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L’utiliser, en parler, le critiquer
10 avril 2011La première attitude à adopter est d’en parler, soit directement avec les personnes impliquées dans son développement, soit autour de vous pour convaincre de nouvelles personnes à l’utiliser.
Plus la communauté sera nombreuse et plus les évolutions seront rapides ...
Une liste de discussion est disponible pour tout échange entre utilisateurs. -
D’autres logiciels intéressants
12 avril 2011, parOn ne revendique pas d’être les seuls à faire ce que l’on fait ... et on ne revendique surtout pas d’être les meilleurs non plus ... Ce que l’on fait, on essaie juste de le faire bien, et de mieux en mieux...
La liste suivante correspond à des logiciels qui tendent peu ou prou à faire comme MediaSPIP ou que MediaSPIP tente peu ou prou à faire pareil, peu importe ...
On ne les connais pas, on ne les a pas essayé, mais vous pouvez peut être y jeter un coup d’oeil.
Videopress
Site Internet : (...) -
Librairies et logiciels spécifiques aux médias
10 décembre 2010, parPour un fonctionnement correct et optimal, plusieurs choses sont à prendre en considération.
Il est important, après avoir installé apache2, mysql et php5, d’installer d’autres logiciels nécessaires dont les installations sont décrites dans les liens afférants. Un ensemble de librairies multimedias (x264, libtheora, libvpx) utilisées pour l’encodage et le décodage des vidéos et sons afin de supporter le plus grand nombre de fichiers possibles. Cf. : ce tutoriel ; FFMpeg avec le maximum de décodeurs et (...)
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Audio object type 3 is not supported
3 octobre 2013, par ArdoramorI am running into an issue with a custom-built ffmpeg. Basically, its a vanilla ffmpeg with x264 and fdk-aac libs enabled. I am feeding an SDP file through ffmpeg API and get the following error while passing the info to
avcodec_open2
:Audio object type 3 is not supported
My SDP file :
v=0
o=- 0 0 IN IP4 127.0.0.1
s=Unnamed
i=N/A
c=IN IP4 192.168.1.1
t=0 0
a=recvonly
m=audio 8204 RTP/AVP 96
a=rtpmap:96 mpeg4-generic/32000
a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr; config=1a88; SizeLength=13; IndexLength=3; IndexDeltaLength=3;
a=control:trackID=0
m=video 8202 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=428028;sprop-parameter-sets=Z0KAKJWgKA9E,aM48gA==;
a=control:trackID=1
m=application 8206 RTP/AVP 96
a=rtpmap:96 smpte336m/10
a=control:trackID=2The following is the output I get from ffmpeg as I open the input, dump its format, and try to open input streams :
Format sdp probed with size=2048 and score=50
audio codec set to : aac
audio samplerate set to : 32000
audio channels set to : 1
video codec set to : h264
RTP Packetization Mode : 1
RTP Profile IDC : 42 Profile IOP : 80 Level : 28
Extradata set to 0xc8d3dc0 (size : 21) !
File position before avformat_find_stream_info() is 522
Audio object type 3 is not supported.
Could not find codec parameters for stream 1 (Video : h264) : unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
File position after avformat_find_stream_info() is 522
Dumping input AV format
Input #0, sdp, from 'conf/test.sdp' :Metadata :
title :Unnamed
comment :N/ADuration:N/A, bitrate:N/A
Stream #0:0, 0, 1/32000: Audio: aac, 32000 Hz, 1 channels, fltp
Stream #0:1, 0, 1/90000: Video: h264, 90k tbr, 90k tbn, 180k tbc
Stream #0:2, 0, 1/90000: Data: none, 0/1Audio object type 3 is not supported.
I also see the error reported for not finding codec parameters for stream 1 :
Could not find codec parameters for stream 1 (Video : h264) : unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' optionsHowever, I think that is due to me not running an actual rtp at the moment of the test.
In any case, what may be the reason for getting audio object type 3 not supported exception ?
