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  • 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 (...)

  • 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 (...)

  • 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 (...)

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  • Merge two MP3 files and maintain high bitrate and original properties of MP3s

    19 juillet 2019, par sigur7

    I have two MP3 files that were created from the same source, with different audio within them. Here are the properties from ffprobe

    Duration: 00:00:08.86, bitrate: 384 kb/s
    
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, 1 channels, s16, 384 kb/s

    NOTE : Even though the file is an MP3 it shows as pcm_s16le

    When I try and join the two files together using
    ffmpeg -i download.mp3 -i download1.mp3 -filter_complex [0:a:0][1:a:0]concat=n=2:v=0:a=1[outa] -map [outa] joineddownloads.mp3

    I get the following result and a big drop in bitrate(quality)

    Duration: 00:00:10.42, start: 0.046042, bitrate: 32 kb/s
    
    Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s

    How can I maintain the high 320kbs bitrate and all the other properties that were present before I created the joined file ?

  • ffserver leave original stream size

    28 novembre 2014, par ihnatkuk

    Hope you guys will help me, because I have got stuck and can’t find solution for this problem by myself.
    I am trying to stream video from webcam to users using ffmpeg+ffserver. But I have faced with a problem :

    ffmpeg gets stream from camera and pushes it to feed of ffserver:
    
ffmpeg -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -y -vcodec libvpx http://127.0.0.1:8090/1.ffm

    ffserver stream options :

    <stream>
    
Feed 1.ffm
    
Format webm
    
NoAudio
    
#VideoCodec libvpx
    
#VideoSize 480x320
    
VideoFrameRate 24
    
AVOptionVideo flags +global_header
    
AVOptionVideo cpu-used 0
    
AVOptionVideo qmin 1
    
AVOptionVideo qmax 31
    
AVOptionVideo quality good
    
PreRoll 0
    
StartSendOnKey
    
VideoBitRate 128
    
</stream>

    (note, videoSize option is commented). But even with default VideoSize (160x128), ffserver doesn’t respond for each request. Browser always gets :

    HTTP/1.0 200 OK
    
Pragma: no-cache
    
Content-Type: video/webm

    But sometimes video content is not sent.

    If I uncomment VideoSize option - the same problem but much less successfull requests comparing with default video size.

    ffserver log looks regular with no errors. But as you can see that sometimes it sends only headers to client :

    Thu Nov 27 12:49:11 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    
Thu Nov 27 12:49:25 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    
Thu Nov 27 12:49:36 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    
Thu Nov 27 12:50:52 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    
Thu Nov 27 12:53:54 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
    
Thu Nov 27 13:30:19 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
    
Thu Nov 27 13:30:34 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 385731
    
Thu Nov 27 13:30:34 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 458752
    
Thu Nov 27 13:30:36 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
    
Thu Nov 27 13:30:58 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 493
    
Thu Nov 27 13:30:58 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 622592

    Does anybody know what could it be ? Actually I need to save original VideoSize for stream. I am trying to override ffserver stream options with ffmpeg using the command (passing the same parameters as in ffserver’s stream) :

    ffmpeg -re -override_ffserver -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -an -r 24 -qmin 1 -qmax 31 -cpu-used 0 -quality good -flags:v +global_header -b:v 128 -vcodec libvpx -f webm -y http://127.0.0.1:8090/1.ffm

    But at the momment I still have error message ’Output file is empty, nothing was encoded’. Here is ffmpeg’s output :

    ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers
    
  built on Oct  6 2014 17:33:05 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
    
  configuration: --prefix=/opt/ffmpeg --libdir=/opt/ffmpeg/lib/ --enable-shared --enable-avresample --disable-stripping --enable-gpl --enable-version3 --enable-runtime-cpudetect --build-suffix=.ffmpeg --enable-postproc --enable-x11grab --enable-libcdio --enable-vaapi --enable-vdpau --enable-bzlib --enable-gnutls --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libfaac --enable-libvo-aacenc --enable-nonfree --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfdk_aac --enable-libopus --enable-pthreads --enable-zlib --enable-libvpx --enable-libfreetype --enable-libpulse --enable-debug=3
    
  libavutil      54.  7.100 / 54.  7.100
    
  libavcodec     56.  1.100 / 56.  1.100
    
  libavformat    56.  4.101 / 56.  4.101
    
  libavdevice    56.  0.100 / 56.  0.100
    
  libavfilter     5.  1.100 /  5.  1.100
    
  libavresample   2.  1.  0 /  2.  1.  0
    
  libswscale      3.  0.100 /  3.  0.100
    
  libswresample   1.  1.100 /  1.  1.100
    
  libpostproc    53.  0.100 / 53.  0.100
    
Guessed Channel Layout for  Input Stream #0.1 : mono
    
Input #0, rtsp, from 'rtsp://admin:admin@192.168.10.76:80':
    
  Metadata:
    
    title           : RTSP Session/2.0
    
  Duration: N/A, start: 0.000000, bitrate: 128 kb/s
    
    Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709), 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 100 tbr, 90k tbn, 50 tbc
    
    Stream #0:1: Audio: pcm_alaw, 16000 Hz, 1 channels, s16, 128 kb/s
    
[swscaler @ 0x197f7a0] deprecated pixel format used, make sure you did set range correctly
    
[libvpx @ 0x1a0c080] Bitrate 128 is extremely low, maybe you mean 128k
    
[libvpx @ 0x1a0c080] v1.3.0
    
The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
    
Output #0, webm, to 'http://127.0.0.1:8090/1.ffm':
    
  Metadata:
    
    title           : RTSP Session/2.0
    
    encoder         : Lavf56.4.101
    
    Stream #0:0: Video: vp8 (libvpx), yuv420p, 480x320 [SAR 32:27 DAR 16:9], q=1-31, 0 kb/s, 24 fps, 1k tbn, 24 tbc
    
    Metadata:
    
      encoder         : Lavc56.1.100 libvpx
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (h264 (native) -> vp8 (libvpx))
    
Press [q] to stop, [?] for help
    
frame=   33 fps= 22 q=0.0 size=       0kB time=00:00:00.00 bitrate=N/A dup=0 droframe=   43 fps= 22 q=0.0 Lsize=       0kB time=00:00:00.00 bitrate=N/A dup=0 drop=1    
    
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
    
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
    
Received signal 2: terminating.

    Thanks in advance.

  • avformat/avidec : don't save a copy of the packet's AVBufferRef on DV streams

    2 mai 2021, par James Almer
    avformat/avidec : don't save a copy of the packet's AVBufferRef on DV streams

    It's no longer needed.
    

    Signed-off-by : James Almer <jamrial@gmail.com>
    • [DH] libavformat/avidec.c

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