webrtc h264

Recently, I am working on an online tutorial Network project. I need to upload videos in any format to automatically play h264 standard videos using html5. Finally, PHP + FFMPEG is used for implementation. Here we will share our detailed solutions with you! Configure the php. ini file The Code is as follows: File_uploads = on; // whether to allow file upload over HTTP. ON is enabled by default.Upload_tmp_dir; // upload the file to the place where

Php+ffmpeg implements the automatic transcoding of video into H264 standard MP4 file, Configuring the php.ini File File_uploads = on;//whether to allow the switch to upload files over HTTP. The default is on is open upload_tmp_dir;//File upload to the server where temporary files are stored, if not specified will be used the system default temporary folder Upload_max_filesize = 1024m;//Wang Wen Business, that is, the maximum size allowed to upload fil

H264 is a new generation of coding standards, high compression high-quality and support a variety of network streaming media transmission is known, in terms of coding, I understand his theoretical basis is: reference for a period of time, the statistical results show that in several adjacent images, there is generally a difference in pixels only 10% of the points, The difference in brightness does not change beyond 2%, and the chroma difference is onl

Have such a demand, give you a h264 raw data file, let you play directly out, how to implement?The idea is that the H264 raw data format is 0x00000001 back with 0x67 0x68 0x65 0x41 Such data, decoding requires a complete NAL data unit, We need to read the data from each 0x00000001 and the next 0x00000001 before decoding it to the decoder.Read the file I will not wordy, this article mainly explains how to ob

---------------------- Objective ----------------------- H264 is a new generation of coding standards, with high compression quality and support for a variety of network streaming media transmission is known, in terms of coding, I understand his theoretical basis is: the reference to a period of time the image of the statistical results show that in the adjacent several images, the general difference in pixels only 10% points, The brightness differenc

----------------------- Objective ----------------------- H264 is a new generation of coding standards, high compression high-quality and support a variety of network streaming media transmission is known, in terms of coding, I understand his theoretical basis is: reference for a period of time, the statistical results show that in several adjacent images, there is generally a difference in pixels only 10% of the points, The difference in brightness d

H264 is a new generation of coding standards, high compression high-quality and support a variety of network streaming media transmission is known, in terms of coding, I understand his theoretical basis is: reference for a period of time, the statistical results show that in several adjacent images, there is generally a difference in pixels only 10% of the points, The difference in brightness does not change beyond 2%, and the chroma difference is onl

RTP h264 PS MPEG-2 Preface: RTP analysis. I found a lot of information on the Internet, but it was not complete. So I tried to make a comprehensive analysis, I have referenced many articles and listed them at the end of the article. I would like to express my gratitude. The development of the Internet is inseparable from the selfless dedication of everyone. I decided to start from me and hope everyone can support it. Original is not easy, reprint Ple

One, H264 video encoded into MP4 file See: H264 video encoded into MP4 file See: Compilation of Mp4v2 under VS2010 and its use in projectsRecently do the project needs to be H264 file encapsulated as MP4 file, from the Internet to find the MP4V2 library, downloaded down do not know where to start, the official website https://code.google.com/p/mp4v2/in the comp

Recently doing a mobile end with mobile, web-side text, video, voice chat features. Text chat using WebSocket, a lot of information on the Internet, there is no difficulty. But in the video, voice chat encountered a small difficulty. have been looking for some of the SDK to quickly develop, such as Opentok, cloud communications, etc., but the project is used in the intranet, these SDKs must be used in an external network, you need to obtain signaling on their servers. Later, I will try to use

Crosswalk QuickStart, using WEBRTC (HTML) to start developing video callsInstall PythonDownload the installer from http://www.python.org/downloads/After the installation is complete, add the environment variable again.Installing Oracle JDK Download page:http://www.oracle.com/technetwork/java/javase/downloads/Select the Java version to download (recommended Java 7). Select a JDK to download and accept the license agreement. Once downlo

Some time ago in the audio and video version of iOS, so the title changed to Android IOS WebRTC audio and Video development summary, the following summarizes some of the experience in the development process:1. iOS WEBRTC audio and video compilation and download: have android WebRTC compile download experience and then go to get IOS, you will find a lot easier, a

Some time ago in the audio and video version of iOS, so the title changed to the Android iOS WEBRTC audio and Video development summary, the following summary of some experience in the development process: 1. iOS WEBRTC audio and video compilation and download: Have the android WEBRTC compile download experience to get IOS, you will find that more simple, then th

1. Macro Block matching pixel precision: In MPEG2, the motion estimation accuracy is 12 pixels, which can be implemented through linear insertion (which may be simply corrected ); both H264 and MPEG4 support 14-pixel precision. 2. Number of reference frames: MPEG2. P frames of MPEG4 can only have one reference frame, and B frames can have two reference frames. H. 264 can have up to 15 reference frames, one 1. Macro Block matching pixel precision: In M

Options rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 1User-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 1Date: sat, Jan 01 2000 00:05:11 GMTPublic: Options, describe, setup, teardown, play, pauseDescribe rtsp: // 192.168.1.154: 8557/h264 RTSP/1.0CSeq: 2Accept: Application/SDPUser-Agent: VLC Media Player (live555 streaming media v2010.05.28)RTSP/1.0 200 OKCSeq: 2Date:

Recently, the tutor asked to study WebRTC, hoping to use our ICT2 system in the future.But never did the foundation of the web, whether front-end or back-end, HTML, JS all learn from the beginning. HTML is good to say, not too complicated things.JS is a bit difficult, roughly turned over the JS authoritative guide book, understand the basic grammar, also is enough to deal with. But it's completely out of the picture of the various objects built into t

In the next is WEBRTC development novice, at present encountered a problem, turned over to have not understood. Maybe English is not good, look at the document to see blindfolded, so has not found a solution.Development environment:node. JS Server builtI'm using Socket.io to do communications now.Development Purpose:A classmate to B students to initiate a request, B received after the two sides live video.If there is a clear classmate trouble tell me

The development of video conferencing based on the third party WEBRTC open source platform is not very difficult, mainly the business aspects. However, once involved in the core of the underlying issues need to read the source code, to find out the bug, the difficulty is not small.The project needs to analyze the creation process of peerconnection.assuming clienta,clientb is divided into offer and answer. Offer end PC =new rtcpeerconnec

write on the frontA: The purpose of writing a blog1. Self-study of the hard self-evident.2. All kinds of information on the Internet is a mixed bag, many are outdated.3. Based on the latest WEBRTC source to share some experience in their work.4. If you write a good people clap, write bad don't spray. Money to hold a field, no money ...Two: Compile compile or compile1. It is best to prepare a VPN, do not think of someone to copy the code to upload to t

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4401075.htmlThe first two blog posts complete the WEBRTC audio and video collection module, and the next step is to introduce the key audio and video coding modules. However, before introducing the audio and video coding module, we need to introduce the channel concept, and the transmission flow of each WEBRTC data is encapsulated into a channel