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Autres articles (37)
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Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
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 (...) -
Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)
Sur d’autres sites (5244)
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How to compile ffmpeg via Alchemy gcc ?
3 juillet 2014, par RellaSo I created ffmpeg configuration file that makes it pure C (platform independent, but only theoretically)
So my config is simple (0.6.1,0.6.3 tested) :
./configure --disable-doc --disable-ffplay --disable-ffprobe --disable-ffserver --disable-avdevice --disable-avfilter --disable-pthreads --disable-everything --enable-muxer=flv --enable-encoder=flv --enable-encoder=h263 --disable-mmx --disable-shared --prefix=bin/ --disable-protocols --disable-network --disable-optimizations --disable-debug --disable-asm --disable-stripping
Compiling this on Linux will resolve in 4 libs with totall size of 1 mb.
But I need to compile ffmpeg with custom compiler (opensource gcc analog called Adobe Alchemy, lets us compile C/c++ into Flash)
It gives me errors on nearly each and every file during standart Make :
Array @ARGV missing the @ in argument 1 of shift() at /home/rupert/Downloads/alchemy-ubuntu-v0.5a/achacks/gcc line 218.
cc1: error: unrecognized command line option "-Wtype-limits"
cc1: error: unrecognized command line option "-fno-signed-zeros"So what shall I do - how to compile ffmpeg (at least smallest part of it) via alchemy ?
Update
If we would fix that errors manually (2 in configure.mak and one in alchemy gcc) we would get a really messy and long output like :> $ make -w install
make: Entering directory `/home/rupert/Downloads/ffmpeg-0.6.1'
AR libavformat/libavformat.a
llvm-ld: error opening 'avformat/libavformat.l.bc' for writing!
llvm-ranlib: Archive file does not exist
INSTALL libavformat/libavformat.a
install: cannot stat `libavformat/libavformat.a': No such file or directory
llvm-ranlib: Archive file does not exist
AR libavcodec/libavcodec.a
llvm-ld: error opening 'avcodec/libavcodec.l.bc' for writing!
llvm-ranlib: Archive file does not exist
INSTALL libavcodec/libavcodec.a
install: cannot stat `libavcodec/libavcodec.a': No such file or directory
llvm-ranlib: Archive file does not exist
AR libswscale/libswscale.a
llvm-ld: error opening 'swscale/libswscale.l.bc' for writing!
llvm-ranlib: Archive file does not exist
INSTALL libswscale/libswscale.a
install: cannot stat `libswscale/libswscale.a': No such file or directory
llvm-ranlib: Archive file does not exist
AR libavutil/libavutil.a
llvm-ld: error opening 'avutil/libavutil.l.bc' for writing!
llvm-ranlib: Archive file does not exist
INSTALL libavutil/libavutil.a
install: cannot stat `libavutil/libavutil.a': No such file or directory
llvm-ranlib: Archive file does not exist
INSTALL libavformat/avformat.h
INSTALL libavformat/avio.h
INSTALL libavformat/libavformat.pc
INSTALL libavcodec/avcodec.h
INSTALL libavcodec/avfft.h
INSTALL libavcodec/dxva2.h
INSTALL libavcodec/opt.h
INSTALL libavcodec/vaapi.h
INSTALL libavcodec/vdpau.h
INSTALL libavcodec/xvmc.h
INSTALL libavcodec/libavcodec.pc
INSTALL libswscale/swscale.h
INSTALL libswscale/libswscale.pc
INSTALL libavutil/adler32.h
INSTALL libavutil/attributes.h
INSTALL libavutil/avstring.h
INSTALL libavutil/avutil.h
INSTALL libavutil/base64.h
INSTALL libavutil/common.h
INSTALL libavutil/crc.h
INSTALL libavutil/error.h
INSTALL libavutil/fifo.h
INSTALL libavutil/intfloat_readwrite.h
INSTALL libavutil/log.h
INSTALL libavutil/lzo.h
INSTALL libavutil/mathematics.h
INSTALL libavutil/md5.h
INSTALL libavutil/mem.h
INSTALL libavutil/pixdesc.h
INSTALL libavutil/pixfmt.h
INSTALL libavutil/rational.h
INSTALL libavutil/sha1.h
INSTALL libavutil/avconfig.h
INSTALL libavutil/libavutil.pc
LD ffmpeg_g
WARNING: While resolving call to function 'main' arguments were dropped!
Cannot yet select: 0x8e707e8: i32 = ConstantPool < i64 6881500230622117888> 0
llc[0x86c7dec]
make: *** [ffmpeg_g] Error 6
make: Leaving directory `/home/rupert/Downloads/ffmpeg-0.6.1' -
C# library for audio resampling that has the same abilities as FFmpeg
21 avril 2013, par DesignationI have to use a pure C# solution for resampling audio, which can produce me the exact same results as FFmpeg's audio sampling can.
FFmpeg first builds some kind of polyphase filter bank, and then uses that for the sampling process (sorry for the vague phrasing, but I'm not too familiar with this topic). According to this brief documentation, the initialization can be customized this way :
AVResampleContext* av_resample_init(
int out_rate,
int in_rate,
int filter_length,
int log2_phase_count,
int linear,
double cutoff
)The parameters are :
- out_rate : output sample rate
- in_rate : input sample rate
- filter_length : length of each FIR filter in the filterbank relative to the cutoff freq
- log2_phase_count : log2 of the number of entries in the polyphase filterbank
- linear : if 1 then the used FIR filter will be linearly interpolated between the 2 closest, if 0 the closest will be used
- cutoff : cutoff frequency, 1.0 corresponds to half the output sampling rate
I'd need to use a C# library that is configurable in the same depth. I've been trying to use NAudio (more specifically, its
WaveFormatConversionStream
class), but there, I could only set the input and output sample rates, so I didn't get the expected results.So, is there a C# lib that could resample with the same settings as FFmpeg can ? Or one that has almost all of these settings or similar ones ? Note : I need a C# solution, not a wrapper !
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configure : Maintain alphabetical order of components
24 janvier 2016, par Timothy Gu