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  • 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 (9921)

  • Visual Studio LNK2001 Error, despite using extern "C" for ffmpeg libraries in C++ [duplicate]

    1er février 2020, par Faizan Cassim

    This is what it looks like :

    ’’’

    extern "C"
    
{
    
#include "libavcodec/avcodec.h"
    
#include "libavformat/avformat.h"
    
#include "libavutil/avutil.h"
    
};

    ’’’

    But I still get these errors when using the FFmpeg libraries in Visual Studio 2019

    1>AudioFile.obj : error LNK2001: unresolved external symbol avformat_open_input
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_read_frame
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_free
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_get_sample_fmt_name
    
1>AudioFile.obj : error LNK2001: unresolved external symbol avformat_close_input
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_init_packet
    
1>AudioFile.obj : error LNK2001: unresolved external symbol avcodec_receive_frame
    
1>AudioFile.obj : error LNK2001: unresolved external symbol avcodec_open2
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_sample_fmt_is_planar
    
1>AudioFile.obj : error LNK2001: unresolved external symbol avcodec_close
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_get_bytes_per_sample
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_packet_unref
    
1>AudioFile.obj : error LNK2001: unresolved external symbol avformat_find_stream_info
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_find_best_stream
    
1>AudioFile.obj : error LNK2001: unresolved external symbol av_frame_alloc

    I would be greatfull to anyone who could help me resolve this. I am using C++ for Windows.

  • FFMPEG video concatenation returning error : `Error reinitializing filters !`, `Invalid argument`, `Conversion failed !`

    2 février 2020, par PureStress

    I am attempting to concatenate .mp4 files. I am using fluent-ffmpeg and @ffmpeg-installer/ffmpeg Here is the pseudocode :

    let mergeClips = ffmpeg();
    
for (let i = 0; i < 5; i++) {
    
    await downloadVideo(newUrl, './testVideos/0/' + i + '.mp4');
    
    mergeClips = mergeClips.mergeAdd('./testVideos/0/' + i + '.mp4');
    
}
    
mergeClips.mergeToFile('compilation.mp4', './temp');

    This results in errors. The following is the command sent by fluent-ffmpeg :

    ffmpeg -i ./testVideos/0/0.mp4 -i ./testVideos/0/6.mp4 -i ./testVideos/0/14.mp4 -i ./testVideos/0/15.mp4 -y -filter_complex concat=n=4:v=1:a=0 ./testVideos/0/compilation.mp4

    When I run the command manually, I receive the following :

    PS C:\path> ffmpeg -i ./testVideos/0/0.mp4 -i ./testVideos/0/6.mp4 -i ./testVideos/0/14.mp4 -i ./testVideos/0/15.mp4 -y -filter_complex concat=n=4:v=1:a=0 ./testVideos/0/compilation.mp4
    
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 9.1.1 (GCC) 20190807
    
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
    
  libavutil      56. 31.100 / 56. 31.100
    
  libavcodec     58. 54.100 / 58. 54.100
    
  libavformat    58. 29.100 / 58. 29.100
    
  libavdevice    58.  8.100 / 58.  8.100
    
  libavfilter     7. 57.100 /  7. 57.100
    
  libswscale      5.  5.100 /  5.  5.100
    
  libswresample   3.  5.100 /  3.  5.100
    
  libpostproc    55.  5.100 / 55.  5.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './testVideos/0/0.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:00:04.49, start: 0.000000, bitrate: 257 kb/s
    
    Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 460x460, 254 kb/s, 22.05 fps, 100 tbr, 12800 tbn, 200 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from './testVideos/0/6.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:00:05.04, start: 0.000000, bitrate: 1088 kb/s
    
    Stream #1:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 720x404, 1086 kb/s, 24.01 fps, 100 tbr, 12800 tbn, 200 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
Input #2, mov,mp4,m4a,3gp,3g2,mj2, from './testVideos/0/14.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:00:26.99, start: 0.000000, bitrate: 52 kb/s
    
    Stream #2:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 250x444, 52 kb/s, 6 fps, 100 tbr, 12800 tbn, 200 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
Input #3, mov,mp4,m4a,3gp,3g2,mj2, from './testVideos/0/15.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf58.12.100
    
  Duration: 00:00:00.19, start: 0.000000, bitrate: 17599 kb/s
    
    Stream #3:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1920x1080, 17564 kb/s, 36.84 fps, 33.33 tbr, 12800 tbn, 200 tbc (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
Stream mapping:
    
  Stream #0:0 (h264) -> concat:in0:v0
    
  Stream #1:0 (h264) -> concat:in1:v0
    
  Stream #2:0 (h264) -> concat:in2:v0
    
  Stream #3:0 (h264) -> concat:in3:v0
    
  concat -> Stream #0:0 (libx264)
    
Press [q] to stop, [?] for help
    
[Parsed_concat_0 @ 000001768445cf40] Input link in1:v0 parameters (size 720x404, SAR 0:1) do not match the corresponding output link in0:v0 parameters (460x460, SAR 0:1)     
    
[Parsed_concat_0 @ 000001768445cf40] Failed to configure output pad on Parsed_concat_0
    
Error reinitializing filters!
    
