How to implement live video function in HTML5-H5 Tutorial-php.cn

Video live broadcast has become very popular recently. I found that the current mainstream video live broadcast solutions on the WEB are HLS and RTMP. The mobile WEB side is currently mainly HLS, and the PC side is mainly RTMP. The real-time performance is better. Next, we will focus on these two This video streaming protocol is used to share the H5 live broadcast theme. Let’s share with you the detailed explanation of HTML5 video live broadcast ideas through this article. Let’s take a look together

Preface

Not long ago, I took the time to do some research and exploration on the currently popular video live broadcast, understand its overall implementation process, and explore the feasibility of HTML5 live broadcast on the mobile terminal.

It is found that the current mainstream video live broadcast solutions on the WEB are HLS and RTMP. The mobile WEB end is currently based on HLS (HLS has latency issues, and you can also use RTMP with the help of video.js). The PC end uses RTMP. In order to achieve better real-time performance, H5 live broadcast theme sharing will be carried out around these two video streaming protocols.

1. Video streaming protocol HLS and RTMP

1. HTTP Live Streaming

HTTP Live Streaming (HLS for short) is a video streaming protocol based on HTTP, implemented by Apple. QuickTime, Safari on Mac OS and Safari on iOS all support HLS well. Higher versions of Android also add support for HLS. . Some common clients such as MPlayerX and VLC also support the HLS protocol.

HLS protocol is based on HTTP, and a server that provides HLS needs to do two things:

Encoding: encode images in H.263 format, and encode sounds in MP3 or HE-AAC Encoding, and finally packaged into an MPEG-2 TS (Transport Stream) container; segmentation: split the encoded TS file into small files with the suffix ts of equal length, and generate a .m3u8 plain text index file; the browser uses are m3u8 files. m3u8 is very similar to the audio list format m3u. You can simply think of m3u8 as a playlist containing multiple ts files. The player plays them one by one in order, and then requests the m3u8 file after playing them all, and obtains the playlist containing the latest ts file to continue playing, and the cycle starts again. The entire live broadcast process relies on a constantly updated m3u8 and a bunch of small ts files. m3u8 must be dynamically updated, and ts can be processed through a CDN. A typical m3u8 file format is as follows:

#EXTM3U #EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=200000 gear1/prog_index.m3u8 #EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=311111 gear2/prog_index.m3u8 #EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=484444 gear3/prog_index.m3u8 #EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=737777 gear4/prog_index.m3u8

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You can see that the essence of the HLS protocol is still one HTTP request/response, so it has good adaptability and will not Affected by firewall. But it also has a fatal weakness: the delay is very obvious. If each ts is divided into 5 seconds, and an m3u8 puts 6 ts indexes, it will cause at least a 30-second delay. If you reduce the length of each ts and reduce the number of indexes in m3u8, the delay will indeed be reduced, but it will lead to more frequent buffering and the request pressure on the server will also increase exponentially. So we can only find a compromise point based on the actual situation.

For browsers that support HLS, just write this to play:

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Note: HLS is only supported on PC Safari browser, similar to Chrome browser, cannot play m3u8 format using HTML5 video

tag. You can directly use some more mature solutions on the Internet, such as: sewise-player, MediaElement, videojs-contrib-hls, jwplayer.

2. Real Time Messaging Protocol

Real Time Messaging Protocol (RTMP for short) is a set of video live broadcast protocols developed by Macromedia and now belongs to Adobe. This solution requires building a specialized RTMP streaming service such as Adobe Media Server, and only Flash can be used to implement the player in the browser. Its real-time performance is very good and the delay is very small, but its shortcoming is that it cannot support mobile WEB playback.

Although it cannot be played on the H5 page of iOS, you can write your own decoding and parsing for native iOS applications. RTMP has low latency and good real-time performance. On the browser side, the HTML5 video

tag cannot play RTMP protocol videos, which can be achieved through video.js.

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3. Comparison of video streaming protocol HLS and RTMP

2. Live broadcast format

##The current live broadcast display format is usually YY live broadcast and Yingke live broadcast. There are many types of pages, and you can see that their structure can be divided into three layers:

① Background video layer

② Follow and comment module

③ Like animation

The current H5 is similar to a live broadcast page, and the implementation technical difficulty is not big. It can be divided into implementation methods:

① The video background at the bottom uses the video video tag to achieve playback

② Follow and comment The module uses WebScoket to send and receive new messages in real time through DOM and CSS3.

③ Like and use CSS3 animation

After understanding the live broadcast format, let’s understand the live broadcast process as a whole.

3. The overall process of live broadcast

The overall process of live broadcast can be roughly divided into:

Video collection terminal: it can be a computer The audio and video input device on the mobile phone, or the camera or microphone on the mobile phone, is currently dominated by mobile phone video.

