The loop attribute should do it: <video width=”320″ height=”240″ autoplay loop muted> <source src=”movie.mp4″ type=”video/mp4″ /> <source src=”movie.ogg” type=”video/ogg” /> Your browser does not support the video tag. </video> The addition of the unintuitive muted attribute is required by Chrome as documented on their dev site. Should you have a problem with the loop attribute … Read more
For your fast playback – frame rate issue, there is nothing you have to do here. Since it is a streaming solution the other side has to be told the frame rate in advance or timestamps with each frame. Both of these are not part of elementary stream. Either pre-determined framerate is chosen or you … Read more
The request for support for http byte-range was open at the time of this answer but is fixed in Spring 4.2.RC1. Check jira SPR-10805 or, the PR here. But based on the code that you linked in your question, Davin Kevin built a solution that could work for your request. Code for MultipartFileSender: import org.apache.commons.lang3.StringUtils; … Read more
Is WebSockets over TCP a fast enough protocol to stream a video of, say, 30fps? Yes.. it is, take a look at this project. Websockets can easily handle HD videostreaming.. However, you should go for Adaptive Streaming. I explain here how you could implement it. Currently we’re working on a webbased instant messaging application with … Read more
For rtsp streaming you can also try following servers: Darwin Streaming Server – linux package is available Windows Media Services – can be installed on Windows Server Trial VLC – standalone application For testing purposes of your application i would also recommend you to use existing mobile video services like: m.youtube.tv m.wp.tv You can extract … Read more
You could divide your recording to separate files with a length of say, 10sec, then send them separately. If you use AVCaptureSession‘s beginConfiguration and commitConfiguration methods to batch your output change you shouldn’t drop any frames between the files. This has many advantages over frame by frame upload: The files can be directly used for … Read more
A possible alternative for that: Use an encoder (e.g. VLC or FFmpeg) into packetize your input stream to OGG format. For example, in this case I used VLC to packetize screen capture device with this code: C:\Program Files\VideoLAN\VLC\vlc.exe -I dummy screen:// :screen-fps=16.000000 :screen-caching=100 :sout=#transcode{vcodec=theo,vb=800,scale=1,width=600,height=480,acodec=mp3}:http{mux=ogg,dst=127.0.0.1:8080/desktop.ogg} :no-sout-rtp-sap :no-sout-standard-sap :ttl=1 :sout-keep Embed this code into a <video> tag … Read more
Technologies and Requirements The only web-based technology set really geared toward low latency is WebRTC. It’s built for video conferencing. Codecs are tuned for low latency over quality. Bitrates are usually variable, opting for a stable connection over quality. However, you don’t necessarily need this low latency optimization for all of your users. In fact, … Read more
Once published live stream to SRS, over RTMP by OBS, FFmpeg or other encoder, you could play the stream on other devices. Generally there are serval players, on both H5 or mobile: HLS, about 5~10s latency, easy to play, you only need a H5 video tag, or hls.js. HTTP-FLV, about 1~3s latency, also very easy, … Read more