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  • Submit bugs and patches

    13 avril 2011

    Unfortunately a software is never perfect.
    If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
    If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
    You may also (...)

  • (Dés)Activation de fonctionnalités (plugins)

    18 février 2011, par

    Pour gérer l’ajout et la suppression de fonctionnalités supplémentaires (ou plugins), MediaSPIP utilise à partir de la version 0.2 SVP.
    SVP permet l’activation facile de plugins depuis l’espace de configuration de MediaSPIP.
    Pour y accéder, il suffit de se rendre dans l’espace de configuration puis de se rendre sur la page "Gestion des plugins".
    MediaSPIP est fourni par défaut avec l’ensemble des plugins dits "compatibles", ils ont été testés et intégrés afin de fonctionner parfaitement avec chaque (...)

  • Le plugin : Podcasts.

    14 juillet 2010, par

    Le problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
    Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
    Types de fichiers supportés dans les flux
    Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...)

Sur d’autres sites (6039)

  • Reduce size of APK when including .so files

    13 septembre 2014, par Arslan

    I am using a 3rd party API for camera library that is using ffmpeg frame recorder and .so files along with javacv.

    I am also using a 3rd party library for extracting meta data out from a video which also has some .so files

    When i merged all these files into only one folder "armeabi" my application did not work.
    So I have to copy all these files to all other folders "armeabi-v7a, mips, x86". Which works perfectly fine but obviously the size of the apk is now too large.

    Total size of these file are 20MB per folder. That make 80MB for all folder. My .apk size is 41MB. Please suggest ma what I can do to remove duplicate files/folders or reduce .apk size.

    Why I need all these folders armeabi, armeabi-v7a, mips, x86
    enter image description here

    .so files in folder
    enter image description here

  • ffmpeg can't stream to remote client

    4 septembre 2014, par KFL

    I’m building a simple ffmpeg command line on my laptop to stream from its camera. The command line reads (verbose output at the botton) :

    host1> ffmpeg -v verbose \
                 -f dshow \
                 -i video="Camera":audio="Microphone" \
                 -r 30 -g 0 -vcodec h264 -acodec libmp3lame \
                 -tune zerolatency \
                 -preset ultrafast \
                 -f mpegts udp://12.34.56.78:12345

    Firstly, it works locally. I.e., I can view the output by using ffplay on the same host :

    host1> ffplay -hide_banner -v udp://12.34.56.78:12345

    Now what is NOT working is when I do this from another machine in the same network. It shows a nan progress :

    host2> ffplay -hide_banner -v udp://12.34.56.78:12345
       nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0  

    I used ncat to dump the raw content. But there’s no output :

    host2>\ncat\ncat -v -u 12.34.56.78 12345
    Ncat: Version 5.59BETA1 ( http://nmap.org/ncat )
    Ncat: Connected to 12.34.56.78:12345.
    (...and nothing happen...)

    Note that I can exclude firewall issues as I used ncat to communicate with each other across the wire using the same port and protocol (UDP). This works and they can chat to each other :

    host1> ncat -l -u -p 12345
    host2> ncat -u 12.34.56.78 12345

    Any hint ?

    I’m using Windows x64 with FFMPEG 64bit installed from here. Below is the Output of my ffmpeg command :

