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Médias (1)
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MediaSPIP Simple : futur thème graphique par défaut ?
26 septembre 2013, par
Mis à jour : Octobre 2013
Langue : français
Type : Video
Autres articles (63)
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Demande de création d’un canal
12 mars 2010, parEn fonction de la configuration de la plateforme, l’utilisateur peu avoir à sa disposition deux méthodes différentes de demande de création de canal. La première est au moment de son inscription, la seconde, après son inscription en remplissant un formulaire de demande.
Les deux manières demandent les mêmes choses fonctionnent à peu près de la même manière, le futur utilisateur doit remplir une série de champ de formulaire permettant tout d’abord aux administrateurs d’avoir des informations quant à (...) -
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...)
Sur d’autres sites (13706)
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C - FFmpeg streaming from a C program ?
16 août 2017, par golmschenkI’m looking to replicate an FFmpeg command-line command in my C code. Specifically I would like to be able to run :
ffmpeg -re -i video.mp4 -f mpegts udp://localhost:7777
One thing I’ve noticed when looking at people’s code who have used the libraries of FFmpeg in their own code is that they often have a few hundred lines of code for a single command similar to an FFmpeg command-line command. I’m guessing this is just because they are doing something very specific, because if I can run that short command on my command line and get what I want it should probably only take about ten lines of code to do the same thing in my C code. This should only take about that much work right ? Why would it take much more ?
I’m having a bit of difficulty finding explanations on how to use the streaming capabilities of the FFmpeg libraries that aren’t overly complex because they’re for a very specific purpose. Can anyone explain how I might go about writing the code for the above command ? Or at the very least point me to some documentation explaining how to write such a script/program ? Thank you much !
EDIT : I do hope to run this from an iPhone app eventually so I won’t just be able to straight up call FFmpeg from my program. I’ll need to use the libraries used by FFmpeg.
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Can not play local rtsp URL using FFmpeg in iOS
12 juin 2015, par PriyankaI am using
FFmpeg
for streamrtsp
URL in iOS.
I am trying to stream a local url but my app is failed to open url
avformat_open_input
method always return -5I have played the same url
rtsp://172.16.1.226:5544/1
on VLC media player on my iPhone and macbook it works on both.After few research i have found there is some problem with
rtsp_transport
I was using
av_dict_set(&serverOpt, "rtsp_transport", "tcp", 0);
for the server configuration while opening url and the result is can not open feed.When I changed it to
av_dict_set(&serverOpt, "rtsp_transport", "udp", 1);
I am able to open url successfully but I continuously getting error rtsp 1 missing packet and so on.Can anybody help what should be the right configuration while opening a local
rtsp url
usingffmpeg
.
Should i need to updateav_dict_set(&serverOpt, "rtsp_transport", "udp", 1)
Thanks in advance
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IframeExtractor don't output sound with rtsp
9 janvier 2013, par KamaxI use IframeExtractor from the git mooncatventure, it play nice the .mov file.
But when i try to read a rtsp stream, i hear no sound.This is the FFMEG dump from the rtsp stream :
Metadata:
title : unknown
comment : unknown
Duration: N/A, start: 49435.000589, bitrate: 258 kb/s
Program 3223
No Program
Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0:1(fra): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 142 kb/s
Stream #0:2(fra): Subtitle: dvb_teletext ([6][0][0][0] / 0x0006)
Stream #0:3(qad): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, mono, fltp, 47 kb/s
Stream #0:4(qaa): Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 68 kb/sAnd this is the dump from the local .mov file that work :
Metadata:
major_brand : qt
minor_version : 0
compatible_brands: qt
creation_time : 2010-01-17 21:52:33
model : iPhone 3GS
model-eng : iPhone 3GS
date : 2010-01-17T16:52:33-0500
date-eng : 2010-01-17T16:52:33-0500
encoder : 3.1.2
encoder-eng : 3.1.2
make : Apple
make-eng : Apple
Duration: 00:00:03.25, start: 0.000000, bitrate: 3836 kb/s
Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 640x480, 3695 kb/s, 30.02 fps, 30 tbr, 600 tbn, 1200 tbc
Metadata:
rotate : 90
creation_time : 2010-01-17 21:52:33
handler_name : Core Media Data Handler
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 63 kb/s
Metadata:
creation_time : 2010-01-17 21:52:33
handler_name : Core Media Data HandlerThe audio class that manage sounds contain a codec detector which say that the codec CODEC_ID_AAC is found for the two input :
audioStreamBasicDesc_.mFormatFlags = 0;
switch (_audioCodecContext->codec_id) {
case CODEC_ID_MP3:
audioStreamBasicDesc_.mFormatID = kAudioFormatMPEGLayer3;
break;
case CODEC_ID_AAC:
audioStreamBasicDesc_.mFormatID = kAudioFormatMPEG4AAC;
audioStreamBasicDesc_.mFormatFlags = kMPEG4Object_AAC_Main;
NSLog(@"audio format aac %s (%d) is supported", _audioCodecContext->codec_name, _audioCodecContext->codec_id);
break;
}I see data going into the buffer but i hear nothing. It's maybe audioStreamBasicDesc_ which has wrong settings but i can't find what.
Is it possible that it's not the same AAC codec ?
Has someone experienced the same issue ?
Any help are welcome, i'm on this problem since some days now.
Edit :
I have found a error that i had not before, i don't know how to resolve it. If i change audioStreamBasicDesc.mFramesPerPacket to 0 or divided by 2, the error message dissapear.AudioConverterNew returned 'fmt?'
Prime failed ('fmt?'); will stop (72000/0 frames)