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Valkaama DVD Label
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Autres articles (46)
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La file d’attente de SPIPmotion
28 novembre 2010, parUne file d’attente stockée dans la base de donnée
Lors de son installation, SPIPmotion crée une nouvelle table dans la base de donnée intitulée spip_spipmotion_attentes.
Cette nouvelle table est constituée des champs suivants : id_spipmotion_attente, l’identifiant numérique unique de la tâche à traiter ; id_document, l’identifiant numérique du document original à encoder ; id_objet l’identifiant unique de l’objet auquel le document encodé devra être attaché automatiquement ; objet, le type d’objet auquel (...) -
List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...) -
Contribute to documentation
13 avril 2011Documentation is vital to the development of improved technical capabilities.
MediaSPIP welcomes documentation by users as well as developers - including : critique of existing features and functions articles contributed by developers, administrators, content producers and editors screenshots to illustrate the above translations of existing documentation into other languages
To contribute, register to the project users’ mailing (...)
Sur d’autres sites (4919)
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FFmpeg hangs when using -vsync 1 for libvpx
31 mars 2014, par user3466413I'm trying to transcode a
.mov
, ultimately generated by GoToMeeting, to a.webm
. I've found to get the audio and video in sync for other formats that I need-vsync 1
and encoding at a constant framerate using-r:v 30
. This causes FFmpeg to take the input.mov
file, which is variable framerate, and transform it to constant framerate at 30 frames per second. In doing so it duplicates frames, but appears to just make many reference frames so the file size doesn't get massive.This approach works perfectly fine when encoding using
libx264
andaac
to a.mp4
container. This makes a file playable in all browsers that support MPEG-4. But, when I encode using usinglibvpx
andlibtheora
to a.webm
container for other browsers, particularly Chrome, FFmpeg just hangs or runs ridiculously slowly. It doesn't respond to pressing Q to abort it, either ; I have to kill the process.So, I get output that looks like this :
user@virtualbox:~/Desktop$ ffmpeg -y -i original.mov -vcodec libvpx -acodec libvorbis -r:v 30 -vsync 1 omgwork.webm
ffmpeg version N-61874-g57e939d Copyright (c) 2000-2014 the FFmpeg developers
built on Mar 27 2014 11:00:31 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
configuration: --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab
libavutil 52. 69.100 / 52. 69.100
libavcodec 55. 55.100 / 55. 55.100
libavformat 55. 35.101 / 55. 35.101
libavdevice 55. 11.100 / 55. 11.100
libavfilter 4. 3.100 / 4. 3.100
libswscale 2. 5.102 / 2. 5.102
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x254b940] decoding for stream 0 failed
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'original.mov':
Metadata:
major_brand : qt
minor_version : 537199360
compatible_brands: qt
creation_time : 2014-03-20 02:09:22
Duration: 03:08:26.72, start: 0.000000, bitrate: 75 kb/s
Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 1368x728 [SAR 1:1 DAR 171:91], 20 kb/s, 0.60 fps, 300 tbr, 600 tbn, 1200 tbc (default)
Metadata:
creation_time : 2014-03-20 02:09:22
handler_name : Apple Alias Data Handler
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 47 kb/s (default)
Metadata:
creation_time : 2014-03-20 02:09:22
handler_name : Apple Alias Data Handler
Stream #0:2(eng): Data: none (rtp / 0x20707472), 6 kb/s
Metadata:
creation_time : 2014-03-20 02:09:22
handler_name : Apple Alias Data Handler
Stream #0:3(und): Data: none (rtp / 0x20707472), 0 kb/s
Metadata:
creation_time : 2014-03-20 02:09:22
handler_name : Apple Alias Data Handler
[libvpx @ 0x255e460] v1.2.0
Output #0, webm, to 'omgwork.webm':
Metadata:
major_brand : qt
minor_version : 537199360
compatible_brands: qt
encoder : Lavf55.35.101
Stream #0:0(und): Video: vp8 (libvpx), yuv420p, 1368x728 [SAR 1:1 DAR 171:91], q=-1--1, 200 kb/s, 1k tbn, 30 tbc (default)
Metadata:
creation_time : 2014-03-20 02:09:22
handler_name : Apple Alias Data Handler
Stream #0:1(eng): Audio: vorbis (libvorbis), 16000 Hz, mono, fltp (default)
Metadata:
creation_time : 2014-03-20 02:09:22
handler_name : Apple Alias Data Handler
Stream mapping:
Stream #0:0 -> #0:0 (h264 -> libvpx)
Stream #0:1 -> #0:1 (aac -> libvorbis)
Press [q] to stop, [?] for help
frame= 117 fps= 10 q=0.0 size= 4kB time=00:00:03.90 bitrate= 8.0kbits/s dup=111 drop=0Note how it says it's three minutes into transcoding...and it stays there for minutes on end. This video takes only about 10 minutes to completely transcode to MPEG-4 and there is no delay whatsoever from the
Press [q] to stop, [?] for help
message and to the first progress message. For WebM, the progress message only appears after a few minutes, and then just crawls along so slowly that it would take literally days to complete.When this is happening, FFmpeg maxes out all the CPU cores I give it (for example
-threads 4
uses all four CPU cores). Aside from making my CPU burst into flames, it doesn't speed things up at all.I've tried :
- Using Windows and Linux. Both do the same thing.
