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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • 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 (...)

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  • How to reduce audio frequency in mp3 format through FFMPEG [on hold]

    25 octobre 2014, par internetdoping

    I want to reduce audio bitrate of my movie in FFMPEG when I use libmp3lame -b:a 16k -ac 1 -ar 44100 it works correctly but when I reduce those to libmp3lame -b:a -16k -ac 1 -ar 8000 I get the error :

    Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input

    please advise me how to reduce audio frequency from 44100 to 8000 mp3lame.

    The following code is working

      ffmpeg -re -ss 00:30:00 -i j:\movie\need4speed.mp4 -preset ultrafast -threads 1 -vcodec libx264 -b:v 20000k -r 24 -g 24 -keyint_min 4 -x264opts "keyint=4:min-keyint=48:no-scenecut" -s 10265*4320 -acodec libmp3lame -b:a 16k -ac 1 -ar 44100 -f flv rtmp://12.11.1.2/livepkgr/need4speed4320p?adbe-live-event=liveevent

    The following code IS NOT working

      ffmpeg -re -ss 00:30:00 -i j:\movie\need4speed.mp4 -preset ultrafast -threads 1 -vcodec libx264 -b:v 20000k -r 24 -g 24 -keyint_min 4 -x264opts "keyint=4:min-keyint=48:no-scenecut" -s 10265*4320 -acodec libmp3lame -b:a 16k -ac 1 -ar 8000 -f flv rtmp://12.11.1.2/livepkgr/need4speed4320p?adbe-live-event=liveevent

    Thanks

  • FFMPEG - Stream discovered after head already parsed [on hold]

    7 février 2014, par John Doe

    I am trying to live transcode an RTMP stream to another RTMP HLS stream using the following command :

    ffmpeg -re -i rtmp://localhost/videochat/testing -c:v libx264 -c:a:0 libfaac -b:a:0 480k -f flv rtmp://localhost:12345/hls/mystream;

    However I receive the following error and the transcoding never begins :

    ffmpeg version git-2014-02-06-474db7a Copyright (c) 2000-2014 the FFmpeg developers
    
built on Feb  6 2014 22:20:14 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
    
configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --               enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3
    
libavutil      52. 63.100 / 52. 63.100
    
libavcodec     55. 49.101 / 55. 49.101
    
libavformat    55. 30.100 / 55. 30.100
    
libavdevice    55.  7.100 / 55.  7.100
    
libavfilter     4.  1.102 /  4.  1.102
    
libswscale      2.  5.101 /  2.  5.101
    
libswresample   0. 17.104 /  0. 17.104
    
libpostproc    52.  3.100 / 52.  3.100
    
Metadata:
    
description           Chat using VideoChat example.
    
[flv @ 0x1ec89e0] Stream discovered after head already parsed
    
^C[flv @ 0x1ec89e0] Could not find codec parameters for stream 0 (Video: none):    unspecified size
    
Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
Input #0, flv, from 'rtmp://localhost/videochat/testing':
    
Metadata:
    
description     : Chat using VideoChat ex   ?5P
    
Duration: N/A, start: 0.000000, bitrate: N/A
    
Stream #0:0: Video: none, 1k tbr, 1k tbn, 1k tbc
    
Stream #0:1: Data: none
    
Codec AVOption b (set bitrate (in bits/s)) specified for output file #0 (rtmp://localhost:12345/hls/mystream) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
    
Output #0, flv, to 'rtmp://localhost:12345/hls/mystream':

    If anyone has ever dealt/solved this problem before, can you please share as I have been trying to solve this for 2 days but to no avail !

  • ffmepg always get broken video streaming from rtsp protocol [on hold]

    6 janvier 2014, par poc

    I'm trying to capture rtsp streaming from Ip Camera by the command

    ffmpeg -t 10 -i  rtsp://172.19.1.42/live.sdp  -ss 00:00:02.500  -c:v copy download.mp4

    However I always get the broken streaming, but if I used the VLC or Qucicktime to watch the RTSP streaming, it worked fine , there was no broken streaming.

    Is there any setting or options for ffmpeg can improve the streaming quality ? Thanks

    [rtsp @ 0x7f9a84017c00] Estimating duration from bitrate, this may be inaccurate
    
Input #0, rtsp, from 'rtsp://172.19.1.42/live.sdp':
    
  Metadata:
    
    title           : RTSP server
    
  Duration: N/A, start: 0.000000, bitrate: N/A
    
    Stream #0:0: Video: h264 (Main), yuv420p, 2048x1536, 13.33 tbr, 90k tbn, 180k tbc
    
-t is not an input option, keeping it for the next output; consider fixing your command line.
    
Output #0, mp4, to 'c0_s1_h264_1920x1536_10_cbr_500_6000000_imagequality.mp4':
    
  Metadata:
    
    title           : RTSP server
    
    encoder         : Lavf54.63.104
    
    Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 2048x1536, q=2-31, 90k tbn, 90k tbc
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)
    
Press [q] to stop, [?] for help
    
[NULL @ 0x7f9a83805200] RTP: missed 3 packets
    
    Last message repeated 1 times
    
RTP: missed 3 packets=-1.0 size=     521kB time=00:00:00.80 bitrate=5303.8kbits/s
    
[NULL @ 0x7f9a83805200] RTP: missed 3 packets
    
RTP: missed 3 packets=-1.0 size=     895kB time=00:00:01.40 bitrate=5224.0kbits/s
    
[NULL @ 0x7f9a83805200] RTP: missed 3 packets
    
RTP: missed 3 packets=-1.0 size=    1271kB time=00:00:02.00 bitrate=5192.3kbits/s
    
[NULL @ 0x7f9a83805200] RTP: missed 3 packets
    
RTP: missed 3 packets=-1.0 size=    1646kB time=00:00:02.60 bitrate=5175.1kbits/s
    
RTP: missed 2 packets=-1.0 size=    2029kB time=00:00:03.20 bitrate=5183.5kbits/s
    
RTP: missed 3 packets=-1.0 size=    2420kB time=00:00:03.80 bitrate=5207.3kbits/s
    
RTP: missed 2 packets=-1.0 size=    2801kB time=00:00:04.40 bitrate=5206.0kbits/s
    
[NULL @ 0x7f9a83805200] RTP: missed 3 packets
    
RTP: missed 2 packets=-1.0 size=    3180kB time=00:00:05.00 bitrate=5201.6kbits/s
    
[NULL @ 0x7f9a83805200] RTP: missed 3 packets
    
RTP: missed 3 packets=-1.0 size=    4708kB time=00:00:07.41 bitrate=5204.6kbits/s
    
RTP: missed 2 packets=-1.0 size=    5088kB time=00:00:08.01 bitrate=5202.7kbits/s
    
RTP: missed 2 packets=-1.0 size=    5857kB time=00:00:09.21 bitrate=5207.3kbits/s
    
[NULL @ 0x7f9a83805200] RTP: missed 2 packets
    
frame=  100 fps= 10 q=-1.0 Lsize=    6382kB time=00:00:09.96 bitrate=5246.4kbits/s

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