

4 messages • Page 1 sur 1

ModÃ©rateurs: Staff Univers Casques, Staff HauteFidÃ©litÃ©, Staff Juridique • Utilisateurs parcourant ce forum: mroboto, phil 13320 et 21 invitÃ©s
petite question sur les convertisseurs en passant...
Quelle est la diffÃ©rence entre upsampling et oversampling ?
 Choussar
 Messages: 2220
 Inscription: 08 FÃ©v 2002 2:00
 Localisation: Aixlesbains
Je vais peutetre dire une connerie mais j'ai toujours vu "upsampling" pour de la video et oversampling pour de l'audio... les 2 se traduisant par surechantillonnage...
Pour une fois que la langue anglaise est plus riche la langue francaise.
olivier
Pour une fois que la langue anglaise est plus riche la langue francaise.
olivier
 reivilo
 Messages: 738
 Inscription: 26 Juil 2001 2:00
 Localisation: paris, france
http://www.aslgroup.com/dcs/upandover.htm
je ne rajoute rien vu que c'est tres bien explique
( colonne de droite )
lien tire de www.sacd.fr.st
duc
ci dessous un copiercoller
UPSAMPLING AND OVERSAMPLING
While there is certainly overlap (but never ‘uplap’!) in the use of the terms ‘oversampling’ and ‘upsampling’, some guidelines can be given to differentiate the processes.
Oversampling is typically used to describe a technique used when transferring beiween the analogue and digital domain, where a signal is sampled many times over and above that actually required by the sampling frequency.
Oversampling in the context of the DA process involves multiplying the sampling frequency by a whole number, typically between 4 and 32, or even higher. For example, in ‘8x oversampling’, CD’s base rate of 4.4.1kHz is raised to 352.8kHz by introducing seven new ‘empty’ samples between the original data samples. These new samples, though, are often not just empty strings of noughts, but based on mathematical models to assist the DAC to work more linearly with the extracted data.
Oversampling, as well as easing the workload of the antialiasing filter, which can now operate more gently at a higher frequency, can also reduce distortion created when those analogue signals are first turned from continuous, analogue waveforms into stepped, digital, stairlike curves. This quantization noise is now spread over a larger band after oversampling, and can even be somewhat shifted out of the audible envelope by the technique of noiseshaping. Sony/Philips’ Direct Stream Digital, as used
in SACD, takes this idea to its limit, in order to dump high levels of digital noise up to higher frequencies than are not directly audible.
Upsampling is a solely digital domain process where the data stream is also stretched out by interpolation — guessing the points in between, again mathematically — and is typically used to refer to small, noninteger changes, such as from 44.1 kHz to 48kHz. When the change is larger than this, such as 44.1 kHz to 192kHz, ‘upsampling’ is a more popular term.
'There is apparently no extra information in the upsampled signal that was not present in the initial signal,’ says Mike Story of dCS. ‘With a 44.1 kS/s input, both the input data stream and the upsampled data stream will only contain a spectrum that must be between 0 and 22.05 kHz and is probably only between 0 and 20kHz.'
'This conventional analysis starts from the viewpoint that the behaviour of the ear can be described in mathematical terms using Fourier analysis. This assumption is probably pretty good — it means we are interested in frequency responses, for example, and these do provide good guides to the performance of equipment and to descriptions of what we hear. The analysis was right at the heart of the definition of the audio coding used on CDs.'
‘For those working with audio, it is also apparent that thearies based on these descriptions are not completely adequate, and that there can be significant differences in the performances of pieces of equipment with similar "conventional" specifications. It seems that two things are going on here: the ear may have more than one mechanism at work; and sine waves may not be the best function to use as the basis for analysis. On the mechanism front, it seems highly likely that the ear has a sound localisation mechanism ("where is it?") that is fast, and independent of the mechanism that says "it’s a violin", and that is related to transient response. There may also be a third mechanism at work. On the analysis front, it may be that some form of wavelet is the best basis for mathematical modelling. The problem here is that sinewave theory is relatively simple, and has been fully worked out by generations of mathematicians, following on from Fourier. Wavelet maths is just plain hard work, and does not yet have anything like such a solid core of mathematical results to call upon. Our ears, however, are not waiting