    C:\ffmpeg\bin>ffmpeg -v verbose -f dshow -i video="Integrated Camera":audio="Microphone (Realtek High Definition Audio)" -r 30 -g 0 -vcodec h264 -acodec libmp3lame -tune zerolatency -preset ultrafast -f mpegts udp://12.34.56.78:12345
    ffmpeg version N-66012-g97b8809 Copyright (c) 2000-2014 the FFmpeg developers
     built on Sep  1 2014 00:21:15 with gcc 4.8.3 (GCC)
     configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug -enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
     libavutil      54.  7.100 / 54.  7.100
     libavcodec     56.  1.100 / 56.  1.100
     libavformat    56.  3.100 / 56.  3.100
     libavdevice    56.  0.100 / 56.  0.100
     libavfilter     5.  0.103 /  5.  0.103
     libswscale      3.  0.100 /  3.  0.100
     libswresample   1.  1.100 /  1.  1.100
     libpostproc    53.  0.100 / 53.  0.100
    Guessed Channel Layout for  Input Stream #0.1 : stereo
    Input #0, dshow, from 'video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)':
     Duration: N/A, start: 171840.657000, bitrate: N/A
       Stream #0:0: Video: rawvideo, bgr24, 640x480, 30 fps, 30 tbr, 10000k tbn, 30 tbc
       Stream #0:1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
    Matched encoder 'libx264' for codec 'h264'.
    [graph 0 input from stream 0:0 @ 0000000000470aa0] w:640 h:480 pixfmt:bgr24 tb:1/10000000 fr:10000000/333333 sar:0/1 sws_param:flags=2
    [auto-inserted scaler 0 @ 0000000004326d00] w:iw h:ih flags:'0x4' interl:0
    [format @ 0000000004325a00] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format'
    [auto-inserted scaler 0 @ 0000000004326d00] w:640 h:480 fmt:bgr24 sar:0/1 -> w:640 h:480 fmt:yuv444p sar:0/1 flags:0x4
    No pixel format specified, yuv444p for H.264 encoding chosen.
    Use -pix_fmt yuv420p for compatibility with outdated media players.
    [graph 1 input from stream 0:1 @ 0000000000460c20] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
    [audio format for output stream 0:1 @ 00000000004601a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
    [auto-inserted resampler 0 @ 00000000004604a0] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
    [libx264 @ 000000000081bb20] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
    [libx264 @ 000000000081bb20] profile High 4:4:4 Intra, level 3.0, 4:4:4 8-bit
    [mpegts @ 000000000081abe0] muxrate VBR, pcr every 3 pkts, sdt every 200, pat/pmt every 40 pkts
    Output #0, mpegts, to 'udp://12.34.56.78:12345':
     Metadata:
       encoder         : Lavf56.3.100
       Stream #0:0: Video: h264 (libx264), yuv444p, 640x480, q=-1--1, 30 fps, 90k tbn, 30 tbc
       Metadata:
         encoder         : Lavc56.1.100 libx264
       Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
       Metadata:
         encoder         : Lavc56.1.100 libmp3lame
    Stream mapping:
     Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
     Stream #0:1 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help
    *** 1 dup!
    frame=  241 fps= 31 q=28.0 Lsize=    3439kB time=00:00:08.03 bitrate=3506.4kbits/s dup=1 drop=0
    video:3035kB audio:125kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.791966%
    Input file #0 (video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)):
     Input stream #0:0 (video): 240 packets read (221184000 bytes); 240 frames decoded;
     Input stream #0:1 (audio): 16 packets read (1411200 bytes); 16 frames decoded (352800 samples);
     Total: 256 packets (222595200 bytes) demuxed
    Output file #0 (udp://12.34.56.78:12345):
     Output stream #0:0 (video): 241 frames encoded; 241 packets muxed (3108187 bytes);
     Output stream #0:1 (audio): 306 frames encoded (352512 samples); 307 packets muxed (128313 bytes);
     Total: 548 packets (3236500 bytes) muxed
    [libx264 @ 000000000081bb20] frame I:241   Avg QP:27.97  size: 12897
    [libx264 @ 000000000081bb20] mb I  I16..4: 100.0%  0.0%  0.0%
    [libx264 @ 000000000081bb20] coded y,u,v intra: 26.3% 0.5% 0.0%
    [libx264 @ 000000000081bb20] i16 v,h,dc,p: 19% 28% 21% 31%
    [libx264 @ 000000000081bb20] kb/s:3095.29
    [dshow @ 0000000000467720] real-time buffer[Integrated Camera] too full (90% of size: 3041280)! frame dropped!
    Received signal 2: terminating. (I pressed CTRL-C)
  • Timelapse (1/6 fps) from slo-mo (240 fps) with ffmpeg

    4 avril 2022, par baskak

    I recorded slo-mo video on an iPhone SE (2) by mistake instead of timelapse.

    


    I know there's a lot of answers to this question here, but I'm trying again and again and always something's wrong (like a video that has a correct total no. of frames, but lasts 3 hours and is basically a freeze :D )
My recent command was

    


    


    ffmpeg -i IMG_2174.MOV -vf framestep=1440,setpts=N/120/TB -c:v libx264
-preset slow -crf 22 -an -r 30 IMG_2174.timelapse.MOV

    


    


    but it resulted in a one-second-long video, so way over-timelapsed. Should be several seconds IINM. The source video is 30 minutes long @240fps, 17GB.
Thx.