webrtc h264

Transferred from: Http://www.oschina.net/p/kurentoKurento is a WebRTC streaming media server and some client APIs, which makes it easier to develop advanced video applications for WWW and smart phone platforms. The types of applications that can be developed using Kurento include video conferencing, audio and video broadcasting, audio and video recording, transcoding, and more.kurento/kurento-media-serverwatch151 Fork50Kurento Media server-more ...Mas

attend the meeting2, A and B establish A connection3, B and C establish the connection4, B forward a audio and video to c,b forward C audio and video to aThis situation in the case of B equipment performance is high, and a and C performance is weak, with B as a bridge to achieve 3-party calls, thus reducing the burden on the server. applicable Scenario : This model is only suitable for meetings of 3 people.B. forwarding via server synthesisEveryone attending the meeting sent the audio and video

This article original from Http://blog.csdn.net/voipmaker reprint annotated source.The latest version number TELEMCU added Android phone-side WEBRTC video conferencing capabilities, Android phone installed Chrome/firefox browser after loading TELEMCU Webrtcclientteleweb can directly participate in video conferencing,At the same time, TeleWeb can support up to two webrtcclient-to-peer communication, such as the following:TeleWeb Test Address:Http://ope

anyway.-Not really, the project over there is not finished yet ...Sad and hurried, so:As party A: the choice is most important to the person. When you do not know whether or not to choose the right person, you can only start small projects test water or have a backup plan. In addition to the actual cost, once you find the right person, do not be too stingy and timely payment, so as to have a winning result.As party B: must be delivered on schedule, must be honest. If you can't deliver it on sch

services; loss of use, DATA, or profits; * or business interruption) HOWEVER CAUSED AND ON ANY TH Eory of liability, * whether in contract, strict liability, or tort (including negligence or * OTHERWISE) arising in any way out of the use of this software, even if * advised of the possibility of such damage. */# ifndef TALK_BASE_BASICTYPES_H _ # define TALK_BASE_BASICTYPES_H _ # include The above Code defines the basic types, as well as the hardware architecture, in byte order. It will be use

WebRTC Code read (10): rtp_rtcp module analysis, webrtcrtp_rtcp1. Call interface RtpReceiverImpl: IncomingRtpPacket call interface ModuleRtpRtcpImpl: RtpData2. the main processing class ModuleRtpRtcpImpl, control Module, is a Module, you can independently process RtpPacketizer/RtpPacketizerH264/handler specific Format Decoding handler class RtpDepacketizer/Resolver/handler/specific format parsing RTP Header Processing class RtpReceiverImpl accept RTP

according to the RTMP protocol, the RTMP video stream can be published. To facilitate video packaging and publishing, an RTMPStream is encapsulated. Currently, only videos of H264 can be sent. You can directly send H264 data frames or H264 files. The interface provided by RTMPStream is as follows. Class CRTMPStream {public: CRTMPStream (void );~ CRTMPStream (vo

H264 decoder source code, porting the H264 decoding part of the FFmpeg to Android, depth pruning optimization, in the simulator (320x480) validated through. The application of the JNI architecture. Interface section, file reading, video display is done in Java, the bottom of the video decoding with C to do to meet the speed requirements. In this release, the NAL from the

;??????????????????????????????????????????????????????? Encoder deviceAvcodeccontext *cotx = NULL;????????????????????????????????? The encoder context, the main parameter setting, determines the operation of the encoderAvframe *pframe;????????????????????????????????????????????????????? Frame, fill in the data to be encoded and then give it to the coding method for processingAvpacket PKT;????????????????????????????????????????????????????????????? Package, the original frame encoding gets pa

The original post link is as follows: http://blog.csdn.NET/ren65432/article/details/43449391 Sometimes we need to get h264 naked stream for analysis. This article describes how to obtain a h264 stream by FFmpeg. The H264 code stream files obtained can be played directly through the player such as VLC. I. H264 file Dat

