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  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • Other interesting software

    13 avril 2011, par

    We don’t claim to be the only ones doing what we do ... and especially not to assert claims to be the best either ... What we do, we just try to do it well and getting better ...
    The following list represents softwares that tend to be more or less as MediaSPIP or that MediaSPIP tries more or less to do the same, whatever ...
    We don’t know them, we didn’t try them, but you can take a peek.
    Videopress
    Website : http://videopress.com/
    License : GNU/GPL v2
    Source code : (...)

  • À propos des documents

    21 juin 2013, par

    Que faire quand un document ne passe pas en traitement, dont le rendu ne correspond pas aux attentes ?
    Document bloqué en file d’attente ?
    Voici une liste d’actions ordonnée et empirique possible pour tenter de débloquer la situation : Relancer le traitement du document qui ne passe pas Retenter l’insertion du document sur le site MédiaSPIP Dans le cas d’un média de type video ou audio, retravailler le média produit à l’aide d’un éditeur ou un transcodeur. Convertir le document dans un format (...)

Sur d’autres sites (8107)

  • avcodec/mpegutils : Move definitions to better places

    15 octobre 2023, par Andreas Rheinhardt
    avcodec/mpegutils : Move definitions to better places

    FRAME_SKIPPED -> h263dec.h
    
CANDIDATE_MB_TYPE_* -> mpegvideoenc.h
    
INPLACE_OFFSET -> mpegvideoenc.h
    
enum OutputFormat -> mpegvideo.h
    

    Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
    • [DH] libavcodec/h263dec.c
    • [DH] libavcodec/h263dec.h
    • [DH] libavcodec/intelh263dec.c
    • [DH] libavcodec/mpeg4videoenc.c
    • [DH] libavcodec/mpegpicture.c
    • [DH] libavcodec/mpegutils.h
    • [DH] libavcodec/mpegvideo.h
    • [DH] libavcodec/mpegvideoenc.h
    • [DH] libavcodec/nvdec_mpeg12.c
    • [DH] libavcodec/ratecontrol.c
    • [DH] libavcodec/rv10.c
    • [DH] libavcodec/vc1dec.c

  • avformat/matroskadec : Assert that num_levels is non negative

    2 juin 2024, par Michael Niedermayer
    avformat/matroskadec : Assert that num_levels is non negative

    Maybe Closes : CID1452496 Uninitialized scalar variable
    

    Sponsored-by : Sovereign Tech Fund
    
Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
    • [DH] libavformat/matroskadec.c

  • 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.

    &#xA;

    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.

    &#xA;

    timestamp   2024/6/13 16:34:01&#xA;bytesSent   5592&#xA;[bytesSent_in_bits/s]   176.2108579387652&#xA;packetsSent 243&#xA;[packetsSent/s] 1.001198056470257&#xA;bytesReceived   69890594&#xA;[bytesReceived_in_bits/s]   0&#xA;packetsReceived 49678&#xA;[packetsReceived/s] 0&#xA;dtlsState   connected&#xA;selectedCandidatePairId CPeVYPKUmD_FoU/ff10&#xA;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&#xA;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&#xA;tlsVersion  FEFD&#xA;dtlsCipher  TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256&#xA;dtlsRole    client&#xA;srtpCipher  AES_CM_128_HMAC_SHA1_80&#xA;selectedCandidatePairChanges    1&#xA;iceRole controlling&#xA;iceLocalUsernameFragment    R5DR&#xA;iceState    connected&#xA;

    &#xA;

    video recv info

    &#xA;

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

    &#xA;

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

    &#xA;

    enter image description here

    &#xA;

    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 ?

    &#xA;

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