Live streaming video server: an Nginx server, which collects the video stream (H264/ACC encoding) transmitted by the video recording end, parses and encodes it on the server side, and pushes RTMP /HLS format video is streamed to the video player.

Video player: It can be the player on the computer (QuickTime Player, VLC), the native player on the mobile phone, and the video tag of H5, etc. Currently, the native player on the mobile phone is the main one.

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4. H5 recording video

For H5 video recording, you can use the powerful webRTC (Web Real-Time Communication), which is a technology that supports web browsers for real-time voice conversations or video conversations. The disadvantage is that it is only supported on Chrome on PC and is supported on mobile terminals. Not ideal.

Basic process of using webRTC to record video

① Call window.navigator.webkitGetUserMedia()

Get the user’s PC camera video data.

② Convert the obtained video stream data into window.webkitRTCPeerConnection

(a video stream data format).

③ Use WebScoket

to transmit video stream data to the server.

Note:

Although Google has been promoting WebRTC, and many mature products have appeared, most mobile browsers do not yet support webRTC (the latest iOS 10.0 does not support it either) , so the real video recording still needs to be achieved by the client (iOS, Android), and the effect will be better.

WebRTC support

iOS native application calls the camera to record video process

① Audio and video Collection, the original audio and video data stream can be collected using AVCaptureSession and AVCaptureDevice.

② Encode H264 for video and AAC for audio. There are encapsulated encoding libraries (x264 encoding, faac encoding, ffmpeg encoding) in iOS to encode audio and video.

③ Assemble the encoded audio and video data into packets.

④ Establish RTMP connection and push it to the server.

5. Build Ng

5. Build Nginx Rtmp live streaming service

Install nginx, nginx-rtmp-module

① First clone the nginx project to the local:

brew tap homebrew/nginx

② Execute the installation nginx-rtmp-module

brew install nginx-full –with-rtmp-module

2. nginx.conf configuration file, configure RTMP, HLS

Find the nginx.conf configuration file (path/usr /local/etc/nginx/nginx.conf), configure RTMP and HLS.

① Add the rtmp configuration content before the http node:

② Add the hls configuration in http

3. 重启nginx服务

重启nginx服务，浏览器中输入 http://localhost:8080，是否出现欢迎界面确定nginx重启成功。

nginx -s reload

六、直播流转换格式、编码推流

当服务器端接收到采集视频录制端传输过来的视频流时，需要对其进行解析编码，推送RTMP/HLS格式视频流至视频播放端。通常使用的常见编码库方案，如x264编码、faac编码、ffmpeg编码等。鉴于 FFmpeg 工具集合了多种音频、视频格式编码，我们可以优先选用FFmpeg进行转换格式、编码推流。

1.安装 FFmpeg 工具

brew install ffmpeg

2.推流MP4文件

视频文件地址：/Users/gao/Desktop/video/test.mp4

推流拉流地址：rtmp://localhost:1935/rtmplive/home，rtmp://localhost:1935/rtmplive/home

//RTMP 协议流 ffmpeg -re -i /Users/gao/Desktop/video/test.mp4 -vcodec libx264 -acodec aac -f flv rtmp://10.14.221.17:1935/rtmplive/home //HLS 协议流 ffmpeg -re -i /Users/gao/Desktop/video/test.mp4 -vcodec libx264 -vprofile baseline -acodec aac -ar 44100 -strict -2 -ac 1 -f flv -q 10 rtmp://10.14.221.17:1935/hls/test

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注意：

当我们进行推流之后，可以安装VLC、ffplay（支持rtmp协议的视频播放器）本地拉流进行演示

3.FFmpeg推流命令

① 视频文件进行直播

ffmpeg -re -i /Users/gao/Desktop/video/test.mp4 -vcodec libx264 -vprofile baseline -acodec aac -ar 44100 -strict -2 -ac 1 -f flv -q 10 rtmp://192.168.1.101:1935/hls/test ffmpeg -re -i /Users/gao/Desktop/video/test.mp4 -vcodec libx264 -vprofile baseline -acodec aac -ar 44100 -strict -2 -ac 1 -f flv -q 10 rtmp://10.14.221.17:1935/hls/test

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② 推流摄像头＋桌面+麦克风录制进行直播

ffmpeg -f avfoundation -framerate 30 -i “1:0″ \-f avfoundation -framerate 30 -video_size 640x480 -i “0” \-c:v libx264 -preset ultrafast \-filter_complex ‘overlay=main_w-overlay_w-10:main_h-overlay_h-10′ -acodec libmp3lame -ar 44100 -ac 1 -f flv rtmp://192.168.1.101:1935/hls/test

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更多命令，请参考：

FFmpeg处理RTMP流媒体的命令大全

FFmpeg常用推流命令

七、H5 直播视频播放

移动端iOS和 Android 都天然支持HLS协议，做好视频采集端、视频流推流服务之后，便可以直接在H5页面配置 video 标签播放直播视频。