WebRTC VideoEngine综合应用示例(一)——视频通话的基本流程

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。


各位看官,如果您觉得本人的博客对您有所帮助,可以扫描如下二维码进行打赏,打赏多少您随意~


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版权声明:本文为博主原创文章,未经博主允许不得转载。 https://blog.csdn.net/nonmarking/article/details/47375849
个人分类: webrtc
上一篇HEVC码率控制算法研究与HM相应代码分析(三)——算法及代码分析
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