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Publier sur MédiaSpip
13 juin 2013Puis-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 -
Librairies et logiciels spécifiques aux médias
10 décembre 2010, parPour un fonctionnement correct et optimal, plusieurs choses sont à prendre en considération.
Il est important, après avoir installé apache2, mysql et php5, d’installer d’autres logiciels nécessaires dont les installations sont décrites dans les liens afférants. Un ensemble de librairies multimedias (x264, libtheora, libvpx) utilisées pour l’encodage et le décodage des vidéos et sons afin de supporter le plus grand nombre de fichiers possibles. Cf. : ce tutoriel ; FFMpeg avec le maximum de décodeurs et (...) -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...)
Sur d’autres sites (5460)
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How to record a specific window using ffmpeg ?
14 février 2023, par Matin LotfalieeI use ffmpeg to record a window using this code :



ffmpeg.exe
-f dshow 
-y 
-i video="screen-capture-recorder":audio="virtual-audio-capturer":audio="Microphone (USB Audio Device)" 
-framerate 15 
-vcodec libx264 
-crf 0 
-preset ultrafast 
-acodec pcm_s16le 
-vf crop=Width:Height:Left:Top 
output.flv




But the problem is i might move the window, this leads to recording an area without the window i want.



How can i capture a specific window that I am able to move it ?





Edit : I also used
gdigrab
to capture my window (Skype
for instance) instead ofdshow
:


ffmpeg.exe
-y
-f dshow
-i audio="virtual-audio-capturer":audio="Microphone (USB Audio Device)"
-f gdigrab
-draw_mouse 0
-i title="Skype"
-framerate 30
-vcodec libx264
-crf 0
-preset ultrafast
-acodec pcm_s16le
output.flv




But the conference is black...


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Trim Overlay memory issue
4 juillet 2019, par Костянтин ТюртюбекGreetings fellow FFmpeg users,
We are experiencing a strange memory leek issue (or maybe just doing something really wrong) and need some directions on how to debug it.
What are we trying to achieve :
Process a conference recording that includes multiple user streams, each in its own separate file (all files are mp4/opus).
- Make a dynamic scene from a set of recordings, based on their volume level at set point of time.
- The scene must include two parts : smaller grid of all the participants videos, bigger grid of currently talking people. Something like Google Hangouts or Skype does in their applications.
What went wrong :
- Memory footprint started unpredictably skyrocketing for some reason during montage
What are we using :
First FFmpeg command that reads filter_complex_script from file and adds drawbox as a talking indication on each video source file, when its volume is over a set threshold.
Second FFmpeg command that reads filter_complex_script from file and :
- takes an input file (using 0:v),
- trims a part of it, when the user was talking,
- scales it according to the amount of concurrently talking users,
- pads to that resolution (in case if user video is smaller)
filter_complex command using SELECT :
[0]select='between(t, 1, 2)', scale=762:428:force_original_aspect_ratio=decrease,pad=763:429:(ow-iw)/2:(oh-ih)/2[stream-0-workspace-scale-1-1];
[block-2-grid][stream-0-workspace-scale-1-1]
overlay=repeatlast=1:shortest=0:x=10:y=316:enable='between(t, 1, 2)'
[block-2-workspace-1];filter_complex command using TRIM :
[input-file-tag]
trim=start=#{start}:duration=#{duration},
setpts=PTS-STARTPTS,
scale=#{w-1}:#{h-1}:force_original_aspect_ratio=decrease,
pad=#{w}:#{h}:(ow-iw)/2:(oh-ih)/2
[input-file-trimmed];
[previous-block-tag]
overlay=repeatlast=1:shortest=0:x=#{x}:y=#{y}:enable='between(t, #{from}, #{to})'
[next-block-tag]We have tried going the TRIM command way, tried the SELECT command way. Problem is, both take insane amounts of ram during execution.
Examples and more description :
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Lets assume that only two of the five inputs have the volume above a
certain volume threshold from second two to five. -
We are trying to display only them according to some overlay math.
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Cropped commands in human readable form : https://pastebin.com/YwrnRgnA
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Full FFmpeg command is way too long to read through, that is the reason we started using filter_complex_script and loading it from file
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Sometimes one block of video conference may have up to 300+
intermediate overlays, which leads to the memory issue described. We were expecting the memory footprint to be similar to the amount of input files or maybe two-to-three times higher. However, we reach 15Gb or RAM usage withing the first two minutes of montage, while the input files are no bigger than 200Mb.
What have we done in terms of debugging :
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We had been using split at first, but quickly figured out that split
does in fact copy each input and load it in memory, so we had to
ditch that approach. -
As matter of fact we moved to using the input files themselves, so
the problem lies elsewhere. -
To clarify, we have split our ffmpeg command into two separate ones.
First one overlays the talking box animation using drawbox as well as
user avatar and name. It outputs new video files which we than use in
the command described above as direct input files tags, like 0:v, 1:v
etc.
Thank you for taking time reading through our issue.
We sure hope that you can help us narrow it down.
Please feel free to ask for any additional information or descriptions if needed.Have a good day !
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Using FFMPEG To Fill in Gaps of a Raw Audio UDP Stream
14 février 2020, par WallaceI have a software defined radio (SDR) that picks up audio from emergency services and with the help of software, streams raw audio using UDP. The audio is PCM signed 16-bit little-endian. The UDP stream is also not constant and only has data when audio is detected.
The problem I’m trying to solve is that I would like the gaps in recorded audio to be filled with silent or Null audio. Below are just a couple of my attempts at resolving this :
- ffmpeg -f s16le -ar 8000 -i udp ://127.0.0.1:23456 -af aresample=async=1 -acodec libmp3lame -
f rtp rtp ://127.0.0.1:1234 - ffmpeg -re -f lavfi -i anullsrc -f s16le -ar 8000 -i udp ://127.0.0.1:23456 -filter_complex
amix=inputs=2:duration=first -acodec libmp3lame -f rtp rtp ://127.0.0.1:1234
I guess my questions are regarding the best way to resolve this and whether or not ffmpeg can be used in a way to accomplish this ?
- ffmpeg -f s16le -ar 8000 -i udp ://127.0.0.1:23456 -af aresample=async=1 -acodec libmp3lame -