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Autres articles (48)
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MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs
Sur d’autres sites (7953)
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Can ffmpeg transcode an audio track and add it as a second audio track at the same time, or if not, how to do it as separate commands ?
31 mai 2020, par wb6vpmA bit of history. I am using Plex as my media server, but for reasons unknown, it has issues transcoding the DTS-HD MA 7.1 audio to EAC3 stereo and keeps buffering (the server has plenty of horsepower on all fronts, CPU/RAM/drive space & speed, gigabit networks connections for all devices. The playback device (TCL Roku TV, with a 3rd party soundbar connected via HDMI ARC) doesn't support the built-in 7.1 audio, so I get silence if I play it back directly by putting the file on a USB stick.



Also, I am by no means a ffmpeg guru, I figured out what I do know by Google University and asking questions, so please be kind and forgive me if I ask follow-up questions that may seem n00b-ish, and please provide example commands (preferably in the context of my command below so that I can have a known point of reference to start with).



I have a movie with 4K (HEVC Main 10 HDR) video and DTS-HD MA 7.1 audio that I am looking to leave the video and audio untouched, but to add a 2nd audio track in either EAC3 or if necessary, just AC3 in stereo



So what I am looking for is as follows :



video.mkv



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- Existing->4k video file (no change)
- Existing->7.1 audio (no change)
- Convert and add->stereo audio as a 2nd audio track to the output.mkv file









Below is the command I've historically used with ffmpeg to convert and replace the audio file with the stereo audio, but since I'd prefer to leave the 7.1 audio in place, this doesn't work :

ffmpeg -i "D:\video.mkv" -c:v copy -c:a aac -b:a 128k "D:\output.mkv"



And if this cannot be done as a single command, please also let me know what steps I do need to take to be able to do it.



Thanks in advace,
Mike


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ffmpeg : VLC won’t open .sdp files generated by ffmpeg
8 mai 2020, par tiredamage42TLDR : VLC or Quicktime won’t open .sdp video stream files generated by ffmpeg, even though ffplay does.



Web development and ffmpeg noob, so apologies if I’m using the wrong terminology :



I’m trying to stream my desktop capture (on OSX) using ffmpeg, sending it out via rtp protocol. 
As of right now I’m just testing it out by streaming it over a port in my localhost (4000). And trying to play it locally.



The problem is that when I try and open the .sdp file generated by the ffmpeg command, VLC opens it and immediately stops, no errors or anything, and shows that it has a duration of 0:00. Quicktime won’t event open the file in the first place.



ffplay does play the stream and I can see my desktop in the player window (with a significant loss in quality though). Even so there are a ton of warning and errors that show up intermittently (outlined below)



I’m not sure if it’s a problem in the way I start the ffmpeg stream, the command is after a ton of iterations of trying to just make it work, so my options might be way wrong.



command to 'serve' the desktop capture :



./ffmpeg -f avfoundation -s 1920x1080 -r 60 -i "1" -an \
-vcodec libx264 -preset ultrafast -tune zerolatency -pix_fmt yuv420p \
-sdp_file video.sdp -rtsp_transport tcp -f rtp rtp://127.0.0.1:4000




SDP file that’s generated with the ffmpeg command :



SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.29.100
m=video 4000 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1




ffplay command used to play the stream :



./ffplay -probesize 32 -analyzeduration 0 -sync ext \
-fflags nobuffer -fflags discardcorrupt -flags low_delay -framedrop \
-strict experimental -avioflags direct \
-protocol_whitelist file,rtp,udp -I video.sdp




For a while before ffplay starts I see a bunch of these errors repeating (in red) :



[h264 @ 0x7ff6b788de00] non-existing PPS 0 referenced
[h264 @ 0x7ff6b788de00] decode_slice_header error
[h264 @ 0x7ff6b788de00] no frame!




then the window seems to 'catch up' to the stream and actually shows the desktop capture, and I get these errors and warnings on a regular interval :



1- in a yellow warning color :



[sdp @ 0x7fc85b830600] RTP: missed 4 packets
[sdp @ 0x7fc85b830600] max delay reached. need to consume packet




2-in a red error color :



[h264 @ 0x7fc85b02aa00] out of range intra chroma pred mode
[h264 @ 0x7fc85b02aa00] error while decoding MB 132 32




(I have a feeling that the above errors have to do with previewing the desktop capture in the desktop I’m capturing and causing pixels in the display to overflow)



Edit :
So, I solved the issue soon after posting, but will leave this up in case anyone runs into the same problem.



The solution was to remove the top line in the .sdp file that said
SDP:


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Discord Music Bot joins voice channel, light up green but didnt has any audio. Worked well for 2 weeks before. No errors in console
8 juin 2021, par FeXI coded a bot with node.js. I used the example by Crawl for his music bot. I did everything similar to him. After I finished my build everything worked. Every other command and the
play
command. But now after 2 weeks the bot joins the voice channel, light up green but has no sound. I updatedffmpeg
,@discordjs/opus
,ffmpeg-static
and downloaded the completed version from ffmpeg but the bot still has no audio. Thequeue
,volume
,nowplaying
,skip
,shuffle
,loop
everything works. But after I got the video or playlist with the play command the bot only joins light up green but has no audio. So the bot definitly get the url, get the video, get everything he needs to play. But after joining he doesnt use the informations to play. Also he doesnt leave the voicechannel after the song should end.


function play(guild, song) {

 try {

 const ServerMusicQueue = queue.get(guild.id);

 if (!song) {

 ServerMusicQueue.textchannel.send(`ퟎ