = ptrViERtpRtcp->SetRTCPStatus(videoChannel, webrtc::kRtcpCompound_RFC4585); if (error == -1) { printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n"); return -1; }
设置接收端解码器出问题的时候，比如关键帧丢失或损坏，如何重新请求关键帧的方式
error = ptrViERtpRtcp->SetKeyFrameRequestMethod(videoChannel, webrtc::kViEKeyFrameRequestPliRtcp); if (error == -1) { printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n"); return -1; }
设置是否为当前channel使用REMB（Receiver Estimated Max Bitrate）包，发送端可以用它表明正在编码当前channel
接收端用它来记录当前channel的估计码率
error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true); if (error == -1) { printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n"); return -1; }
设置rendering用于显示
webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE); if (ptrViERender == NULL) { printf("ERROR in ViERender::GetInterface\n"); return -1; }
显示本地摄像头数据，这里的window1和下面的window2都是显示窗口，更详细的内容后面再说
error = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0); if (error == -1) { printf("ERROR in ViERender::AddRenderer\n"); return -1; } error = ptrViERender->StartRender(captureId); if (error == -1) { printf("ERROR in ViERender::StartRender\n"); return -1; }
显示接收端收到的解码数据
error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0, 1.0); if (error == -1) { printf("ERROR in ViERender::AddRenderer\n"); return -1; } error = ptrViERender->StartRender(videoChannel); if (error == -1) { printf("ERROR in ViERender::StartRender\n"); return -1; }
设置编解码器
webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE); if (ptrViECodec == NULL) { printf("ERROR in ViECodec::GetInterface\n"); return -1; } VideoCodec videoCodec; int numOfVeCodecs = ptrViECodec->NumberOfCodecs(); for (int i = 0; i<numOfVeCodecs; ++i) { if (ptrViECodec->GetCodec(i, videoCodec) != -1) { if (videoCodec.codecType == kVideoCodecVP8) { break; } } } videoCodec.targetBitrate = 256; videoCodec.minBitrate = 200; videoCodec.maxBitrate = 300; videoCodec.maxFramerate = 25; error = ptrViECodec->SetSendCodec(videoChannel, videoCodec); assert(error != -1); error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec); assert(error != -1);
设置接收和发送地址，然后开始发送和接收
webrtc::ViENetwork* ptrViENetwork = webrtc::ViENetwork::GetInterface(ptrViE); if (ptrViENetwork == NULL) { printf("ERROR in ViENetwork::GetInterface\n"); return -1; } //VideoChannelTransport是由我们自己定义的类，后面将会详细介绍 VideoChannelTransport* video_channel_transport = NULL; video_channel_transport = new VideoChannelTransport(ptrViENetwork, videoChannel); const char* ipAddress = "127.0.0.1"; const unsigned short rtpPort = 6000; std::cout << std::endl; std::cout << "Using rtp port: " << rtpPort << std::endl; std::cout << std::endl; error = video_channel_transport->SetLocalReceiver(rtpPort); if (error == -1) { printf("ERROR in SetLocalReceiver\n"); return -1; } error = video_channel_transport->SetSendDestination(ipAddress, rtpPort); if (error == -1) { printf("ERROR in SetSendDestination\n"); return -1; } error = ptrViEBase->StartReceive(videoChannel); if (error == -1) { printf("ERROR in ViENetwork::StartReceive\n"); return -1; } error = ptrViEBase->StartSend(videoChannel); if (error == -1) { printf("ERROR in ViENetwork::StartSend\n"); return -1; }
设置按下回车键即停止通话
printf("\n call started\n\n"); printf("Press enter to stop..."); while ((getchar()) != '\n') { }
停止通话后的各种stop
error = ptrViEBase->StopReceive(videoChannel); if (error == -1) { printf("ERROR in ViEBase::StopReceive\n"); return -1; } error = ptrViEBase->StopSend(videoChannel); if (error == -1) { printf("ERROR in ViEBase::StopSend\n"); return -1; } error = ptrViERender->StopRender(captureId); if (error == -1) { printf("ERROR in ViERender::StopRender\n"); return -1; } error = ptrViERender->RemoveRenderer(captureId); if (error == -1) { printf("ERROR in ViERender::RemoveRenderer\n"); return -1; } error = ptrViERender->StopRender(videoChannel); if (error == -1) { printf("ERROR in ViERender::StopRender\n"); return -1; } error = ptrViERender->RemoveRenderer(videoChannel); if (error == -1) { printf("ERROR in ViERender::RemoveRenderer\n"); return -1; } error = ptrViECapture->StopCapture(captureId); if (error == -1) { printf("ERROR in