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  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Possibilité de déploiement en ferme

    12 avril 2011, par

    MediaSPIP peut être installé comme une ferme, avec un seul "noyau" hébergé sur un serveur dédié et utilisé par une multitude de sites différents.
    Cela permet, par exemple : de pouvoir partager les frais de mise en œuvre entre plusieurs projets / individus ; de pouvoir déployer rapidement une multitude de sites uniques ; d’éviter d’avoir à mettre l’ensemble des créations dans un fourre-tout numérique comme c’est le cas pour les grandes plate-formes tout public disséminées sur le (...)

Sur d’autres sites (4836)

  • Gaps when recording using MediaRecorder API(audio/webm opus)

    9 août 2018, par Jack Juiceson

    ----- UPDATE HAS BEEN ADDED BELOW -----

    I have an issue with MediaRecorder API (https://www.w3.org/TR/mediastream-recording/#mediarecorder-api).

    I’m using it to record the speech from the web page(Chrome was used in this case) and save it as chunks.
    I need to be able to play it while and after it is recorded, so it’s important to keep those chunks.

    Here is the code which is recording data :

    navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(function(stream) {
     recorder = new MediaRecorder(stream, { mimeType: 'audio/webm; codecs="opus"' })
     recorder.ondataavailable = function(e) {
       // Read blob from `e.data`, decode64 and send to sever;
     }
     recorder.start(1000)
    })

    The issue is that the WebM file which I get when I concatenate all the parts is corrupted(rarely) !. I can play it as WebM, but when I try to convert it(ffmpeg) to something else, it gives me a file with shifted timings.

    For example. I’m trying to convert a file which has duration 00:36:27.78 to wav, but I get a file with duration 00:36:26.04, which is 1.74s less.

    At the beginning of file - the audio is the same, but after about 10min WebM file plays with a small delay.

    After some research, I found out that it also does not play correctly with the browser’s MediaSource API, which I use for playing the chunks. I tried 2 ways of playing those chunks :

    In a case when I just merge all the parts into a single blob - it works fine.
    In case when I add them via the sourceBuffer object, it has some gaps (i can see them by inspecting buffered property).
    697.196 - 697.528 ( 330ms)
    996.198 - 996.754 ( 550ms)
    1597.16 - 1597.531 ( 370ms)
    1896.893 - 1897.183 ( 290ms)

    Those gaps are 1.55s in total and they are exactly in the places where the desync between wav & webm files start. Unfortunately, the file where it is reproducible cannot be shared because it’s customer’s private data and I was not able to reproduce such issue on different media yet.

    What can be the cause for such an issue ?

    ----- UPDATE -----
    I was able to reproduce the issue on https://jsfiddle.net/96uj34nf/4/

    In order to see the problem, click on the "Print buffer zones" button and it will display time ranges. You can see that there are two gaps :
    0 - 136.349, 141.388 - 195.439, 197.57 - 198.589

    1. 136.349 - 141.388
    2. 195.439 - 197.57

    So, as you can see there are 5 and 2 second gaps. Would be happy if someone could shed some light on why it is happening or how to avoid this issue.

    Thank you

  • avcodec/mediacodecdec : fix SEGV on modern nvidia decoders

    31 juillet 2018, par Aman Gupta
    avcodec/mediacodecdec : fix SEGV on modern nvidia decoders
    

    This code came originally from gstreamer, where it was added in [1]
    as a work-around for the Tegra 3. (The alignment was changed in [2]
    as a response to [3], from 32-bit to 16-bit).

    gstreamer only used this workaround in the case where the decoder
    didn't return a slice-height property, but when the code was copied
    into avcodec the conditional got lost. This commit restores the guard
    and prefers the slice-height from the decoder when it is available.

    This fixes segfaults decoding 1920x1080 h264 and mpeg2 videos on the
    NVidia SHIELD after upgrading to Android Oreo.

    [1] https://github.com/GStreamer/gst-plugins-bad/commit/a870e6a5c30dd85240fe75c7409cc1cf1b86541d
    [2] https://github.com/GStreamer/gst-plugins-bad/commit/21ff3ae0b0127bd82951d278ca24f2d54133b7cd
    [3] https://bugzilla.gnome.org/show_bug.cgi?id=748867

    Signed-off-by : Aman Gupta <aman@tmm1.net>

    • [DH] libavcodec/mediacodecdec_common.c
  • Updated(reproducible) - Gaps when recording using MediaRecorder API(audio/webm opus)

    25 mars 2019, par Jack Juiceson

    ----- UPDATE HAS BEEN ADDED BELOW -----

    I have an issue with MediaRecorder API (https://www.w3.org/TR/mediastream-recording/#mediarecorder-api).

    I’m using it to record the speech from the web page(Chrome was used in this case) and save it as chunks.
    I need to be able to play it while and after it is recorded, so it’s important to keep those chunks.

    Here is the code which is recording data :

    navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(function(stream) {
     recorder = new MediaRecorder(stream, { mimeType: 'audio/webm; codecs="opus"' })
     recorder.ondataavailable = function(e) {
       // Read blob from `e.data`, decode64 and send to sever;
     }
     recorder.start(1000)
    })

    The issue is that the WebM file which I get when I concatenate all the parts is corrupted(rarely) !. I can play it as WebM, but when I try to convert it(ffmpeg) to something else, it gives me a file with shifted timings.

    For example. I’m trying to convert a file which has duration 00:36:27.78 to wav, but I get a file with duration 00:36:26.04, which is 1.74s less.

    At the beginning of file - the audio is the same, but after about 10min WebM file plays with a small delay.

    After some research, I found out that it also does not play correctly with the browser’s MediaSource API, which I use for playing the chunks. I tried 2 ways of playing those chunks :

    In a case when I just merge all the parts into a single blob - it works fine.
    In case when I add them via the sourceBuffer object, it has some gaps (i can see them by inspecting buffered property).
    697.196 - 697.528 ( 330ms)
    996.198 - 996.754 ( 550ms)
    1597.16 - 1597.531 ( 370ms)
    1896.893 - 1897.183 ( 290ms)

    Those gaps are 1.55s in total and they are exactly in the places where the desync between wav & webm files start. Unfortunately, the file where it is reproducible cannot be shared because it’s customer’s private data and I was not able to reproduce such issue on different media yet.

    What can be the cause for such an issue ?

    ----- UPDATE -----
    I was able to reproduce the issue on https://jsfiddle.net/96uj34nf/4/

    In order to see the problem, click on the "Print buffer zones" button and it will display time ranges. You can see that there are two gaps :
    0 - 136.349, 141.388 - 195.439, 197.57 - 198.589

    1. 136.349 - 141.388
    2. 195.439 - 197.57

    So, as you can see there are 5 and 2 second gaps. Would be happy if someone could shed some light on why it is happening or how to avoid this issue.

    Thank you