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Set axe fx2 to other frequencies than 48khz?

carloszeke

Experienced
As far as I remember, 16bit, 44.1kHz was developped for Audio CD in 1980 at least 12 years before MP3 was released, no ?
You are correct. I should have said that first...It was then used as the basic MP3 format to fuel the market of compact players using solid state storage. (remember how expensive 1 Gig used to be?)
 

Stillbruch

Experienced
Cagey said:
44.1Khz is where the Nyquist theorem says you've done as much as you can plus a little bit to prevent aliasing.
+1
Nyquist-Shannon samping theorem states that fs > 2*fa, where fs is the sampling frequency and fa the highest incoming frequency. The human ear is able to perceive up to 20 kHz, which means that, in theory, a sampling rate of 40 kHz would cover everything up to 20 kHz. Now there are some reasons why one would still use higher sampling frequencies, but it's a little bit to mathematical for me to explain.
 

barhrecords

Axe-Master
Sampling theory is not the same as A/D implementation.

Since the implementation of the A/D hardware and software is not perfect, higher than 44.1kHz sampling rates can sound better. Its because of the implementation of the hardware and software not because of the theoretical capability of sampling. All convertors are not created equal.

Bit depth can give more headroom and also allow fewer cumulative errors when doing digital audio processing or digital audio mixing. E.g. a multi-track DAW using plugins for processing and digital summing to a stereo out.

IIRC Cliff recently added some Output mode options that make analog re-amping easier? I'm not a big re-amper though. I've done it with USB a few times no worries. If re-amping is critical, you could always get an external re-amp box from Little Labs, Creation Labs or Radial.
 
If you are for some reason forced to bring down the number of bits in your samples, 24 -> 16, it's really recommended to use dithering to avoid nasty noice in low level parts. But this shall be done in the absolute last phase of your project. Sample rate is not that important, but working with at least 48kHz does a lot of improvement. Rule of thumb is using twice the frequency than the final result, i.e 88kHz if CD quality is the goal and 96kHz if DVD/mp3 is the goal. Working with 32 bit (or higher) is really great since it's impossible to reach the limit (digital distorsion).
 

FractalAudio

Administrator
Fractal Audio Systems
Moderator
So much misinformation here...

48 kHz is considered "pro" sampling rate. The reason for 44.1 kHz on CD's is subject to debate. Some maintain that the sample rate was lowered so that Beethoven's 9th would fit on a single CD. Others claim that it was because that rate was compatible with video equipment.

IMO 44.1 kHz is insufficient for professional audio. Personally I would prefer 64 kHz. Whilst Nyquist theorem is all well and good most people don't understand the details and simply state "the sample rate must be twice the highest desired frequency". The problem with this is as you approach Nyquist the filter demands become extreme. The more extreme the filter demands the more taps are needed, the more precision is needed, the more latency is incurred, etc. A 64 kHz sample rate would give you a nice, smooth roll-off from 20 kHz to 32 kHz rather than the brick wall you get with 44.1 kHz.

There is no hardware advantage to using 48 vs. 44.1. The costs would be the same in either case. Modern converters use over-sampling techniques to implement the necessary anti-aliasing filters thereby reducing off-chip filtering to simple circuits.

MP3s have no native sample rate but are typically 44.1 kHz because they are usually derived from CDs. MP3 is a psycho-acoustic compression format that exploits frequency masking to lower the data required to store audio information.
 

vinnieRice

Inspired
Notice that sample rates of 48k,96k and 192k are multiples of 2. They can be converted from one to another provided your equipment has the versatility to arrive at a common setting. The Axe FX, being fixed at 24bit, 48k is quite acceptable. The higher the sample rate, the purer your high-pitch notes and percussion (cymbals) will sound. Keep in mind that the all too familiar 16bit, 44.1kHz format was developed to accommodate the pocket-sized MP3 players where size, cost, and memory capacity were the main concern. High fidelity audio can't be fully appreciated with ear buds anyway. Equipment and DAW's exists that can up-sample a lower order signal, but that does not improve the sound quality. It only builds a bigger word that can be processed as a higher order sample.
wow...
 

