mkv h264

FFmpeg extract H264 Nalu from mp4 http://blog.csdn.net/gavinr/article/details/7183499 639/* bitstream filters */ 640 REGISTER_BSF (AAC_ADTSTOASC, AAC_ADTSTOASC); 641 REGISTER_BSF (CHOMP, CHOMP); 642 REGISTER_BSF (Dump_extradata, dump_extradata); 643 REGISTER_BSF (H264_MP4TOANNEXB, H264_MP4TOANNEXB); 644 REGISTER_BSF (HEVC_MP4TOANNEXB, HEVC_MP4TOANNEXB); 645 REGISTER_BSF (Imx_dump_header, Imx_dump_header); 646 REGISTER_BSF (Mjpeg2jpeg, mj

* 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----------------------

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,

;??????????????????????????????????????????????????????? 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

H264 video packet loss is inevitable during transmission in the group network, especially when the network environment is poor, transmission of h264 code streams, packet loss will lead to the decoding terminal screen, mosaic is serious, this cutting-edge technology is FEC, NACK, the former is the forward correction technology, and the latter is the retransmission. The two can work well to solve the visual e

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, I found a strange problem. I can call the SIP client of the lower computer by using the Linphone client of the upper computer to work normally, but in turn there is a problem. Packet Capture found that Linphone sent a large number of IP fragmentation data packets, so google knows that when the data found is larger than MTU, it will generate IP fragmentation data packets. I have already split the RTP package? This should not happen normally. Linphone uses RTP to package