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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (67)
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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 (...) -
Gestion générale des documents
13 mai 2011, parMédiaSPIP ne modifie jamais le document original mis en ligne.
Pour chaque document mis en ligne il effectue deux opérations successives : la création d’une version supplémentaire qui peut être facilement consultée en ligne tout en laissant l’original téléchargeable dans le cas où le document original ne peut être lu dans un navigateur Internet ; la récupération des métadonnées du document original pour illustrer textuellement le fichier ;
Les tableaux ci-dessous expliquent ce que peut faire MédiaSPIP (...)
Sur d’autres sites (13094)
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Encoding videos for use with Adobe Live Streaming
10 mai 2016, par FiddleMeRaggedI have an original video coded at 20Mbps, 1920x1080, 30fps and want to convert it down to be 640x480 30fps at a range of (3 different) bitrates for use by Adobe Live Streaming.
Should I use ffmpeg to resize and encode at the 3 bitrates then use f4fpackager to create the f4m f4f and f4x files or just use ffmpeg to reduce the resolution and then f4fpackager to encode the relevant bitrates ?
I’ve had several tries so far, but when encoded the videos seem to play at a much larger bitrate than they’ve been encoded at. For example, if I set up the OSMF to play from my webserver, I’d be expecting my best encoded video to play at 1,500kbps but it’s way above that.
Has anyone had any experience of encoding for use like this ?
I’m using the following options to f4fpackager
--bitrate=1428 --segment-duration 30 --fragment-duration 2
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Linphone OSx msx264 encryption VGA takes 97% CPU, why ?
11 septembre 2013, par Maxim ShoustinI have problem and today don't know how to fix it or even from where to start.
I have Linphone application that uses msx264 plugin.
All stuff I run on OSx and my ffmpeg version installed from
port
, I didn't using selfupdate for portbash-3.2# port installed ffmpeg-devel
The following ports are currently installed:
ffmpeg-devel @20130205_0+gpl2
ffmpeg-devel @20130328_0 (active)
ffmpeg-devel @20130328_0+gpl2So I compiled and build msx264, no errors.
Now I try to send video over CIP resolution VGA (640x480) and get huge delay 8-9 seconds, even self-view I see in big delay.
when I configure CIF (352x288), all seems fine.
It's really strange that self-view camera has delay 4-5 sec.
So from logs during the session I found that msx264 plugin takes 97% CPU
On PC (windows 7) the same code runs fine, even HD I don't see any problems.
What is the problem should be ?
warning: Video MSTicker: We are late of 32146 miliseconds.
message: Filter MSRtpRecv is not scheduled; nothing to do.
message: ===========================================================
message: AUDIO SESSION'S RTP STATISTICS
message: -----------------------------------------------------------
message: sent 2344 packets
message: 403168 bytes
message: received 2038 packets
message: 350536 bytes
message: incoming delivered to the app 325080 bytes
message: lost 0 packets
message: received too late 123 packets
message: bad formatted 0 packets
message: discarded (queue overflow) 17 packets
message: ===========================================================
message: ms_filter_unlink: MSAuRead:0x7fb5a34955b0,0-->MSResample:0x7fb5aa917820,0
message: ms_filter_unlink: MSResample:0x7fb5aa917820,0-->MSSpeexEC:0x7fb5a34f6d20,1
message: ms_filter_unlink: MSSpeexEC:0x7fb5a34f6d20,1-->MSVolume:0x7fb5a3493450,0
message: ms_filter_unlink: MSVolume:0x7fb5a3493450,0-->MSTee:0x7fb5a3498e40,0
message: ms_filter_unlink: MSTee:0x7fb5a3498e40,0-->MSUlawEnc:0x7fb5a3499410,0
message: ms_filter_unlink: MSUlawEnc:0x7fb5a3499410,0-->MSRtpSend:0x7fb5aa910ba0,0
message: ms_filter_unlink: MSRtpRecv:0x7fb5a3400170,0-->MSUlawDec:0x7fb5a34933c0,0
message: ms_filter_unlink: MSUlawDec:0x7fb5a34933c0,0-->MSGenericPLC:0x7fb5aa91b040,0
message: ms_filter_unlink: MSGenericPLC:0x7fb5aa91b040,0-->MSDtmfGen:0x7fb5a6585f00,0
message: ms_filter_unlink: MSDtmfGen:0x7fb5a6585f00,0-->MSVolume:0x7fb5aa917790,0
message: ms_filter_unlink: MSVolume:0x7fb5aa917790,0-->MSTee:0x7fb5aa914fc0,0
message: ms_filter_unlink: MSTee:0x7fb5aa914fc0,0-->MSEqualizer:0x7fb5a3498f50,0
message: ms_filter_unlink: MSEqualizer:0x7fb5a3498f50,0-->MSSpeexEC:0x7fb5a34f6d20,0
message: ms_filter_unlink: MSSpeexEC:0x7fb5a34f6d20,0-->MSResample:0x7fb5aa9178b0,0
message: ms_filter_unlink: MSResample:0x7fb5aa9178b0,0-->MSAuWrite:0x7fb5a3499380,0
message: ms_filter_unlink: MSTee:0x7fb5a3498e40,1-->MSAudioMixer:0x7fb5aa914df0,0
message: ms_filter_unlink: MSTee:0x7fb5aa914fc0,1-->MSAudioMixer:0x7fb5aa914df0,1
message: ms_filter_unlink: MSAudioMixer:0x7fb5aa914df0,0-->MSFileRec:0x7fb5aa911020,0
message: Audio MSTicker thread exiting
message: ===========================================================
message: FILTER USAGE STATISTICS
message: Name Count Time/tick (ms) CPU Usage
message: -----------------------------------------------------------
message: MSX264Enc 321 138.147 97.1677
