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  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

  • 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 (...)

  • Script d’installation automatique de MediaSPIP

    25 avril 2011, par

    Afin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
    Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
    La documentation de l’utilisation du script d’installation (...)

Sur d’autres sites (12157)

  • Android + ffmpeg + AudioTrack produces bad audio output

    12 septembre 2014, par Goddchen

    here is what I am trying to do : use an AudioRecord and "pipe" the output of AudioRecord.read(byte[],...) to an ffmpeg process’ stdin that will convert to a 3gp (AAC) file.

    The ffmpeg call is as follows :

           ProcessBuilder processBuilder = new ProcessBuilder(BINARY.getAbsolutePath(),
                   "-y",
                   "-ar", "44100", "-c:a", "pcm_s16le", "-ac", "1","-f","s16le",
                   "-i", "-",
                   "-strict", "-2", "-c:a", "aac",
                   outFile.getAbsolutePath());

    The AudioRecord is setup as follows :

    AudioRecord record = new AudioRecord(/*AudioSource.VOICE_RECOGNITION,*/ AudioSource.MIC,
               SAMPLING_RATE,
               AudioFormat.CHANNEL_IN_MONO,
               AudioFormat.ENCODING_PCM_16BIT,
               bufferSize);

    SAMPLING_RATE = 44100 and bufferSize is the one returned by AudioRecord.getMinBufferSize(...)

    I am writing the data to ffmpeg like this :

    try {
                           IOUtils.write(data, getFFmpegHelper().getCurrentProcessOutputStream());
                       } catch (Exception e) {
                           Log.e(Application.LOG_TAG, "Error writing data to ffmpeg process", e);
                           //TODO notify user, stop the recording, etc...
                       }

    So far so good, the ffmpeg runs and created a proper 3gp file. But the audio in the file is totally off. It seems "choppy" (not sure if this is the correct english word ;) ) and also the pace is wrong, is plays too fast.

    Check out this sample : http://goddchen.de/android/tmp/tmp.3gp

    This is the output of the ffmpeg process :

       [s16le @ 0x23634d0] Estimating duration from bitrate, this may be inaccurate
       Guessed Channel Layout for  Input Stream #0.0 : mono
       Input #0, s16le, from 'pipe:':
       Duration: N/A, start: 0.000000, bitrate: 705 kb/s
       Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
       [aformat @ 0x2363100] auto-inserting filter 'auto-inserted resampler 0' between the filter 'src' and the filter 'aformat'
       [aresample @ 0x235b0a0] chl:mono fmt:s16 r:44100Hz -> chl:mono fmt:flt r:44100Hz
       Output #0, 3gp, to '/data/data/com.test.audio/files/tmp.3gp':
       Metadata:
       encoder         : Lavf54.6.100
       Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, flt, 128 kb/s
       Stream mapping:
       Stream #0:0 -> #0:0 (pcm_s16le -> aac)
       size=       3kB time=00:00:00.18 bitrate= 132.5kbits/s    
    size=       8kB time=00:00:00.55 bitrate= 120.9kbits/s    
    size=      12kB time=00:00:00.83 bitrate= 121.8kbits/s    
    size=      16kB time=00:00:01.04 bitrate= 122.8kbits/s    
    size=      20kB time=00:00:01.32 bitrate= 122.5kbits/s    
    size=      23kB time=00:00:01.53 bitrate= 121.6kbits/s    
    size=      27kB time=00:00:01.81 bitrate= 121.0kbits/s    
    size=      31kB time=00:00:02.11 bitrate= 120.7kbits/s    
    size=      35kB time=00:00:02.32 bitrate= 123.4kbits/s
       video:0kB audio:34kB global headers:0kB muxing overhead 3.031610%
  • passing script variable of filename with spaces in bash to external program (ffmpeg) fails

    13 janvier 2016, par BostonScott

    Short story : I’m trying to write a script that will use FFmpeg to convert the many files stored in one directory to a "standard" mp4 format and save the converted files in another directory. It’s been a learning experience (a fun one !) since I haven’t done any real coding since using Pascal and FORTRAN on an IBM 370 mainframe was in vogue.

    Essentially the script takes the filename, strips the path and extension off it, reassembles the filename with the path and an mp4 extension and calls FFmpeg with some set parameters to do the conversion. If the directory contains only video files with without spaces in the names, then everything works fine. If the filenames contain spaces, then FFmpeg is not able to process the file and moves on to the next one. The error indicates that FFMpeg is only seeing the filename up to the first space. I’ve included both the script and output below.

    Thanks for any help and suggestions you may have. If you think I should be doing this in another way, please by all means, give me your suggestions. As I said, it’s been a long time since I did anything like this. I’m enjoying it though.

    I’ve include the code first followed by example output.

    for file in ./TBC/*.mp4
       do

       echo "Start of iteration"
       echo "Full text of file name:" $file

       #Remove everything up to  "C/" (filename without path)
       fn_orig=${file#*C/}
       echo "Original file name:" $fn_orig

       #Length of file name
       fn_len=${#fn_orig}
       echo "Filename Length:" $fn_len

       #file name without path or extension
       fn_base=${fn_orig:0:$fn_len-4}
       echo "Base file name:" $fn_base

       #new filename suffix
       newsuffix=".conv.mp4"

       fn_out=./CONV/$fn_base$newsuffix
       echo "Converted file name:" $fn_out

       ffmpeg -i $file -metadata title="$fn_orig" -c:v libx264 -c:a libfdk_aac -b:a 128k $fn_out

       echo "End of iteration"
       echo
       done
    echo "Script completed"

    With the ffmpeg line commented out, and two files in the ./TBC directory, this is the output that I get

       Start of iteration
       Full text of file name: ./TBC/Test file with spaces.mp4
       Original filename: Test file with spaces.mp4
       Filename Length: 25
       Base filename: Test file with spaces
       Converted file name: ./CONV/Test file with spaces.conv.mp4
       End of iteration

       Start of iteration
       Full text of file name: ./TBC/Test_file_with_NO_spaces.mp4
       Original file name: Test_file_with_NO_spaces.mp4
       Filename Length: 28
       Base file name: Test_file_with_NO_spaces
       Converted file name: ./CONV/Test_file_with_NO_spaces.conv.mp4
       End of iteration

       Script completed

    I won’t bother to post the results when ffmpeg is uncommented, other than to state that it fails with the error :
    ./TBC/Test : No such file or directory

    The script then continues to the next file which completes successfully because it has no spaces in its name. The actual filename is "Test file with spaces.mp4" so you can see that ffmpeg stops after the word "Test" when it encounters a space.

    I hope this has been clear and concise and hopefully someone will be able to point me in the right direction. There is a lot more that I want to do with this script such as parsing subdirectories and ignoring non-video files, etc.

    I look forward to any insight you can give !

  • avcodec/dovi_rpuenc : add ff_dovi_rpu_generate()

    30 mars 2024, par Niklas Haas
    avcodec/dovi_rpuenc : add ff_dovi_rpu_generate()
    

    This function takes a decoded AVDOVIMetadata struct and turns it back
    into a binary RPU. Verified using existing tools, and matches the
    bitstream in official reference files.

    I decided to just roll the EMDF and NAL encapsulation into this function
    because the end user will need to do it otherwise anyways.

    • [DH] libavcodec/dovi_rpu.h
    • [DH] libavcodec/dovi_rpuenc.c