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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.

  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

Sur d’autres sites (10810)

  • 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 !

  • FPS drop in FFMPEG streaming processes to FB from production server

    30 janvier 2017, par Aakash Gupta

    I have made a rails app that can stream live videos to facebook rtmp server and deployed it on AWS. I have used nginx as web server. The major problem that I am encountering after viewing log files of FFMpeg processes is that sometimes the FPS of FFmpeg process starts to drop. In some cases, it remains stable at 25 FPS but in some cases, it remains at 25 only for sometime, and after that it starts to drop and sometimes it falls to even 3-4 FPS which is unacceptable during live streaming. As FFMpeg process is quite heavy, I would also like to share my CPU info as well.

    CPU information is :

    cat /proc/cpuinfo
    processor   : 0
    vendor_id   : GenuineIntel
    cpu family  : 6
    model       : 63
    model name  : Intel(R) Xeon(R) CPU E5-2676 v3 @ 2.40GHz
    stepping    : 2
    microcode   : 0x25
    cpu MHz     : 2400.070
    cache size  : 30720 KB
    physical id : 0
    siblings    : 1
    core id     : 0
    cpu cores   : 1
    apicid      : 0
    initial apicid  : 0
    fpu     : yes
    fpu_exception   : yes
    cpuid level : 13
    wp      : yes
    flags       : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx rdtscp lm constant_tsc rep_good nopl xtopology eagerfpu pni pclmulqdq ssse3 fma cx16 pcid sse4_1 sse4_2 x2apic movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand hypervisor lahf_lm abm xsaveopt fsgsbase bmi1 avx2 smep bmi2 erms invpcid
    bogomips    : 4800.14
    clflush size    : 64
    cache_alignment : 64
    address sizes   : 46 bits physical, 48 bits virtual
    power management:

    FFMPEG log file with unstable fps : https://drive.google.com/open?id=0B1gtp1iXJppkUndFamk4M0lRYzA

    FFMPEG log file with stable fps : https://drive.google.com/open?id=0B1gtp1iXJppkMkVCZEJjYWJrVTA

    When FPS was stable, I also tried to run another parallel FFMpeg process from the same server which resulted in FPS dropping of both the processes to 13-14 FPS.

    I am currently using this FFMPEG command :

    ffmpeg -loop 1 -re -y -f image2 -i "image_path" -i "audio_path.aac" -acodec copy -bsf:a aac_adtstoasc -pix_fmt yuv420p -profile:v high -s 1280x720 -vb 400k -maxrate 400k -minrate 400k -bufsize 600k -deinterlace -vcodec libx264 -preset veryfast -g 30 -r 30 -t 14400 -strict -2 -f flv "rtmp_server_link"

    I never face this problem when I try to stream to FB using app on my localhost.

    So, my questions are :

    1. What can be the reason for this FPS drop ?
    2. Can upscaling production server help me fix this issue ?
    3. Can I run multiple FFMpeg processes for streaming from same server without performance drop ?

    Thanks in advance :)