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Somos millones 1
21 juillet 2014, par
Mis à jour : Juin 2015
Langue : français
Type : Video
Autres articles (18)
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HTML5 audio and video support
13 avril 2011, parMediaSPIP 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 (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...) -
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
Sur d’autres sites (8236)
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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
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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!