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  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (7846)

  • 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%

  • doc/encoders : partially rewrite and reformat libx264 docs

    23 juillet 2013, par Timothy Gu
    doc/encoders : partially rewrite and reformat libx264 docs

    Format is based on the thread :
    
"[PATCH] doc/encoders : Add libopus encoder doc" (06-28-2013)
    
http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/165368/
    

    Also merge the two option sections (Mapping and Private options).
    

    Patch partially edited by Stefano Sabatini.
    

    Signed-off-by : Stefano Sabatini <stefasab@gmail.com>
    • [DH] doc/encoders.texi

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