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  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 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, par

    MediaSPIP 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, par

    MédiaSPIP 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)

  • Encoding videos for use with Adobe Live Streaming

    10 mai 2016, par FiddleMeRagged

    I 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
  • Linphone OSx msx264 encryption VGA takes 97% CPU, why ?

    11 septembre 2013, par Maxim Shoustin

    I 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 port

    bash-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+gpl2

    So 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

  • Streaming without Content-Length in response

    29 août 2011, par kain

    I'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 ?