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
");
    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
");
    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
");
    return -1;
}

webrtc::ViERTP_RTCP* ptrViERtpRtcp = webrtc::ViERTP_RTCP::GetInterface(ptrViE);
if (ptrViERtpRtcp == NULL)
{
    printf("ERROR in ViERTP_RTCP::GetInterface
");
    return -1;
}

int videoChannel = -1;
error = ptrViEBase->CreateChannel(videoChannel);
if (error == -1)
{
    printf("ERROR in ViEBase::CreateChannel
");
    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
");
    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:
");
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
");
        return -1;
    }
    printf("	 %d. %s
", captureIdx + 1, deviceName);
}
printf("
Choose capture device: ");

if (scanf("%d", &captureIdx) != 1)
{
    printf("Error in scanf()
");
    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
");
    return -1;
}

int captureId = 0;
error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength, captureId);
if (error == -1)
{
    printf("ERROR in ViECapture::AllocateCaptureDevice
");
    return -1;
}

error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
if (error == -1)
{
    printf("ERROR in ViECapture::ConnectCaptureDevice
");
    return -1;
}

error = ptrViECapture->StartCapture(captureId);
if (error == -1)
{
    printf("ERROR in ViECapture::StartCapture
");
    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
");
    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
");
    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
");
    return -1;
}

设置rendering用于显示

webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
if (ptrViERender == NULL)
{
    printf("ERROR in ViERender::GetInterface
");
    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
");
    return -1;
}

error = ptrViERender->StartRender(captureId);
if (error == -1)
{
    printf("ERROR in ViERender::StartRender
");
    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
");
    return -1;
}

error = ptrViERender->StartRender(videoChannel);
if (error == -1)
{
    printf("ERROR in ViERender::StartRender
");
    return -1;
}

设置编解码器

webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
if (ptrViECodec == NULL)
{
    printf("ERROR in ViECodec::GetInterface
");
    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
");
    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
");
    return -1;
}
error = video_channel_transport->SetSendDestination(ipAddress, rtpPort);
if (error == -1)
{
    printf("ERROR in SetSendDestination
");
    return -1;
}

error = ptrViEBase->StartReceive(videoChannel);
if (error == -1)
{
    printf("ERROR in ViENetwork::StartReceive
");
    return -1;
}

error = ptrViEBase->StartSend(videoChannel);
if (error == -1)
{
    printf("ERROR in ViENetwork::StartSend
");
    return -1;
}

设置按下回车键即停止通话

printf("
 call started

");
printf("Press enter to stop...");
while ((getchar()) != '
')
{

}

停止通话后的各种stop

error = ptrViEBase->StopReceive(videoChannel);
if (error == -1)
{
    printf("ERROR in ViEBase::StopReceive
");
    return -1;
}

error = ptrViEBase->StopSend(videoChannel);
if (error == -1)
{
    printf("ERROR in ViEBase::StopSend
");
    return -1;
}

error = ptrViERender->StopRender(captureId);
if (error == -1)
{
    printf("ERROR in ViERender::StopRender
");
    return -1;
}

error = ptrViERender->RemoveRenderer(captureId);
if (error == -1)
{
    printf("ERROR in ViERender::RemoveRenderer
");
    return -1;
}

error = ptrViERender->StopRender(videoChannel);
if (error == -1)
{
    printf("ERROR in ViERender::StopRender
");
    return -1;
}

error = ptrViERender->RemoveRenderer(videoChannel);
if (error == -1)
{
    printf("ERROR in ViERender::RemoveRenderer
");
    return -1;
}
error = ptrViECapture->StopCapture(captureId);
if (error == -1)
{
    printf("ERROR in ViECapture::StopCapture
");
    return -1;
}

error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
if (error == -1)
{
    printf("ERROR in ViECapture::DisconnectCaptureDevice
");
    return -1;
}

error = ptrViECapture->ReleaseCaptureDevice(captureId);
if (error == -1)
{
    printf("ERROR in ViECapture::ReleaseCaptureDevice
");
    return -1;
}

error = ptrViEBase->DeleteChannel(videoChannel);
if (error == -1)
{
    printf("ERROR in ViEBase::DeleteChannel
");
    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
");
    return -1;
}

bool deleted = webrtc::VideoEngine::Delete(ptrViE);
if (deleted == false)
{
    printf("ERROR in VideoEngine::Delete
");
    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#

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原文地址：https://www.cnblogs.com/happykoukou/p/5764129.html