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  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • Contribute to translation

    13 avril 2011

    You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
    To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
    MediaSPIP is currently available in French and English (...)

Sur d’autres sites (6611)

  • How to merge two audio files while retaining correct timings with ffmpeg

    31 août 2017, par andy

    Disclaimer : I don’t understand ffmpeg and mostly just copy commands

    I have two webm video files. Their duration is off by about a second.

    I converted these to mp4 and everything was fine :

    ffmpeg -acodec libopus -i 1.webm -r 10 -cpu-used 5 -c:v libx264 -crf 20 -c:a aac -strict experimental -loglevel error 1.mp4

    I then extracted the audio from each video and everything was fine. By fine, I mean, when I playback the audio files by themselves they match the timings on the original video. I used this command to extract the audio :

    fmpeg -i 1.mp4 -map 0:a -vn -acodec copy 1audio.m4a

    I now have two audio tracks and I want to combine them into one, I want to "overlay" them, AND I want to keep the timings the same as they were.

    BUT, whenever I try and combine the audio the timings are off. The "seconds" don’t match the original single audio files.

    I have tried these commands to merge the audio :

    ffmpeg -i 1audio.m4a -i 2audio.m4a -filter_complex "[0:a][1:a]amerge=inputs=2[a]" -map [a] -c:a libfdk_aac mergedaudio.m4a

    ffmpeg -i 2audio.m4a -i 1audio.m4a -filter_complex amix=inputs=2:duration=first mergedaudio.m4a

    ffmpeg -i 1audio.m4a -i 2audio.m4a -filter_complex amerge -ac 2 -c:a libfdk_aac -vbr 4 mergedaudio.m4a

    All the above command have resulted in a merged audio file... but the timings are off.

    What am I doing wrong ?

  • lavc/cbs_av1 : restore CodedBitstreamAV1Context when AVERROR(ENOSPC)

    28 septembre 2022, par Haihao Xiang
    lavc/cbs_av1 : restore CodedBitstreamAV1Context when AVERROR(ENOSPC)
    

    The current pbc might be small for an obu frame, so a new pbc is
    required then parse this obu frame again. Because
    CodedBitstreamAV1Context has already been updated for this obu frame, we
    need to restore CodedBitstreamAV1Context, otherwise
    CodedBitstreamAV1Context doesn't match this obu frame when parsing obu
    frame again, e.g. CodedBitstreamAV1Context.order_hint.

    $ ffmpeg -i input.ivf -c:v copy -f null -
    [...]
    [av1_frame_merge @ 0x558bc3d6f880] ref_order_hint[i] does not match
    inferred value : 20, but should be 22.
    [av1_frame_merge @ 0x558bc3d6f880] Failed to write unit 1 (type 6).
    [av1_frame_merge @ 0x558bc3d6f880] Failed to write packet.
    [obu @ 0x558bc3d6e040] av1_frame_merge filter failed to send output
    packet

    Reviewed-by : James Almer <jamrial@gmail.com>
    Reviewed-by : Wenbin Chen <wenbin.chen@intel.com>
    Signed-off-by : Haihao Xiang <haihao.xiang@intel.com>

    • [DH] libavcodec/cbs_av1.c
  • checkasm : updated tests for sw_scale

    13 août 2022, par Swinney, Jonathan
    checkasm : updated tests for sw_scale
    

    Change the reference to exactly match the C reference in swscale,
    instead of exactly matching the x86 SIMD implementations (which
    differs slightly). Test with and without SWS_ACCURATE_RND - if this
    flag isn't set, the output must match the C reference exactly,
    otherwise it is allowed to be off by 2.

    Mark a couple x86 functions as unavailable when SWS_ACCURATE_RND
    is set - apparently this discrepancy hasn't been noticed in other
    exact tests before.

    Add a test for yuv2plane1.

    Signed-off-by : Jonathan Swinney <jswinney@amazon.com>
    Signed-off-by : Martin Storsjö <martin@martin.st>

    • [DH] libswscale/x86/swscale.c
    • [DH] tests/checkasm/sw_scale.c