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Autres articles (103)

  • 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 (...)

  • Formulaire personnalisable

    21 juin 2013, par

    Cette page présente les champs disponibles dans le formulaire de publication d’un média et il indique les différents champs qu’on peut ajouter. Formulaire de création d’un Media
    Dans le cas d’un document de type média, les champs proposés par défaut sont : Texte Activer/Désactiver le forum ( on peut désactiver l’invite au commentaire pour chaque article ) Licence Ajout/suppression d’auteurs Tags
    On peut modifier ce formulaire dans la partie :
    Administration > Configuration des masques de formulaire. (...)

  • Qu’est ce qu’un masque de formulaire

    13 juin 2013, par

    Un masque de formulaire consiste en la personnalisation du formulaire de mise en ligne des médias, rubriques, actualités, éditoriaux et liens vers des sites.
    Chaque formulaire de publication d’objet peut donc être personnalisé.
    Pour accéder à la personnalisation des champs de formulaires, il est nécessaire d’aller dans l’administration de votre MediaSPIP puis de sélectionner "Configuration des masques de formulaires".
    Sélectionnez ensuite le formulaire à modifier en cliquant sur sont type d’objet. (...)

Sur d’autres sites (3852)

  • ffmpeg concatenate image and audio to beginning of file

    18 mai 2017, par swg1cor14

    So I want to try and use the ffmpeg MAP command to do the following :

    Merge audio and jpg
    Concatenate mp4 to end

    Still.jpg
    
audio.mp3   ->    video.mp4     =    output.mp4

    Right now I have

    /home/admin/ffmpeg/ffmpeg -i still.jpg -i audio.mp3 render1.mp4

    But I want to also go ahead and add on the video.mp4 as well. I tried reading the map function but I’m so confused as to how you know how many streams a file has, etc

    Heres the output of the command....note I have different file names

    Input #0, image2, from 'slide_2.jpg':
    
  Duration: 00:00:00.04, start: 0.000000, bitrate: 8276 kb/s
    
    Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1280x720 [SAR 1:1 DAR 16:9], 25 tbr, 25 tbn, 25 tbc
    
Input #1, mp3, from '/home/admin/web/admin.simplewebevents.com/public_html/cron/steveng1.mp3':
    
  Metadata:
    
    encoder         : Lavf57.56.100
    
  Duration: 00:00:05.78, start: 0.023021, bitrate: 64 kb/s
    
    Stream #1:0: Audio: mp3, 48000 Hz, mono, s16p, 64 kb/s
    
File 'introFile62.mp4' already exists. Overwrite ? [y/N] y
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264))
    
  Stream #1:0 -> #0:1 (mp3 (native) -> aac (native))
    
Press [q] to stop, [?] for help
    
No pixel format specified, yuvj420p for H.264 encoding chosen.
    
Use -pix_fmt yuv420p for compatibility with outdated media players.
    
[libx264 @ 0x3589a00] using SAR=1/1
    
[libx264 @ 0x3589a00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
    
[libx264 @ 0x3589a00] profile High, level 3.1
    
[libx264 @ 0x3589a00] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 2003-2017 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    
Output #0, mp4, to 'introFile62.mp4':
    
  Metadata:
    
    encoder         : Lavf57.72.101
    
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuvj420p(pc, progressive), 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 25 fps, 12800 tbn, 25 tbc
    
    Metadata:
    
      encoder         : Lavc57.96.101 libx264
    
    Side data:
    
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
    
    Stream #0:1: Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, mono, fltp, 69 kb/s
    
    Metadata:
    
      encoder         : Lavc57.96.101 aac
    
frame=    1 fps=0.2 q=28.0 Lsize=      70kB time=00:00:05.76 bitrate=  99.2kbits/s speed=1.27x    
    
video:18kB audio:49kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3.483214%
    
[libx264 @ 0x3589a00] frame I:1     Avg QP:25.71  size: 18268
    
[libx264 @ 0x3589a00] mb I  I16..4: 25.6% 63.4% 10.9%
    
[libx264 @ 0x3589a00] 8x8 transform intra:63.4%
    
[libx264 @ 0x3589a00] coded y,uvDC,uvAC intra: 14.3% 18.5% 8.9%
    
[libx264 @ 0x3589a00] i16 v,h,dc,p: 81%  9% 10%  1%
    
[libx264 @ 0x3589a00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 50% 12% 33%  1%  1%  1%  0%  1%  1%
    
[libx264 @ 0x3589a00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 46% 21% 13%  3%  4%  5%  2%  4%  2%
    
[libx264 @ 0x3589a00] i8c dc,h,v,p: 76%  8% 15%  1%
    
[libx264 @ 0x3589a00] kb/s:3653.60
    
[aac @ 0x358aea0] Qavg: 118.057

  • How can I play libvorbis(ogg) streams using ffplay received with udp ?

    31 mars 2017, par crisman

    ffmpeg

    ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a libvorbis -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111

    ffplay

    ffplay -codec:a libvorbis -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111

    not playing audio.

    Other codecs play back working, example below.

    aac working

    ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a aac -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111
    
ffplay -codec:a aac -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111

    opus codec working

    ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a libopus -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111
    
ffplay -codec:a libopus -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111

    mp3 working

    ffmpeg -f dshow -i audio="virtual-audio-capturer" -codec:a mp3 -b:a 128k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1111
    
ffplay -codec:a mp3 -b:a 128k -ar 48000 -ac 2 udp://127.0.0.1:1111

    why libvorbis not working ?

    transport success, but can not play, i think.

