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Médias (33)
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Stereo master soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Audio
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#7 Ambience
16 octobre 2011, par
Mis à jour : Juin 2015
Langue : English
Type : Audio
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#6 Teaser Music
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#5 End Title
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#3 The Safest Place
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#4 Emo Creates
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
Autres articles (32)
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Soumettre améliorations et plugins supplémentaires
10 avril 2011Si vous avez développé une nouvelle extension permettant d’ajouter une ou plusieurs fonctionnalités utiles à MediaSPIP, faites le nous savoir et son intégration dans la distribution officielle sera envisagée.
Vous pouvez utiliser la liste de discussion de développement afin de le faire savoir ou demander de l’aide quant à la réalisation de ce plugin. MediaSPIP étant basé sur SPIP, il est également possible d’utiliser le liste de discussion SPIP-zone de SPIP pour (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...)
Sur d’autres sites (5899)
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Bit-field badness
30 janvier 2010, par Mans — Compilers, OptimisationConsider the following C code which is based on an real-world situation.
struct bf1_31 unsigned a:1 ; unsigned b:31 ; ;
void func(struct bf1_31 *p, int n, int a)
int i = 0 ;
do
if (p[i].a)
p[i].b += a ;
while (++i < n) ;
How would we best write this in ARM assembler ? This is how I would do it :
func : ldr r3, [r0], #4 tst r3, #1 add r3, r3, r2, lsl #1 strne r3, [r0, #-4] subs r1, r1, #1 bgt func bx lr
The
add
instruction is unconditional to avoid a dependency on the comparison. Unrolling the loop would mask the latency of theldr
instruction as well, but that is outside the scope of this experiment.Now compile this code with
gcc -march=armv5te -O3
and watch in horror :func : push r4 mov ip, #0 mov r4, r2 loop : ldrb r3, [r0] add ip, ip, #1 tst r3, #1 ldrne r3, [r0] andne r2, r3, #1 addne r3, r4, r3, lsr #1 orrne r2, r2, r3, lsl #1 strne r2, [r0] cmp ip, r1 add r0, r0, #4 blt loop pop r4 bx lr
This is nothing short of awful :
- The same value is loaded from memory twice.
- A complicated mask/shift/or operation is used where a simple shifted add would suffice.
- Write-back addressing is not used.
- The loop control counts up and compares instead of counting down.
- Useless
mov
in the prologue ; swapping the roles orr2
andr4
would avoid this. - Using
lr
in place ofr4
would allow the return to be done withpop {pc}
, saving one instruction (ignoring for the moment that no callee-saved registers are needed at all).
Even for this trivial function the gcc-generated code is more than twice the optimal size and slower by approximately the same factor.
The main issue I wanted to illustrate is the poor handling of bit-fields by gcc. When accessing bitfields from memory, gcc issues a separate load for each field even when they are contained in the same aligned memory word. Although each load after the first will most likely hit L1 cache, this is still bad for several reasons :
- Loads have typically two or three cycles result latency compared to one cycle for data processing instructions. Any bit-field can be extracted from a register with two shifts, and on ARM the second of these can generally be achieved using a shifted second operand to a following instruction. The ARMv6T2 instruction set also adds the
SBFX
andUBFX
instructions for extracting any signed or unsigned bit-field in one cycle. - Most CPUs have more data processing units than load/store units. It is thus more likely for an ALU instruction than a load/store to issue without delay on a superscalar processor.
- Redundant memory accesses can trigger early flushing of store buffers rendering these less efficient.
No gcc bashing is complete without a comparison with another compiler, so without further ado, here is the ARM RVCT output (
armcc --cpu 5te -O3
) :func : mov r3, #0 push r4, lr loop : ldr ip, [r0, r3, lsl #2] tst ip, #1 addne ip, ip, r2, lsl #1 strne ip, [r0, r3, lsl #2] add r3, r3, #1 cmp r3, r1 blt loop pop r4, pc
This is much better, the core loop using only one instruction more than my version. The loop control is counting up, but at least this register is reused as offset for the memory accesses. More remarkable is the push/pop of two registers that are never used. I had not expected to see this from RVCT.
Even the best compilers are still no match for a human.
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Batch splitting large audio files into small fixed-length audio files in moments of silence
26 juillet 2023, par Haldjärvito train the SO-VITS-SVC neural network, we need 10-14 second voice files. As a material, let's say I use phrases from some game. I have already made a batch script for decoding different files into one working format, another batch script for removing silence, as well as a batch script for combining small audio files into files of 13-14 seconds (I used Python, pydub and FFmpeg). To successfully automatically create a training dataset, it remains only to make one batch script - Cutting audio files lasting more than 14 seconds into separate files lasting 10-14 seconds, cutting in places of silence or close to silence is highly preferable.


So, it is necessary to batch cut large audio files (20 seconds, 70 seconds, possibly several hundred seconds) into segments of approximately 10-14 seconds, however, the main task is to look for the quietest place in the cut areas so as not to cut phrases in the middle of a word (this is not very good for model training). So, is it really possible to do this in a very optimal way, so that the processing of a 30-second file does not take 15 seconds, but is fast ? Quiet zone detection is required only in the area of cuts, that is, 10-14 seconds, if counted from the very beginning of the file.


I would be very grateful for any help.


I tried to write a script together with ChatGPT, but all options gave completely unpredictable results and were not even close to what I needed... I had to stop at the option with a sharp cut of files for exactly 14000 milliseconds. However, I hope there is a chance to make a variant with cutting exactly in quiet areas.


import os
from pydub import AudioSegment

input_directory = ".../RemSilence/"
output_directory = ".../Split/"
max_duration = 14000

def split_audio_by_duration(input_file, duration):
 audio = AudioSegment.from_file(input_file)
 segments = []
 for i in range(0, len(audio), duration):
 segment = audio[i:i + duration]
 segments.append(segment)
 return segments

if __name__ == "__main__":
 os.makedirs(output_directory, exist_ok=True)
 audio_files = [os.path.join(input_directory, file) for file in os.listdir(input_directory) if file.endswith(".wav")]
 audio_files.sort(key=lambda file: len(AudioSegment.from_file(file)))
 for file in audio_files:
 audio = AudioSegment.from_file(file)
 if len(audio) > max_duration:
 segments = split_audio_by_duration(file, max_duration)
 for i, segment in enumerate(segments):
 output_filename = f"output_{len(os.listdir(output_directory))+1}.wav"
 output_file_path = os.path.join(output_directory, output_filename)
 segment.export(output_file_path, format="wav")
 else:
 output_filename = f"output_{len(os.listdir(output_directory))+1}.wav"
 output_file_path = os.path.join(output_directory, output_filename)
 audio.export(output_file_path, format="wav")



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Revision 36454 : uniformiser les inputs du formulaire de login
19 mars 2010, par brunobergot@… — Loguniformiser les inputs du formulaire de login