webrtc h264

rtmp Server for output H264 live streamingRTMP (Real time Messaging Protocol) is a common streaming media protocol, used to transmit audio and video data, combined with flash, widely used in live, on-demand, chat and other applications, as well as PC, mobile, embedded and other platforms, is to do streaming media development often come into contact with the protocol. I have previously written an article, "rtmp protocol to send H. E encoding and AAC en

rtmp Server for output H264 live streamingRTMP (Real time Messaging Protocol) is a common streaming media protocol, used to transmit audio and video data, combined with flash, widely used in live, on-demand, chat and other applications, as well as PC, mobile, embedded and other platforms, is to do streaming media development often come into contact with the protocol. I have previously written an article, "rtmp protocol to send H. E encoding and AAC en

Os:centos6.4-64bit--------compiling FFMPEG with h265--------1. Installing HG Tool#yum Install HG2. Download x265 source code and make#hg Clone https://bitbucket.org/multicoreware/x265#cd X265/build/linux#make#make Install3. Download x264 and FFmpeg source and makePlease refer to one of my previous blog posts:Use Nginx+ffmpeg to build HLS live Transcoding server (http://blog.csdn.net/wutong_login/article/details/42292787)4. Compile support for h265 FFMPEG#PKG_CONFIG_PATH =/usr/local/lib/pkgconfig

The raw stream data obtained from h264 is. Generally, the bitstream structure is SPS, PPS, I frame, P frame ...... SPS, PPS, I frame, P frame ............ When we use RTP to package h264 data, SPS and PPS can directly send I and P frames without sending them. It also depends on the size of I frame and P frame. If it is smaller than MTU, it can be sent directly with the RTP package. If it is larger than MTU,

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! 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 htm

exponential GRUMB algorithm decoding process:When parsing the K-Order index Columbus Code, the first non-0-bit leadingzerobits is first searched from the current position of the bitstream, and then the value of Code_num is calculated according to the formula:Code_num = 2^ (leadingzerobits + k)-2^k + values;Where values are the first non-0-bit followed by (leadingzerobits + K) bits.Example (1-order exponential GRUMB algorithm decoding 001011): leadingzerobits=2;values=011b=3;code_num=2^3-2^1+3=9

Previous HiSilicon H264 decoding library ,It realizes the simple decoding of H264 frame, but after replacing the camera, there is a watermark problem in the center of decoding video, such asFind the network, the basic related to this article, has not given a good solution.http://bbs.csdn.net/topics/390325547which warrior knows: HiSilicon H. A PC decoding library Hi_h264dec_w.dll, video decoding, how to remo

It took me nearly three weeks to implement HTTP live streaming on the crtmpserver to package h264 and aac into ts streams and play them on the iPad through HTML5, because there is no ready-to-useCodeFor reference, the packaging code is all handwritten. For the packaging format, refer to ISO/ice 18318-1.pdf. During this period, I encountered many problems and took a lot of detours. The standard-compliant ts may not be able to play on the iPad, but the

H264 SVC is a technology that divides video streams into multiple layers of resolution, quality, and frame rate. It is an extension of H.264 Video Coding/decoding standard adopted by most video conferencing Devices today. Video conferencing equipment uses SVC technology to send and receive multi-layer video streams consisting of a small base layer and multiple other optional layers that can improve resolution, frame rate, and quality. This layered met

Last Post hardware video encoding, after the default parameters are set, encode 1000 frames 640x480 H264 file size is about 180m, very large, you must set the parametersAfter the test discovery is Enablembratecontrol, the encoded data begins to shrink significantly:Here the first 54 control parameters are all initialized-1, in these 54 parameters, including h264,h263 and MPEG video format settings, for each

Thanks to Dr. Rai for the Chinese video codec in the pay, http://blog.csdn.net/leixiaohua1020 Recently to do some video streaming things, to parse H264 bare stream and get OPENCV format of mat data to the algorithm engineers to run algorithms. Related resources have been difficult to find, often too old to make the API version of the replacement, no way to let new people or laymen [such as me] Quick validation of the code is feasible. In the referenc

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC clients establish a call through interactive candidate to achieve NAT traversal, if these interactions candidate in the Offer/anwser SDP will lead to through the establishment time is very slow,Because the WEBRTC client needs to communicate with the Stun/turn server to get these candidate, the entire process is block, in order to increase the speed of call establ

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The WebRTC-Peer part uses the ice framework, the ICE framework includes the Stun,turn, and one problem with the actual use of WebRTC to develop audio and video applications is that calls are built very slowly because the ice process takes too much time for the client to communicate with the stun server before initiating the c

This article mainly introduces the last encounter of some machines do not see the video problem, the article from the blog Park Rtc.blacker, reproduced please indicate the source.Before doing the video chat app has been running well, a few days ago customer feedback said in Samsung 9100. You can't see your own image on the Android4.0.3.After a search to find the WEBRTC is the bottom of the bug, has been repaired and feedback to the community, the foll

Transferred from: http://www.xuebuyuan.com/1248366.htmlThe bandwidth adaptive algorithm in WEBRTC is divided into two types:1, the originator bandwidth control, the principle is the RTCP in the packet loss statistics to dynamically increase or decrease the bandwidth, in the reduction of bandwidth using the TFRC algorithm to increase the smoothness.2, the receiver bandwidth estimation, the principle is and by the receipt of RTP data, the estimated band

based on WebRTC of the MCU Open Source Projects Licode the environment to buildDue to the needs of the project, we need to build multi-person communication and investigate three common structures of multi-person communication:1. The previous blog has been based on Codelab for three people chatting, a multi-person system based on Mesh structure. Specifically, the fake has n+1 client, then for each client needs to establish peerconnection with other N o

Recently the major live sites are compared to fire, want to explore how to play. But read a few Daniel's answer, feel there are too many unfamiliar things, try to get up a little higher cost. Found that there is a thing called WEBRTC, someone has analyzed he is not suitable for the flow of large numbers of live. But I'm just playing with it and feeling the video connectivity.The first thing I saw on GitHub was a demo of all the APIs, and an example of

Recently in the docking WebRTC to Android phone, there is a demand is the mobile phone horizontal screen when the other side of the image rotation, study the code of WEBRTC Video_render found that the remote video rendering using OPENGLES20 or Surfaceview implementation, Where OPENGLES20 uses hardware rendering, so performance is better, so simply add the VideoRenderOpenGles20 class in the video_render_open

1. Introduction It is well known that the browser itself does not support the direct establishment of communication channels between each other through the server to relay. For example, there are now two clients, A and B, they want to communicate, first need a and server, B and server to establish a channel between. A to send a message to B, a first send the message to the server, the server to a message to relay, sent to B, the reverse is the same. So a message between A and b through

WEBRTC FEC (forward error-correcting code) is an important part of its QoS, which can be used to recover original data packets when packet loss is lost, reduce retransmission times, reduce latency and improve video quality. It is an implementation of the RFC 5109 standard. Below, we will delve into its rationale. Redundant coding To understand the FEC in WEBRTC, you need to first understand red Packet. The