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La conservation du net art au musée. Les stratégies à l’œuvre
26 mai 2011
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (59)
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Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
Publier sur MédiaSpip
13 juin 2013Puis-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 -
Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP 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 (...)
Sur d’autres sites (12333)
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Batch script to convert all avi to mp4 and delete after conversion using HandBrake command line
26 février 2014, par RarLinesSo easy and cool batch question. Sorry for this ultimate newbie question !
I've a folder which contains hundreds of videos like that :
Video001 - Introduction.avi
Video002 - History of Stack Overflow.avi
Video003 - Before Asking.avi
...
Video999 - Conclusion.aviI need re-encode all of them with x264 codec. Ffmpeg is very slow so I tried to use HandBrake. There is a command line edition of HB and great default presents. I could write this command for converting only one file with "Normal" present :
HandBrakeCLI.exe -i "Video001 - Introduction.avi" -o "Video001 - Introduction.mp4" -Z Normal
My question : How can I convert all of them in folder and delete after conversion process ? Thank you !
Note : If you think ffmpeg is better solution I can give my fav present. Handbrake says about Normal present :
Normal: -e x264 -q 20.0 -a 1 -E faac -B 160 -6 dpl2 -R Auto -D 0.0 --audio-copy-mask aac,ac3,dtshd,dts,mp3 --audio-fallback ffac3 -f mp4 --loose-anamorphic --modulus 2 -m --x264-preset veryfast --h264-profile main --h264-level 4.0
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avformat/matroskadec : fix setting channel layout using the Channels element
4 juillet 2022, par James Almeravformat/matroskadec : fix setting channel layout using the Channels element
If the stream's channel layout is first set into a native layout using codec
private parameters, this code here could potentially result in an invalid
native layout where popcnt(ch_layout.u.mask) != ch_layout.nb_channels being
propagated.Fixes : Timeout printing a billion channels
Fixes : 48099/clusterfuzz-testcase-minimized-ffmpeg_dem_MATROSKA_fuzzer-6754782204788736Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Tested-by : Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by : James Almer <jamrial@gmail.com> -
ffmpeg, v4l, snd_aloop ... sound asyncron (alsa buffer xrun)
28 janvier 2019, par TobiasI’m trying to create a stream that automatically reloads random inputs. I would like to extend this to a database later.
Each time ffmpeg finishes and starts again, so the input changes, the connection to the rtmp is interrupted briefly causing the whole connection breaks down. I then tried to separate audio and video, to send them to virtual devices and read from there again. Split the stream on virtual devices, reassemble them directly and send them to rtmp. If the input is then exchanged, the sending to the devices interrupts what does not bother the second ffmpeg. As soon as I stop sending to the devices the fps go very slowly (10 - 20 sec) from 25 to 0. Only then does the transmitter ffmpeg break the connection to the rtmp. The script which exchanges the inputs needs only one second. A practical test showed that everything works as desired.
I can quite comfortably change the input while the second ffmpeg maintains the stream ...
The joy did not last long. The sound is good 1 sec delayed. But sporadically. Sometimes everything works great. Sometimes the sound is offset.
I wrote several scripts for this.
Background :
- File is selected by random
- Media file is split and written to / dev / video0 (v4l loopback) and alsa default (snd_aloop loopback)
- Put the splits together again and stream them to a rtmp server
Code that selects the input and sends to / dev / video0 and alsa default
#!/bin/bash
cat /dev/null > log
while true;
do
WATERMARK="watermark.png";
dir='/homeXXXXXXXXXX/mix'
file=`/bin/ls -1 "$dir" | sort --random-sort | head -1`
DATEI=`readlink --canonicalize "$dir/$file"` # Converts to full path
if [ -z $DATEI ]
then
echo "Keine Datei gefunden" >> log;
else
START=$(date +%s);
echo "Sende $DATEI" >> log;
ffmpeg -re -y -i "$DATEI" -c:v libx264 -vf "fps=25,scale=640:480,setdar=4:3" -async 1 -pix_fmt yuv420p -preset ultrafast -map 0:0 -f v4l2 -vcodec rawvideo /dev/video0 -f alsa default
fi
DOKILL=`cat kill`;
if [ "$DOKILL" = "1"]
then
break;
fi
doneThe Output
./run.sh
ffmpeg version 3.2.12-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 34.101 / 55. 34.101
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.101 / 57. 56.101
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/mix/XXXXXXXXXXXXX.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
title : XXXXXXXXXXXXXXX
encoder : Lavf57.41.100
Duration: 00:03:53.48, start: 0.000000, bitrate: 2705 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 2573 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 127 kb/s (default)
Metadata:
handler_name : SoundHandler
Codec AVOption preset (Configuration preset) specified for output file #0 (/dev/video0) 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.
