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    21 juin 2013, par

    Présentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
    Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
    Vous pouvez personnaliser le formulaire de création d’une actualité.
    Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...)

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Sur d’autres sites (6158)

  • Transcode HLS Segments individually using FFMPEG

    27 mai 2013, par rayh

    I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

    Here is an example ffmpeg command line :

    ffmpeg -threads 1 -nostdin -loglevel verbose \
      -nostdin -y -i input.ts -c:a libfdk_aac \
      -ac 2 -b:a 64k -y -metadata -vn output.ts

    Inspecting an example sound file shows that there is a gap at the end of the audio :

    End

    And the start of the file looks suspiciously attenuated (although this may not be an issue) :

    Start

    My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

    Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?

    ** UPDATE 1 **

    Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)

    Original Start
    Original End

    ** UPDATED 2 **

    So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :

    Side-by-side start
    Side-by-side end

    I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

    ** UPDATE 3 **

    According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

    For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.

  • What are the gotchas of using statically linked libraries in serverless platforms such as Google Cloud Functions ?

    5 septembre 2017, par Dzh

    Libraries such as ffmpeg-static upload statically linked binaries onto container.

    I wonder what are the drawbacks of using this approach ?

    Does the library size counts against your memory use (it’s billed by GCloud) ?

    Does it slow down the container ? Perhaps some future-proofing issues ?

    Edit : Found something of a related (I wanted to setup OpenCV) on AWS blog. It doesn’t explain drawbacks, just shows how to do it exactly.

  • what codec to specify to accessing my HDMI-to-USB adaptor, under Linux ? [closed]

    14 mai 2022, par David

    A week or so ago,I bought a HDMI-to-USB adapter, to use to capture video
TV content from my TV's set-top box. (Xfinity, if it matters. Box generically is :
"XiD X1"...I have both the Pace and the Cisco models available here in this house.)

    


    Specifically, here's the adapter I bought,from Amazon :
https://www.amazon.com/gp/product/B09FLN63B3

    


    So, I'm fluent in both Windows (Win-11) and Linux (Debian 'Bullseye', on my chromebook).

    


    The adapter does not come with any recommendations for what software/drivers to (try to) use, but I was prepared for that.
After some google searches, I decided to first try using the cmd-line
'ffmpeg' program, because I'm
quite familiar with that (excellent !) piece of open-source software !

    


    So, after a day or two of (mostly) success recordings under Win-11, using
ffmpeg's Microsoft-based 'dshow' (aka 'DirectShow'), I decided to attempt to get
up to the same level of accomplishment on my Chromebook, under Linux, also
using 'ffmpeg'.

    


    [Ok...a very brief explanation of 'mostly' successful. I'll
post another separate question here, about the specifics of my glitches,
using 'dshow' on Windows. But, essentially, when I try to record to a MP4
file, I get 2 scenarios of glitch : #1 : Suddenly, dropped packets surges up,
and I get "1000 dropped' yellow msg #2 : On other trials, I get '...contains
no image...'.) So, I figured I should first give a Linux a chance,
before spending more effort trying to resolve the glitches on Windows.]

    


    My first snag, was learning that 'dshow' seems to be specific to 'Windows',
and thus ffmpeg is getting 'unknown' for my reference to 'dshow'. After more hours of 'guessing', I've finally learned/concluded that there are other things
(something call "DeckLink" is one such alternative ?) for Linux, but I'm unclear
what extra Linux packages might exist for ffmpeg support, or whether I will need to built a more complete 'ffmpeg' (e.g. from source code), to get things going under Linux ?

    


    Is my device able to be accessed from some tools other than 'ffmpeg' ?
(e.g. VLC or Handbrake or whatever ?) more easily, on the Linux platform ?

    


    [If I had to, I'd probably invest another $20-$50 in some other hardware
device that goes from HDMI-to-USB (USB-A/B), if it were ]

    


    All ideas are welcome...(TIA)

    


    — Dave