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Spitfire Parade - Crisis
15 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
Autres articles (56)
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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 (...) -
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 -
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
Sur d’autres sites (7791)
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avcodec_decode_video2 takes long time to decode wmv
22 février 2013, par Mike VersteegI have written a player using the ffmpeg libraries that plays several avi, mp4 and mpg files fine but not wmv. Debugging has shown that the culprit is avcodec_decode_video2 which for some reason takes half a second to decode a single wmv frame. I have tested and confirmed this with multiple wmv files, even those with much lower resolution then other files that play successfully. Anyone has a clue what I'm doing wrong ?
PS : log shows "Extra data : 8 bits left, value : 0" and "parser not found for codec wmav2, packets or times may be invalid." for WMV files.
UPDATE :
Further examination has shown the call to avcodec_decode_video2 is not slow, instead it seems to return the same frame some 16 times or so, giving the visual impression that it is slow. It does not return an error code so I have no clue why it won't decode all frames in the case of wmv.Relevant code :
AVPacket packet;
av_init_packet(&packet);
done = pOwner->av_read_frame(pFormatCtx, &packet) < 0;
if (!done && packet.stream_index==videostream)
bool error = pOwner->avcodec_decode_video2(pVCodecCtx, pFrameYUV, &finished, &packet) < 0; -
PyAV/ffmpeg : How to resolve 'Reinit context to WxH ... deprecated pixel format'
26 mai 2023, par hi im BaconI am using the
pyav
library to read and process video data and as I am reading through the videos I am getting lots (and lots) of the following output.

I am creating the videos by segmenting either a much larger source file or RTSP/HTTP IP camera.


I would love to understand what this output actually means, and what I can do to prevent its output


ffmpeg creating videos :


ffmpeg -y -i https://manifest.googlevideo.com/api/... -pix_fmt yuvj422p -c:v libx264 -an -vf scale='min(1280,iw)':'min(720,ih)':force_original_aspect_ratio=decrease -r 12 -vsync cfr -x264opts keyint=12:min-keyint=12 -f segment -segment_format matroska -segment_time 5 -initial_offset 1656497474.686672 -strftime 1 /tmp/tmp87cc1n2o/%Y-%m-%dT%H%M%S.mkv -loglevel error



python :


container = av.open(io.BytesIO(...))
for frame in container.decode(video=0):
 ...



2022-06-30 14:45:00,964 DEBUG:****:Creating <***.core.generators.av.AVGenerator object at 0x7f13340abd50> with target path: <_io.BytesIO object at 0x7f13341c93b0>
2022-06-30 14:45:00,966 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
2022-06-30 14:45:00,966 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuvj422p
2022-06-30 14:45:00,984 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
2022-06-30 14:45:01,045 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
2022-06-30 14:45:01,046 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuvj422p
deprecated pixel format used, make sure you did set range correctly
deprecated pixel format used, make sure you did set range correctly
 (repeated 9 more times)
2022-06-30 14:45:01,194 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
deprecated pixel format used, make sure you did set range correctly
deprecated pixel format used, make sure you did set range correctly
2022-06-30 14:45:01,215 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
deprecated pixel format used, make sure you did set range correctly
deprecated pixel format used, make sure you did set range correctly
 (repeated 17 more times)
2022-06-30 14:45:01,384 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
deprecated pixel format used, make sure you did set range correctly
deprecated pixel format used, make sure you did set range correctly
 (repeated 3 more times)
2022-06-30 14:45:01,398 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
deprecated pixel format used, make sure you did set range correctly
deprecated pixel format used, make sure you did set range correctly
 (repeated 22 more times)
2022-06-30 14:45:01,550 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p
deprecated pixel format used, make sure you did set range correctly
deprecated pixel format used, make sure you did set range correctly
 (repeated 2 more times)
2022-06-30 14:45:01,592 DEBUG:libav.h264:Reinit context to 1280x720, pix_fmt: yuv420p



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Making volume adjustments at precise time with ffmpeg -filter_complex
25 février 2023, par a1s2d3f4I am using the following command to mute parts of the audio file with ffmpeg :


ffmpeg.exe -y -i "C:\temp\inputfile3.wav" -filter_complex_script "C:\temp\filter_complex_cmds.txt" "C:\temp\outputfile3.wav"



Inside filter_complex_cmd.txt I have :


[0]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume='if(between(t,0.95,1.05),0*(t-0.95) + 0,1)':eval=frame,volume='if(between(t,1.15,1.25),0*(t-1.15) + 0,1)':eval=frame,volume='if(between(t,1.41,1.49),0*(t-1.41) + 0,1)':eval=frame,volume='if(between(t,2.10,2.35),0*(t-2.10) + 0,1)':eval=frame,volume='if(between(t,2.75,2.85),0*(t-2.75) + 0,1)':eval=frame, etc. x1000...



You get the idea - many, many calls to reduce volume at specific times.
However, when I look at outputfile3.wav, the volume is not reduced from 0.95000 to 1.05000 seconds, but instead from 0.952018 to 1.068118 (i.e. off by 2 and 18 milliseconds respectively) and not from 1.150000 to 1.250000 seconds but from 1.160998 to 1.253878 milliseconds, etc. It can sometimes be off by as many as 20 milliseconds.


Can anyone tell me what's going on and what to do to make it precise ?


Also, while I can reserve it for a separate question, but I also want to be able to find a fade out/fade in commands that would make this change to and from silence a bit more smooth by dialing it down and up over the course of 20 milliseconds around my "volume 0" times.