That's great, Arny, though I don't know that I would necessarily center the discussion around acoustic recording. Lots of things can go direct these days.
And are you 100% sure that the plugin is samplerate aware???I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates.
Also, if you actually see how most top bands work, their drums and percussion are also acoustic.
Please tell me how to do vocals without going acoustic. ;-)
As a rule, to sell recordings modern groups tour, and as a rule touring groups are still very reliant on microphones.
If nonlinear processing is done in the digital domain, certain high harmonics can reflect back down into the audible range. Any signal that is generated in the digital domain above the Nyquist frequency aliases around the Nyquist frequency. For example, an 8 KHz tone in a 44.1 KHz sampled sytem is distorted by a fourth order digitally-implemented nonlinearity and would be expected to produce a fourth harmonic at 32 KHz. Since 32 KHz is 10 Khz higher than the 44.1 KHz Nyquist frequency of 22 KHz, the 4th harmonic is aliased down to 12 KHz where it is far more audible than it would be in a digital system with a much higher sample rate or an analog system. In a 96 KHz system, the same nonlinearity would produce a foruth harmonic at 32 KHz where it would not be audible.
A simple non-linear transfer function (i.e. literally mapping sample values to new sample values using a look-up table which, if plotted input vs output on a graph, would show a curve) - that can produce harmonics above the noise into the MHz range. You don't prevent audible aliasing just be using a "slightly" higher sample rate.It helps a bit, but "just" 96kHz isn't really a solution.Cheers,David.
This forum clearly has lost its edge, I really feel that the world has turned - but without you.
One should sample at the double of the highest frequency.The reverse holds too, if you sample at 44, there shouldn't be any signal above 22kHz in the source.Practical consequence, the input must be band limited.If you want to cover everything up to 20 kHz, the only thing you can do is using a pretty steep low pass filter (brick wall). In general this type of filter produces artifacts like pre-ringing.
Question:Shouldn't the rest of the worlds acoustics be considered as well?I mean, non audible frequencies pass through materials and become audible. So they need to be there if you want a natural sound reproduction.It's not just our eardrums in the room, is it?To make a comparison with light:Your day at the office would be pretty dark if you were to take the non visible part of the spectra out of the equation.As the flourecent lights fully depends on ultraviolet light, a higher frequency light just outside the visible spectra.NOTE: I'm a musician and an audiophile, not a scientist. So I could be horribly wrong.And maybe this has been said a million times before.But to me it feels pretty obvious.You should start to slow the soundwaves down right at the speaker covers./Levi
To make a comparison with light:Your day at the office would be pretty dark if you were to take the non visible part of the spectra out of the equation.
As the flourecent lights fully depends on ultraviolet light, a higher frequency light just outside the visible spectra.
Question:Shouldn't the rest of the worlds acoustics be considered as well?I mean, non audible frequencies pass through materials and become audible.
Far more real (i.e. measurable and sometimes audible) is unwanted intermodulation in the speakers themselves. That never exited in the original performance, and can be measurably reduced by removing all ultrasonics cleanly.
First: Thanks for your replies!And just to tell you. Personally, I'm only curious. I have no preference (but vinyl).I've never participated in this kind of discussion before and I haven't done much more reading than the physics in school (15 years ago).Though I did do some reading before writing this and all I can say is: This is waaay too complicated for me.Turns out there are actually several ultrasonic speakers. Where the air it passes through slows down the soundwaves, so that it will be audible at a certain point.You can find them at disneyworld and in your local mall. Aimed at you.But really, I can't understand how anyone without a degree in physics can say either YES or NO to my question.You really need to know your way around Nonlinear Acoustics.Here's a place to start: Nonlinear Acoustics WikiSo unless anybody can explain this in a simple (and informed) way, I'm just gonna have to leave this alone.Really sorry for wasting your time./Levi
So unless anybody can explain this in a simple (and informed) way, I'm just gonna have to leave this alone.
Any hypothetical intermodulation of ultrasonics from the original instruments, mixing in air to create audible components, will be strongest near the instruments themselves - and will be captured, in the audible range, by the microphones at the original event. So you don't need to create them in the listening room - they're already in the recording if they exist.
My understanding is ultrasonic speakers work similar to AM radio - you have an (inaudible) ultrasonic wave modulated by the audio you want to produce. You never hear the ultrasonic signal itself; it's just a carrier for the audible stuff.
But still, it leaves out all the sounds that hasn't been recorded through a mic. Synths, or a lined bass.And more important (i'm guessing) it leaves out all the sounds together, in a mix.
Quote from: drewfx on 08 November, 2011, 04:42:51 PMMy understanding is ultrasonic speakers work similar to AM radio - you have an (inaudible) ultrasonic wave modulated by the audio you want to produce. You never hear the ultrasonic signal itself; it's just a carrier for the audible stuff.Well that's one kind. Where you have a reciever remodulating the ultrasonic sound.The kind I'm refering to needs no reciever. It's just air slowing the soundwaves down and at a desired distance the sound becomes audible. As I understand it they shape the ultrasonic sound so that when the air has slowed it down to certain wavelengths interferance amongst the short soundwaves produces longer wavelengths and thus a high fidelity audible sound.The mechanism is called Parametric Array.Here's some reading: Parametric array - WikiHere's an interesting link of an implementation in an art installation: Link
I bet most of these 16/44.1 fanatics rips/ripped their best tapes/vinyls using least 24/96.Juha
[24-bit] keeps rounding errors inaudible when you're passing the audio through a digital processing/production workflow.
Quote from: krabapple on 15 November, 2011, 11:29:44 AM[24-bit] keeps rounding errors inaudible when you're passing the audio through a digital processing/production workflow.I'm betting they were inaudible anyway. Post ABX results and details of your workflow if you disagree