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  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

  • 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 (...)

  • Organiser par catégorie

    17 mai 2013, par

    Dans MédiaSPIP, une rubrique a 2 noms : catégorie et rubrique.
    Les différents documents stockés dans MédiaSPIP peuvent être rangés dans différentes catégories. On peut créer une catégorie en cliquant sur "publier une catégorie" dans le menu publier en haut à droite ( après authentification ). Une catégorie peut être rangée dans une autre catégorie aussi ce qui fait qu’on peut construire une arborescence de catégories.
    Lors de la publication prochaine d’un document, la nouvelle catégorie créée sera proposée (...)

Sur d’autres sites (7299)

  • FFMPEG - Stream discovered after head already parsed [on hold]

    7 février 2014, par John Doe

    I am trying to live transcode an RTMP stream to another RTMP HLS stream using the following command :

    ffmpeg -re -i rtmp://localhost/videochat/testing -c:v libx264 -c:a:0 libfaac -b:a:0 480k -f flv rtmp://localhost:12345/hls/mystream;

    However I receive the following error and the transcoding never begins :

    ffmpeg version git-2014-02-06-474db7a Copyright (c) 2000-2014 the FFmpeg developers
    built on Feb  6 2014 22:20:14 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
    configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --               enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3
    libavutil      52. 63.100 / 52. 63.100
    libavcodec     55. 49.101 / 55. 49.101
    libavformat    55. 30.100 / 55. 30.100
    libavdevice    55.  7.100 / 55.  7.100
    libavfilter     4.  1.102 /  4.  1.102
    libswscale      2.  5.101 /  2.  5.101
    libswresample   0. 17.104 /  0. 17.104
    libpostproc    52.  3.100 / 52.  3.100
    Metadata:
    description           Chat using VideoChat example.
    [flv @ 0x1ec89e0] Stream discovered after head already parsed
    ^C[flv @ 0x1ec89e0] Could not find codec parameters for stream 0 (Video: none):    unspecified size
    Consider increasing the value for the 'analyzeduration' and 'probesize' options
    Input #0, flv, from 'rtmp://localhost/videochat/testing':
    Metadata:
    description     : Chat using VideoChat ex   ?5P
    Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: none, 1k tbr, 1k tbn, 1k tbc
    Stream #0:1: Data: none
    Codec AVOption b (set bitrate (in bits/s)) specified for output file #0 (rtmp://localhost:12345/hls/mystream) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
    Output #0, flv, to 'rtmp://localhost:12345/hls/mystream':

    If anyone has ever dealt/solved this problem before, can you please share as I have been trying to solve this for 2 days but to no avail !

  • Recording screencast using ffmpeg produces audio that is out of sync with the video

    7 décembre 2014, par AgilE

    I’m on Fedora 20. I got it pre-compiled from the repo.

    I use the follow command to record screencasts :

    ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1366x768 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 -threads 0 filename.mkv

    The audio is perfectly synced with the video at the beginning. However, as the video progresses, the audio starts lagging behind severely (say a lag of more than 4 seconds). I’ve included a sample output that ffmpeg shows while recording a video.

    ffmpeg version 2.1.3 Copyright (c) 2000-2013 the FFmpeg developers
     built on Jan 25 2014 15:11:42 with gcc 4.8.2 (GCC) 20131212 (Red Hat 4.8.2-7)
     configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
     libavutil      52. 48.101 / 52. 48.101
     libavcodec     55. 39.101 / 55. 39.101
     libavformat    55. 19.104 / 55. 19.104
     libavdevice    55.  5.100 / 55.  5.100
     libavfilter     3. 90.100 /  3. 90.100
     libavresample   1.  1.  0 /  1.  1.  0
     libswscale      2.  5.101 /  2.  5.101
     libswresample   0. 17.104 /  0. 17.104
     libpostproc    52.  3.100 / 52.  3.100
    Guessed Channel Layout for  Input Stream #0.0 : stereo
    Input #0, alsa, from 'pulse':
     Duration: N/A, start: 1391533629.323015, bitrate: 1536 kb/s
       Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    [x11grab @ 0x153ad80] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 1366 height: 768
    [x11grab @ 0x153ad80] shared memory extension found
    Input #1, x11grab, from ':0.0':
     Duration: N/A, start: 1391533629.365274, bitrate: 1007124 kb/s
       Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768, 1007124 kb/s, 30 tbr, 1000k tbn, 30 tbc
    [swscaler @ 0x1520460] deprecated pixel format used, make sure you did set range correctly
    No pixel format specified, yuv444p for H.264 encoding chosen.
    Use -pix_fmt yuv420p for compatibility with outdated media players.
    [libx264 @ 0x1551420] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
    [libx264 @ 0x1551420] profile High 4:4:4 Predictive, level 3.2, 4:4:4 8-bit
    [libx264 @ 0x1551420] 264 - core 138 r2363 c628e3b - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=0 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=0 chroma_qp_offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=cqp mbtree=0 qp=0
    Output #0, matroska, to 'testing.mkv':
     Metadata:
       encoder         : Lavf55.19.104
       Stream #0:0: Video: h264 (libx264) (H264 / 0x34363248), yuv444p, 1366x768, q=-1--1, 1k tbn, 30 tbc
       Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
    Stream mapping:
     Stream #1:0 -> #0:0 (rawvideo -> libx264)
     Stream #0:0 -> #0:1 (pcm_s16le -> pcm_s16le)
    Press [q] to stop, [?] for help
    [swscaler @ 0x1520460] Warning: data is not aligned! This can lead to a speedloss
    frame=  213 fps= 27 q=-1.0 Lsize=    3572kB time=00:00:08.00 bitrate=3653.6kbits/s    
    video:2025kB audio:1501kB subtitle:0 global headers:0kB muxing overhead 1.291094%
    [libx264 @ 0x1551420] frame I:1     Avg QP: 0.00  size:692585
    [libx264 @ 0x1551420] frame P:212   Avg QP: 0.00  size:  6509
    [libx264 @ 0x1551420] mb I  I16..4: 100.0%  0.0%  0.0%
    [libx264 @ 0x1551420] mb P  I16..4: 47.2%  0.0%  0.0%  P16..4:  0.1%  0.0%  0.0%  0.0%  0.0%    skip:52.8%
    [libx264 @ 0x1551420] coded y,u,v intra: 0.7% 0.7% 0.7% inter: 0.1% 0.1% 0.1%
    [libx264 @ 0x1551420] i16 v,h,dc,p: 100%  0%  0%  0%
    [libx264 @ 0x1551420] kb/s:2335.32

    Is there something I’m missing ?

  • Green tint color shift converting RGBA to YUV420p ffmpeg libavcodec

    2 août 2017, par Michael B

    I was wondering someone could help. I’m currently taking converting RBGA bitmaps to YUV420p before encoding the frames with h264 codec and dumping the encoded packets to file successfully.

    I’m also able to playback the h264 video file in VLC. However, there seems to be a color shift which is more apparent where grey is actually light tint of screen. I’m used a an animation video as my example which is supposed to show a blue sky in the background, however after converting the image, I’m getting an orange sky background. Do you have any ideas how I can fix this please ?

    Do you happen to know if it’s possible to convert rgba to yuv BEFORE calling sws_scale and before encoding packet ?