Freddie H
 
Over and out!

dmmi


Jose....sorry....you misinterpreted my last comment.  I was trying to say lets not dispute anymore....I don't want to argue because my stance is really solid on this, and so is yours.

We will really get no-where....OK

I pitched in trying to help....others can take it if they want.....or they can read the recommended books and take whatever side they want....based on research, feeling, hearing, etc.

I state my posts as facts for my reasons which no-one here knows of.....maybe I have read the books and don't believe them....who cares...I stand by my comments.

My point is we are all here to help....I tried to help, and so did you (many times on other threads as well).  We've reached the point that cannot go any further, so I was just trying to calmly get back to Freddie's thread topic.

Peace my friend!

drewfx1


OK, for anyone who does not believe 3 samples contain all of the information about a sin wave below Nyquist, and thus doesn't believe that adding more samples can't increase accuracy,

I hearby issue the following challenge to prove it:

If anyone here has SoundForge, create a pure sin wave in SoundForge and tell me only the sampling frequency, bit depth and 3 consecutive actual sample values that SF will show you. From this information, I will tell you your sin wave's frequency, amplitude and the phase of each sample.

How to do it (in SF9):
1. Create a pure sin wave of any frequency and amplitude (Tools-Synthesis-Simple). Record the frequency and amplitude, but don't share them with me.

2.  Zoom all the way in until you can see the individual sample points (it won't look anything remotely like a sin wave if you chose a high frequency). Select any individual sample and do File-Properties (or ALT+Enter). On the Format Tab give me Sample Rate, Bit depth, and Sample Value. Hit Cancel, then right arrow and give me the next Sample Value. Repeat once more and give me the 3rd Sample Value.

I will tell you the frequency, amplitude, and phase (at each sample point) of the sin wave you created.

As a free bonus, I will also predict the next sample value for you. And if I can predict the next value accurately, note that I could also predict any other values in between the samples as well.

drewfx

Wait a minute now!!

Think about it!

kicksville

I can't tell you how many times I have been pitch correcting with Melodyne and I'm like. "Oh that's a ton better!" and I keep moving in the project until I finally get to a note that I can't seem to pitch correct and then I realize..."Oh wait. Melodyne is Bypassed?"

Exactly! That's a good example of what I'm getting at. Here's another:

I'm the chief engineer at one of the largest theater & arts complexes in the US. The main space in the complex has L/C/R FOH arrays, subs, 4 balcony levels each with its own underbalcony fill speakers, lip fills, loge fills, hearing assist feeds, and multiple program feeds. All told, I have to manage almost 20 different delay zones. One more than one occasion, I've been working on timing or voicing with a house system tech, make a change, and we BOTH go "Oh, that's a lot better!" Then we realize the control system was offline and didn't actually implement the change.

Another example at someone else's expense: Without naming the guilty party, I'm working on the sound design for a major opera. During a rehearsal the other day, the conductor complained that he heard distortion from the pit monitors. THEY WEREN'T ON! But that didn't matter to him - he heard distortion, so therefore there was a problem. He really didn't want to engage in a discussion of the physics involved with a distorted signal emanating from a monitor that wasn't receiving any signal of any kind. And he most definitely did NOT want my opinion that the distortion was his eardrums' reaction to the incredibly shrill soprano bellowing at 110dbSPL ten feet from his head......

So when folks respond to the facts about sampling by saying "I don't care what the reality is. I BELIEVE 2+2=5", I say "Yes, Maestro, I'll work on it."

And go on about my business 

John

bitflipper


Think it makes any difference if this song is 24-, 32- or 64-bit?



P.S. The above picture is not one of my songs, it's the title track from a multi-platinum, Grammy-winning album from 2004.

Gee what a great stereo file so much separation. How did you get the track on the edges like that? Such a nice green background too.

Freddie H
  Its cool Jose!
What I mean is: how do we know what we see on the display is what we actually hear? Some audio programs doesn't display the audio graphic curve 100% correctly. Cubase is one. It look the same in program but isn't the same in real life..
 
