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    5 septembre 2013, par

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    Afin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
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    La documentation de l’utilisation du script d’installation (...)

  • Ajouter des informations spécifiques aux utilisateurs et autres modifications de comportement liées aux auteurs

    12 avril 2011, par

    La manière la plus simple d’ajouter des informations aux auteurs est d’installer le plugin Inscription3. Il permet également de modifier certains comportements liés aux utilisateurs (référez-vous à sa documentation pour plus d’informations).
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Sur d’autres sites (9817)

  • How to add blurred border on top and bottom based on the video

    5 avril 2023, par Noob69

    I am trying to modify a video to 1080x1920 scale and I want to add to the borders on top and bottom, a blurred version of the video based on the pixel on the edge.

    


    import subprocess

input_file = "my_video1.mp4"
output_file = "my_video_processed1.mp4"

command = f'ffmpeg -i {input_file} -vf "scale=1080:1920:force_original_aspect_ratio=decrease,pad=1080:1920:(ow-iw)/2:(oh-ih)/2,eq=saturation=2.0:gamma=1.2:contrast=1.2,unsharp=lx=5:ly=5:la=0.5:cx=5:cy=5:ca=0.5" -c:v libx264 -preset slow -crf 18 -c:a copy {output_file}'
subprocess.call(command, shell=True)



    


    I tried
mode = replicate, however it is not working for the latest version of ffmpeg from Windows builds by BtbN.

    


  • 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

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