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Médias (91)
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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
-
Stereo master soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Audio
-
Elephants Dream - Cover of the soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Image
-
#7 Ambience
16 octobre 2011, par
Mis à jour : Juin 2015
Langue : English
Type : Audio
-
#6 Teaser Music
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
-
#5 End Title
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
Autres articles (73)
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Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
-
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 -
Installation en mode ferme
4 février 2011, parLe mode ferme permet d’héberger plusieurs sites de type MediaSPIP en n’installant qu’une seule fois son noyau fonctionnel.
C’est la méthode que nous utilisons sur cette même plateforme.
L’utilisation en mode ferme nécessite de connaïtre un peu le mécanisme de SPIP contrairement à la version standalone qui ne nécessite pas réellement de connaissances spécifique puisque l’espace privé habituel de SPIP n’est plus utilisé.
Dans un premier temps, vous devez avoir installé les mêmes fichiers que l’installation (...)
Sur d’autres sites (8897)
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ffmpeg with "-pattern_type glob" and variable in bash script
20 juin 2019, par KlausPeterI’m trying to let ffmpeg make a video of all pictures in a directory using the -pattern_type glob switch and "/foo/bar/*.jpg". This works well, if I execute the command manually für just one directory. For example :
ffmpeg -framerate 35 -pattern_type glob -i '/root/webcam_test/2018-07-21/*.jpg' -vf scale=1280:-1 -c -c:v libx264 -pix_fmt yuv420p /root/clips/out01_cut.mp4
However, if I do it in a bash script and set the path via a variable, according to ffmpegs output, the variable gets substituted correctly, but ffmpeg states that
’/root/webcam_test/2018-07-21/*.jpg’ : No such file or directory
The part of the script looks like this :
for D in `find /root/webcam_test/ -type d`
do
[...]
cmd="ffmpeg -framerate 35 -pattern_type glob -i '$D/*.jpg' -vf scale=1280:-1 -c -c:v libx264 -pix_fm t yuv420p /root/clips/$d_cut.mp4"
echo $cmd
[...]
doneDoes anyone know how to make ffmpeg do its wildcard interpretation even if the path is constructed by a script and not just try to plainly use the given path ?
Best regards and thanks in advance -
ffmpeg demux into audio and video resets PTS
30 juillet 2018, par Mukund ManikarnikeDemuxing
I am demuxing TS segments into audio and video as follows.
ffmpeg -y -i input.ts -vcodec copy -an output_video.ts
ffmpeg -y -i input.ts -acodec copy -vn output_audio.aacInspecting Input
The
start_pts
andstart_time
oninput.ts
are as shown below. I was able to inspect these values usingffprobe -show_streams -print_format json input.ts
"start_pts": 8306558438,
"start_time": "92295.093756",Inspecting output video
The output .ts has some default
start_pts
andstart_time
values as shown below. These were also obtained using the sameffprobe
command as indicated above."start_pts": 126000,
"start_time": "1.400000",Inspecting output audio
The same
ffprobe
command onoutput_audio.aac
shows that the output aac has invalidcodec_tag
andcodec_tag_string
as shown below. Thestart_pts
andstart_time
are not present in theoutput_audio.aac
."codec_tag_string": "[0][0][0][0]", (should have been [15][0][0][0])
"codec_tag": "0x0000", (should have been 0xf000)Questions
- Wondering if this difference in the
start_pts
,start_time
,codec_tag
is expected ? - If it is expected, what can I do to ensure that the all of these parameters get retained on the output ?
- If it is not expected, is there some more information I can share to track this down ?
Note
There were other outputs that I found inconsistent in the
ffprobe
command for theoutput_audio.aac
likeduration etc.
. I shared what I thought are most valuable at this point. If required I can share complete outputs from all of the above executions.[EDIT 07/30/2018 - 08:00 MST]
logs forffmpeg -y -i input.ts -vcodec copy -an output_video.ts -acodec copy -vn output_audio.aac
are as shown below.ffmpeg version 4.0.2 Copyright (c) 2000-2018 the FFmpeg developers
built with Apple LLVM version 9.0.0 (clang-900.0.39.2)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.0.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-ffplay --enable-frei0r --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-librtmp --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma --enable-libopenjpeg --disable-decoder=jpeg2000 --extra-cflags=-I/usr/local/Cellar/openjpeg/2.3.0/include/openjpeg-2.3 --enable-nonfree
libavutil 56. 14.100 / 56. 14.100
libavcodec 58. 18.100 / 58. 18.100
libavformat 58. 12.100 / 58. 12.100
libavdevice 58. 3.100 / 58. 3.100
libavfilter 7. 16.100 / 7. 16.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 1.100 / 5. 1.100
libswresample 3. 1.100 / 3. 1.100
libpostproc 55. 1.100 / 55. 1.100
[mpegts @ 0x7f88ed803000] start time for stream 0 is not set in estimate_timings_from_pts
Input #0, mpegts, from 'i7h9456s_media_46185.ts':
Duration: 00:00:06.05, start: 86216.852667, bitrate: 2898 kb/s
Program 1
Stream #0:0[0x102]: Data: timed_id3 (ID3 / 0x20334449)
Stream #0:1[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p(tv, smpte170m, progressive), 640x360 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:2[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 190 kb/s
Output #0, mpegts, to '../output_video.ts':
Metadata:
encoder : Lavf58.12.100
Stream #0:0: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p(tv, smpte170m, progressive), 640x360 [SAR 1:1 DAR 16:9], q=2-31, 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
Output #1, adts, to '../output_audio.aac':
Metadata:
encoder : Lavf58.12.100
Stream #1:0: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 190 kb/s
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Stream #0:2 -> #1:0 (copy)
Press [q] to stop, [?] for help
frame= 180 fps=0.0 q=-1.0 Lsize= 2088kB time=00:00:06.03 bitrate=2833.8kbits/s speed= 904x
video:1918kB audio:142kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.349750% - Wondering if this difference in the
-
Reading file stream from Google Cloud Storage to ffmpeg (using fluent-ffmpeg)
27 juillet 2018, par ekuusiI’m trying to run ffmpeg on a NodeJS backend with fluent-ffmpeg, reading input files from Google Cloud Storage. Everything works fine if I download the file first :
const file = storage
.bucket('example_bucket')
.file('examplefile.mp4');
file.download({destination: 'test.mp4'}, (err) => {
let command = ffmpeg()
.input('test.mp4')
.duration(10)
.format('mp4');
command.save('test_out.mp4');
});
res.json([{
message: 'Command sent!'
