Recherche avancée

Médias (0)

Mot : - Tags -/diogene

Aucun média correspondant à vos critères n’est disponible sur le site.

Autres articles (30)

  • Pas question de marché, de cloud etc...

    10 avril 2011

    Le vocabulaire utilisé sur ce site essaie d’éviter toute référence à la mode qui fleurit allègrement
    sur le web 2.0 et dans les entreprises qui en vivent.
    Vous êtes donc invité à bannir l’utilisation des termes "Brand", "Cloud", "Marché" etc...
    Notre motivation est avant tout de créer un outil simple, accessible à pour tout le monde, favorisant
    le partage de créations sur Internet et permettant aux auteurs de garder une autonomie optimale.
    Aucun "contrat Gold ou Premium" n’est donc prévu, aucun (...)

  • Keeping control of your media in your hands

    13 avril 2011, par

    The vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
    While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
    MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
    MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...)

  • Le plugin : Podcasts.

    14 juillet 2010, par

    Le problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
    Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
    Types de fichiers supportés dans les flux
    Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...)

Sur d’autres sites (4832)

  • ffmpeg audio conversion distorted - half rate

    6 novembre 2013, par user1688971

    I'm trying to convert an asf audio to mp3 using ffmpeg.
    But I have one specific audio that gets distorted in the middle and starts like if the person was talking in slow motion (at half rate).

    The command I'm using is :

    ffmpeg - i input.asf -ac 2 output.mp3

    I've tried a lot of options, but about the middle of the audio is when it fails.
    The raw file sounds good, so it's not the recording. It is af in the middle of the transmission the frame rate went down for some reason.

    Thanks all !

    [EDIT]

    I'm adding the console response after running the command as a suggestion from LordNeckbeard :

    [root@mynasserver home]# ffmpeg -i recording-8532-1.asf -ac 2 -ab 64k -ar 44100 recording-8532-ac2-ar44100.mp3
    FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
    built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
    configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
    libavutil     50.15. 1 / 50.15. 1
    libavcodec    52.72. 2 / 52.72. 2
    libavformat   52.64. 2 / 52.64. 2
    libavdevice   52. 2. 0 / 52. 2. 0
    libavfilter    1.19. 0 /  1.19. 0
    libswscale     0.11. 0 /  0.11. 0
    libpostproc   51. 2. 0 / 51. 2. 0
    [flv @ 0x86a4850]max_analyze_duration reached
    [flv @ 0x86a4850]Estimating duration from bitrate, this may be inaccurate
    Input #0, flv, from 'recording-8532-1.asf':
    Metadata:
    source          : STW MediaProxy v3.3.7.19894
    Duration: 04:00:08.49, start: 0.000000, bitrate: N/A
    Stream #0.0: Audio: aac, 44100 Hz, 2 channels (FC), s16
    Output #0, mp3, to 'recording-8532-ac2-ar44100.mp3':
    Metadata:
    TSSE            : Lavf52.64.2
    Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
    Stream mapping:
    Stream #0.0 -> #0.0
    Press [q] to stop encoding
    size=  150906kB time=19315.93 bitrate=  64.0kbits/s    
    video:0kB audio:150906kB global headers:0kB muxing overhead 0.000021%

    So from the data above, you can see the input file is about 4hrs. The output ends up being around 5 hrs 20 mins.

  • Overlaying jpg onto Mp4 using -VF

    16 février 2013, par brux

    I am using the following command to overlay a jpg :

    ffmpeg -i in.mp4 -vf "movie=bb.png [movie] ; [in] [movie] overlay=0:0 [out]" -vcodec libx264 -acodec copy out.mp4

    This works as expected with the first file (listed below) but it doesnt work with the second file. There is no error when I try with the second file, rather is creates an unusually large file that would not open :

    File 1 :

