Recherche avancée

Médias (1)

Mot : - Tags -/epub

Autres articles (26)

  • 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 (...)

  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Gestion de la ferme

    2 mars 2010, par

    La ferme est gérée dans son ensemble par des "super admins".
    Certains réglages peuvent être fais afin de réguler les besoins des différents canaux.
    Dans un premier temps il utilise le plugin "Gestion de mutualisation"

Sur d’autres sites (3702)

  • ffempg editing metadata major_brand

    29 juillet 2013, par circler

    Hello I have an MP4 video, I want to change the creation_time using ffmepg, without changing the rest of metadata. but I am facing some problems..

    The major_brand and more stuff are changed. I want those to be the same. if I put it in ffprobe.exe this is what I see :

        Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '6.mp4':
    
  Metadata:
    
    major_brand     : mp42
    
    minor_version   : 1
    
    compatible_brands: mp41mp42isom
    
    creation_time   : 2013-03-23 16:25:53
    
  Duration: 00:00:06.55, start: 0.000000, bitrate: 919 kb/s
    
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 480x480,
    
862 kb/s, 29.97 fps, 29.97 tbr, 600 tbn, 1200 tbc
    
    Metadata:
    
      creation_time   : 2013-03-23 16:25:53
    
      handler_name    : Core Media Video
    
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 62 k
    
b/s
    
    Metadata:
    
      creation_time   : 2013-03-23 16:25:53
    
      handler_name    : Core Media Audio

    I want to change the creation_time of the video, when I run :

    ffmpeg.exe -i 6.mp4 -metadata creation_time="2013-06-22 15:00:00" -acodec copy -vcodec copy output.mp4

    I get :

    Output #0, mp4, to 'output.mp4':
    
  Metadata:
    
    major_brand     : mp42
    
    minor_version   : 1
    
    compatible_brands: mp41mp42isom
    
    creation_time   : 2013-06-22 15:00:00
    
    encoder         : Lavf55.12.102
    
    Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 480x480, q=
    
2-31, 862 kb/s, 29.97 fps, 19200 tbn, 600 tbc
    
    Metadata:
    
      creation_time   : 2013-03-23 16:25:53
    
      handler_name    : Core Media Video
    
    Stream #0:1(und): Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, mono, 62 kb
    
/s
    
    Metadata:
    
      creation_time   : 2013-03-23 16:25:53
    
      handler_name    : Core Media Audio
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)
    
  Stream #0:1 -> #0:1 (copy)
    
Press [q] to stop, [?] for help
    
frame=  194 fps=0.0 q=-1.0 Lsize=     738kB time=00:00:06.61 bitrate= 913.9kbits
    
/s

    According to the output and because I chose "-acodec copy -vcodec copy" everything should stay the same. But when I run ffprobe for the newly created .mp4 file. Here is the output :

    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    creation_time   : 2013-06-22 15:00:00
    
    encoder         : Lavf55.12.102
    
  Duration: 00:00:06.62, start: 0.000000, bitrate: 913 kb/s
    
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 480x480,
    
862 kb/s, 29.97 fps, 30 tbr, 19200 tbn, 38400 tbc
    
    Metadata:
    
      creation_time   : 2013-06-22 15:00:00
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 62 k
    
b/s
    
    Metadata:
    
      creation_time   : 2013-06-22 15:00:00
    
      handler_name    : SoundHandler

    As you see everything has changed, even though i chose to keep everything the same.

    Please help me on this. Thanks !

  • can't decode RTMP stream from adobe FMS

    25 juillet 2013, par Mike Versteeg

    I have written code to decode RTMP streams but ran into a problem decoding a stream from FMS. Same stream from Wowza server works fine, but when using Adobe FMS I
    keep getting the same error (note it works fine in a flash player).

    I can confirm the problem using ffmpeg.exe, here's the output of the latest git, anyone have an idea ?

    ffmpeg version N-54901-g55db06a Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jul 23 2013 18:01:29 with gcc 4.7.3 (GCC)
    
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
    
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
    
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
    
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
    
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
    
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
    
enable-libxvid --enable-zlib
    
  libavutil      52. 40.100 / 52. 40.100
    
  libavcodec     55. 19.100 / 55. 19.100
    
  libavformat    55. 12.102 / 55. 12.102
    
  libavdevice    55.  3.100 / 55.  3.100
    
  libavfilter     3. 81.102 /  3. 81.102
    
  libswscale      2.  4.100 /  2.  4.100
    
  libswresample   0. 17.103 /  0. 17.103
    
  libpostproc    52.  3.100 / 52.  3.100
    
Parsing...
    
Parsed protocol: 0
    
Parsed host    : [removed for privacy reasons]
    
Parsed app     : vidlivestream/_definst_/stream
    
RTMP_Connect1, ... connected, handshaking
    
HandShake: Type Answer   : 03
    
HandShake: Server Uptime : 506058230
    
HandShake: FMS Version   : 4.5.5.1
    
HandShake: Handshaking finished....
    
