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  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 0.2 is the first MediaSPIP stable release.
    Its official release date is June 21, 2013 and is announced here.
    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 (...)

  • Librairies et binaires spécifiques au traitement vidéo et sonore

    31 janvier 2010, par

    Les logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
    Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
    Binaires complémentaires et facultatifs flvtool2 : (...)

Sur d’autres sites (4860)

  • Video files recorded in Google Chrome have stuttering audio

    4 juin 2018, par maxpaj

    Background

    I’m developing a platform where users can record videos of themselves or their screen and send them as video messages to customers / clients.

    I have limited users to only using my application with Google Chrome and I’m using the MediaRecorder API to record the video data from the users screen or webcamera. The codecs that are used for recording are VP8/OPUS (WEBM container).

    I need the videos to run in as many browsers as possible, so I’m using a 3rd party service to transcode videos from whatever format I’m getting from the users to a H.265/AAC MP4 container (caniuse MPEG-4/H.264).

    Issue

    Lately I’ve seen that some videos recorded on Mac OSX machines have the video and audio out of sync or that the video and audio stutters, depending on which player I’m using. I call these video files corrupt, for lack of a better word. Playing a corrupt file in Google Chrome renders smooth playing audio. Playing the video in VLC on my Windows machine renders stuttering audio.

    When I run the corrupt video files through the transcoding service I get video files with stuttering audio, no matter which player I’m using.

    This is an unwanted result and pretty much unacceptable since the audio needs to be smooth in order for the recipient of a video to not be bothered with the quality.

    Debugging

    According to the transcoding service support, this happens because of their mechanisms that try to sync up the audio and video from the corrupt file :

    Inspecting our encoding logs, I’ve noticed the following kind of
    warnings :

    [2018-05-16 14:08:38.009] [pcm_s16le @ 0x1d608c0] pcm_encode_frame :
    filling in for 5856 missing samples (122 ms) before pts 40800 to
    correct sync ! [2018-05-16 14:08:38.009] [pcm_s16le @ 0x1d608c0]
    pcm_encode_frame : dropping 2880 samples (60 ms) at pts 43392 to help
    correct sync to -3168 samples (-66 ms) !

    The problem here comes from the way that the audio in the original
    source file is encoded.

    -

    you should ensure that the audio is not out of sync (audio timestamps
    are correct) in your source file before submitting the job

    Running a corrupt file through ffmpeg on my own machine, re-encoding with the same codecs, produces the same kind of stuttering video. The logs produce an alarming amount of errors. Here is a sample of the log output :

    [libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 194.8kbits/s dup=0 drop=5 speed=0.31x
    [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15434, current: 15394; changing to 15434. This may result in incorrect timestamps in the output file.
    [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15434, current: 15414; changing to 15434. This may result in incorrect timestamps in the output file.
    [libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 193.3kbits/s dup=0 drop=5 speed=0.309x
    [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15539, current: 15499; changing to 15539. This may result in incorrect timestamps in the output file.
    [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15539, current: 15519; changing to 15539. This may result in incorrect timestamps in the output file.
    [libopus @ 0000029938e24d80] Queue input is backward in timeitrate= 192.0kbits/s dup=0 drop=5 speed=0.308x
    [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15667, current: 15627; changing to 15667. This may result in incorrect timestamps in the output file.
    [webm @ 0000029938e09b00] Non-monotonous DTS in output stream 0:1; previous: 15667, current: 15647; changing to 15667. This may result in incorrect timestamps in the output file.
    [libopus @ 0000029938e24d80] Queue input is backward in time

    I tried running the same inputs through another transcoding service and those outputs worked a lot better - video was still stuttering but the audio played smoothly, which is more important to the use case of my application.

    To my knowledge, this have so far only occurred for users on Mac OSX machines.

    Questions

    1. Is there anything I can do to have the files work better ? Or is this entirely a consequence of how encoding of video and audio in Google Chrome works ?

    2. One step in the right direction would be to just be able to detect when the video is corrupt. How can I do that ?

  • ffmpeg with AudioUnit

    27 octobre 2012, par deimus

    What I have

    My aim is to parse some media file using ffmpeg and provide video and audio playback. Which I do successfully using the OpenGL for video and AudioQueue for audio.

    What I need to do
    I need to change AudioQueue to Audio Unit service, because it does provide several nasty features for Audio manipulations.

    Basically I'm confused on integration of Audio Units into ffmpeg run loop.
    So would like to have some references/samples from you guys where Audio Unit is intergrated with ffmpeg media playback loop i.e. media packet extraction and its pushing into some buffer which Audio Unit can play.

  • Get "Delay relative to video" value using ffmpeg

    22 avril 2023, par user3449922

    I like to known if is possible detect via ffmpeg the value of property "Delay relative to video" showed here :

    


    Audio #2
ID                                       : 3
Format                                   : AC-3
Format/Info                              : Audio Coding 3
Commercial name                          : Dolby Digital
Codec ID                                 : A_AC3
Duration                                 : 10 min 0 s
Bit rate mode                            : Constant
Bit rate                                 : 320 kb/s
Channel(s)                               : 2 channels
Channel layout                           : L R
Sampling rate                            : 48.0 kHz
Frame rate                               : 31.250 FPS (1536 SPF)
Bit depth                                : 32 bits
Compression mode                         : Lossy
Delay relative to video                  : 1 s 78 ms
Stream size                              : 22.8 MiB (1%)
Language                                 : rom
Service kind                             : Complete Main
Default                                  : No
Forced                                   : No


    


    Thank you !