- Using the latest available Windows binary and compiling FFmpeg from the Git repository. No improvement or change.
- Using
-vsync 0
and-vsync 2
. The file does transcode when I do this, but the audio and video remain out of sync just as it did with other formats I tried.-vsync 1
seems absolutely essential for this to work, - Searching, and the closest I could find was bug #3208. But, I'm not specifying
-fflags
at all ; you can see the exact command I'm running.
Ideas ?
-
In FFMpeg Ubuntu “rtmp ://url live=1” not working
13 juillet 2015, par Yuvraj KakkarTrying to excute following ffmpeg command in ubuntu.
*
ffmpeg -i "rtmp://IP/live/1234 live=1" -f flv rtmp://IP/live/1234_56
*ffmpeg version 2.2.git Copyright (c) 2000-2014 the FFmpeg developers
built on Apr 8 2014 13:15:21 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
configuration: --prefix=/home/encoder/ffmpeg_build --extra-cflags=-I/home/encoder/ffmpeg_build/include --extra-ldflags=-L/home/encoder/ffmpeg_build/lib --bindir=/home/encoder/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab
libavutil 52. 74.100 / 52. 74.100
libavcodec 55. 58.102 / 55. 58.102
libavformat 55. 36.102 / 55. 36.102
libavdevice 55. 11.100 / 55. 11.100
libavfilter 4. 3.100 / 4. 3.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
[rtmp @ 0x3c5a140] Detected librtmp style URL parameters, these aren't supported by the libavformat internal RTMP handler currently enabled. See the documentation for the correct way to pass parameters.
^C[flv @ 0x3c59b00] Could not find codec parameters for stream 0 (Video: none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[flv @ 0x3c59b00] Could not find codec parameters for stream 1 (Audio: none, 0 channels): unspecified sample format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
rtmp://IP/live/1234 live=1: could not find codec parameters
Received signal 2: terminating.I have installed avserver (Libav) earlier to ffmpeg.Kindly suggest any alternative to live=1 or provide suggstion.
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ffmpeg extracts about 30 sec of audio and stops [on hold]
30 mars 2014, par user2957951I'm trying to extract audio from an mp4 video stream. I use ffmpeg but it seems to stop after about 30 sec or so ! Ideally, I'd like to extract the audio as mp3, but ffmpeg crashes if I specify .mp3 as the file extension.
Stream : http://vimeo.com/82975217
Here is the command I use and the output I get :
c:\>ffmpeg -i "http://av.vimeo.com/23628/941/105695865.mp4?token2=1396166650_12b
05b5c828c98a2feec82fa76230f64&aksessionid=c18fdc4c061823af&ns=4" -vn -acodec cop
y audiofile1.aac
ffmpeg version N-59742-gd9481dc Copyright (c) 2000-2014 the FFmpeg developers
built on Jan 9 2014 22:04:35 with gcc 4.8.2 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3
--disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --ena
ble-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --e
nable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-lib
modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw
b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge
r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en
able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libx264 --enable-libxavs --enable-
libxvid --enable-zlib
libavutil 52. 62.100 / 52. 62.100
libavcodec 55. 47.101 / 55. 47.101
libavformat 55. 22.103 / 55. 22.103
libavdevice 55. 5.102 / 55. 5.102
libavfilter 4. 1.100 / 4. 1.100
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 17.104 / 0. 17.104
libpostproc 52. 3.100 / 52. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'http://av.vimeo.com/23628/941/105695865
.mp4?token2=1396166650_12b05b5c828c98a2feec82fa76230f64&aksessionid=c18fdc4c0618
23af&ns=4':
Metadata:
major_brand : isom
minor_version : 1
compatible_brands: isomavc1
creation_time : 2012-06-29 21:30:28
Duration: 01:05:34.86, start: 0.000000, bitrate: 594 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x360 [
SAR 1:1 DAR 16:9], 490 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
Metadata:
creation_time : 2012-06-29 21:30:28
handler_name : GPAC ISO Video Handler
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 99
kb/s (default)
Metadata:
creation_time : 2012-06-29 21:30:32
handler_name : GPAC ISO Audio Handler
Output #0, adts, to 'audiofile1.aac':
Metadata:
major_brand : isom
minor_version : 1
compatible_brands: isomavc1
encoder : Lavf55.22.103
Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, 99 kb/s
(default)
Metadata:
creation_time : 2012-06-29 21:30:32
handler_name : GPAC ISO Audio Handler
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Press [q] to stop, [?] for help
size= 344kB time=00:00:27.02 bitrate= 104.2kbits/sAs you can see it just hangs after extracting the 1st 20-30 sec, and after about min or so, the window shuts down !
Note : That extracting video (i.e. mp4 works great, but audio does not)
Your help/insight are much appreciated !
Cheers !