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Valkaama DVD Label
4 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Image
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Podcasting Legal guide
16 mai 2011, par
Mis à jour : Mai 2011
Langue : English
Type : Texte
Autres articles (111)
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Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Script d’installation automatique de MediaSPIP
25 avril 2011, parAfin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
La documentation de l’utilisation du script d’installation (...) -
Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
Sur d’autres sites (11307)
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Why frame->pts increases by 20, rather than by 1 ?
19 mars 2013, par user1914692Following the exmaples of ffmpeg : decoding_encoding.c and filtering_video.c, I process one video file taken by iPhone. The video file : .mov, video dimensions ; 480x272, video Codec : H.264/AVC, 30 frames per second, bitrate : 605 kbps.
I first extract each frame, which is YUV.
I convert YUV to RGB24, and process the RGB24, then write the RGB24 to a .ppm file. It shows the .ppm file is correct.Then I plan to encode processed RGB24 frames to a video file.
Since MPEG does not support RGB24 picture format, I used AV_CODEC_ID_HUFFYUV.
But the output video file (showing 18.5 MB) does not play. Movie Player on Ubuntu claims an error : Could not determine type of stream.
I also tried it on VCL. It simply does not work, without any error information.My second questions is :
For each extracted fram from the input video file, I get its pts as follows according to filtering_video.c :frame->pts = av_frame_get_best_effort_timestamp(frame);I print out each frame's pts, and find that it increases by 20, like below :
pFrameRGB_count: 0, frame->pts: 0
pFrameRGB_count: 1, frame->pts: 20
pFrameRGB_count: 2, frame->pts: 40
pFrameRGB_count: 3, frame->pts: 60Where frame is the extracted frame from the input video, and pFrameRGB_count is the count for processed frame in RGB24 form.
Why are they wrong ?
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How do I convert a .wav file to 16bit 44.1kz using ffmpeg or other utility [closed]
26 mai 2023, par Seth EdwardsA preface :
I am building an environment for a my own streaming box. Since building the UI. I turned to the now obsolete MSNTV box to find its UI sound effects.


I found the dump on GitHub. I downloaded and located where the sounds where located.


I listened to them one by one. I noticed that they are wave files. But they sound like they were low quality and may have been compressed before being turned into a wave file.


I was using the Apple Files app on an iPhone 6s running iOS 15.7.1.


They play back fine.


I try importing them into GarageBand for iOS and it gives me an error saying that it only allows 16bit 44.1khz files. This confirmed my suspicion of it being low quality.


I then tried playing them on a Dell Chromebook 3100 running ChromeOS. Chrome’s player would also not play the files.


I need to find out how to convert them to 16bit 44.1khz wave files.


My guess is that since the MSNTV had a small amount of storage space that they compressed the audio.


I tried converting the files to mp3. And they are Noticeably worse.


Does anyone know how to convert these files so they can be played back normally.


In the end I plan to use these files and play them using the pygame library.


I have tried changing the metadata


Converting to mp3


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ffmpeg concat demuxer dropping audio after first clip
6 septembre 2020, par marcmanI'm trying to concatenate a video collection that all should be the same type and format. It seems to work like expected when the original sources are the same type, but when they're different types the demuxer drops audio. I understand that the demux requires all inputs to have the same codecs, but I believe I am doing that already.


This is my workflow (pseudocode with Python-like for loop) :


for i, video in enumerate(all_videos):
 # Call command for transcoding and filtering
 # I allow this command to be called on mp4, mov, and avi files
 # The point of this filter is:
 # (1) to superimpose a timestamp on the bottom right of the video
 # (2) to scale and pad the videos to a common output resolution (the specific numbers below are just copied from a video I ran, but they are filled in automatically for each given video by the rest of my script)
 # (3) To transcode all videos to the same common format
 ffmpeg \
 -y \
 -loglevel quiet \
 -stats \
 -i video_<i>.{mp4, mov, avi} \
 -vcodec libx264 \
 -acodec aac \
 -vf "scale=607:1080, pad=width=1920:height=1080:x=656:y=0:color=black, drawtext=expansion=strftime: basetime=$(date +%s -d'2020-08-27 16:42:26')000000 : fontcolor=white : text='%^b %d, %Y%n%l\\:%M%p' : fontsize=36 : y=1080-4*lh : x=1263-text_w-2*max_glyph_w" \
 tmpdir/video_<i>.mp4


# create file_list.txt, e.g.
#
# file '/abspath/to/tmpdir/video_1.mp4'
# file '/abspath/to/tmpdir/video_2.mp4'
# file '/abspath/to/tmpdir/video_3.mp4'
# ...


ffmpeg \
 -f concat \
 -safe 0 \
 -i file_list.txt \
 -c copy \
 all_videos.mp4
</i></i>

In my test case, my inputs are 3 videos in this order :


- 

- a camcorder video output in H.264+aac in a mp4
- an iphone video in mov format
- an iphone video in mp4 format








When I review each of the intermediate mp4-transcoded videos in
tmpdirthey all playback audio and video just fine and the filtering works as expected. However, when I review the final concatenated output, only the first clip (the camcorder video) has sound. When all the videos are from the camcorder, there is no audio issue—they all have sound.

When I output ffmpeg warnings and errors, the only thing that shows up is an expected warning about the timestamp :


[mp4 @ 0x55997ce85300] Non-monotonous DTS in output stream 0:1; previous: 5909504, current: 5430298; changing to 5909505. This may result in incorrect timestamps in the output file.
[mp4 @ 0x55997ce85300] Non-monotonous DTS in output stream 0:1; previous: 5909505, current: 5431322; changing to 5909506. This may result in incorrect timestamps in the output file.
[mp4 @ 0x55997ce85300] Non-monotonous DTS in output stream 0:1; previous: 5909506, current: 5432346; changing to 5909507. This may result in incorrect timestamps in the output file.
[mp4 @ 0x55997ce85300] Non-monotonous DTS in output stream 0:1; previous: 5909507, current: 5433370; changing to 5909508. This may result in incorrect timestamps in the output file.
...


What might I be doing wrong here ? I'm testing in both the Ubuntu 20.04 "Videos" applications as well as VLC Player and both demonstrate the same problem. I'd prefer to use the demuxer if possible for speed as re-encoding during concatenation is quite expensive.


NOTE : This is a different issue than laid out here, in which some of the videos had no audio. In my case, all videos have both video and audio.


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