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Autres articles (91)
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Mise à jour de la version 0.1 vers 0.2
24 juin 2013, parExplications des différents changements notables lors du passage de la version 0.1 de MediaSPIP à la version 0.3. Quelles sont les nouveautés
Au niveau des dépendances logicielles Utilisation des dernières versions de FFMpeg (>= v1.2.1) ; Installation des dépendances pour Smush ; Installation de MediaInfo et FFprobe pour la récupération des métadonnées ; On n’utilise plus ffmpeg2theora ; On n’installe plus flvtool2 au profit de flvtool++ ; On n’installe plus ffmpeg-php qui n’est plus maintenu au (...) -
Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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Ecrire une actualité
21 juin 2013, parPrésentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
Vous pouvez personnaliser le formulaire de création d’une actualité.
Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...)
Sur d’autres sites (15968)
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How would I add an audio channel to an rtsp stream ?
9 septembre 2022, par PlayerWetGood companions. It turns out that my Raspberry does not give more than itself when it comes to joining video with audio at good quality and sending it by rtsp. But I think I could send the video in rtsp format and then the audio in mp3, but I would need to join it again on another computer (nas Debian) on my home lan where I have the Shinobi program (security camera manager) installed.


I would need something that can somehow grab the rtsp stream and another mp3 audio and merge them into a new rtsp stream. Is this possible ? or is it a crazy idea.


On the one hand I send this, which is the transmission of the camera by rtsp through v4l2rtspserver :


v4l2rtspserver -H 1080 -W 1920 -F 25 -P 8555 /dev/video0



And separately I send an audio in mp3 with sound from a usb microphone through ffmpeg :


ffmpeg -ac 1 -f alsa -i hw:1,0 -acodec libmp3lame -ab 32k -ac 1 -f rtp rtp://192.168.1.77:12348



My idea is to put both things together on a nas server in a new rtsp stream (or another idea).


But I don't know if with ffmpeg I can capture the video from an rtsp video stream and then be able to join it with the mp3 audio and form another new rtsp stream.


Merge these two streams into one and reassemble an rtsp :


rtsp://192.168.1.57:8555/unicast

rtp://192.168.1.77:12348



I have tried this way but it gives me an error :


ffmpeg \
 -i rtsp://192.168.1.57:8555/unicast \
 -i rtp://192.168.1.37:12348 \
 -acodec copy -vcodec libx264 \
 -f rtp_mpegts "rtp://192.168.1.77:5000?ttl=2"



Error :


[h264 @ 0x55ac6acaf4c0] non-existing PPS 0 referenced
 Last message repeated 1 times
[h264 @ 0x55ac6acaf4c0] decode_slice_header error
[h264 @ 0x55ac6acaf4c0] no frame!
[h264 @ 0x55ac6acaf4c0] non-existing PPS 0 referenced
 Last message repeated 1 times
[h264 @ 0x55ac6acaf4c0] decode_slice_header error
[h264 @ 0x55ac6acaf4c0] no frame!
[h264 @ 0x55ac6acaf4c0] non-existing PPS 0 referenced
 Last message repeated 1 times



What am I doing wrong ?


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Extract Video Frames from SDP Output
1er août 2022, par user1446797Does anyone know how to extract image frames from a SDP video output ? I'm using a Nest battery camera. The wired version gave me an RTSP stream which was easy to extract frames. However, the battery version gave me a SDP output which is hard to make sense of. I've looked at a few posts on stackoverflow but none seemed too promising :




Executing FFmpeg recording using in-line SDP


Even being able to stream SDP to a mp4 file using ffplay would be a nice start. But ultimately I would like to run a python script to extract frames from SDP output.


I must admit, SDP (session description protocol) seems pretty long and complicated compared to working with RTSP streams. Anyway to simply convert an SDP stream to a RTSP stream ?


https://andrewjprokop.wordpress.com/2013/09/30/understanding-session-description-protocol-sdp/


Thanks !
Jacob


SDP output looks something like this :


v=0\r\no=- 0 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 2 1\r\na=msid-semantic : WMS 16733765853514488918/633697675 virtual-6666\r\na=ice-lite\r\nm=audio 19305 UDP/TLS/RTP/SAVPF 111\r\nc=IN IP4 142.250.9.127\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=candidate : 1 udp 2113939711 2607:f8b0:4002:c11::7f 19305 typ host generation 0\r\na=candidate : 1 tcp 2113939710 2607:f8b0:4002:c11::7f 19305 typ host tcptype passive generation 0\r\na=candidate : 1 ssltcp 2113939709 2607:f8b0:4002:c11::7f 443 typ host generation 0\r\na=candidate : 1 udp 2113932031 142.250.9.127 19305 typ host generation 0\r\na=candidate : 1 tcp 2113932030 142.250.9.127 19305 typ host tcptype passive generation 0\r\na=candidate : 1 ssltcp 2113932029 142.250.9.127 443 typ host generation 0\r\na=ice-ufrag:UVDO0GOJASABT95E\r\na=ice-pwd:FRILJDCJZCH+51YNWDGZIN0K\r\na=fingerprint:sha-256 24:53:14:34:59:50:89:52:72:58:04:57:71:BB:C4:89:91:3A:52:EF:C0:5A:A5:EC:B5:51:64:80:AC:13:89:8A\r\na=setup:passive\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=sendrecv\r\na=msid:virtual-6666 virtual-6666\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10 ;useinbandfec=1\r\na=ssrc:6666 cname:6666\r\nm=video 9 UDP/TLS/RTP/SAVPF 108 109\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:UVDO0GOJASABT95E\r\na=ice-pwd:FRILJDCJZCH+51YNWDGZIN0K\r\na=fingerprint:sha-256 24:53:14:34:59:50:89:52:72:58:04:57:71:BB:C4:89:91:3A:52:EF:C0:5A:A5:EC:B5:51:64:80:AC:13:89:8A\r\na=setup:passive\r\na=mid:1\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:13 urn:3gpp:video-orientation\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=sendrecv\r\na=msid:16733765853514488918/633697675 16733765853514488918/633697675\r\na=rtcp-mux\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack\r\na=rtcp-fb:108 nack pli\r\na=rtcp-fb:108 goog-remb\r\na=fmtp:108 level-asymmetry-allowed=1 ;packetization-mode=1 ;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=ssrc-group:FID 633697675 3798748564\r\na=ssrc:633697675 cname:633697675\r\na=ssrc:3798748564 cname:633697675\r\nm=application 9 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:UVDO0GOJASABT95E\r\na=ice-pwd:FRILJDCJZCH+51YNWDGZIN0K\r\na=fingerprint:sha-256 24:53:14:34:59:50:89:52:72:58:04:57:71:BB:C4:89:91:3A:52:EF:C0:5A:A5:EC:B5:51:64:80:AC:13:89:8A\r\na=setup:passive\r\na=mid:2\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n


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FFmpeg - sharpening the high pass filter
11 août 2024, par kubinka0505I'm writing the command that is resampling the audio file and adding the high-pass filter to it.


In the software that I'm using, the high pass filter is very sharp - the problem is that I don't know how to sharpify the filter in FFmpeg.


I've experimented with some options but none succeded. How can I achieve that ?


ffmpeg -i "Audio.wav" -ar 44100 -af highpass=f=5000 "Audio_HP.wav" -y








 What FFmpeg does 

What I want