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  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • 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 (...)

  • 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 (12491)

  • FFMPEG, faulty input with only a few keyframes ?

    9 août 2020, par browncatmegazord

    I have this video I can watch on MPC-HC alright, but I've been trying to re-encode it with FFMPEG to no avail.

    


    The video unerringly stops 5 minutes into the clip and freezes. Checking the key frames it would seem like the last keyframe is indeed at around the 5 minute mark, even though the video is completely watchable for its entire duration of upwards an hour. I've been looking into finding a way to repopulate the index of keyframes or something, but the answer eludes me.

    


    Both Handbrake and Premiere Pro fails to handle this video properly as well. Premiere Pro just imports it as though it's 5 minutes long, and Handbrake freezes when the encoding reaches the 5 minute mark.

    


    Even doing a -c copy would give me an output that terminates at the 5 minute mark.

    


    What can I do to fix this ?

    


    Edit : Added log as requested.

    


    ffmpeg version git-2020-02-27-9b22254 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-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 42.100 / 56. 42.100
  libavcodec     58. 73.102 / 58. 73.102
  libavformat    58. 39.101 / 58. 39.101
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 77.100 /  7. 77.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000019d5f659b00] st: 0 edit list: 1 Missing key frame while searching for timestamp: 20
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000019d5f659b00] st: 0 edit list 1 Cannot find an index entry before timestamp: 20.
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '[REDACTED].mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 1
    compatible_brands: isom
    creation_time   : 2009-07-27T12:06:40.000000Z
  Duration: 01:13:58.42, start: 0.000000, bitrate: 2671 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 716x480 [SAR 8:9 DAR 179:135], 2504 kb/s, SAR 29127:32768 DAR 1007794:760071, 29.97 fps, 29.97 tbr, 48k tbn, 59.94 tbc (default)
    Metadata:
      rotate          : 0
      creation_time   : 2007-09-08T18:58:09.000000Z
      encoder         : AVC Coding
    Side data:
      displaymatrix: rotation of -0.00 degrees
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 159 kb/s (default)
    Metadata:
      creation_time   : 2009-07-27T12:06:47.000000Z
      handler_name    : GPAC ISO Audio Handler
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
[libx264 @ 0000019d5f65de40] using SAR=8/9
[libx264 @ 0000019d5f65de40] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000019d5f65de40] profile High 4:4:4 Predictive, level 3.0, 4:2:0, 8-bit
[libx264 @ 0000019d5f65de40] 64 - 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=0 mixed_ref=1 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=0 chroma_qp_offset=0 threads=9 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc=cqp mbtree=0 qp=0
Output #0, mp4, to '[REDACTED].mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 1
    compatible_brands: isom
    encoder         : Lavf58.39.101
    Stream #0:0(und): Video: h264 (libx264) (avc1 / 0x31637661), yuv420p(progressive), 716x480 [SAR 29127:32768 DAR 1007794:760071], q=-1--1, 29.97 fps, 11988 tbn, 29.97 tbc (default)
    Metadata:
      encoder         : Lavc58.73.102 libx264
      creation_time   : 2007-09-08T18:58:09.000000Z
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
      displaymatrix: rotation of -0.00 degrees
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 192 kb/s (default)
    Metadata:
      creation_time   : 2009-07-27T12:06:47.000000Z
      handler_name    : GPAC ISO Audio Handler
      encoder         : Lavc58.73.102 aac
frame= 9958 fps= 70 q=-1.0 Lsize=  960472kB time=00:11:13.32 bitrate=11685.6kbits/s speed=4.71x
video:944292kB audio:15836kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.035744%
[libx264 @ 0000015f858dde40] frame I:46    Avg QP: 0.00  size:113701
[libx264 @ 0000015f858dde40] frame P:9912  Avg QP: 0.00  size: 97026
[libx264 @ 0000015f858dde40] mb I  I16..4: 52.6% 10.7% 36.7%
[libx264 @ 0000015f858dde40] mb P  I16..4: 22.5%  5.5% 10.5%  P16..4: 31.8% 15.2% 11.1%  0.0%  0.0%    skip: 3.4%
[libx264 @ 0000015f858dde40] 8x8 transform intra:14.3% inter:40.9%
[libx264 @ 0000015f858dde40] coded y,uvDC,uvAC intra: 95.6% 84.9% 84.0% inter: 81.2% 83.0% 82.5%
[libx264 @ 0000015f858dde40] i16 v,h,dc,p: 53% 44%  2%  1%
[libx264 @ 0000015f858dde40] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 40% 47%  8%  1%  1%  1%  1%  1%  1%
[libx264 @ 0000015f858dde40] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 33% 44%  5%  3%  5%  4%  2%  3%  1%
[libx264 @ 0000015f858dde40] i8c dc,h,v,p: 12% 44% 43%  1%
[libx264 @ 0000015f858dde40] Weighted P-Frames: Y:1.0% UV:0.2%
[libx264 @ 0000015f858dde40] ref P L0: 59.1% 10.2% 27.8%  2.9%  0.0%
[libx264 @ 0000015f858dde40] kb/s:23281.49
[aac @ 0000015f85b5b880] Qavg: 182.070


