rtsp rtmp

-ndk-r9d/toolchains/arm-linux-androideabi-4.8/prebuilt/linux-x86/bin/ ARM-LINUX-ANDROIDEABI-GCC--cross-prefix=/home/wangrui/java/android-ndk-r9d/toolchains/ arm-linux-androideabi-4.8/prebuilt/linux-x86/bin/arm-linux-androideabi---nm=/home/wangrui/java/android-ndk-r9d /TOOLCHAINS/ARM-LINUX-ANDROIDEABI-4.8/PREBUILT/LINUX-X86/BIN/ARM-LINUX-ANDROIDEABI-NM--sysroot=/home/wangrui/ java/android-ndk-r9d/platforms/android-9/arch-arm/--extra-cflags= '-o3-fpic-dandroid-dhave_sys_uio_h=1- Mfloat-abi=softfp-

are also modified. 6. Connect the phone with the data cable, turn on the USB mode, install the ADB tool sudo apt-get install android-tools-adb 7, go to the SDK directory, my is/home/wangrui/java/sdk/platform-tools Execute command: ADB shellWill go into the phone and execute the command: CD SDcardGo to the SD card and locate the Arm-nginx directory 8. The complete command is: Perform ./nginxPrompt error: Nginx: [alert] could not open error log file:open () "/home/wangrui/arm-nginx/logs/error.

I tried it in version 1.0, and I can support the MP3,MP4,FLV,F4V format. This will be very powerful, you can do music website, do video website. Regardless of the format of the media files in the background of the website, the front end is played by the RTMP protocol with the Flash player. If it's live, use Adobe's own flash live Media Encoder to do provider Red5 uses Java development, function and performance are very good, personally feel no less

Battle the people of Russia to develop a nginx streaming plug-in, in addition to live streaming audio and video streaming server with the common functions of the media such as push-pull streaming media resources Flv/mp4 VOD on-demand based on HTTP HLS (HTTP Live streaming) M3U8 's support HTTP-based operations (publish, play, record) Can work well with existing streaming media servers and players Online call FFmpeg streaming media for transcoding H264/AAC audio

,//7 Mixedarray = 08,//8 end = 09,//9 Array = 10 ,//Date = 11,//longstring = 12,//Typeasobject = 13,//Recordset = 14,//-Xml = 15,//Typedobject = 16, Amf3data = 17// RTMP the data in the protocol is big-end, so the data should be turned into big-endian form before putting the data. Number type is actually double, accounting for 8bytes. For example: 00 00 00 00 00 00 00 00, the first byte is the AMF type, and then the 8bytes is the double value 0.0.

See such a post in the water Wood community: http://www.newsmth.net/nForum/#!article/VideoCoding/26094 The answer is as follows: "In Ohcabinet's masterpiece, he mentions:" Flash Video content delivered using HTTP lacks intellectual property (IP) protection. There is no-to-limit offline access to Flash video content served over http:the Web browser caches Flash video files In its temporary storage, which can is accessed by the user after browsing the site. As such, content producers and lic

Installing Virtual-audio-capturerDownload Address Https://github.com/jiao808813/virtual-audio-capture-grabber-deviceCan download the source code to compile itself, can also download the installation package directly installed Record screen and record and push the stream command line as follows Ffmpeg-f dshow-i audio= "Microphone (Realtek high Definition Au"-F dshow-i audio= "Virtual-audio-capturer"-filter_complex Amix =inputs=2:duration=first:dropout_transition=2-f gdigrab-i Desktop -vcodec l

= "$LD _library_path:/usr/local/ffmpeg/lib/" souce/etc/profile 4. Download opencv2.4.13, after decompression into the directory, the author is using opencv-2.4.13, execute the following command CD ~/opencv-2.4.13/ mkdir build CD The following make command because the author installs the Cuda, therefore inside uses cuda_generation=auto this compilation option, does not use Cuda compiles the child shoe to be possible to delete this piece configuration to compile. cmake-d cmake_build_type=release

To achieve network playback, find the protocol and find the open source file. Finally find a usable I was shocked when the broadcast came! This is too easy. The steps are as follows:   1, download live555, URL: http://www.live555.com/liveMedia/ 2

Use the library in live555 to write a simple rtspserver program for learning purposes only. The Code in the following example shows the function call process of the rtspserver. # Include # include "livemedia. HH "static servermediasession *

http://ubuntuforums.org/showthread.php?t=1544946Http://www.live555.com/liveCaster/liveInput.htmlhttps://forum.videolan.org/viewtopic.php?t=109421http://stackoverflow.com/questions/22680558/take-input-from-a-live-source-instead-of-from-a-file-in-live5

Http://blog.csdn.net/taixinlfx/article/details/7557103 To achieve network playback, find the protocol and find the open source file. Finally find a usable I was shocked when the broadcast came! This is too easy. The steps are as follows:  

using system; using system. collections. generic; using system. LINQ; using system. text; using system. security. cryptography; namespace rtsp_digest_authentication {class program {// /// MD5 32-bit encryption /// /// /// static string MD5

Implementation of Some RTSPSource code Name side? Lang URL-----------------------------------------------------------------------* Live.com C/s c ++ http://www.live555.com* Darwins C ++ http://www.opensource.apple.com/darwinsource/* Vlc c

I. H264 Basic Concepts 1.1, NAL, slice and frame introduction and interrelated NAL refers to the network extraction layer, which put some network-related information. Slice is the meaning of the film, 264 in the image into a frame (frame) or two

After receiving the 34bytes sent by the client, remove the 12bytes header and then use 22bytesDeserialize to decode [cpp] switch (H_MT (header )) {... case when: {message [RM_INVOKE] [RM_INVOKE_IS_FLEX] = (bool) (H_MT (header) =

Remove the 12bytes Header [cpp] bool RTMPProtocolSerializer: Deserialize (header & header, IOBuffer & buffer, Variant & message) {switch (H_MT (Header )) {case when: {return DeserializeUsrCtrl (buffer, message [RM_USRCTRL]) ;}} [cpp] bool

said in front. See a lot of people reprint this article, but certainly compile does not pass, oneself need to add some header file. In fact, for me, I just need to compile from 264 to FLV (vacant audio part does not need), so this code I did not

HLS: value on|off switch HLS hls_path: Sets the position of the playlist (m3u8) and the media block Hls_fragment: followed by time to set the size of each block. The default is 5 seconds. can only be integer hls_playlist_length: Sets the length of

HLS: Value On|off Toggle HLS hls_path: Sets the location of the playlist (m3u8) and the media block Hls_fragment: After time, used to set the size of each block. The default is 5 seconds. can only be integer hls_playlist_length: Set the length of