UPDATE - Oct 02, 2013
Static ffmpeg build gives the following flags :
Static ffmpeg 2.0.1 build flag output :
ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
built on Sep 30 2013 22:24:00 with gcc 4.1.2 (GCC) 20080704 (Red Hat 4.1.2-54)
configuration: --enable-gpl --enable-nonfree --enable-shared
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libdc1394 --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-strippingMy build is from ffmpeg 2.0.1 snapshot with the following flags :
--enable-static --disable-shared --enable-gpl --enable-nonfree --enable-libfdk-aac --enable-libfaac --enable-libmp3lame --enable-libx264 --disable-symver --disable-doc --disable-ffplay --enable-ffmpeg --disable-ffprobe --disable-ffserver --disable-avdevice --disable-avfilter --disable-filters --disable-devices --disable-everything --enable-avformat --enable-avcodec --enable-muxers --enable-protocols --enable-parsers --enable-demuxers --enable-decoders --enable-encoders --enable-bsfs --enable-network --enable-swscale --disable-demuxer=sbg --disable-demuxer=dts --disable-parser=dca --disable-decoder=dca --enable-asm --enable-pthreads --enable-version3
The fdk-aac library version I'm using :
fdk-aac - 0.1.2
So it seems that the same .sdp file causes Audio object type 3 error with my build but not with static but I really do not see what causes it (granted I am not an expert with ffmpeg API and may simply miss important details).
Simplified code for opening .sdp file that doesn't include error checks and logic separation :
AVFormatContext *inputContext = NULL;
av_register_all();
avformat_network_init();
avformat_open_input(&inputContext, "test.sdp", NULL, NULL);
avformat_find_stream_info(inputContext, NULL);
int inAudioStreamIdx = av_find_best_stream(inputContext, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
AVStream *st = inputContext->streams[inAudioStreamIdx];
AVCodecContext *dec_ctx = st->codec;
AVCodec *dec = avcodec_find_decoder(dec_ctx->codec_id);
avcodec_open2(dec_ctx, dec, NULL); // Fails here with Audio object type 3Am I opening the sdp input improperly ?
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Using Node.js module fluent-ffmpeg to convert video but my files end up corrupt
19 septembre 2013, par El Guapoi am writing a node based media encoding tool and have found a few good node packages that will help me to do this, but the output files are either totally corrupt or it only encodes half the video.
The main node package i am using is fluent-ffmpeg, and i am trying it with the following code :
var ffmpeg = require('fluent-ffmpeg');
var proc = new ffmpeg({ source: 'uploads/robocop-tlr1_h480p.mov', nolog: false})
.withVideoCodec('libx264')
.withVideoBitrate(800)
.withAudioCodec('libvo_aacenc')
.withAudioBitrate('128k')
.withAudioChannels(2)
.toFormat('mp4')
.saveToFile('output/robocop.mp4',
function(retcode, error){
console.log('file has been converted succesfully');
});There is not a problem with the source video as i encoded it just fine using FFmpeg normally with the following comand line string (i run it from a batch file) :
"c:\ffmpeg\bin\ffmpeg.exe" -i %1 -acodec libvo_aacenc -b:a 128k -ac 2 -vcodec libx264 -b:v 800k -f mp4 "../output/robocop2.mp4"
Any ideas what i am doing wrong here ?
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stream video with transcoded audio only with fluent-ffmpeg and nodejs
9 septembre 2013, par TristanI'm putting together a simple media server to be able to serve my mkv files to web browsers. The problem is most my files have ac3 audio and I don't really want to recode all my files. So I'm streaming them to the browser with ffmpeg but while doing so they all take a hit to image quality no matter the bitrate I attempt to use.
What I would ideally like to achieve is using ffmpeg to transcode the audio only and have the video left alone ! See below for the code im currently using :
new ffmpeg({
source: sFileName, timeout: 432000
}).toFormat('webm').withVideoBitrate(10000).addOptions(["-sn"]).writeToStream(cResponse, function(){
console.log(arguments);
});