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Médias (1)
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Video d’abeille en portrait
14 mai 2011, par
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (46)
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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 -
Librairies et logiciels spécifiques aux médias
10 décembre 2010, parPour un fonctionnement correct et optimal, plusieurs choses sont à prendre en considération.
Il est important, après avoir installé apache2, mysql et php5, d’installer d’autres logiciels nécessaires dont les installations sont décrites dans les liens afférants. Un ensemble de librairies multimedias (x264, libtheora, libvpx) utilisées pour l’encodage et le décodage des vidéos et sons afin de supporter le plus grand nombre de fichiers possibles. Cf. : ce tutoriel ; FFMpeg avec le maximum de décodeurs et (...) -
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 (...)
Sur d’autres sites (8547)
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not able to convert a specific .wav file to mp3 or m4a with sox, avconv
25 juillet 2017, par astrographAt the office we have a project where we apply IoT technologies to a real bee hive.
One of the features is to detect specific sounds the bees make when a new queen hatches. We have a special microphone in place, the algorithm is also implemented. For now we get a lot of false positives, and want to quickly be able to identify them, by listening to the audio files in the browser. Therefore I want to convert the .wav files to either .mp3 or .m4a
The .wav file format seems to be quite strange, as I was not able to convert it to mp3 using avconv, sox or even audacity. The funny thing is, the Microsoft media player can play the .wav file fine.
Here is the information soxi gives about the wav file :
pi@raspberrypi:~ $ soxi Channel1.wav
soxi WARN wav: wave header missing extended part of fmt chunk
Input File : 'Channel1.wav'
Channels : 1
Sample Rate : 6250
Precision : 24-bit
Duration : 00:01:21.00 = 506250 samples ~ 6075 CDDA sectors
File Size : 2.03M
Bit Rate : 200k
Sample Encoding: 32-bit Floating Point PCMThis is the avconv command I am trying to use :
avconv -y -v quiet -i Channel1.wav -strict experimental -ar 44100 -ab 160k Channel1.m4a
I also tried with sox :
sox -v 0.60 Channel1.wav -r 22050 Channel1.m4a
but the output is mostly silent, with some random noise.
The question is how can a wav file like this : https://drive.google.com/open?id=0B9YVh-jkOMLsQThERlI2emN2QWM be converted to an audio format using a raspberry pi that can be played in the browser ?
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Anomalie #4643 : {logo} ne marche pas sur un logo inséré automatiquement par le référencement d’un...
28 janvier 2021Précision : alors que le logo devrait être dans IMG/logo/fichierdulogo il se trouve dans IMG/siteon10.png
Il manque donc le dossier logo/. -
ac3enc_fixed : convert to 32-bit sample format
9 janvier 2021, par Lynneac3enc_fixed : convert to 32-bit sample format
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.SIZE SAVINGS :
ARM32 :
HARDCODED TABLES :
BASE - 10709590
DROP DSP - 10702872 - diff : -6.56KiB
DROP MDCT - 10667932 - diff : -34.12KiB - both : -40.68KiB
DROP FFT - 10336652 - diff : -323.52KiB - all : -364.20KiB
SOFTCODED TABLES :
BASE - 9685096
DROP DSP - 9678378 - diff : -6.56KiB
DROP MDCT - 9643466 - diff : -34.09KiB - both : -40.65KiB
DROP FFT - 9573918 - diff : -67.92KiB - all : -108.57KiBARM64 :
HARDCODED TABLES :
BASE - 14641112
DROP DSP - 14633806 - diff : -7.13KiB
DROP MDCT - 14604812 - diff : -28.31KiB - both : -35.45KiB
DROP FFT - 14286826 - diff : -310.53KiB - all : -345.98KiB
SOFTCODED TABLES :
BASE - 13636238
DROP DSP - 13628932 - diff : -7.13KiB
DROP MDCT - 13599866 - diff : -28.38KiB - both : -35.52KiB
DROP FFT - 13542080 - diff : -56.43KiB - all : -91.95KiBx86 :
HARDCODED TABLES :
BASE - 12367336
DROP DSP - 12354698 - diff : -12.34KiB
DROP MDCT - 12331024 - diff : -23.12KiB - both : -35.46KiB
DROP FFT - 12029788 - diff : -294.18KiB - all : -329.64KiB
SOFTCODED TABLES :
BASE - 11358094
DROP DSP - 11345456 - diff : -12.34KiB
DROP MDCT - 11321742 - diff : -23.16KiB - both : -35.50KiB
DROP FFT - 11276946 - diff : -43.75KiB - all : -79.25KiBPERFORMANCE (10min random s32le) :
ARM32 - before - 39.9x - 0m15.046s
ARM32 - after - 28.2x - 0m21.525s
Speed : -30%ARM64 - before - 36.1x - 0m16.637s
ARM64 - after - 36.0x - 0m16.727s
Speed : -0.5%x86 - before - 184x - 0m3.277s
x86 - after - 190x - 0m3.187s
Speed : +3%- [DH] doc/encoders.texi
- [DH] libavcodec/Makefile
- [DH] libavcodec/ac3enc.c
- [DH] libavcodec/ac3enc.h
- [DH] libavcodec/ac3enc_fixed.c
- [DH] libavcodec/ac3enc_float.c
- [DH] libavcodec/ac3enc_template.c
- [DH] libavcodec/version.h
- [DH] tests/fate/ac3.mak
- [DH] tests/fate/ffmpeg.mak
- [DH] tests/ref/fate/unknown_layout-ac3
- [DH] tests/ref/lavf/rm