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Autres articles (28)
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D’autres logiciels intéressants
12 avril 2011, parOn ne revendique pas d’être les seuls à faire ce que l’on fait ... et on ne revendique surtout pas d’être les meilleurs non plus ... Ce que l’on fait, on essaie juste de le faire bien, et de mieux en mieux...
La liste suivante correspond à des logiciels qui tendent peu ou prou à faire comme MediaSPIP ou que MediaSPIP tente peu ou prou à faire pareil, peu importe ...
On ne les connais pas, on ne les a pas essayé, mais vous pouvez peut être y jeter un coup d’oeil.
Videopress
Site Internet : (...) -
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 (...) -
De l’upload à la vidéo finale [version standalone]
31 janvier 2010, parLe chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
Upload et récupération d’informations de la vidéo source
Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)
Sur d’autres sites (4726)
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How to decode mp3 to pcm by ffmpeg
30 janvier 2017, par Meph-I need decode mp3 audio data to pcm. I have data which starts with mp3 header. Api-example.c doesn’t work, output is strange :
command
ffmpeg -i input.mp3 output.wav
is great, this is what i need. But I cant find way how to do that in code. Does anybody know, where some tutorial with ffmpeg library is ? ThanksEdit 2.7.13 :
Hi again,
I rebuilt the audio decode example method from ffmpeg and my problem is probably here :len = avcodec_decode_audio4(avCodecContext,avFrame, &got_frame,&avPacket);
int data_size = av_samples_get_buffer_size(NULL,avFrame->channels,avFrame->nb_samples,AV_SAMPLE_FMT_S16P,1);data_size is size of data frame from decoder, it depends on number of channels, number of data samples and data type(my data are 16bit PCM stereo encoded to mp3 to 1152 samples of mp3 frame)
If I open an output file in audacity, correct parameters, which give correct output, are stereo (right), 8bit pcm (wrong) and half sample rate (also wrong), what’s it happened ?
data before encoding :
16bit PCM 44100Hz, stereodata after decoding :
8bit PCM 22050Hz, stereo ---> ???!!!I’m tired of this....
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bash variable changes in loop with ffmpeg
17 septembre 2018, par MikeI wrote a skript to quickly create short preview clips from vides I recorded on timestamps that I found worth checking out later for cutting.
My file with the timestamps is written like thisFILE_NAME1#MM:SS MM:SS
FILE_NAME2#MM:SS MM:SS MM:SS MM:SSexample :
MAH01728#02:47 03:34 03:44 05:00 06:08 06:55
The script looks like this :
#!/bin/bash
while read f
do
file=$(echo $f | cut -d"#" -f1)
filename=${file}".MP4"
timestamps=$(echo $f | cut -d"#" -f2)
for time in $timestamps
do
ffmpeg -ss 00:${time}.0 -i "orig/${filename}" -c copy -t 10 "preview/${file}_${time}.MP4"
done
done < $1The script gets half of the previews that I want and on the other the filename is messed up and ffmpeg complains that the file is not found :
orig/714.MP4: No such file or directory
orig/00:58 01:25.MP4: No such file or directorySo I modified the script for trouble shooting and just put an echo in front of the ffmpeg command - now all file names are correct. What am I missing ?
ffmpeg -ss 00:01:47.0 -i orig/MAH01714.MP4 -c copy -t 10 preview/MAH01714_01:47.MP4
ffmpeg -ss 00:02:00.0 -i orig/MAH01713.MP4 -c copy -t 10 preview/MAH01713_02:00.MP4
ffmpeg -ss 00:00:58.0 -i orig/MAH01712.MP4 -c copy -t 10 preview/MAH01712_00:58.MP4
ffmpeg -ss 00:01:25.0 -i orig/MAH01712.MP4 -c copy -t 10 preview/MAH01712_01:25.MP4 -
Converting variable bitrate on the fly with ffmpeg
4 septembre 2016, par Łukasz Bezwerchnyi do cut some video files with either avidemux or virtualdub using directstream copy, some of the files comes with variable bitrate, the problem is that after such cutting my default video cataloging software shows those files lenght as 0:0, i managed to fix this problem with ffmpeg using "-vcodec copy -acodec copy". The output seem to be fine now but i have another problem, these new files have sound problem, on pot player everything is working fine but on mpc and my cataloging software which also plays media sound gets cut off at the half of the movie, for example 2h movie has only 30min of sound. I did look at the using mediainfo software and it shows me something like : video lenght 1:55, audio lenght 32min. I think the problem is the vbr, i did manage to solve this problem by extracting mp3 file from the troublesome video and convert it to 128kb constant bitrate and again merge the video and new audio file and it seems fine. Video and audio match at the whole video lenght. It’s just a bit of tedious to get things done for a big amount of files to fix, is there a parameter that i could use in one command line to do the job in one process, i mean like :
ffmpeg -i test.avi -vcodec copy -"convert vbr to 128kb" "save to test2.avi" ?