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  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (5561)

  • 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/dvbsubdec : support returning exact end times

    22 juin 2014, par Anshul Maheshwari
    avcodec/dvbsubdec : support returning exact end times

    fixess part of ticket #2024
    

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
    • [DH] libavcodec/dvbsubdec.c

  • lavu/opt : Clarify the scope of AVOptions

    24 avril 2024, par Andrew Sayers
    lavu/opt : Clarify the scope of AVOptions

    See discussion on the mailing list :
    
https://ffmpeg.org/pipermail/ffmpeg-devel/2024-April/326054.html
    

    Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
    • [DH] libavutil/opt.h

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