基于ffmpeg的rtmp推流小工具，采用think做GUI，直接python运行命令行调用ffmpeg.zip

# NGINX-based Media Streaming Server ## nginx-rtmp-module ### Project blog http://nginx-rtmp.blogspot.com ### Wiki manual https://github.com/arut/nginx-rtmp-module/wiki/Directives ### Google group https://groups.google.com/group/nginx-rtmp https://groups.google.com/group/nginx-rtmp-ru (Russian) ### Donation page (Paypal etc) http://arut.github.com/nginx-rtmp-module/ ### Features * RTMP/HLS/MPEG-DASH live streaming * RTMP Video on demand FLV/MP4, playing from local filesystem or HTTP * Stream relay support for distributed streaming: push & pull models * Recording streams in multiple FLVs * H264/AAC support * Online transcoding with FFmpeg * HTTP callbacks (publish/play/record/update etc) * Running external programs on certain events (exec) * HTTP control module for recording audio/video and dropping clients * Advanced buffering techniques to keep memory allocations at a minimum level for faster streaming and low memory footprint * Proved to work with Wirecast, FMS, Wowza, JWPlayer, FlowPlayer, StrobeMediaPlayback, ffmpeg, avconv, rtmpdump, flvstreamer and many more * Statistics in XML/XSL in machine- & human- readable form * Linux/FreeBSD/MacOS/Windows ### Build cd to NGINX source directory & run this: ./configure --add-module=/path/to/nginx-rtmp-module make make install Several versions of nginx (1.3.14 - 1.5.0) require http_ssl_module to be added as well: ./configure --add-module=/path/to/nginx-rtmp-module --with-http_ssl_module For building debug version of nginx add `--with-debug` ./configure --add-module=/path/to-nginx/rtmp-module --with-debug [Read more about debug log](https://github.com/arut/nginx-rtmp-module/wiki/Debug-log) ### Windows limitations Windows support is limited. These features are not supported * execs * static pulls * auto_push ### RTMP URL format rtmp://rtmp.example.com/app[/name] app - should match one of application {} blocks in config name - interpreted by each application can be empty ### Multi-worker live streaming Module supports multi-worker live streaming through automatic stream pushing to nginx workers. This option is toggled with rtmp_auto_push directive. ### Example nginx.conf rtmp { server { listen 1935; chunk_size 4000; # TV mode: one publisher, many subscribers application mytv { # enable live streaming live on; # record first 1K of stream record all; record_path /tmp/av; record_max_size 1K; # append current timestamp to each flv record_unique on; # publish only from localhost allow publish 127.0.0.1; deny publish all; #allow play all; } # Transcoding (ffmpeg needed) application big { live on; # On every pusblished stream run this command (ffmpeg) # with substitutions: $app/${app}, $name/${name} for application & stream name. # # This ffmpeg call receives stream from this application & # reduces the resolution down to 32x32. The stream is the published to # 'small' application (see below) under the same name. # # ffmpeg can do anything with the stream like video/audio # transcoding, resizing, altering container/codec params etc # # Multiple exec lines can be specified. exec ffmpeg -re -i rtmp://localhost:1935/$app/$name -vcodec flv -acodec copy -s 32x32 -f flv rtmp://localhost:1935/small/${name}; } application small { live on; # Video with reduced resolution comes here from ffmpeg } application webcam { live on; # Stream from local webcam exec_static ffmpeg -f video4linux2 -i /dev/video0 -c:v libx264 -an -f flv rtmp://localhost:1935/webcam/mystream; } application mypush { live on; # Every stream published here # is automatically pushed to # these two machines push rtmp1.example.com; push rtmp2.example.com:1934; } application mypull { live on; # Pull all streams from remote machine # and play locally pull rtmp://rtmp3.example.com pageUrl=www.example.com/index.html; } application mystaticpull { live on; # Static pull is started at nginx start pull rtmp://rtmp4.example.com pageUrl=www.example.com/index.html name=mystream static; } # video on demand application vod { play /var/flvs; } application vod2 { play /var/mp4s; } # Many publishers, many subscribers # no checks, no recording application videochat { live on; # The following notifications receive all # the session variables as well as # particular call arguments in HTTP POST # request # Make HTTP request & use HTTP retcode # to decide whether to allow publishing # from this connection or not on_publish http://localhost:8080/publish; # Same with playing on_play http://localhost:8080/play; # Publish/play end (repeats on disconnect) on_done http://localhost:8080/done; # All above mentioned notifications receive # standard connect() arguments as well as # play/publish ones. If any arguments are sent # with GET-style syntax to play & publish # these are also included. # Example URL: # rtmp://localhost/myapp/mystream?a=b&c=d # record 10 video keyframes (no audio) every 2 minutes record keyframes; record_path /tmp/vc; record_max_frames 10; record_interval 2m; # Async notify about an flv recorded on_record_done http://localhost:8080/record_done; } # HLS # For HLS to work please create a directory in tmpfs (/tmp/hls here) # for the fragments. The directory contents is served via HTTP (see # http{} section in config) # # Incoming stream must be in H264/AAC. For iPhones use baseline H264 # profile (see ffmpeg example). # This example creates RTMP stream from movie ready for HLS: # # ffmpeg -loglevel verbose -re -i movie.avi -vcodec libx264 # -vprofile baseline -acodec libmp3lame -ar 44100 -ac 1 # -f flv rtmp://localhost:1935/hls/movie # # If you need to transcode live stream use 'exec' feature. # application hls { live on; hls on; hls_path /tmp/hls; } # MPEG-DASH is similar to HLS application dash { live on; dash on; dash_path /tmp/dash; } } } # HTTP can be used for accessing RTMP stats http { server { listen 8080; # This URL provides RTMP statistics in XML location /stat {