    


  • Trouble with converting webm into mp3 with pydub in python

    15 août 2020, par rc_marty

    so basically I want to convert song what I downloaded from youtube in webm and convert to into mp3

    


    when I wanted export song just with song.export("neco.mp3") it didn't work too

    


    I have in workfolder ffmpeg.exe and ffprobe.exe

    


    here is the code

    


    from pydub import AudioSegment

song = AudioSegment.from_file(downloaded.webm,"webm")
print("Loaded")
song.export("neco.mp3", format="mp3", bitrate="320k")
print("Converted and saved")


    


    here is the output of the console

    


    Loaded&#xA;Traceback (most recent call last):&#xA;  File "e:/martan/projekty/Python/programek na pisnicky/songDownloader.py", line 188, in <module>&#xA;    song.export("neco.mp3", format="mp3", bitrate="320k")&#xA;  File "C:\Users\BIBRAIN\AppData\Local\Programs\Python\Python38\lib\site-packages\pydub\audio_segment.py", line 911, in export&#xA;    raise CouldntEncodeError(&#xA;pydub.exceptions.CouldntEncodeError: Encoding failed. ffmpeg/avlib returned error code: 1&#xA;&#xA;Command:[&#x27;ffmpeg&#x27;, &#x27;-y&#x27;, &#x27;-f&#x27;, &#x27;wav&#x27;, &#x27;-i&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpo20ooz_z&#x27;, &#x27;-b:a&#x27;, &#x27;320k&#x27;, &#x27;-f&#x27;, &#x27;mp3&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpiqpl57g7&#x27;]&#xA;&#xA;Output from ffmpeg/avlib:&#xA;&#xA;ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers&#xA;  built with gcc 10.2.1 (GCC) 20200726&#xA;  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-libsrt --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-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libgsm --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf&#xA;  libavutil      56. 51.100 / 56. 51.100&#xA;  libavcodec     58. 91.100 / 58. 91.100&#xA;  libavformat    58. 45.100 / 58. 45.100&#xA;  libavdevice    58. 10.100 / 58. 10.100&#xA;  libavfilter     7. 85.100 /  7. 85.100&#xA;  libswscale      5.  7.100 /  5.  7.100&#xA;  libswresample   3.  7.100 /  3.  7.100&#xA;  libpostproc    55.  7.100 / 55.  7.100&#xA;Guessed Channel Layout for Input Stream #0.0 : stereo&#xA;Input #0, wav, from &#x27;C:\Users\BIBRAIN\AppData\Local\Temp\tmpo20ooz_z&#x27;:&#xA;  Duration: 00:03:54.71, bitrate: 3072 kb/s&#xA;    Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (mp3_mf))&#xA;Press [q] to stop, [?] for help&#xA;[mp3_mf @ 00000000004686c0] could not find any MFT for the given media type&#xA;[mp3_mf @ 00000000004686c0] could not create MFT&#xA;Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height&#xA;Conversion failed!&#xA;</module>

    &#xA;

    I think it is something with codec but I have no idea what to do

    &#xA;

  • FFMPEG Transcode VP8 to H264 from rtp stream

    5 août 2020, par Akil

    I have a rtp stream, the server is receiving audio and video on 2 separate ports, the video is in VP8 and the audio is in Opus.