[Parsed_setdar_2 @ 0x5571234fe020] num:den syntax is deprecated, please use num/den or named options instead
-async is forwarded to lavfi similarly to -af aresample=async=1:min_hard_comp=0.100000:first_pts=0.
Output #0, v4l2, to '/dev/video0':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
title : XXXXXXXXXXX
encoder : Lavf57.56.101
Stream #0:0(und): Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc (default)
Metadata:
handler_name : VideoHandler
encoder : Lavc57.64.101 rawvideo
Output #1, alsa, to 'default':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
title : XXXXXXXXXX
encoder : Lavf57.56.101
Stream #1:0(und): Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)
Metadata:
handler_name : SoundHandler
encoder : Lavc57.64.101 pcm_s16le
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> rawvideo (native))
Stream #0:1 -> #1:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
frame= 736 fps= 24 q=-0.0 Lsize=N/A time=00:00:29.67 bitrate=N/A speed=0.979x
video:331200kB audio:5112kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Exiting normally, received signal 2.The send script
#!/bin/bash
IP="XXXXXXXXX";
ffmpeg -f video4linux2 -i /dev/video0 -f alsa -acodec pcm_s16le -i default -f flv -async 1 -pix_fmt yuv420p -preset ultrafast -vcodec libx264 -r 25 -s 640x260 -acodec aac rtmp://$IP:1935/live/testThe Output
./send_stream.sh
ffmpeg version 3.2.12-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 34.101 / 55. 34.101
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.101 / 57. 56.101
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
Input #0, video4linux2,v4l2, from '/dev/video0':
Duration: N/A, start: 1548393682.674066, bitrate: 110592 kb/s
Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480, 110592 kb/s, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
Guessed Channel Layout for Input Stream #1.0 : stereo
Input #1, alsa, from 'default':
Duration: N/A, start: 1548393682.677901, bitrate: 1536 kb/s
Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
-async is forwarded to lavfi similarly to -af aresample=async=1:min_hard_comp=0.100000:first_pts=0.
[libx264 @ 0x55e22cfa4f00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x55e22cfa4f00] profile Constrained Baseline, level 2.1
[libx264 @ 0x55e22cfa4f00] 264 - core 148 r2748 97eaef2 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
Output #0, flv, to 'rtmp://XXXXXXXXXXX:1935/live/test':
Metadata:
encoder : Lavf57.56.101
Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 640x260, q=-1--1, 25 fps, 1k tbn, 25 tbc
Metadata:
encoder : Lavc57.64.101 libx264
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 48000 Hz, stereo, fltp, 128 kb/s
Metadata:
encoder : Lavc57.64.101 aac
Stream mapping:
Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
[alsa @ 0x55e22cf87300] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
[video4linux2,v4l2 @ 0x55e22cf84fe0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
Past duration 0.613319 too large 7344kB time=00:01:05.85 bitrate= 913.5kbits/s speed=1.04x
Past duration 0.614372 too large 7644kB time=00:01:08.39 bitrate= 915.6kbits/s speed=1.04x
Past duration 0.609749 too large 7834kB time=00:01:10.91 bitrate= 905.0kbits/s speed=1.04x
Past duration 0.604362 too large 8038kB time=00:01:12.92 bitrate= 903.0kbits/s speed=1.04x
