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Autres articles (25)
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Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
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) (...)
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List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)
Sur d’autres sites (6422)
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Extracting audio from video using fluent-ffmpeg
12 mars, par Idi FavourIm trying to extract the audio from a video, an error is occuring
Error converting file : Error : ffmpeg exited with code 234 : Error opening output file ./src/videos/output-audio.mp3.
Error opening output files : Invalid argument


I use this same directory for my video compression that runs before this one and it works.


ffmpeg()
 .input(url)
 .audioChannels(0)
 .format("mp3")
 .output("./src/videos/output-audio.mp3")
 .on("error", (err) => console.error(`Error converting file: ${err}`))
 .on("end", async () => {
 console.log("audio transcripts");
 
 const stream = fs.createReadStream("./src/videos/output-audio.mp3");
 const transcription = await openai.audio.transcriptions.create({
 file: stream,
 model: "whisper-1",
 response_format: "verbose_json",
 timestamp_granularities: ["word"],
 });
 transcripts = transcription.text;
 console.log(transcription.text);
 })
 .run();



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Revision bfc27bb614 : tx-skip experiment : improve entropy coding of coeff tokens This patch allows th
31 mars 2015, par hui suChanged Paths :
Modify /vp9/common/vp9_entropy.c
Modify /vp9/common/vp9_entropy.h
Modify /vp9/common/vp9_onyxc_int.h
Modify /vp9/common/vp9_quant_common.c
Modify /vp9/decoder/vp9_detokenize.c
Modify /vp9/encoder/vp9_encodemb.c
Modify /vp9/encoder/vp9_rdopt.c
Modify /vp9/encoder/vp9_tokenize.c
tx-skip experiment : improve entropy coding of coeff tokensThis patch allows the prediction residues of tx-skipped blocks
to use probs that are different from regular transfrom
coefficients for token entropy coding. Prediction residues are
assumed as in band 6.The initial value of probs is obtained with stats from limited
tests. The statistic model for constrained token nodes has not
been optimized. The probs for token extra bits have not been
optimized. These can be future work.Certain coding improvment is observed :
derflr with all experiments : +6.26% (+0.10%)
screen_content with palette : +22.48% (+1.28%)Change-Id : I1c0d78178ee9f3655febb6f30cdaef8ee9f8e3cc
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FFMPEG android very slow converting raw aac to mp3
24 novembre 2013, par FerasI managed to compile ffmpeg for android from this repo :
https://github.com/aksalj/ffmpeg-android
So you can see the configuration options in here :
https://github.com/aksalj/ffmpeg-android/blob/master/config.shThe only changes I did were to change the darwin-x86_54 to darwin-x86 since the new r9b NDK only comes with it and also change the androideabi-4.4.3 to androideabi-4.6 again because of the new NDK. I am encoding a raw ADTS aac to mp3, the decoder for aac is libav and encoder for mp3 is libmp3lame. Even though the process works a simple 5 minute aac files (44100, 2ch) is taking about 3 minutes to process. The same file on my mac (with the macports ffmpeg) takes a few seconds.
I am not sure but are there any optimizations that can be made to the ffmpeg config script or maybe i need to compile ffmpeg with something like libaacplus(even though i think aacplus is used for encoding only, so shouldnt make a difference).Anyways any performance optimizations will be highly appreciated !
The android device is a US Galaxy S3 (i747m model which has the Snapdragon S4).