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Autres articles (82)
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Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...) -
Menus personnalisés
14 novembre 2010, parMediaSPIP utilise le plugin Menus pour gérer plusieurs menus configurables pour la navigation.
Cela permet de laisser aux administrateurs de canaux la possibilité de configurer finement ces menus.
Menus créés à l’initialisation du site
Par défaut trois menus sont créés automatiquement à l’initialisation du site : Le menu principal ; Identifiant : barrenav ; Ce menu s’insère en général en haut de la page après le bloc d’entête, son identifiant le rend compatible avec les squelettes basés sur Zpip ; (...) -
Gestion de la ferme
2 mars 2010, parLa ferme est gérée dans son ensemble par des "super admins".
Certains réglages peuvent être fais afin de réguler les besoins des différents canaux.
Dans un premier temps il utilise le plugin "Gestion de mutualisation"
Sur d’autres sites (8089)
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washed out colors when converting h264 into vp9 using ffmpeg [closed]
31 juillet 2024, par apes-together-strongI'm trying to convert an h264 video to vp9/opus webm.
Every attempt so far has had washed out colors.
Is there a fix for this issue or is it just the way h264->vp9 conversion is ?


ffprobe of the source file :


Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test1.mp4':
 Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: isommp42
 creation_time : 2024-07-30T17:03:10.000000Z
 com.android.version: 10
 Duration: 00:01:04.07, start: 0.000000, bitrate: 20198 kb/s
 Stream #0:0[0x1](eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, bt470bg/bt470bg/smpte170m, progressive), 1920x1080, 20001 kb/s, SAR 1:1 DAR 16:9, 30 fps, 30 tbr, 90k tbn (default)
 Metadata:
 creation_time : 2024-07-30T17:03:10.000000Z
 handler_name : VideoHandle
 vendor_id : [0][0][0][0]
 Side data:
 displaymatrix: rotation of -90.00 degrees
 Stream #0:1[0x2](eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 96 kb/s (default)
 Metadata:
 creation_time : 2024-07-30T17:03:10.000000Z
 handler_name : SoundHandle
 vendor_id : [0][0][0][0]



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There is no data in the inbound-rtp section of WebRTC. I don't know why
13 juin 2024, par qytI 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. Thetransport
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.




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 ?


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Revision 30295 : Amélioration de l’encodage multiple
28 juillet 2009, par kent1@… — LogAmélioration de l’encodage multiple