message: MSResample 8076 0.0550274 0.97085
message: MSSpeexEC 4302 0.0873765 0.821276
message: MSH264Dec 291 0.880267 0.561463
message: MSRtpSend 6174 0.012353 0.166623
message: MSRtpRecv 6174 0.0115132 0.155295
message: MSOSXGLDisplay 375 0.0376117 0.0308912
message: MSAudioMixer 4695 0.00249638 0.0256072
message: MSV4m 1480 0.00740446 0.0239537
message: MSUlawEnc 4038 0.0019542 0.0172411
message: MSTee 6540 0.000698976 0.00998688
message: MSAuRead 4695 0.00095017 0.0097466
message: MSUlawDec 1890 0.00205553 0.00849059
message: MSVolume 5928 0.000633159 0.00820007
message: MSFileRec 4695 0.000722743 0.00741371
message: MSDtmfGen 4695 0.0005 0.00512887
message: MSGenericPLC 4695 0.000429514 0.00440585
message: MSAuWrite 4038 0.000364199 0.00321319
message: MSEqualizer 1890 0.000250661 0.00103538
message: MSSizeConv 322 0.00104334 0.000736128
message: MSJpegWriter 290 0.000694158 0.00044124
message: MSPixConv 322 0.000405573 0.000286151
message: MSFilePlayer 0 0 0
message: MSVoidSink 0 0 0
message: ===========================================================
warning: Video MSTicker: We are late of 32256 miliseconds.
message: v4m video device closed.
message: Filter MSRtpRecv is not scheduled; nothing to do.
message: ===========================================================
message: VIDEO SESSION'S RTP STATISTICS
message: -----------------------------------------------------------
message: sent 1311 packets
message: 1517528 bytes
message: received 1783 packets
message: 1049010 bytes
message: incoming delivered to the app 986868 bytes
message: lost 0 packets
message: received too late 0 packets
message: bad formatted 0 packets
message: discarded (queue overflow) 0 packets
message: ===========================================================In addition the application shows me delay status, from logs :
message:: Dialog [0x7fb5a7634940]: now updated by transaction [0x7fb5aa9685d0].
warning: Video MSTicker: We are late of 20415 miliseconds.
warning: Video MSTicker: We are late of 20564 miliseconds.
message:: A SPS is being sent.
message:: A PPS is being sent.
warning: Video MSTicker: We are late of 20609 miliseconds.
warning: Video MSTicker: We are late of 20636 miliseconds.
warning: Video MSTicker: We are late of 20694 miliseconds.
warning: Video MSTicker: We are late of 20784 miliseconds.
warning: Video MSTicker: We are late of 20894 miliseconds.
warning: Video MSTicker: We are late of 21016 miliseconds.
warning: echo canceller: we are accumulating too much reference signal, need to throw out 1216 samples
message:: audio_stream_iterate(): local statistics available
Local's current jitter buffer size:77.440002 ms
message:: bandwidth usage: audio=[d=80.1,u=80.1] video=[d=305.3,u=441.8] kbit/sec
message:: Thread processing load: audio=2.135499 video=1268.186768
warning: Video MSTicker: We are late of 21134 miliseconds.
warning: Video MSTicker: We are late of 21256 miliseconds.
warning: Video MSTicker: We are late of 21382 miliseconds.
warning: Video MSTicker: We are late of 21506 miliseconds.
warning: Video MSTicker: We are late of 21638 miliseconds.
warning: Video MSTicker: We are late of 21781 miliseconds.
warning: Video MSTicker: We are late of 21921 miliseconds.
message:: bandwidth usage: audio=[d=81.6,u=80.0] video=[d=271.9,u=185.5] kbit/sec
message:: Thread processing load: audio=1.971647 video=1342.125000
warning: Video MSTicker: We are late of 22068 miliseconds.
message:: audio_stream_iterate(): remote statistics available
remote's interarrival jitter=68
remote's lost packets percentage since last report=0.390625
round trip time=0.258850 seconds
warning: Video MSTicker: We are late of 22216 miliseconds.Please, help me to find the problem.
Thanks,
this is a msx264 git repository :
git clone git://git.linphone.org/msx264.git
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Streaming without Content-Length in response
29 août 2011, par kainI'm using Node.js, Express (and connect), and fluent-ffmpeg.
We want to stream audio files that are stored on Amazon S3 through http.
We have all working, except that we would like to add a feature, the on-the-fly conversion of the stream through ffmpeg.
This is working well, the problem is that some browsers checks in advance before actually getting the file.
Incoming requests containing the Range header, for which we reply with a 206 with all the info from S3, have a fundamental problem : we need to know in advance the content-length of the file.
We don't know that since it is going through ffmpeg.
One solution might be to write out the resulting content-length directly on S3 when storing the file (in a special header), but this means we have to go through the pain of having queues to encode after upload just to know the size for future requests.
It also means that if we change compressor or preset we have to go through all this over again, so it is not a viable solution.We also noticed big differencies in the way Chrome and Safari request the audio tag src, but this may be discussion for another topic.
Fact is that without a proper content-length header in response everything seems to break or browsers goes in an infinite loop or restart the stream at pleasure.
Ideas ?