je ne rajoute rien vu que c'est tres bien explique
( colonne de droite )
lien tire de www.sacd.fr.st
duc
ci dessous un copiercoller
UPSAMPLING AND OVERSAMPLING
While there is certainly overlap (but never ‘uplap’!) in the use of the terms ‘oversampling’ and ‘upsampling’, some guidelines can be given to differentiate the processes.
Oversampling is typically used to describe a technique used when transferring beiween the analogue and digital domain, where a signal is sampled many times over and above that actually required by the sampling frequency.
Oversampling in the context of the DA process involves multiplying the sampling frequency by a whole number, typically between 4 and 32, or even higher. For example, in ‘8x oversampling’, CD’s base rate of 4.4.1kHz is raised to 352.8kHz by introducing seven new ‘empty’ samples between the original data samples. These new samples, though, are often not just empty strings of noughts, but based on mathematical models to assist the DAC to work more linearly with the extracted data.
Oversampling, as well as easing the workload of the antialiasing filter, which can now operate more gently at a higher frequency, can also reduce distortion created when those analogue signals are first turned from continuous, analogue waveforms into stepped, digital, stairlike curves. This quantization noise is now spread over a larger band after oversampling, and can even be somewhat shifted out of the audible envelope by the technique of noiseshaping. Sony/Philips’ Direct Stream Digital, as used
in SACD, takes this idea to its limit, in order to dump high levels of digital noise up to higher frequencies than are not directly audible.
Upsampling is a solely digital domain process where the data stream is also stretched out by interpolation — guessing the points in between, again mathematically — and is typically used to refer to small, noninteger changes, such as from 44.1 kHz to 48kHz. When the change is larger than this, such as 44.1 kHz to 192kHz, ‘upsampling’ is a more popular term.
'There is apparently no extra information in the upsampled signal that was not present in the initial signal,’ says Mike Story of dCS. ‘With a 44.1 kS/s input, both the input data stream and the upsampled data stream will only contain a spectrum that must be between 0 and 22.05 kHz and is probably only between 0 and 20kHz.'
'This conventional analysis starts from the viewpoint that the behaviour of the ear can be described in mathematical terms using Fourier analysis. This assumption is probably pretty good — it means we are interested in frequency responses, for example, and these do provide good guides to the performance of equipment and to descriptions of what we hear. The analysis was right at the heart of the definition of the audio coding used on CDs.'
‘For those working with audio, it is also apparent that thearies based on these descriptions are not completely adequate, and that there can be significant differences in the performances of pieces of equipment with similar "conventional" specifications. It seems that two things are going on here: the ear may have more than one mechanism at work; and sine waves may not be the best function to use as the basis for analysis. On the mechanism front, it seems highly likely that the ear has a sound localisation mechanism ("where is it?") that is fast, and independent of the mechanism that says "it’s a violin", and that is related to transient response. There may also be a third mechanism at work. On the analysis front, it may be that some form of wavelet is the best basis for mathematical modelling. The problem here is that sinewave theory is relatively simple, and has been fully worked out by generations of mathematicians, following on from Fourier. Wavelet maths is just plain hard work, and does not yet have anything like such a solid core of mathematical results to call upon. Our ears, however, are not waiting
 duc
 Messages: 480
 Inscription: 04 Jan 2002 2:00
 Localisation: Paris
merci beaucoup duc !
 Choussar
 Messages: 2220
 Inscription: 08 FÃ©v 2002 2:00
 Localisation: Aixlesbains

4 messages
• Page 1 sur 1
Retourner vers Discussions GÃ©nÃ©rales
