AmpliTube Metal - Released THIS month!!!

[redshift factor]

Lunch Money, the question you asked was about the target demographic of X-GEAR. I pointed out that the cost of a couple Amplitube powered plugins is not that steep compared to the cost guitarists regularly shell out for quality kit. Just one of the vintage pedals that IKMM modeled for their AT products costs a lot more. That was the point I was trying to make. If you disagree that's fine, but that is no reason to be insulting. However, While I take pride in the guitars I have, I have nothing to prove when it comes to playing them. They're just a few of the many tools I use to create my music. Whether or not I'm a "true guitarist" is the least of my concerns.

[hibidy]

48khz is fine for recording most things.

It's not fine for an 88.2kHz or 96kHz project.

that is so true!

I don't know about anyone else.....but many projects I've been called in on are 96k (pt hd systems almost always)

At any rate they always have the kick butt TDM stuff........but still, If you want to go with YOUR sounds (and despite if anyone like me or not, I'm kickass at getting tones/making presets/etc) this isn't going to work as far as digital i/o (which btw, the sound guy would want in preference!)

It's not a stomp i/o whine as much as an fyi for serious players!

[hibidy]

sorry, I did forget to add that my home shit is at 48k........which is probably the "thrust" of the thinking.......(me, two minds.......? na........ )

[redshift factor]

You should record at double your target samplerate in order to bypass your A/D converters' anti-aliasing filters.

[Amberience]

You should record at double your target samplerate in order to bypass your A/D converters' anti-aliasing filters.

Is that what you've heard then?

[redshift factor]

I wouldn't put your ignorance on display like that.

[Amberience]

Well show me some hardcore science then, because I have never heard what you claim until now.

I have heard that the reason 44.1khz is adequate is because it is double the range of human hearing, and therefore is capable of capturing all of those frequencies.

But there is a debate: Some people say even though they cannot hear frequencies outside of 20hz-22khz, that they still have an effect on the sound, hence the need for bigger sampling rates. But other people have dismissed this as poppycock. I don't dismiss it, but I don't think I can really afford to run my projects in 96khz, which is why I stick with 44.1khz.

[Amberience]

Perhaps you should ask Aleksey Vaneev to explain. I'm sure he can tell you all about it.

Aleksey didn't make the comment, you did. The onus to explain what the hell you're talking about is on you, not Aleksey.

[redshift factor]

You should research the phase distortion that anti-alias filters introduce when applied close to the audible spectrum.

Perhaps you should ask Aleksey Vaneev to explain. I'm sure he can tell you all about it. This is a well known phenomenon and one of the reasons high end samplerate converters like r8brain PRO exist.

[redshift factor]

You're talking about he Nyquist frequency. However, your extremely basic understanding of what you've "heard" shows that you're out of your depth here. There is a lot more involved when it comes to A/D converters. There is an anti-aliasing filter applied to 44.1kHz and 48kHz which is not phase linear by the nature of being a realtime physical filter and it distorts the audible spectrum into 20kHz and below. Cheaper "prosumer" converters have lower quality anti-aliasing filters which introduce more phase distortion. By recording well above the audible range (88.2kHz and above) the antialiasing filters are either not engaged, or is applied too far from the audible range to affect it. This is a good way to compensate for cheaper converters.

Then you would use something like r8brain PRO to downsample by half, and have a clean recording with minimal phase distortion near 20kHz.

[Amberience]

Well I've never been bothered by any so called distortion, even if it is indeed there. Which I assume to be correct, because as you said, my knowledge is indeed basic.

So ok, no problem. I've learnt something new here, and that is good. But I sit here right now strumming chords through an instance of Amplitube 2. 96khz and all of the oversampling options inside of Amplitube 2 turned on. I'm getting crackles and glitches, and I really could not work like this. So for my purposes going down to 44.1 or 48khz suits my creativity much better - because I can actually work at those rates without worrying about processing power.

And at the end of the day, all technical splargon aside, these tools exist to help our creativity, we really shouldn't be arguing about how good or how bad they are.

[Amberience]

You're talking about he Nyquist frequency. However, your extremely basic understanding of what you've "heard" shows that you're out of your depth here. There is a lot more involved when it comes to A/D converters. There is an anti-aliasing filter applied to 44.1kHz and 48kHz which is not phase linear by the nature of being a realtime physical filter and it distorts the audible spectrum into 20kHz and below. Cheaper "prosumer" converters have lower quality anti-aliasing filters which introduce more phase distortion. By recording well above the audible range (88.2kHz and above) the antialiasing filters are either not engaged, or is applied too far from the audible range to affect it. This is a good way to compensate for cheaper converters.

Then you would use something like r8brain PRO to downsample by half, and have a clean recording with minimal phase distortion near 20kHz.

That's good information. What you mean by "cheaper converters" I don't really follow though. Is a £300 soundcard a cheap unit with cheap converters, because my wallet says different!

I sit here now with a Tascam FW1804. Quite an aging unit now, but to my ears sounds fantastic, in your estimation is this a cheap unit? I know nothing of the converters, I just know that it sounds good to my ears and can record a bunch of stuff at once!

Also I just lowered my project sample-rate to 88.2khz with all of the oversampling options in Amplitube 2 turned on. No crackles or pops when I play, which is much much better for my creative flow.

[redshift factor]

Right, doubling your samplerate doubles your CPU usage, but it also reduces your latency. When recording, you should track to a stereo 88.2/96kHz mixdown of your song if your CPU is giving you trouble with realtime multitrack projects. After recording you can downsample with a high quality converter (Voxengo r8brain PRO, SRC Crystal Resampler in Wavelab 6, or Audio Ease BarbaBatch) and import into your 44.1/48kHz multitrack project. It should help make your highs clearer and less harsh. Most of the brittleness you hear people talk about in 44.1kHz is actually due to the anti-alias filters. They were especially bad on '90s digital equipment.

By "cheap converters" I mean anything below Apogee...

[stag]

I found odd only having 48Khz at hand because if something is enhanced by higher resolution are distortion and compressions, the core of an ampsim.

On top it's an efficient way of reducing latencies, if the minimum latency setting was 64 samples 2,5ms. at 44.5, it would be about 1ms. at 96Khz or 0.5ms.(or less) at 192K.

Resolution has nothing to to do with the frequency of pitch. Its a measure of the "refresh"(frequency) of those bits.
On clean sound, not clipping is hard to tell but on dirtier stuff one can tell.

As a solution for guitarists, let me repeat: Stomp I\O sucks. The marketing on the side, sucks even more.

Good luck on selling it doe. Guitar players are not drummers.

[Amberience]

Well, from my perspective, the software is brilliant. Amplitube and Revalver lead the pack imo. But Stomp is a big fat waste of time. I mean, it looks decent enough, but for the money... well, I'd rather get a midi pedalboard and work out some way of using it with the software.