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Les Miserables
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Pourquoi Obama lit il mes mails ?
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Autres articles (66)
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Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
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Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...)
Sur d’autres sites (11185)
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FFmpeg youtube stream get corrupted until restart
19 février 2023, par unpwnI am streaming images from python to youtube using ffmpeg.
It's working fine with just one problem.
Sometimes the stream get somehow corrupted an the stream looks like this :




Or you can see for yourself in the stream itself : https://www.youtube.com/watch?v=PG6NLtr0jjM (Scroll back to the beginning)


Then I have to restart ffmpeg and its working fine. Sometimes an error occurs regarding the network and I have to automatically restart the stream. Then there is the possibility it is corrupted again.


The parameters for ffmpeg are the following :


cmd_out = ['ffmpeg',
 '-re',
 '-f', 'concat',
 '-i', os.path.join(BASE_DIR, 'music', 'music.txt').replace('\\', '/'), 
 '-r', '1',
 '-f', 'image2pipe',
 '-vcodec', 'png',
 '-r', '1',
 '-i', '-',
 '-c:v', 'libx264',
 '-c:a', 'copy',
 '-g', '4',
 '-f', 'flv',
 'rtmp://a.rtmp.youtube.com/live2/{}'.format(stream_key)] 



I never had that problem testing it on windows. It just appears when I use my linux server.


I am using ffmpeg version 3.2.18-0+deb9u1 and the generated images are fine, when I save them to a normal file.


Does someone know how to fix it ? Or where the problem is comming from ?



Solution : An old ffmpeg version was available while using docker. I solved the problem by using a static build from https://www.johnvansickle.com/ffmpeg/


I used this to use it in docker :


WORKDIR /app
ADD ./ffmpeg /app
RUN mv ffmpeg /usr/local/ffmpeg
RUN chmod +x /usr/local/ffmpeg



Until now the error seems to be solved.


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FFmpeg Change audio filter parameters during runtime
31 janvier 2021, par cs guyI am making a music app where the user can add FX to playing music through a pad. I am using FFmpeg in C++ to dsp. In FFmpeg you can create an audio filter and set it parameters just like the following :


beat_fx = avfilter_get_by_name("aecho");
 beat_fx_ctx = avfilter_graph_alloc_filter(filterGraph, beat_fx, "echo");
 av_opt_set(beat_fx_ctx, "decays", std::to_string(std::max(0.00001, (beatFX.xSoundPerc + 1) / (double)2)).c_str(), AV_OPT_SEARCH_CHILDREN);
 av_opt_set(beat_fx_ctx, "delays", std::to_string(beatFX.ms).c_str(), AV_OPT_SEARCH_CHILDREN);
 av_opt_set_double(beat_fx_ctx, "in_gain", 1.0, AV_OPT_SEARCH_CHILDREN);
 av_opt_set_double(beat_fx_ctx, "out_gain", 1.0, AV_OPT_SEARCH_CHILDREN);

 if (avfilter_init_str(beat_fx_ctx, nullptr) < 0) {
 LOGE("FXProcessor::FXProcessor Could not initialize the beat_fx_ctx filter!");
 isOff = true;
 return;
 }



My problem is since the user will use a FX pad to change these parameters I need to be able to modify these parameters during runtime.


Looking at ffmpeg filter documentation :




Some options can be changed during the operation of the filter using a
command. These options are marked ’T’ on the output of ffmpeg -h
filter=. The name of the command is the name of the
option and the argument is the new value.




I looked at
aecho, agate, acrusher
and more but nearly all the effects I want have 0 modifiable option which makes my FX pad nonadjustable. I am not using command line to process my effects, so maybe the link above is irrelevant.

Is there a way to make ffmpeg audio filters change their parameters during runtime ?


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ffmpeg/sox audio processing : Merging files with envelope changes
2 octobre 2020, par March HareSo I have two audio files. One is a music bed with an intro that segues into a looping music clip (let's call this *1). The second is the voice over audio track (referenced as *2, length n).


Audio *1 is fixed, while the voice over (*2) is downloaded about 3 times a day, and can vary in length. *1 is longer than we ever expect *2 to be.


What I need to do is


- 

- Alter the overall gain of *1 to -7.5 dB
- Begin merging VO *2 at time m, while reducing the volume envelope of *1 by -11 dB. This is fixed based on the length of the intro.
- Fade everything out to -∞ dB around the end of *2
- Trim off the silence at the end.
For reference, the total length of the final track should be m+n.










Unfortunately, I'm not versed enough with ffmpeg or sox to know exactly what I'm after here, and a lot of the examples tend to do one thing or another and aren't always clear when combining happens. I didn't get a lot of prior notice about this coming down the pipeline, so I'd like to get something relatively quick. We're able to do all of this stuff nicely in Adobe Audition (and I can do something similar in Audacity), but the idea is to automate it. For our envelope adjustments, we were just using linear ramps rather than smoothsteps, and that sounded fine.


The TLDR : The VO track *2 governs how long the file winds up being, while audio bed *1 needs to be ducked when *2 begins, and the whole thing faded out right when *2 ends.


We also have an automation system (radio station automation, specialized for something different than I need), so in a pinch if we have to just cut off the audio at the end of *2, we can get the fadeout from the radio automation system.


I've been using the information at this link to some effect (specifically the bit about ffmpeg volumes), but it still isn't dynamic enough for the situation.
Envelope pattern in SoX (Sound eXchange) or ffmpeg


Anyone have any advice on this one ? I've got Sox and ffmpeg available, and if need be I can probably install other tools as well.