RTMP_Connect1, handshaked
    
Invoking connect
    
HandleServerBW: server BW = 1250000
    
HandleClientBW: client BW = 1250000 2
    
HandleChangeChunkSize, received: chunk size change to 1024
    
HandleCtrl, received ctrl. type: 6, len: 6
    
HandleCtrl, Ping 506058630
    
sending ctrl. type: 0x0007
    
RTMP_ClientPacket, received: invoke 242 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: 
    
(object begin)
    
Property: 4,5,5,4013>
    
Property: 
    
Property: 
    
(object end)
    
Property: 
    
(object begin)
    
Property: 
    
Property: 
    
Property: 
    
Property: 
    
Property: 
    
(object begin)
    
Property: 
    
(object end)
    
(object end)
    
(object end)
    
HandleInvoke, server invoking <_result>
    
HandleInvoke, received result for method call <connect>
    
sending ctrl. type: 0x0003
    
Invoking createStream
    
RTMP_ClientPacket, received: invoke 21 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: NULL
    
(object end)
    
HandleInvoke, server invoking <onbwdone>
    
Invoking _checkbw
    
RTMP_ClientPacket, received: invoke 29 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: NULL
    
Property: 
    
(object end)
    
HandleInvoke, server invoking &lt;_result>
    
HandleInvoke, received result for method call <createstream>
    
SendPlay, seekTime=0, stopTime=0, sending play: test
    
Invoking play
    
sending ctrl. type: 0x0003
    
RTMP_ClientPacket, received: invoke 16419 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: NULL
    
Property:  K H 7 ~ + $ K Z #   ! v 1 &lt; m N % h 9 n G t % J M p 1 f # t %
    
^ u ( I ^ ) &lt; 5 : ? @ a V O &lt; S n [ * y N y T e * 3 P 1 F ! 6 #   + ( w > W \ -
    
: = ` _ 6 q $ - 0 e x G . &#39; 4 [ * / 0 / &amp; _ l ] @ k 8 )v>
    
Property: 
    
(object end)
    
HandleInvoke, server invoking &lt;_onbwcheck>
    
Invoking _result
    
HandleChangeChunkSize, received: chunk size change to 1024
    
RTMP_ClientPacket, received: invoke 142 bytes
    
(object begin)
    
Property: 
    
Property: 
    
Property: NULL
    
Property: 
    
(object begin)
    
Property: 
    
Property: 
    
Property: 
    
Property: 
    
(object end)
    
(object end)
    
HandleInvoke, server invoking <onstatus>
    
HandleInvoke, onStatus: NetStream.Play.Failed
    
Closing connection: NetStream.Play.Failed
    
</onstatus></createstream></onbwdone></connect>

    PS : although there is some resemblance to this topic, it is very old (certainly in ffmpeg terms) and the suggestions make no difference.

  • Why audio element currentTime on ffmpeg encoded mp3 file in Chrome browser does not work

    26 juillet 2013, par Peter

    I have an HTML5 audio element :

    <audio preload="auto">
    
    <source src="./Sound/recording.mp3" type="audio/mpeg">
    
</source></audio>

    and I need to be able to play last 4 seconds from mp3 recording. My javaScript is :

    audio.currentTime = audio.duration-4;
    
audio.play();

    Works ok in IE10 and Firefox, but Chrome starts playing from a wrong place. The difference between reported audio.currentTime and actual playback position is about 20s. The recording.mp3 is created with ffmpeg :

    ffmpeg -i recording.wav -ab 32k recording.mp3

    It works, when I strip the ID3v2 header from the recording.mp3 (deleting the first couple bytes in the file before the audio data).

    It also works when I compress to ogg. Can somebody point me to the right direction (ffmpeg switches, audio element attributes or whatever) to get it work also in chrome ?

    Thanks in advance

    EDIT :
    the ffmpeg output :

    ffmpeg version N-53528-g160ea26 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on May 27 2013 15:20:09 with gcc 4.7.3 (GCC)
    
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
    
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
    
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
    
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
    
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
    
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
    
enable-libxvid --enable-zlib
    
  libavutil      52. 34.100 / 52. 34.100
    
  libavcodec     55. 12.100 / 55. 12.100
    
  libavformat    55.  7.100 / 55.  7.100
    
  libavdevice    55.  1.101 / 55.  1.101
    
  libavfilter     3. 72.100 /  3. 72.100
    
  libswscale      2.  3.100 /  2.  3.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  3.100 / 52.  3.100
    
[wav @ 0433e840] max_analyze_duration 5000000 reached at 5015510 microseconds
    
Guessed Channel Layout for  Input Stream #0.0 : mono
    
Input #0, wav, from &#39;recording.wav&#39;:
    
  Duration: 02:30:07.86, bitrate: 176 kb/s
    
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 11025 Hz, mono, s16, 176 kb/s
    
Output #0, mp3, to &#39;recording.mp3&#39;:
    
  Metadata:
    
    TSSE            : Lavf55.7.100
    
    Stream #0:0: Audio: mp3 (libmp3lame), 11025 Hz, mono, s16p, 32 kb/s
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (pcm_s16le -> libmp3lame)
    
Press [q] to stop, [?] for help
    
size=   35188kB time=02:30:07.86 bitrate=  32.0kbits/s
    
video:0kB audio:35187kB subtitle:0 global headers:0kB muxing overhead 0.000672%

Navigation