ViECapture::StopCapture\n"); return -1; } error = ptrViECapture->DisconnectCaptureDevice(videoChannel); if (error == -1) { printf("ERROR in ViECapture::DisconnectCaptureDevice\n"); return -1; } error = ptrViECapture->ReleaseCaptureDevice(captureId); if (error == -1) { printf("ERROR in ViECapture::ReleaseCaptureDevice\n"); return -1; } error = ptrViEBase->DeleteChannel(videoChannel); if (error == -1) { printf("ERROR in ViEBase::DeleteChannel\n"); return -1; } delete video_channel_transport; int remainingInterfaces = 0; remainingInterfaces = ptrViECodec->Release(); remainingInterfaces += ptrViECapture->Release(); remainingInterfaces += ptrViERtpRtcp->Release(); remainingInterfaces += ptrViERender->Release(); remainingInterfaces += ptrViENetwork->Release(); remainingInterfaces += ptrViEBase->Release(); if (remainingInterfaces > 0) { printf("ERROR: Could not release all interfaces\n"); return -1; } bool deleted = webrtc::VideoEngine::Delete(ptrViE); if (deleted == false) { printf("ERROR in VideoEngine::Delete\n"); return -1; } return 0;
以上就是VideoEngine的基本使用流程，下面说一下显示窗口如何创建
这里使用了webrtc已经为我们定义好的类ViEWindowCreator，它有一个成员函数CreateTwoWindows可以直接创建两个窗口，只需实现定义好窗口名称、窗口大小以及坐标即可，如下
ViEWindowCreator windowCreator; ViEAutoTestWindowManagerInterface* windowManager = windowCreator.CreateTwoWindows(); VideoEngineSample(windowManager->GetWindow1(), windowManager->GetWindow2());
这里的VideoEngineSample就是我们在前面所写的包含全部流程的示例程序，它以两个窗口的指针作为参数。
至于前面提到的VideoChannelTransport定义如下
class VideoChannelTransport : public webrtc::test::UdpTransportData { public: VideoChannelTransport(ViENetwork* vie_network, int channel); virtual ~VideoChannelTransport(); // Start implementation of UdpTransportData. virtual void IncomingRTPPacket(const int8_t* incoming_rtp_packet, const int32_t packet_length, const char* /*from_ip*/, const uint16_t /*from_port*/) OVERRIDE; virtual void IncomingRTCPPacket(const int8_t* incoming_rtcp_packet, const int32_t packet_length, const char* /*from_ip*/, const uint16_t /*from_port*/) OVERRIDE; // End implementation of UdpTransportData. // Specifies the ports to receive RTP packets on. int SetLocalReceiver(uint16_t rtp_port); // Specifies the destination port and IP address for a specified channel. int SetSendDestination(const char* ip_address, uint16_t rtp_port); private: int channel_; ViENetwork* vie_network_; webrtc::test::UdpTransport* socket_transport_; }; VideoChannelTransport::VideoChannelTransport(ViENetwork* vie_network, int channel) : channel_(channel), vie_network_(vie_network) { uint8_t socket_threads = 1; socket_transport_ = webrtc::test::UdpTransport::Create(channel, socket_threads); int registered = vie_network_->RegisterSendTransport(channel, *socket_transport_); } VideoChannelTransport::~VideoChannelTransport() { vie_network_->DeregisterSendTransport(channel_); webrtc::test::UdpTransport::Destroy(socket_transport_); } void VideoChannelTransport::IncomingRTPPacket( const int8_t* incoming_rtp_packet, const int32_t packet_length, const char* /*from_ip*/, const uint16_t /*from_port*/) { vie_network_->ReceivedRTPPacket( channel_, incoming_rtp_packet, packet_length, PacketTime()); } void VideoChannelTransport::IncomingRTCPPacket( const int8_t* incoming_rtcp_packet, const int32_t packet_length, const char* /*from_ip*/, const uint16_t /*from_port*/) { vie_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet, packet_length); } int VideoChannelTransport::SetLocalReceiver(uint16_t rtp_port) { int return_value = socket_transport_->InitializeReceiveSockets(this, rtp_port); if (return_value == 0) { return socket_transport_->StartReceiving(500); } return return_value; } int VideoChannelTransport::SetSendDestination(const char* ip_address, uint16_t rtp_port) { return socket_transport_->InitializeSendSockets(ip_address, rtp_port); }
继承自UdpTransportData类，主要重写了IncomingRTPPacket和IncomingRTCPPacket两个成员函数，分别调用了vie_network的ReceivedRTPPacket和ReceivedRTCPPacket方法，当需要将接收到的RTP和RTCP包传给VideoEngine时就应该使用这两个函数。
该示例程序最后效果如下，我这里是几个虚拟摄像头，然后会有两个窗口，一个是摄像头画面，一个是解码的画面。
源码地址在这里，这是一个可以脱离webrtc那个大项目而独立运行的工程。
原文转自 http://blog.csdn.net/nonmarking/article/details/47375849#
