I think the "light" version of r8brain pro will suit my needs, but to make sure I have a few questions:
(1) In the light version I can input a 96khz 24 bit stereo .wav file and output a 48 khz 24bit stereo .wav file, correct?
(2) The light version only allows linear phase but not " minimum-phase", correct?
And now the 2 big questions:
I have read on these forums that using r8brain pro you can have "perfect" results when going from more audio information to less audio information--for example downsampling from 96(more) -> to -> 44.1 (less):
"Problem lies exactly in 147/640 ratio. It is highly inefficient to upsample original VERY time-consuming. Other than that, there is no difference in resampling from 88.2k or 96k, to 44.1k....That's interesting Aleksey, would you consider releasing a product that can do perfect 96->44.1 downsampling -- even if it takes a very long time to process? Or are we talking like weeks per minute or something?"
(3) BUT what about upsampling? For example, If I start out with a .wav recorded at 44.1 khz (less audio info) and upsample to 48 khz (more audio info) using r8brain pro will it be perfect? And what exactly does "perfect" mean? Does pefect mean, in the case of 96 -> 44.1, that the resulting 44.1 .wav would be the same as if you had originally recorded the sound in 44.1 as opposed to 96? And if 44.1 -> 48 khz is not perfect what is missing?
(4) Last question: If I have two targets: (1) 44.1 16 bit (cd audio) and 96khz 24 bit audio (for use on a dvd player) What sample rate should I record my music at? 96 and then downsample to 44.1? The problem is my drum samples are only 44.1 khz 24 bit... (Drumkit from hell superior)... so mabye I should record at 88.2 and then upsample to 96 and downsample to 44.1? I'm so confused.
But thanks for your help and this great forum--Aleksey you are an audio genius!!
1. Yes, it is correct.
2. Yes, it is.
3. r8brain PRO offers almost perfect conversion quality, in any direction. What is usually lost during conversion is a minimal amount of higher inaudible frequencies. By 'perfect' I mean that conversion is mathematically correct and the resulting audio file is very close to a theoretically possible quality.
4. You can work at whatever sample rate you wish. Of course, sampling at 96kHz and converting it to 44.1kHz theoretically should deliver a better quality than if you sampled at 44.1kHz initially. Although, best soundcards usually negate this rule.
Thank you for your help.
I have been testing r8brain pro vs. Sounds Logic ReSample 1.1, which uses
"double-precision (64-bit) foating-point computations throughout [MATLAB libraries] ... [providing] very high numerical precision."
But r8brain pro, in my tests, preserves more higher frequences than ReSample 1.1--so I think you have a superior filter design!
Btw, does R8brain use (or even need) double-precison 64 bit floating-point computations?
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