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  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

  • Installation en mode ferme

    4 février 2011, par

    Le mode ferme permet d’héberger plusieurs sites de type MediaSPIP en n’installant qu’une seule fois son noyau fonctionnel.
    C’est la méthode que nous utilisons sur cette même plateforme.
    L’utilisation en mode ferme nécessite de connaïtre un peu le mécanisme de SPIP contrairement à la version standalone qui ne nécessite pas réellement de connaissances spécifique puisque l’espace privé habituel de SPIP n’est plus utilisé.
    Dans un premier temps, vous devez avoir installé les mêmes fichiers que l’installation (...)

  • Les formats acceptés

    28 janvier 2010, par

    Les commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
    ffmpeg -codecs ffmpeg -formats
    Les format videos acceptés en entrée
    Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
    Les formats vidéos de sortie possibles
    Dans un premier temps on (...)

Sur d’autres sites (4299)

  • libavfilter - inputs and outputs confusion in avfilter_graph_parse_ptr

    11 juillet 2020, par Rahul

    I am learning libavfilter library. I have used transcoding.c and filtering_video.c examples from the ffmpeg source code. They are working fine and I understood everything except one part. I have some confusion between inputs and outputs parameters of the avfilter_graph_parse_ptr

    


    If you look at here, I was expecting buffersink to go in the outputs parameter and buffersrc to the inputs. But it does it in reverse - as if inputs are outputs of the filter graph.

    


    https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/transcoding.c#L340

    


    Same done here in another example,
https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/filtering_video.c#L149

    


    Can someone help me understanding that particular part, why inputs and outputs are used reverse ?

    


    Thank you for the help and insight

    


  • find the timestamp of a sound sample of an mp3 with linux or python

    23 juin 2020, par cardamom

    I am slowly working on a project which where it would be very useful if the computer could find where in an mp3 file a certain sample occurs. I would restrict this problem to meaning a fairly exact snippet of the audio, not just for example the chorus in a song on a different recording by the same band where it would become more some kind of machine learning problem. Am thinking if it has no noise added and comes from the same file, it should somehow be possible to locate the time at which it occurs without machine learning, just like grep can find the lines in a textfile where a word occurs.

    


    In case you don't have an mp3 lying around, can set up the problem with some music available on the net which is in the public domain, so nobody complains :

    


    curl https://web.archive.org/web/20041019004300/http://www.navyband.navy.mil/anthems/ANTHEMS/United%20Kingdom.mp3 --output godsavethequeen.mp3


    


    It's a minute long :

    


    exiftool godsavethequeen.mp3 | grep Duration
Duration                        : 0:01:03 (approx)


    


    Now cut out a bit between 30 and 33 seconds (the bit which goes la la la la..) :

    


    ffmpeg -ss 30 -to 33 -i godsavethequeen.mp3 gstq_sample.mp3


    


    both files in the folder :

    


    $ ls -la
-rw-r--r-- 1 cardamom cardamom   48736 Jun 23 00:08 gstq_sample.mp3
-rw-r--r-- 1 cardamom cardamom 1007055 Jun 22 23:57 godsavethequeen.mp3


    


    This is what am after :

    


    $ findsoundsample gstq_sample.mp3 godsavethequeen.mp3
start 30 end 33


    


    Am happy if it is a bash script or a python solution, even using some kind of python library. Sometimes if you use the wrong tool, the solution might work but look horrible, so whichever tool is more suitable. This is a one minute mp3, have not thought yet about performance just about getting it done at all, but would like some scalability, eg find ten seconds somewhere in half an hour.

    


  • Android : Recording and Streaming at the same time

    23 avril 2020, par Bruno Siqueira

    This is not really a question as much as it is a presentation of all my attempts to solve one of the most challenging functionalities I was faced with.

    



    I use libstreaming library to stream realtime videos to Wowza Server and I need to record it at the same time inside the SD card. I am presenting below all my attempts in order to collect new ideias from the community.

    



    Copy bytes from libstreaming stream to a mp4 file

    



    Development

    



    We created an interception in libstreaming library to copy all the sent bytes to a mp4 file. Libstreaming sends the bytes to Wowza server through a LocalSocket. It users MediaRecorder to access the camera and the mic of the device and sets the output file as the LocalSocket's input stream. What we do is create a wrapper around this input stream extending from InputStream and create a File output stream inside it. So, every time libstreaming executes a reading over the LocaSocket's input stream, we copy all the data to the output stream, trying to create a valid MP4 file.

    



    Impediment

    



    When we tried to read the file, it is corrupted. We realized that there are meta information missing from the MP4 file. Specifically the moov atom. We tried to delay the closing of the streaming in order to give time to send this header (this was still a guessing) but it didn't work. To test the coherence of this data, we used a paid software to try to recover the video, including the header. It became playable, but it was mostly green screen. So this became an not trustable solution. We also tried using "untrunc", a free open source command line program and it couldn't even start the recovery, since there was no moov atom.

    



    Use ffmpeg compiled to android to access the camera

    



    Development

    



    FFMPEG has a gradle plugin with a java interface to use it inside Android apps. We thought we could access the camera via command line (it is probably in "/dev/video0") and sent it to the media server.

    



    Impediment

    



    We got the error "Permission Denied" when trying to access the camera. The workaround would be to root the device to have access to it, but it make the phones loose their warranty and could brick them.

    



    Use ffmpeg compiled to android combined with MediaRecorder

    



    Development

    



    We tried to make FFMPEG stream a mp4 file being recorded inside the phone via MediaRecorder

    



    Impediment

    



    FFMPEG can not stream MP4 files that are not yet done with the recording.

    



    Use ffmpeg compiled to android with libstreaming

    



    Development

    



    Libstreaming uses LocalServerSocket as the connection between the app and the server, so we thought that we could use ffmpeg connected with LocalServerSocket local address to copy the streaming directly to a local file inside the SD card. Right after the streaming started, we also ran the ffmpeg command to start recording the data to a file. Using ffmpeg, we believed that it would create a MP4 file in the proper way, which means with the moov atom header included.

    



    Impediment

    



    The "address" created is not readable via command line, as a local address inside the phone. So the copy is not possible.

    



    Use OpenCV

    



    Development

    



    OpenCV is an open-source, cross-platform library that provides building blocks for computer vision experiments and applications. It offers high-level interfaces for capturing, processing, and presenting image data. It has their own APIs to connect with the device camera so we started studding it to see if it had the necessary functionalities to stream and record at the same time.

    



    Impediment

    



    We found out that the library is not really defined to do this, but more as image mathematical manipulation. We got even the recommendation to use libstreaming (which we do already).

    



    Use Kickflip SDK

    



    Development

    



    Kickflip is a media streaming service that provides their own SDK for development in android and IOS. It also uses HLS instead of RTMP, which is a newer protocol.

    



    Impediment

    



    Their SDK requires that we create a Activity with camera view that occupies the entire screen of the device, breaking the usability of our app.

    



    Use Adobe Air

    



    Development

    



    We started consulting other developers of app's already available in the Play Store, that stream to servers already.

    



    Impediment

    



    Getting in touch with those developers, they reassured that would not be possible to record and stream at the same time using this technology. What's more, we would have to redo the entire app from scratch using Adobe Air.

    



    UPDATE

    



    Webrtc

    



    Development

    



    We started using WebRTC following this great project. We included the signaling server in our NODEJS server and started doing the standard handshake via socket. We were still toggling between local recording and streaming via webrtc.

    



    Impediment

    



    Webrtc does not work in every network configuration. Other than that, the camera acquirement is all native code, which makes a lot harder to try to copy the bytes or intercept it.