Failed to inject frame into filter network: Invalid argument
    
Error while processing the decoded data for stream #3:0
    
Conversion failed!

    The videos are downloading correctly.

    Does anybody know what’s going on here ?

  • Create HLS streamable audio file from mp3

    15 août 2023, par isADon

    I am using following command to create a hls aac audio file for web streaming

    



    ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8


    



    This command works only with some audio files. With many mp3 files I receive following output :

    



    C:\ffmpeg>ffmpeg -y -i song.mp3 -c:a aac -b:a 128k -f hls -hls_time 7 -hls_list_size 0 -hls_segment_filename file%d.m4a playlist.m3u8
ffmpeg version git-2020-01-31-62d92a8 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.2.1 (GCC) 20200122
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 38.100 / 56. 38.100
  libavcodec     58. 67.100 / 58. 67.100
  libavformat    58. 37.100 / 58. 37.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 72.100 /  7. 72.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[mp3 @ 0000027d800babc0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'song.mp3':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
  Duration: 00:03:21.46, start: 0.000000, bitrate: 325 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
    Stream #0:1: Video: mjpeg (Baseline), yuvj444p(pc, bt470bg/unknown/unknown), 400x400 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
Stream mapping:
  Stream #0:1 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Stream #0:0 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
[hls @ 0000027d80100c40] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
[libx264 @ 0000027d800c1280] using SAR=1/1
[libx264 @ 0000027d800c1280] MB rate (56250000) > level limit (16711680)
[libx264 @ 0000027d800c1280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000027d800c1280] profile High 4:4:4 Predictive, level 6.2, 4:4:4, 8-bit
[libx264 @ 0000027d800c1280] 264 - core 159 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - 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=4 threads=12 lookahead_threads=2 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, hls, to 'playlist.m3u8':
  Metadata:
    TSS             : Logic Pro 8.0.2
    iTunNORM        :  000000EE 000000ED 00000C34 00001135 000088F0 0000B505 000080FA 00007577 00009B82 00018F49
    iTunSMPB        :  00000000 00000210 00000A07 00000000008783E9 00000000 007AD4E6 00000000 00000000 00000000 00000000 00000000 00000000
    genre           : Rock
    TCM             : Kevin MacLeod
    album           : Funk and Blues
    TKE             : C
    TBP             : 101
    title           : Funkorama
    artist          : Kevin MacLeod
    date            : 2008-06-16 18:35
    encoder         : Lavf58.37.100
    Stream #0:0: Video: h264 (libx264), yuvj444p(pc, progressive), 400x400 [SAR 72:72 DAR 1:1], q=-1--1, 90k fps, 90k tbn, 90k tbc (attached pic)
    Metadata:
      comment         : Other
      encoder         : Lavc58.67.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
    Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc58.67.100 aac
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6 speed=68.6x
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -5 -5
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -2 -2
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 1 times
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -3 -3
[mp3float @ 0000027d80146580] overread, skip -6 enddists: -4 -4
[mp3float @ 0000027d80146580] overread, skip -7 enddists: -6 -6
    Last message repeated 2 times
[mp3float @ 0000027d80146580] overread, skip -5 enddists: -4 -4
[hls @ 0000027d80100c40] Opening 'file0.m4a' for writingate=N/A speed=64.1x
[hls @ 0000027d80100c40] Opening 'playlist.m3u8.tmp' for writing
frame=    1 fps=0.3 q=33.0 Lsize=N/A time=00:03:21.45 bitrate=N/A speed=63.7x
video:7kB audio:3209kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[libx264 @ 0000027d800c1280] frame I:1     Avg QP:34.64  size:  6567
[libx264 @ 0000027d800c1280] mb I  I16..4: 19.5% 53.0% 27.5%
[libx264 @ 0000027d800c1280] 8x8 transform intra:53.0%
[libx264 @ 0000027d800c1280] coded y,u,v intra: 46.8% 26.1% 15.3%
[libx264 @ 0000027d800c1280] i16 v,h,dc,p: 38% 39%  9% 14%
[libx264 @ 0000027d800c1280] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 26%  8%  5%  6%  5%  7%  7%
[libx264 @ 0000027d800c1280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 42% 16% 14%  7%  4%  5%  3%  4%  4%
[libx264 @ 0000027d800c1280] kb/s:4728240.00
[aac @ 0000027d800bcc40] Qavg: 2138.508


    



    Notice the "mp3float overread" message.

    



    It results in a single file0.m4a file without splitting it up after every 7 seconds as specified.
This is an example audio file I am trying to convert to a aac hls stream that results the mentioned problem : https://incompetech.com/music/royalty-free/index.html?isrc=USUAN1100474

    



    How can I convert an audio file to a web friendly hls stream with ffmpeg ?

    


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