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Autres articles (93)

  • 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 (...)

  • 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 (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (10107)

  • Twitter gives error on ffmpeg generated video “The media you tried to upload was invalid.”

    9 mai 2018, par TrooPHP Developer

    ffmpeg generated video gives error while sharing on twitter. the error is :

    “Cannot read property ‘code’ of undefined”

    I am generating video from audio.
    my sample code Example is :

    ffmpeg -i audio.webm -i image.png -vcodec libx264 -pix_fmt yuv420p -strict -2 -acodec aac video.mp4

    I am directly trying to upload generated video to twitter website and video size is just 6 seconds.

  • Live stream the screen and microphone to azure media services with ffmpeg

    16 octobre 2020, par Panzer Ihnen

    I'm able to stream a video of my screen to Azure Media Services with ffmpeg, but when I add the audio input, the stream stops with this error :

    


    ffmpeg: frame=   27 fps=0.0 q=0.0 size=       0kB time=00:00:00.36 bitrate=   8.4kbits/s dup=0 drop=7 speed=0.703x
ffmpeg: frame=   43 fps= 42 q=0.0 size=       0kB time=00:00:00.36 bitrate=   8.4kbits/s dup=0 drop=7 speed=0.356x    
ffmpeg: frame=   58 fps= 38 q=27.0 size=      37kB time=00:00:00.36 bitrate= 818.6kbits/s dup=0 drop=7 speed=0.24x    
ffmpeg: av_interleaved_write_frame(): End of file
ffmpeg:     Last message repeated 1 times
ffmpeg: [flv @ 0633fec0] Failed to update header with correct duration.
ffmpeg: [flv @ 0633fec0] Failed to update header with correct filesize.
ffmpeg: Error writing trailer of rtmp://rtmpurl/stream: End of file
ffmpeg: frame=   59 fps= 32 q=26.0 Lsize=      37kB time=00:00:00.39 bitrate= 764.5kbits/s dup=0 drop=7 speed=0.214x    
ffmpeg: video:35kB audio:2kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
ffmpeg: [libx264 @ 0633c6c0] frame I:1     Avg QP:18.27  size: 30491
ffmpeg: [libx264 @ 0633c6c0] frame P:16    Avg QP:20.68  size:  4210
ffmpeg: [libx264 @ 0633c6c0] frame B:42    Avg QP:23.25  size:   152
ffmpeg: [libx264 @ 0633c6c0] consecutive B-frames:  5.1%  0.0%  0.0% 94.9%
ffmpeg: [libx264 @ 0633c6c0] mb I  I16..4: 12.7% 36.9% 50.4%
Ffmpeg Exited
ffmpeg: [libx264 @ 0633c6c0] mb P  I16..4:  0.4%  1.0%  0.6%  P16..4: 45.6% 10.5% 12.1%  0.0%  0.0%    skip:29.8%
ffmpeg: [libx264 @ 0633c6c0] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8: 15.5%  0.3%  0.0%  direct: 0.3%  skip:83.9%  L0:46.5% L1:51.0% BI: 2.5%
ffmpeg: [libx264 @ 0633c6c0] 8x8 transform intra:41.1% inter:59.1%
ffmpeg: [libx264 @ 0633c6c0] coded y,uvDC,uvAC intra: 79.9% 96.5% 88.6% inter: 7.8% 15.8% 5.4%
ffmpeg: [libx264 @ 0633c6c0] i16 v,h,dc,p: 20% 13%  6% 61%
ffmpeg: [libx264 @ 0633c6c0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 29% 21% 22%  4%  4%  5%  3%  3%  8%
ffmpeg: [libx264 @ 0633c6c0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 38% 16% 10%  5%  7%  7%  5%  6%  6%
ffmpeg: [libx264 @ 0633c6c0] i8c dc,h,v,p: 54% 18% 23%  5%
ffmpeg: [libx264 @ 0633c6c0] Weighted P-Frames: Y:0.0% UV:0.0%
ffmpeg: [libx264 @ 0633c6c0] ref P L0: 55.1%  2.9% 29.5% 12.6%
ffmpeg: [libx264 @ 0633c6c0] ref B L0: 79.5% 14.0%  6.6%
ffmpeg: [libx264 @ 0633c6c0] ref B L1: 93.2%  6.8%
ffmpeg: [libx264 @ 0633c6c0] kb/s:333.49
ffmpeg: Conversion failed!


    


    My ffmpeg command is :

    


    -f dshow -i audio="Microphone device" -strict -2 -c:a aac -f gdigrab -framerate 30 -offset_x 161 -offset_y 203 -video_size 430x322 -show_region 1 -i title="Window Title" -b:v 415K -g 60 -keyint_min 60 -b:a 32K -ar 22050 -filter:a "volume=0.8" -pix_fmt yuv420p -c:v libx264 -preset medium -bufsize 400k -maxrate 400k -f flv rtmp://rtmpurl/stream


    


  • HLS : How to detect out of order segments in media playlist ?

    27 juin 2018, par anirudh612

    What would be an efficient way to detect if an http live streaming VOD playlist has segments out of order (and count how many segments are out of order) ? They are ordered correctly based on the #EXT-X-PROGRAM-DATETIME tag but the segment decoding timestamps in some cases are out of order. Currently, the workflow I’m following is :

    1. Convert the HLS stream into an mp4 using ffmpeg :

      ffmpeg -i http://localhost:8080/test/unsorted.m3u8 -c copy -bsf:a aac_adtstoasc test/unsorted.mp4 &> test/unsorted_ffmpeg.log

    2. Inspect the logs and count number of occurrences of "Non-monotonous DTS in output stream" log messages :

      [mp4 @ 0x7fe74f01b000] Non-monotonous DTS in output stream 0:1 ; previous : 12063760, current : 11866128 ; changing to 12063761. This may result in incorrect timestamps in the output file.

      However, this requires downloading and reading all of the ts segments and is an expensive operation. Is there a more efficient way to determine out of order DTS or PTS in chunks using ffmpeg or ffprobe ?

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