Actually it's harder to do right than a static Since many people prefer high voice counts instead of good sounding sample playback quality usually suffers. If you write lots of music that uses sample playback and you change the pitch of samples you should really check into the quality of your sample playback plugins as this could really help the quality of your tracks.
Some resampling methods sacrifice phase linearity to reduce latency but it is never removed. Inter-sample wobbles and peaks are always there even though you can't see them most of the time And.. Quote: Originally Posted by Dan Lavry Good conversion requires attention to capturing and reproducing the range we hear while filtering and keeping out energy in the frequency range outside of our hearing.
Although 60 KHz would be closer to the ideal; given the existing standards, At 96 KHz sampling rate the theoretical bandwidth is 48 KHz. In designing a real world converter operating at 96 KHz, one ends up with a bandwidth of approximately 40 KHz. But if you do hear it and it changes your way of working for the better..
I'm sorry, but while your explaination might seem logical, you don't really understand how the PCM audio works. All the added "resolution" is actually out of the hearing range. If you make a fourier transform of a square wave at 20 kHz or even a bit less, you'll see that it's composed from a sine wave at 20 kHz and higher harmonics that of course have a higher frequency. This is the detail, you're talking about, but the frequency of the detail is higher than what we can hear, therefore it doesn't matter and is removed by LP filters.
Regards Demokid My best resources say that is not the case. Pretty much the same amount of calculation is required. It is not simply dividing by 2! If you can cite sources who have done the math, it would be welcome, but to me it is an old groupie's tale to coin a clunky phrase Working at 96khz uses more resources than Only slightly, but it does.
The same cannot be said for all SRC's however. If your sample libraries are at 44khz they also up sample to even doubles more cleanly. Whether it'll be audible or not depends on circumstances, but why chance it or give it any thought when you can just work at 88? If your end product is to be Ever stop to wonder why all oversampling implementations keep to even multiples?
It's quite odd that anyone would chime in at this point with what they've heard in passing when you've got the clear recommendations and summary of the subject by two of the best authorities around! By using this site, you agree to our use of cookies. Code by Port Forward. Hosted by Nimbus Hosting.
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Page 1 of 3. Why are you recording at 96khz or higher sample rate? Shut up and take my money. I would imagine abletons got similar options…. It usually doubles or 4x the internal sample rate used to process the sound then plays out at the interfaces sample rate but still yields greater detail because of the processing math being done at higher resolution.
I am pretty sure that Herp Alberts Rise album from was the first ever digitally recorded album. That was recorded in 96khz, so technically 96khz will always be the original standard. Is 96kHz the new recording standard? The Lounge General Discussion. Tascam Model No, it means that running at 96kHz effectively adds an oversampling button to plug-ins lacking that option.
It has nothing to do with the frequency range our ears can hear. Pros and cons. Any subtle quality improvement spread over multiple plug-ins has a cumulative and potentially audible, not just theoretical, effect.
However, 96kHz stresses out your computer more and limits the maximum number of audio streams between your USB or FireWire audio interface. You can achieve lower latency—a nice bonus—if your computer can handle it.
Furthermore, a well-designed plug-in might actually sound better when oversampled from a lower sample rate. A higher sample rate oversamples everything, whether it needs it or not. Oversampling is capable of improving resolution and signal-to-noise ratio, and can be helpful in avoiding aliasing and phase distortion by relaxing anti-aliasing filter performance requirements. Random oversampling involves randomly selecting examples from the minority class, with replacement, and adding them to the training dataset.
Random undersampling involves randomly selecting examples from the majority class and deleting them from the training dataset. Begin typing your search term above and press enter to search.
Press ESC to cancel. Ben Davis February 1, What kHz should I record at? What sample rate is best for recording? Is higher kHz better? Does bit sound better? Which is better 48KHz or Is it worth recording at 96kHz?
Is 48KHz good enough? Is 16 bit What sample rate should I use in Pro Tools? Does sample rate matter? What is 48kHz sample rate? Can you change the sample rate of a Pro Tools session?
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