webrtc h264

http://blog.csdn.net/jefry_xdz/article/details/84613431, nal full name Network abstract layer, that is, the net abstraction level.In the H.264/AVC video coding standard, the whole system framework is divided into two levels: Video coding level (VCL) and network abstraction level (NAL). The former is responsible for effectively representing the content of the video data, while the latter is responsible for formatting the data and providing the header information to ensure that the data suitable f

After reading the introduction of H264/AVC codec, I just left the following three articles in my mind:1, H264 does not understand how to implement a codec, just a coded video bit stream syntax, and the Bitstream decoding method, this is similar to MPEG.2, H264 and once the standard (such as H261, H263, MPEG-1, MPEG-4) codec implementation process is not much diff

The previous article described "H264 video via rtmp stream live", below will explain how to H264 live video live through RTSP. The realization of the idea is to send the video stream to live555, by live555 to achieve the H264 data stream live. The video capture module sends the H264 data frame to live555 through a FIFO

Compile and install WebRTCsvncheckouthttp in Ubuntu: // configure. Then gclientconfighttps: // webrtc. Compile and install WebRTC in Ubuntu Svn checkout http://webrtc.googlecode.com/svn/trunk/ After the download is complete, WebRTC will get a folder named trunk by default, which contains the WebRTC source code, which i

Google's first integration of WebRTC in the Chrome Dev release released this January was a source of widespread concern. Today, Google published a roadmap for the development of WebRTC technology in its blog.WebRTC is a technology for real-time video and audio communication inside the browser, and Google acquired a technology in 2010 to acquire Global IP Solutions. The technology is based on the WHATWG prot

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC Technology is committed to the browser to achieve real-time audio and video, multimedia data interoperability, its NAT traversal part of the ice framework, the purpose is to achieve media P2P,SBC called the session Border controller, dedicated to the media, signaling NAT traversal, but SBC technology in the media by the server relay, This violates the original intention of

In WebRTC's example project, there is an Android project called Apprtcdemo, which enables video calling (VoIP) on a wide area network. This article is intended to demonstrate the compilation of Apprtcdemo, with Windows as an example, but also for Mac and Linux. Switch to a Linux environment please specify what platform you are currently using, and if it is Linux, you can ignore this step; otherwise, you will need a virtual machine. I'm using damn windows, and I recommend vagrant, a lightweight v

Previous notes, finishingWEBRTC in the default open RTX for packet loss retransmission, the introduction of RTX can refer to Rfc4588,https://tools.ietf.org/html/rfc4588#section-4RTX uses an additional SSRC transmission, SSRC is identified in the SDP.↵a=rtpmap: rtx/90000↵a2736695910239189782Like this.A RTX packet, in Turnserver, is such that the raw UDP data->turn/stun protocol header->RTP Header1->RTP header2In RTP header1, according to payload type to distinguish RTP, RTX data, if it is rtx, yo

Author: Gustavo Garcia (original link) Translation: Liu Tong Bandwidth estimation is probably the most important part of the WEBRTC video engine. The task of the Bandwidth estimation (BWE) module is to determine how much video stream you can send and not network congestion to ensure that video quality is not reduced. In the previous bandwidth estimation algorithm is very basic, in general, based on the design of packet loss. Usually we start to slowl

Some personal understanding about WEBRTC Just participated in the sound network presided over the first WEBRTC conference in Beijing, coupled with reading "hundred asked Freeswtich" written by Daniel, to it has more understanding, record for later review: 1, simple understanding, WEBRTC is a way to achieve web-to-business dial audio and video telephony technolog

reproduced in the original: Http://www.cnblogs.com/mod109/p/5767867.html#top thank you very much. The WEBRTC's audio processing module is divided into noise reduction ns (NSX), echo cancellation AEC (Echo control Acem), Audio gain AGC, and Mute detection section . In addition WEBRTC has encapsulated a set of audio processing module APM, if it is not a special need, if users want to use the echo cancellation and other more complex modules, it is best

Several questions1, WEBRTC transmit bandwidth is estimated for each stream or the total bandwidth2, WebRTC Remb is the overall bandwidth of statistics.3, if WEBRTC at the same time to watch the multi-channel flow, how to for each stream feedback bandwidth, packet loss and other information5, if the WEBRTC simultaneousl

What is h264 video file format? H.264 is the latest format defined by the MPEG-4 standard, and it is also one of the most technically available video coding formats, representing the latest technology, some also called AVC (Advanced Video coding), a video compression standard. is a widely used format for recording, compressing, and publishing high precision video. H.264 hardware Applications-Hd-dvd/Blu-ray (Blu-ray) The DVD Forum (DVD Forum) and th

FFmpeg installing a third-party encoder (encoder) Library Keywords: ffmpeg, code H264, third party encoder After installing the FFmpeg, if you use the FFmpeg tool to turn a video file into H264 video encoding, MP3 audio encoding or other ffmpeg itself without the XXX encoding type, you will see the error message, unknown encoder ' xxx '. Now all you need is to install the other encoder on the line, essentia

During this period of time, the work needs are converted from FLV stream to standard h264 and aac on rtmp. The server is an open-source project crtmpserver, client flex writing, video encoding h264, and audio encoding AAC, some protocol-related items are recorded as follows. 1. Analyze FLV data Let's take a FLV file to analyze the FLV data format. FLV standard document http://www.cnblogs.com/chef/a

All content in this article is original. For more information, see the source! Preface Some time ago, we have been engaged in video coding and decoding and transmission. Now, basically the whole system can implement the function of real-time server encoding packaging and transmission to the client, and the client can be decoded and displayed in real time, so I will record the work in the previous stage and use x264 for H. 264 the process of encoding, the process of transmitting

This time, the work of the need, on the rtmp to do the FLV stream to the standard H264, AAC conversion, the server is open source project Crtmpserver, client flex writing, Video coding H264, Audio coding AAC, now some of the agreements related to the following things record. I. Analysis of FLV data Let's take an FLV file to briefly analyze the format of the FLV data FLV standard document Download address h

A table that indicates the timing and physical layout of the Sampe. Chunk A track of a number of sample units composed of. three, MP4 package format structure diagram 1 Example Sample From Android Mediarecoder video recording, the platform for Huawei T8300 and TCL968, with Mp4info view as follows: use Eseye to view the following: The analysis by Winhex is as follows: 2 box structure diagram Next, some of the useful h264