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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (12562)

  • avformat/hls : do not iterate to next sequence number if interruption is requested

    26 juin 2015, par schenk michael
    avformat/hls : do not iterate to next sequence number if interruption is requested

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
    • [DH] libavformat/hls.c

  • avcodec/xpmdec : Use LUT for hex char->number conversion

    6 mars 2024, par Andreas Rheinhardt
    avcodec/xpmdec : Use LUT for hex char->number conversion

    Speeds up decoding and also reduces binary size.
    

    Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
    • [DH] libavcodec/xpmdec.c

  • Problem understanding audio stream number of samples when decoded with ffmpeg

    21 janvier 2021, par Michael Brown

    The two streams I am decoding are an audio stream (adts AAC, 1 channel, 44100, 8-bit, 128bps) and a video stream (H264) which are received in an Mpeg-Ts stream, but I noticed something that doesn't make sense to me when I decode the AAC audio frames and try to line up the audio/video stream timestamps. I'm decoding the PTS for each video and audio frame, however I only get a PTS in the audio stream every 7 frames.

    &#xA;

    When I decode a single audio frame I get back 1024 samples, always. The frame rate is 30fps, so I see 30 frames each with 1024 samples which comes equals 30,720 samples and not the expected 44,100 samples. This is a problem when computing the timeline as the timestamps on the frames are slightly different between the audio and video streams. It's very close, but since I compute the timestamps via (1024 samples * 1,000 / 44,100 * 10,000 ticks) it's never going to line up exactly with the 30fps video.

    &#xA;

    Am I doing something wrong here with decoding the ffmpeg audio frames, or misunderstanding audio samples ?&#xA;And in my particular application, these timestamps are critical as I am trying to line up LTC timestamps which are decoded at the audio frame level, and lining those up with video frames.

    &#xA;

    FFProbe.exe :

    &#xA;

    Video:&#xA;r_frame_rate=30/1      &#xA;avg_frame_rate=30/1    &#xA;codec_time_base=1/60&#xA;time_base=1/90000      &#xA;start_pts=7560698279   &#xA;start_time=84007.758656&#xA;&#xA;Audio:&#xA;r_frame_rate=0/0&#xA;avg_frame_rate=0/0&#xA;codec_time_base=1/44100&#xA;time_base=1/90000&#xA;start_pts=7560686278&#xA;start_time=84007.625311&#xA;

    &#xA;

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