Past duration 0.609489 too large 8070kB time=00:01:13.45 bitrate= 900.1kbits/s speed=1.04x
Past duration 0.615013 too large 8094kB time=00:01:13.94 bitrate= 896.8kbits/s speed=1.04x
Past duration 0.610893 too large 8179kB time=00:01:14.94 bitrate= 894.0kbits/s speed=1.04x
Past duration 0.664711 too large
Past duration 0.639565 too large 8263kB time=00:01:15.47 bitrate= 896.8kbits/s speed=1.04x
Past duration 0.668999 too large 8339kB time=00:01:15.94 bitrate= 899.5kbits/s speed=1.04x
Past duration 0.605766 too large
Past duration 0.633049 too large 8399kB time=00:01:16.48 bitrate= 899.6kbits/s speed=1.04x
Past duration 0.674599 too large
Past duration 0.616035 too large 8451kB time=00:01:16.95 bitrate= 899.7kbits/s speed=1.04x
Past duration 0.656136 too large
Past duration 0.604195 too large
Past duration 0.601387 too large 8512kB time=00:01:17.46 bitrate= 900.2kbits/s speed=1.04x
Past duration 0.621895 too large 8565kB time=00:01:17.95 bitrate= 900.1kbits/s speed=1.04x
Past duration 0.670937 too large 8605kB time=00:01:18.46 bitrate= 898.4kbits/s speed=1.04x
Past duration 0.604500 too large 8642kB time=00:01:18.99 bitrate= 896.2kbits/s speed=1.04x
frame= 1913 fps= 25 q=-1.0 Lsize= 8670kB time=00:01:19.48 bitrate= 893.6kbits/s speed=1.04x
video:7290kB audio:1280kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.160292%
[libx264 @ 0x55e22cfa4f00] frame I:8 Avg QP:18.25 size: 15502
[libx264 @ 0x55e22cfa4f00] frame P:1905 Avg QP:20.95 size: 3853
[libx264 @ 0x55e22cfa4f00] mb I I16..4: 100.0% 0.0% 0.0%
[libx264 @ 0x55e22cfa4f00] mb P I16..4: 6.4% 0.0% 0.0% P16..4: 38.1% 0.0% 0.0% 0.0% 0.0% skip:55.5%
[libx264 @ 0x55e22cfa4f00] coded y,uvDC,uvAC intra: 46.0% 30.3% 13.4% inter: 20.1% 9.8% 1.1%
[libx264 @ 0x55e22cfa4f00] i16 v,h,dc,p: 47% 34% 10% 9%
[libx264 @ 0x55e22cfa4f00] i8c dc,h,v,p: 45% 28% 22% 5%
[libx264 @ 0x55e22cfa4f00] kb/s:750.98
[aac @ 0x55e22cfa62a0] Qavg: 579.067
Exiting normally, received signal 2.First everything is fine and then comes
Past duration 0.616035 too large 8451kB time=00:01:16.95 bitrate= 899.7kbits/s speed=1.04x
Past duration 0.656136 too large
Past duration 0.604195 too large
Past duration 0.601387 too large 8512kB time=00:01:17.46 bitrate= 900.2kbits/s speed=1.04xAnd then when that comes, dives in the first window, so in the ffmpeg sends the input :
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> rawvideo (native))
Stream #0:1 -> #1:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
frame= 9 fps=0.0 q=-0.0 size=N/A time=00:00:00.36 bitrate=N/A dup=1 drop=0 spframe= 21 fps= 21 q=-0.0 size=N/A time=00:00:00.84 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.
Last message repeated 1 times
frame= 33 fps= 22 q=-0.0 size=N/A time=00:00:01.32 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.
Last message repeated 1 times
frame= 46 fps= 23 q=-0.0 size=N/A time=00:00:01.84 bitrate=N/A dup=1 drop=0 spframe= 58 fps= 23 q=-0.0 size=N/A time=00:00:02.32 bitrate=N/A dup=1 drop=0 spframe= 71 fps= 24 q=-0.0 size=N/A time=00:00:02.84 bitrate=N/A dup=1 drop=0 spframe= 83 fps= 24 q=-0.0 size=N/A time=00:00:03.32 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.
frame= 96 fps= 24 q=-0.0 size=N/A time=00:00:03.84 bitrate=N/A dup=1 drop=0 sp[alsa @ 0x5643b3293160] ALSA buffer xrun.The sound is then absolutely unsynchronized ...
Does anyone have any advice and can help me ?