}]);But if I try to use a readable stream as the input, it fails :
const file = storage
.bucket('example_bucket')
.file('examplefile.mp4');
var filestream = file.createReadStream()
let command = ffmpeg()
.input(filestream)
.duration(10)
.format('mp4');
command.save('test_out.mp4');
});
res.json([{
message: 'Command sent!'
}]);Here is the full output of ffmpeg when trying to do the conversion. It seems to read the details of the file fine but for some reason it fails, saying "Cannot process video : ffmpeg exited with code 1 : pipe:0 : Invalid data found when processing input"
Spawned Ffmpeg with command: ffmpeg -i pipe:0 -y -t 10 -f mp4 test_out.mp4
Stderr output: ffmpeg version 4.0.1 Copyright (c) 2000-2018 the FFmpeg developers
Stderr output: built with Apple LLVM version 9.1.0 (clang-902.0.39.2)
Stderr output: configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-librsvg --enable-libtheora --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex --enable-libass --enable-libbluray --enable-lzma --enable-gnutls --enable-fontconfig --enable-libfreetype --enable-libfribidi --disable-libjack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --disable-indev=jack --enable-opencl --disable-outdev=xv --enable-audiotoolbox --enable-videotoolbox --enable-sdl2 --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-x86asm --enable-libx265 --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
Stderr output: libavutil 56. 14.100 / 56. 14.100
Stderr output: libavcodec 58. 18.100 / 58. 18.100
Stderr output: libavformat 58. 12.100 / 58. 12.100
Stderr output: libavdevice 58. 3.100 / 58. 3.100
Stderr output: libavfilter 7. 16.100 / 7. 16.100
Stderr output: libavresample 4. 0. 0 / 4. 0. 0
Stderr output: libswscale 5. 1.100 / 5. 1.100
Stderr output: libswresample 3. 1.100 / 3. 1.100
Stderr output: libpostproc 55. 1.100 / 55. 1.100
Stderr output: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb6db80a200] stream 2, offset 0x30: partial file
Stderr output: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb6db80a200] Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), none, 1920x1080, 4647 kb/s): unspecified pixel format
Stderr output: Consider increasing the value for the 'analyzeduration' and 'probesize' options
Stderr output: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pipe:0':
Stderr output: Metadata:
Stderr output: major_brand : isom
Stderr output: minor_version : 512
Stderr output: compatible_brands: isomiso2avc1mp41
Stderr output: encoder : Lavf58.12.100
Stderr output: location-eng : +60.2121+024.8754/
Stderr output: location : +60.2121+024.8754/
Stderr output: Duration: 00:02:23.13, bitrate: N/A
Stderr output: Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), none, 1920x1080, 4647 kb/s, SAR 1:1 DAR 16:9, 59.94 fps, 59.94 tbr, 60k tbn, 120k tbc (default)
Stderr output: Metadata:
Stderr output: handler_name : VideoHandler
Stderr output: Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
Stderr output: Metadata:
Stderr output: handler_name : SoundHandler
Stderr output: Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
Stderr output: Metadata:
Stderr output: handler_name : TimeCodeHandler
Stderr output: Stream mapping:
Input is aac (mp4a / 0x6134706D) audio with h264 (avc1 / 0x31637661) video
Stderr output: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Stderr output: Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Stderr output: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb6db80a200] stream 0, offset 0x34: partial file
Stderr output: pipe:0: Invalid data found when processing input
Stderr output: Cannot determine format of input stream 0:0 after EOF
Stderr output: Error marking filters as finished
Stderr output: Conversion failed!
Stderr output:
Cannot process video: ffmpeg exited with code 1: pipe:0: Invalid data found when processing input
Cannot determine format of input stream 0:0 after EOF
Error marking filters as finished
Conversion failed!