 
Best Regards
Freddie

brundlefly

I was defending the same point as you: higher samplerate = better resolution. I then did some research, as suggested, and after reading this paper below, there was no way I could still argue. I was plain wrong.

I thought I detected your position evolving, and suspected there was some research going on in between posts.  Good work. 

info@tomflair.com

Freddie H

info@tomflair.com

Freddie H

info@tomflair.com


btw. ...freddie - how can you state that rme are some of the best converters in the world????  ahhrgghhh !!!
rme are just entry level in pro field ;-)

 
RME have won a lot of awards of there converters, my friend!
Its just the way it is.. they are great  at--> 119db SNR... Have you tried them or listen on them?
 
Apogee, RME, Cirrus logic and Lynx that's it! Have I missed something?   
 
Regards
Freddie
 

freddie - i own (or owned) a rme digi 8 pad, a hammerfall 9632, a hammerfall hdsp 9632, a hammerfall 9652 hdsp, ánd in the second studio in vienna we have a ff800 + a adi8 pro
...so i "think" i can have an opinion about rme ;-)
btw. i also own a konnekt 48 from tc electronics - and PURELY SUBJECTIVELY i thinjk that the konnekt sounds more pleasant than all the rme stuff i ever heard ...
 
yet rme wins Hands down in matters of support, driver stability, performance, and building quality - so especially this facts makes rme "professional" - even the chief of rme himsels says that its futile to discuss about "converter sound" - since probably all convertes available today (except for really chep stuff) give professionally usable results - so its the features beside the (possibly)  "non existing sound" of a unit that makes it a great piece of hardware
 
cheers

Great my friend!
I have RME hammerfall 9652 too! Great clock, great drivers, don't use it so much! My work horse is E-mu 1616m...sounds awesome, great. Ultra low jitter and stable clock, same sounding as Pro tools HD! Have no need of many channels. I have 2x digital Yamaha 02R (version 2) if I need more channels...to patch and route thru...adat cards and so on...
 
 
Apogee sounds great too but only mac if you don't go Rosetta Series of course.
TC konnekt 48 has it native 64bit drivers yet? I've heard people say it sounds awesome . TC has been slow on support of native 64bit drivers in the past? Same on there hardware Reverb etc...stuff.....you know what I mean!
 
 
Regards
Freddie
 
 
well ...talking about tc makes me feel sick ;-)
they are announcing 64 bit drivers since 2007 - and theyonly have a betadriver that is far from perfection - in fact i just got the 9652 in ored to use the konnekt as frontend via adat (even if i loose quite some of its features this way...)
i should have known better anyway - i used to own a powercore mk1 and an mk2 card and both were also not really "perfect" especially the virus plugin (sounded great) was never ever stabel - not even after years of existance - unfortunately tc is diggibng their own grave i guess - users are EXTREMELY pissed off because of the drivers...so i am happy rme exists ;-)
 
 
cheers
 
btw_ i was considering getting a yamaha dm1000 vcl  - i just cant stand mixing inside the daw...so i envy you for those two desks you have ;-)
 

Only thing I'm consider and are interesting in is the V-Studio 700, but hope I get it soon for free, kind of sponsor deal from Cakewalk... You can always wish Santa brings it around for Christmas, right!

Jose7822

Freddie H
  Its cool Jose!
What I mean is: how do we know what we see on the display is what we actually hear? Some audio programs doesn't display the audio graphic curve 100% correctly. Cubase is one. It look the same in program but isn't the same in real life..
 
 
Best Regards
Freddie

I see. 
 
Well, there are a few plugins that do display intersample peaks like Ozone 4 and the X-ISM freebie from SSL:
 
http://www.solid-state-logic.com/music/X-ISM/index.asp
 
 
I know this is not the answer to your question, cause it would be up to the DAW app to display an interpolated waveform.  But at least, these tools provide for a way to prevent going over when we think we're not (because we actually don't see what is really happening after the DAC process).  Perhaps, it would be too process intensive to interpolate multiple waveforms, especially on big projects with over 100 tracks.  Maybe this is why we don't have such feature in our DAWs.
 