    [me@me ~]$ ffmpeg -i 2013-02-08.mp4
    ffmpeg version 1.0.git Copyright (c) 2000-2012 the FFmpeg developers
     built on Jan 11 2013 00:12:08 with gcc 4.7.2 (GCC) 20120921 (Red Hat 4.7.2-2)
     configuration:
     libavutil      52.  8.100 / 52.  8.100
     libavcodec     54. 74.100 / 54. 74.100
     libavformat    54. 37.100 / 54. 37.100
     libavdevice    54.  3.100 / 54.  3.100
     libavfilter     3. 23.101 /  3. 23.101
     libswscale      2.  1.102 /  2.  1.102
     libswresample   0. 17.101 /  0. 17.101
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '2013-02-08.mp4':
     Metadata:
       major_brand     : isom
       minor_version   : 512
       compatible_brands: isomiso2avc1mp41
       creation_time   : 2013-02-08 20:31:49
       encoder         : Lavf53.24.0
     Duration: 00:00:03.20, start: 0.000000, bitrate: 1030 kb/s
       Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 720x576 [SAR 1:1 DAR 5:4], 1247 kb/s, 8.08 fps, 7.50 tbr, 15 tbn, 15 tbc
       Metadata:
         creation_time   : 2013-02-08 20:31:49
         handler_name    : VideoHandler
       Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 8000 Hz, mono, s16, 12 kb/s
       Metadata:
         creation_time   : 2013-02-08 20:31:49
         handler_name    : SoundHandler
    At least one output file must be specified

    File 2

    [me@me ~]$ ffmpeg -i aq.mp4
    ffmpeg version 1.0.git Copyright (c) 2000-2012 the FFmpeg developers
     built on Jan 11 2013 00:12:08 with gcc 4.7.2 (GCC) 20120921 (Red Hat 4.7.2-2)
     configuration:
     libavutil      52.  8.100 / 52.  8.100
     libavcodec     54. 74.100 / 54. 74.100
     libavformat    54. 37.100 / 54. 37.100
     libavdevice    54.  3.100 / 54.  3.100
     libavfilter     3. 23.101 /  3. 23.101
     libswscale      2.  1.102 /  2.  1.102
     libswresample   0. 17.101 /  0. 17.101
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'aq.mp4':
     Metadata:
       major_brand     : isom
       minor_version   : 512
       compatible_brands: isomiso2avc1mp41
       creation_time   : 2013-02-19 20:33:16
       encoder         : Lavf53.24.0
     Duration: 00:00:03.20, start: 0.000000, bitrate: 1394 kb/s
       Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 720x576 [SAR 1:1 DAR 5:4], 1451 kb/s, 30 fps, 30 tbr, 30 tbn, 60 tbc
       Metadata:
         creation_time   : 2013-02-19 20:33:16
         handler_name    : VideoHandler
       Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 8000 Hz, mono, s16, 12 kb/s
       Metadata:
         creation_time   : 2013-02-19 20:33:16
         handler_name    : SoundHandler
    At least one output file must be specified

    In case it is important I am capturing these videos with Android devices. The first mp4 file is created by a Nexus 7, the second (the file which wont overlay the image) is created using a HTC Desire.

  • FFmpeg audio capturing from mic is not working properly

    5 mars 2013, par Thirumalai murugan

    I am using the ffmpeg-20130205-git-c2dd5a1-win64-static version, I am trying to capture the audio and video and send it to the FMS server, I have tried with the following code initially

    ffmpeg -r 25 -f dshow -i video="Logitech HD Pro Webcam C920":audio="Rear Input (SoundMAX Integrated Digital High Definition Audio)" -vcodec libx264 -b:v 600k -b:a 128k -f flv rtmp://127.0.0.1/live/mystream

    it through the following error

    [dshow @ 00000000023f8920] Could not find audio device.
    video=Logitech HD Pro Webcam C920:audio=SoundMAX Integrated Digital High Definit
    ion Audio): Input/output error

    Then I modified the code as follows its working fine

    ffmpeg -f dshow -i video="Logitech HD Pro Webcam C920":audio="Rear Input (SoundMAX Integrated" -b:v 600k -acodec libmp3lame -b:a 128k -f flv rtmp://127.0.0.1/live/mystream

    I am unable to understand why its not accepting the full name of the audio driver and if I use the libx264 with the Logitech HD Pro Webcam C920 its not giving the video, video is blank (note : while using the iball c2.0 camera I am able to get the video)

    what is the wrong in my code ? how to publish in the libx264 format ?