Clive

Experienced
So much misinformation here...

48 kHz is considered "pro" sampling rate. The reason for 44.1 kHz on CD's is subject to debate. Some maintain that the sample rate was lowered so that Beethoven's 9th would fit on a single CD. Others claim that it was because that rate was compatible with video equipment.

IMO 44.1 kHz is insufficient for professional audio. Personally I would prefer 64 kHz. Whilst Nyquist theorem is all well and good most people don't understand the details and simply state "the sample rate must be twice the highest desired frequency". The problem with this is as you approach Nyquist the filter demands become extreme. The more extreme the filter demands the more taps are needed, the more precision is needed, the more latency is incurred, etc. A 64 kHz sample rate would give you a nice, smooth roll-off from 20 kHz to 32 kHz rather than the brick wall you get with 44.1 kHz.

There is no hardware advantage to using 48 vs. 44.1. The costs would be the same in either case. Modern converters use over-sampling techniques to implement the necessary anti-aliasing filters thereby reducing off-chip filtering to simple circuits.

MP3s have no native sample rate but are typically 44.1 kHz because they are usually derived from CDs. MP3 is a psycho-acoustic compression format that exploits frequency masking to lower the data required to store audio information.
So Cliff, for Axe FX III, will we have 48 or 64 or 96 or 192 kHz ? Will users have the choice this time ?
 

shasha

Fractal Fanatic
I'm not going to get into the entire history of my less than fantastic journey, but I will share the end result of literally years of time invested and thousands of dollars thrown at this "limitation" of 48kHz.

If you absolutely have to use a different sample rate than 48kHz than just use the balanced analog output into your audio interface and forget trying to resample the signal because you will not improve upon the original digital output of the AxeFXII.

I know that it is not super popular, but the only true advantage that being in the digital realm in this case is to ensure that the signal is pristine which means that as soon as you use any SRC whether it is hardware or software based you lose that advantage because it is no longer the original. Now you will get varying degrees of quality depending upon how much money you throw at it, but it is never going to improve upon the original. Cliff told me years ago when I was first whining about this that any hardware SRC would be a step backwards in quality and I proved him right, much to my dismay.

I'm not going to paraphrase or regurgitate what he said a few threads up, but everything that he said about the way the digitization process works has been proven to be truthful and if you doubt that just take some time to do some research about ADC process and focus on the challenges of filtering and then think about how that would affect any design that is sample rate agile. Bottom line is that the best sample rate in every case is the sample rate that the device was designed to operate at optimally. After having gone through a lot of equipment and gear and crap the only times that I hear an improvement based on the sample rate is because it was the one that the device was designed to operate at. Another way to look at it is that going from 44.1kHz to 192kHz was not an improvement as much as it was just "less shitty" because the filtering was designed to operate best at 192kHz. Comparing the AxeFXII's native 48kHz to a 192kHz recording using the balanced analog output into a sound card designed for 192kHz yielded almost no difference at all. That's because in both instances the devices were operating at their optimal sample rates. Filtering has a massive impact on the ADC process and from what I have read it is the costliest and most important part (that could probably be proven true or false by someone much smarter). Of course the other massive issue is latency and in every design there are decisions based on optimal and practical and possible. I mean could we have 192kHz 24 bit out of the AxeFXII? Probably. But if there is a massive amount of time between you hitting a note and hearing it and the return on that investment isn't distinguishable in the quality of the output is it "better"? Hell no.

If you have to use any SRC just do it in software after the fact. It's slower, but it's better quality in almost every scenario and there are some out there that do a really good job (there's some open source thing out there that I can't recall off the top of my head).

I'm not trying to sway anyone or tell anyone not to fight this, but it would be kind of crappy on my part to not at least share some of my experiences and at least give you something to think about. I don't know what it would take Cliff to change the sample rate to something else, but I do know that everything that he has told me over the years about it has been truthful.
 
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