 
Take care!
 
 
 

Analog audio has a theoretically infinite frequency range, whereas digital audio (software and hardware) has a hard limit on high frequencies, as determined by the sample rate.

Many analog processes take “infinite” frequency range for granted, but they can't do the same in the digital domain. For example, a standard analog peaking EQ has a bell curve that is symmetrical around the center frequency; one side slopes to 0 Hz, and the other slopes toward infinite Hz.

You can implement the same EQ in the digital domain, but infinity is suddenly much closer. In fact, what was infinite Hz is now the Nyquist frequency (half the sampling rate) or 22.05 kHz at a sampling rate of 44.1 kHz. This difference in the proximity of infinity results in an EQ curve with a dramatically lopsided shape (see Fig. 6).

Things can get stranger as the EQ's center frequency approaches the Nyquist frequency. At those high frequencies, I've seen digital EQs that started to take on weird globular shapes and even go down in actual frequency as I turned up the frequency knob. I've even encountered a peaking EQ that looked much more like a resonant highpass filter.

Those problems can be avoided or at least minimized by clever programming. The degree to which the programmer is successful defines, to a great extent, the differences between good and bad digital EQs.

...

Compressors and limiters also have frequency-related issues. You're probably familiar with aliasing — it causes audio artifacts when sampled audio contains frequencies higher than the Nyquist frequency. Aliasing doesn't occur only during sampling, however; it can also happen entirely within the digital domain.

For instance, compression and limiting work by modulating one audio-rate signal (the input) with another audio-rate signal (the compressor or limiter's automatic gain control, which operates in the audio range when the attack and release envelope times are fast). When you modulate one audio-rate signal with another, it has the effect of adding the two signals' frequencies; if the total exceeds the Nyquist frequency, you'll get some aliasing.

A full-bandwidth audio signal processed with a limiter or a compressor with fast attack or decay times falls into that category; the faster the attack or release and the greater the compression or limiting amount, the more aliasing you hear. That is the cause of the crunchiness many people hear in digital-dynamics processors. Again, clever programming, especially oversampling, can minimize these aliasing artifacts.

You'll find similar predicaments in synths. Resonant filters suffer from the same infinity-is-much-too-close syndrome as digital EQs, and various synths differ widely in their success at addressing the problem. For instance, standard Chamberlin digital filters (the most common type) only work correctly to about one-sixth of the sampling rate. For a synthesizer running at 44.1 kHz, that means the resonance tops out at about 7 kHz.

Oscillators have problems similar to compressors. For example, a square wave at, say, 4 kHz is actually generating frequencies well above the Nyquist limit, because of the waveform's sharp edges. Untamed, that can cause excruciating aliasing, especially toward the top of the keyboard (as you can hear in some popular products). Similar aliasing can also happen when samples are transposed above their original pitches. Techniques for dealing with these complications vary and account for some of the sonic differences among synthesizers.

pollux

 

http://lavryengineering.com/documents/Sampling_Theory.pdf 

dmmi



1.) It IS impossible to have a perfect circle, curve, or waveform in the digital realm....period

2.) Sampling rate IS frequency, therefore cycles per second.

Put them together.....it DOES make a cleaner, truer, rounder, waveform.....period!

And this has NO RELATION whatsoever to Nyquist/Shannon sampling theorems, mathematical equasions demonstating whether we hear the results or not.

I can't make it any simpler than that.

THE SAMPLING RATE DOES NOT = HIGHEST FREQUENCY.....PERIOD!

I said before......everyone here is right.......including myself

This is NOT "BS"....computers ARE 0's and 1's...... discrete data IS 0's or 1's and in the end EVRYTHING within your computer is comprised of that.....waveforms, pictures, software programs.........EVERYTHING