    // not woring ffplay displayed log
    
ffplay.exe -loglevel debug -codec:a vorbis -b:a 128k -ar 48000 -ac 2 -sync audio -i udp://127.0.0.1:1111
    
ffplay version N-83781-g3016e91 Copyright (c) 2003-2017 the FFmpeg developers
    
built with gcc 6.3.0 (GCC)
    
configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
    
libavutil 55. 48.100 / 55. 48.100
    
libavcodec 57. 82.102 / 57. 82.102
    
libavformat 57. 66.103 / 57. 66.103
    
libavdevice 57. 3.100 / 57. 3.100
    
libavfilter 6. 74.100 / 6. 74.100
    
libswscale 4. 3.101 / 4. 3.101
    
libswresample 2. 4.100 / 2. 4.100
    
libpostproc 54. 2.100 / 54. 2.100
    
[udp @ 0000000000b68700] No default whitelist set sq= 0B f=0/0
    
[udp @ 0000000000b68700] end receive buffer size reported is 65536
    
[mpegts @ 0000000000b68ac0] Format mpegts probed with size=2048 and score=50
    
[mpegts @ 0000000000b68ac0] stream=0 stream_type=6 pid=100 prog_reg_desc=
    
[mpegts @ 0000000000b68ac0] Before avformat_find_stream_info() pos: 0 bytes read:8208 seeks:0 nb_streams:1
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2500
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2499
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2498
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2497
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2496
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2495
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2494= 0B f=0/0
    
[mpegts @ 0000000000b68ac0] Probe with size=10038, packets=7 detected mp3 with score=1
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2493= 0B f=0/0
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2492= 0B f=0/0
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2491= 0B f=0/0
    
[mpegts @ 0000000000b68ac0] Probe with size=18321, packets=10 detected mp3 with score=1
    
[mpegts @ 0000000000b68ac0] probing stream 0 pp:2490= 0B f=0/0

  • Add SRT subtitle to video with ffmpeg

    3 juillet 2015, par jAckOdE

    I use ffmpeg to encode, and add subtitle to a video by following command

    $ ffmpeg -i hifi.avi -i hifi.srt -acodec libfaac -ar 48000 -ab 128k -ac 2 -vcodec libx264 -vpre ipod640 -s 480x240 -b 256k -scodec copy  hifi.m4v -newsubtitle

    Here is the output

    ffmpeg version 0.8.git, Copyright (c) 2000-2011 the FFmpeg developers
    
  built on Aug  4 2011 11:11:39 with gcc 4.5.2
    
  configuration: --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --disable-shared --enable-static --enable-gpl --enable-postproc --enable-pthreads --enable-ffplay --disable-ffserver --enable-memalign-hack --enable-nonfree --enable-libfaac --arch=x86 --enable-swscale --enable-libx264 --enable-avfilter --enable-debug=3
    
  libavutil    51. 11. 1 / 51. 11. 1
    
  libavcodec   53.  9. 1 / 53.  9. 1
    
  libavformat  53.  6. 0 / 53.  6. 0
    
  libavdevice  53.  2. 0 / 53.  2. 0
    
  libavfilter   2. 27. 5 /  2. 27. 5
    
  libswscale    2.  0. 0 /  2.  0. 0
    
  libpostproc  51.  2. 0 / 51.  2. 0
    
Input #0, avi, from 'hifi.avi':
    
  Metadata:
    
    encoder         : VirtualDubMod 1.5.4.1 (build 2178/release)
    
    IAS1            : English
    
  Duration: 01:49:02.20, start: 0.000000, bitrate: 897 kb/s
    
    Stream #0.0: Video: mpeg4 (Simple Profile), yuv420p, 544x304 [SAR 1:1 DAR 34:19], 25 tbr, 25 tbn, 25 tbc
    
    Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 32 kb/s
    
[srt @ 0152c100] Estimating duration from bitrate, this may be inaccurate
    
Input #1, srt, from 'hifi.srt':
    
  Duration: N/A, start: 56.080000, bitrate: N/A
    
    Stream #1.0: Subtitle: srt
    
File 'hifi.m4v' already exists. Overwrite ? [y/N] y
    
[buffer @ 02548920] w:544 h:304 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param:
    
[scale @ 02558ee0] w:544 h:304 fmt:yuv420p -> w:480 h:240 fmt:yuv420p flags:0x4
    
[libx264 @ 02547f20] Default settings detected, using medium profile
    
[libx264 @ 02547f20] using SAR=1/1
    
[libx264 @ 02547f20] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64
    
[libx264 @ 02547f20] profile High, level 3.0
    
[libx264 @ 02547f20] 264 - core 115 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=256 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=10000 vbv_bufsize=10000 nal_hrd=none ip_ratio=1.40 aq=1:1.00
    
Output #0, ipod, to 'hifi.m4v':
    
    Stream #0.0: Video: libx264, yuv420p, 480x240 [SAR 1:1 DAR 2:1], q=2-31, 256 kb/s, 90k tbn, 25 tbc
    
    Stream #0.1: Audio: libfaac, 48000 Hz, 2 channels, s16, 128 kb/s
    
    Stream #0.2: Subtitle: srt
    
    Stream #0.3: Subtitle: [0][0][0][0] / 0x0000, 64 kb/s
    
Stream mapping:
    
  Stream #0.0 -> #0.0
    
  Stream #0.1 -> #0.1
    
  Stream #1.0 -> #0.2
    
  Stream #1.0 -> #0.3
    
Encoder (codec id 0) not found for output stream #0.3

    can anyone tell what wrong with my command ??

    thanks,

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