    &#xA;

    My ultimate goal is to convert the RTP stream to RTMP to stream to Youtube Live, but Youtube Live supports only H264 https://developers.google.com/youtube/v3/live/guides/ingestion-protocol-comparison so first i'm looking to transcode my RTP stream to H264.

    &#xA;

    I've run the below command

    &#xA;

    ffmpeg -analyzeduration 300M -probesize 300M -protocol_whitelist file,udp,rtp -i test.sdp -c:v libx264  -pix_fmt yuv420p -r 25 -c:a aac -f flv youtube_rtmp_url&#xA;

    &#xA;

    My sdp file

    &#xA;

    v=0&#xA;o=- 0 0 IN IP4 127.0.0.1&#xA;s=RTP Video&#xA;c=IN IP4 127.0.0.1&#xA;t=0 0&#xA;a=tool:libavformat 55.2.100&#xA;m=audio 50000 RTP/AVP 111&#xA;a=rtpmap:111 OPUS/48000&#xA;m=video 50002 RTP/AVP 100&#xA;a=rtpmap:100 VP8/90000&#xA;a=fmtp:100 packetization-mode=1&#xA;

    &#xA;

    Where 50000 and 50002 are the ports which receive the rtp video and audio.

    &#xA;

    Log output :

    &#xA;

    ffmpeg version 4.3-3ubuntu1~18.04.sav0 Copyright (c) 2000-2020 the FFmpeg developers&#xA;built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)&#xA;configuration: --prefix=/usr --extra-version=&#x27;3ubuntu1~18.04.sav0&#x27; --toolchain=hardened &#xA;--libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 &#xA;--enable-gpl --disable-stripping --enable-avresample --disable-filter=resample &#xA;--enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray &#xA;--enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d &#xA;--enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi &#xA;--enable-libgme               &#xA;--enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg &#xA;--enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librsvg &#xA;--enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr &#xA;--enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame &#xA;--enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp &#xA;--enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi &#xA;--enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 -- &#xA;enable-pocketsphinx --enable-crystalhd --enable-libmfx --enable-libdc1394 --enable-libdrm -- &#xA;enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 -- &#xA;enable-shared&#xA;  libavutil      56. 51.100 / 56. 51.100&#xA;  libavcodec     58. 91.100 / 58. 91.100&#xA;  libavformat    58. 45.100 / 58. 45.100&#xA;  libavdevice    58. 10.100 / 58. 10.100&#xA;  libavfilter     7. 85.100 /  7. 85.100&#xA;  libavresample   4.  0.  0 /  4.  0.  0&#xA;  libswscale      5.  7.100 /  5.  7.100&#xA;  libswresample   3.  7.100 /  3.  7.100&#xA;  libpostproc    55.  7.100 / 55.  7.100&#xA;  [sdp @ 0x5649eef4b8c0] Could not find codec parameters for stream 1 (Video: vp8, yuv420p): &#xA;  unspecified size&#xA;&#xA;  Consider increasing the value for the &#x27;analyzeduration&#x27; and &#x27;probesize&#x27; options&#xA;    Input #0, sdp, from &#x27;test.sdp&#x27;:&#xA;     Metadata:&#xA;       title           : RTP Video&#xA;       Duration: N/A, start: 0.000000, bitrate: N/A&#xA;         Stream #0:0: Audio: opus, 48000 Hz, mono, fltp&#xA;         Stream #0:1: Video: vp8, yuv420p, 90k tbr, 90k tbn, 90k tbc&#xA;      [rtmp @ 0x5649eefd87c0] Cannot open connection tcp://a.rtmp.youtube.com:1935&#xA;      rtmp://a.rtmp.youtube.com/live2/_______: Immediate exit requested&#xA;

    &#xA;

    I've increased 'analyzeduration' and 'probesize' values, error doesn't change.

    &#xA;