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Médias (16)
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#7 Ambience
16 octobre 2011, par
Mis à jour : Juin 2015
Langue : English
Type : Audio
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#6 Teaser Music
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#5 End Title
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#3 The Safest Place
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#4 Emo Creates
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#2 Typewriter Dance
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
Autres articles (31)
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Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
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Creating farms of unique websites
13 avril 2011, parMediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...) -
Personnaliser les catégories
21 juin 2013, parFormulaire de création d’une catégorie
Pour ceux qui connaissent bien SPIP, une catégorie peut être assimilée à une rubrique.
Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire.
Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...)
Sur d’autres sites (5059)
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libavcodec avcodec_open2 returns -22
28 juillet 2012, par buchtakI am trying to learn how to encode video using libavcodec library. I use the following initialization :
avcodec_register_all();
// This works fine.
AVCodec *avcodec = avcodec_find_encoder( CODEC_ID_H264 );
AVCodecContext *avctx = avcodec_alloc_context3( avcodec );
avctx->bit_rate = 400000;
avctx->width = 640;
avctx->height = 480;
avctx->time_base.den = 15;
avctx->time_base.num = 1;
avctx->gop_size = 10;
avctx->max_b_frames = 1;
avctx->pix_fmt = PIX_FMT_YUV420P;
av_opt_set( avctx->priv_data, "preset", "slow", 0 );
// ret should be zero, but it's negative
int ret = avcodec_open2( avctx, avcodec, NULL );However,
avcodec_open2(...)
always returns a negative value. Theavcodec_find_encoder(...)
works fine and the returned pointer is notNULL
. I use Win7 x64, 64-bit Zeranoe FFmpeg build fromhttp://ffmpeg.zeranoe.com/builds/
According to the readme the FFmpeg version is 2012-06-22 git-c17808c built with
--enable-libx264
. I also triedCODEC_ID_MPEG1VIDEO
and changing some of the initialization parameters, but no matter what I do, theavcodec_open2(...)
always returns value-22
. The decoding/encoding example provided with the Zeranoe build (it's the same one as http://ffmpeg.org/doxygen/trunk/api-example_8c-source.html) does not work either... -
FFmpeg inaccurate outputs [closed]
27 juillet 2012, par user1557780Possible Duplicate :
ffmpeg : videos before and after conversion aren't the same lengthRecently, I've been trying to use FFmpeg for an application which requires a VERY accurate manipulation when it comes to the time parameter (milliseconds resolution). Unfortunately, I was surprised to find out that FFmpeg's manipulation functionalities return some inaccurate results.
Here is the output of 'ffmpeg' :
ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
built on Jul 25 2012 19:55:05 with gcc 4.2.1 (Apple Inc. build 5664)
configuration: --enable-gpl --enable-shared --enable-pthreads --enable-libx264 --enable-libmp3lame
libavutil 51. 54.100 / 51. 54.100
libavcodec 54. 23.100 / 54. 23.100
libavformat 54. 6.100 / 54. 6.100
libavdevice 54. 0.100 / 54. 0.100
libavfilter 2. 77.100 / 2. 77.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 15.100 / 0. 15.100
libpostproc 52. 0.100 / 52. 0.100Now, let's assume I want to rip the audio track of 'foo.mov'. Here is the relevant output of 'ffmpeg -i foo.mov' :
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'foo.mov':
Metadata:
major_brand : qt
minor_version : 0
compatible_brands: qt
creation_time : 2012-07-24 23:16:08
Duration: 00:00:40.38, start: 0.000000, bitrate: 805 kb/s
Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 480x360, 733 kb/s, 24.46 fps, 29.97 tbr, 600 tbn, 1200 tbc
Metadata:
rotate : 90
creation_time : 2012-07-24 23:16:08
handler_name : Core Media Data Handler
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, s16, 63 kb/s
Metadata:
creation_time : 2012-07-24 23:16:08
handler_name : Core Media Data HandlerAs you probably noticed, the video file duration is 00:00:40.38. Using the following command, I ripped it's audio track :
'ffmpeg -i foo.mov foo.wav'
Output :
Output #0, wav, to 'foo.wav':
Metadata:
major_brand : qt
minor_version : 0
compatible_brands: qt
creation_time : 2012-07-24 23:16:08
encoder : Lavf54.6.100
Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
Metadata:
creation_time : 2012-07-24 23:16:08
handler_name : Core Media Data Handler
Stream mapping:
Stream #0:1 -> #0:0 (aac -> pcm_s16le)
Press [q] to stop, [?] for help
size=3482kB time=00:00:40.42 bitrate= 705.6kbits/s
video:0kB audio:3482kB global headers:0kB muxing overhead 0.001290%As you can see, the output file is longer than the file in the input.
Another example is audio (and video) file trimming :
Let's assume I would like to use ffmpeg for audio file trimming. I used the next command :'ffmpeg -t 00:00:10.000 -i foo.wav trimmed_foo.wav -ss 00:00:25.000'
Output :
[wav @ 0x10180e800] max_analyze_duration 5000000 reached at 5015510
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, wav, from 'foo.wav':
Duration: 00:00:40.42, bitrate: 705 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
Output #0, wav, to 'trimmed_foo.wav':
Metadata:
encoder : Lavf54.6.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size=864kB time=00:00:10.03 bitrate= 705.6kbits/s
video:0kB audio:864kB global headers:0kB muxing overhead 0.005199%Again, the output file is 30 milliseconds longer than I expected.
I tried, for a long time, to research the issue without any success. When I use audacity for the same functionality, it does it very accurately !
Does anyone have any idea how to solve this problem ?
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ffmpeg usage - quad core still overloaded
29 juillet 2012, par thevoipmanI have a 3.2GHz Quad Core Xeon E3-1230 CPU, Passmark CPU mark Score of 8,200, 32GB ram - and I can't do more than 4 ffmpeg commands/sessions without being overloaded. When I execute a ffmpeg session, it's telling me that it's not using muliple cores, only one. Why is that ? How can I get ffmpeg to fully use all the cpu cores I have ?
Thank you in advance for any kind of suggestions.
ffmpeg version :
ffmpeg version N-42487-gfedefe4-syslint Copyright (c) 2000-2012 the FFmpeg developers
built on Jul 13 2012 23:18:33 with gcc 4.1.2 20080704 (Red Hat 4.1.2-52)
configuration: --prefix=/usr/local/cpffmpeg --enable-shared --enable-nonfree --enable-gpl --enable-pthreads --enable-libopencore-amrnb --enable-decoder=liba52 --enable-libopencore-amrwb --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --extra-cflags=-I/usr/local/cpffmpeg/include/ --extra-ldflags=-L/usr/local/cpffmpeg/lib --enable-version3 --extra-version=syslint
libavutil 51. 64.100 / 51. 64.100
libavcodec 54. 37.100 / 54. 37.100
libavformat 54. 16.104 / 54. 16.104
libavdevice 54. 1.100 / 54. 1.100
libavfilter 3. 2.100 / 3. 2.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 15.100 / 0. 15.100
libpostproc 52. 0.100 / 52. 0.100
Hyper fast Audio and Video encoder