* ppropertybag = NULL;HR = pmoniker-> bindtostorage (0, 0, iid_ipropertybag, (void **) ppropertybag );If (failed (HR )){Iresult =-1;Continue;}Variant name;Name. Vt = vt_bstr;HR = ppropertybag-> Read (L "friendlyname", name, null); // obtain the friendly name of the deviceIf (succeeded (HR) // The friendly name of the device is obtained successfully.{Char friendlyname [256];Friendlyname [0] = 0;Widechartomultibyte (cp_acp, 0, name. bstrval,-1, friendlyname, 256, null, null );If (strcmp (friend

Recently, the IIS server downloads h264 files. After IIS is deployed, the input in the address bar of the browser indicates that the files do not exist and cannot be downloaded. You can try other types of files. Finally, my solution: set your own IIS server as follows: 1. Set mime to enable IIS to support more file types. skip this step if. h264 already exists in the MIME type. You can also set other ty

After debugging for a while, the RTSP server can finally see the live video. The test environment is ① Rt5350 runs in AP Mode ② The client is PC + VLC, And wifi uses a 54 Mbps USB Nic of TP-link. Busybox v1.12.1 (22:09:57 Cst) built-in Shell (Ash)Enter 'help' for a list of built-in commands.# FreeTotal used free shared BuffersMem: 28116 18524 9592 0 836Swap: 0 0 0Total: 28116 18524 9592# TopMem: 18520 K used, 9596 K free, 0 K shrd, 836 K buff, 2840 K cachedCPU: 9% USR 9% sys 0% nice 81% idle

An h264 video decoding trial program is displayed on the Internet, which can be used to test its logical process. The Code is as follows: Test_display_h264 (){In_fd = open (h1__input_file, o_rdonly); // video file open Fstat (in_fd, S); // get input file sizeFile_size = S. st_size; In_addr = (char *) MMAP (0, file_size, prot_read, map_shared, in_fd, 0); // mapping input file to memoryPp_fd = open (pp_dev_name, o_rdwr); // post processor open, why d

Today, when I access the Internet, I occasionally find the answer to my questions. H.264 video types The following media subtypes are defined for H.264 video. Subtype Fourcc Description Mediasubtype_avc1 'Avc1' H. 264 bitstream without start codes. Mediasubtype_h264 'H264' H. 264 bitstream with start codes. Mediasubtype_h264 'H264' Equivalent to

Today, when I access the Internet, I occasionally find the answer to my questions. H.264 video types The following media subtypes are defined for H.264 video. Subtype Fourcc Description Mediasubtype_avc1 'Avc1' H. 264 bitstream without start codes. Mediasubtype_h264 'H264' H. 264 bitstream with start codes. Mediasubtype_h264 'H264' Equivalent to

The following content is transferred from the Peter Lee h264 code stream structure. H.264's bitstream structure is very different from H.263's. It no longer uses a strict hierarchical structure. The following image is from the network and represents the h264 code stream structure from another perspective:

Twelve h264 RTP packet Timestamp Let's take h264 as an example. Void hsf-videortpsink: dospecialframehandling (unsigned/* fragmentationoffset */, The function first checks whether it is the last packet of a frame. If yes, it marks the 'M' and then sets the timestamp. Where does this timestamp come from? It depends on who calls the function dospecialframehandling (). After searching, it is called by multifr

Http://www.cnblogs.com/haibindev/archive/2012/04/16/2450989.html implements an RTMP server that outputs H264 live streamsRTMP (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

types of fu-a and fu-b. The type values are 28 and 29, respectively.Remember the definition of the previous nal_unit_type, 0~23 is for H264 use, 24~31 unused, in RTP packaging, if a nalu placed in a RTP packet, you can use Nalu Nal_unit_type, However, when you need to package multiple Nalu into a single RTP package, or if you need to package a NALU into multiple RTP packets, define a new type to identify it.Type Packet type name----------------------