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  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

  • Submit enhancements and plugins

    13 avril 2011

    If you have developed a new extension to add one or more useful features to MediaSPIP, let us know and its integration into the core MedisSPIP functionality will be considered.
    You can use the development discussion list to request for help with creating a plugin. As MediaSPIP is based on SPIP - or you can use the SPIP discussion list SPIP-Zone.

  • MediaSPIP v0.2

    21 juin 2013, par

    MediaSPIP 0.2 is the first MediaSPIP stable release.
    Its official release date is June 21, 2013 and is announced here.
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

Sur d’autres sites (6302)

  • There is no data in the inbound-rtp section of WebRTC. I don't know why

    13 juin 2024, par qyt

    I am a streaming media server, and I need to stream video to WebRTC in H.264 format. The SDP exchange has no errors, and Edge passes normally.

    


    These are the log debugging details from edge://webrtc-internals/. Both DTLS and STUN show normal status, and SDP exchange is also normal. I used Wireshark to capture packets and saw that data streaming has already started. The transport section (iceState=connected, dtlsState=connected, id=T01) also shows that data has been received, but there is no display of RTP video data at all.

    


    timestamp   2024/6/13 16:34:01
bytesSent   5592
[bytesSent_in_bits/s]   176.2108579387652
packetsSent 243
[packetsSent/s] 1.001198056470257
bytesReceived   69890594
[bytesReceived_in_bits/s]   0
packetsReceived 49678
[packetsReceived/s] 0
dtlsState   connected
selectedCandidatePairId CPeVYPKUmD_FoU/ff10
localCertificateId  CFE9:17:14:B4:62:C3:4C:FF:90:C0:57:50:ED:30:D3:92:BC:BB:7C:13:11:AB:07:E8:28:3B:F6:A5:C7:66:50:77
remoteCertificateId CF09:0C:ED:3E:B3:AC:33:87:2F:7E:B0:BD:76:EB:B5:66:B0:D8:60:F7:95:99:52:B5:53:DA:AC:E7:75:00:09:07
tlsVersion  FEFD
dtlsCipher  TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
dtlsRole    client
srtpCipher  AES_CM_128_HMAC_SHA1_80
selectedCandidatePairChanges    1
iceRole controlling
iceLocalUsernameFragment    R5DR
iceState    connected


    


    video recv info

    


    inbound-rtp (kind=video, mid=1, ssrc=2124085007, id=IT01V2124085007)
Statistics IT01V2124085007
timestamp   2024/6/13 16:34:49
ssrc    2124085007
kind    video
transportId T01
jitter  0
packetsLost 0
trackIdentifier 1395f18c-6ab9-4dbc-9149-edb59a81044d
mid 1
packetsReceived 0
[packetsReceived/s] 0
bytesReceived   0
[bytesReceived_in_bits/s]   0
headerBytesReceived 0
[headerBytesReceived_in_bits/s] 0
jitterBufferDelay   0
[jitterBufferDelay/jitterBufferEmittedCount_in_ms]  0
jitterBufferTargetDelay 0
[jitterBufferTargetDelay/jitterBufferEmittedCount_in_ms]    0
jitterBufferMinimumDelay    0
[jitterBufferMinimumDelay/jitterBufferEmittedCount_in_ms]   0
jitterBufferEmittedCount    0
framesReceived  0
[framesReceived/s]  0
[framesReceived-framesDecoded-framesDropped]    0
framesDecoded   0
[framesDecoded/s]   0
keyFramesDecoded    0
[keyFramesDecoded/s]    0
framesDropped   0
totalDecodeTime 0
[totalDecodeTime/framesDecoded_in_ms]   0
totalProcessingDelay    0
[totalProcessingDelay/framesDecoded_in_ms]  0
totalAssemblyTime   0
[totalAssemblyTime/framesAssembledFromMultiplePackets_in_ms]    0
framesAssembledFromMultiplePackets  0
totalInterFrameDelay    0
[totalInterFrameDelay/framesDecoded_in_ms]  0
totalSquaredInterFrameDelay 0
[interFrameDelayStDev_in_ms]    0
pauseCount  0
totalPausesDuration 0
freezeCount 0
totalFreezesDuration    0
firCount    0
pliCount    0
nackCount   0
minPlayoutDelay 0


    


    wireshark,I have verified that the SSRC in the SRTP is correct.

    


    enter image description here

    


    This player works normally when tested with other streaming servers. I don't know what the problem is. Is there any way to find out why the web browser cannot play the WebRTC stream that I'm pushing ?

    


  • avcodec/hevc.c : for big negative mvy value, should wait line 0 of ref frame due to...

    11 novembre 2014, par Changjiang Wei
    avcodec/hevc.c : for big negative mvy value, should wait line 0 of ref frame due to edge extending
    

    Reviewed-by : Mickaël Raulet <mraulet@gmail.com>
    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavcodec/hevc.c
  • How can i make drawtext filter start the text scrolling from the center while scrolling from right to left

    16 janvier 2018, par Femzy

    Hello How can i make the text start from the center when the video play ? Currently it start from the left side and people dont see the first words before it goes. I want the text to start from either the center of the video or the right while moving from right to left. Thanks in advance for helping me...
    This is my code that currently works for the text, but it starts from the edge of the of the left while moving from right to left

    echo shell_exec('ffmpeg -loop 1 -i slide2.jpg -vf scale=-2:1080,drawtext="textfile=text.txt:expansion=normal:fontfile=FreeSerif.otf: y=h-line_h-20:x=-200*t: fontcolor=white: fontsize=50" -pix_fmt yuv420p -t '.$words_count.' -movflags +faststart textover10.mp4 -report');