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So what happens to the parts between the samples?

https://www.rane.com/note137.h...

If you use a DAC that creates discrete steps, and feed the output through a perfect 0-22kHz lowpass filter, you get the original signal back.

Because it is impossible to create such perfect filter, a common method is to convert the 44 kHz sample rate to a much higher one, say 1 MHz. Feed that through a DAC, and then use a much simpler lowpass filter to get rid of anything above 500 kHz.

As far as higher harmonics: if you can't hear a pure sine at 30 kHz, you cannot hear the harmonics of a 15 kHz fundamental either.

That would be a Delta-Sigma D/A converter, which is most common nowadays and quite different from the original multibit type (see http://www.rane.com/note137.ht...). I think the point is there is a whole lot of filtering going on, and that is where the subtle differences emerge. That perfect copy is in theory only.

I don't think we even got into the differences between 1-bit (like SACD uses) and multibit D/As. They both have their pluses and minuses (sorry!); but if you crank the sample-rate of the single-bit (Delta-Sigma) converters up high enough, they are essentially the same (until you try to start EDITING in the Digital Domain. But that's another story!).

But I agree: The "Perfect Copy" ONLY applies to Sine Waves, and is ONLY in theory, depending on how "close" you want to look...

If you use a DAC that creates discrete steps, and feed the output through a perfect 0-22kHz lowpass filter, you get the original signal back.

Because it is impossible to create such perfect filter, a common method is to convert the 44 kHz sample rate to a much higher one, say 1 MHz. Feed that through a DAC, and then use a much simpler lowpass filter to get rid of anything above 500 kHz.

As far as higher harmonics: if you can't hear a pure sine at 30 kHz, you cannot hear the harmonics of a 15 kHz fundamental either.

That would be a Delta-Sigma D/A converter, which is most common nowadays and quite different from the original multibit type (see http://www.rane.com/note137.ht...). I think the point is there is a whole lot of filtering going on, and that is where the subtle differences emerge. That perfect copy is in theory only.

the digital sample is a perfect representation of the original analog waveform

Not really, but it is "good enough".

Excellent article here on the details of digital audio; Digital Dharma of Audio A/D Converters

Everything you ever wanted to know about quantization, dither, and more.

The speed of sound is 340.29 meter/sec

If you're going to be that precise and definite about the speed of sound you'd better state what temperature/pressure/moisture level you're talking about... In short - it varies.

I don't need all that fancy equipment. Since I bought Rane's Pseudoacoustic Infector, I'll never need to worry about source quality ever again.

Reminds me of this product:

- Independent Power & Glory Switches
- Continuously Variable This/That Level
- Full-Function Ecstacy Generator
- Variable This to That Crossover Frequency
- Here-There Pan (Back Again Switching)
- Program Dependent Sheen Removal
- Anti-Resonant Concrete Chassis
- Proprietary Paint to Reflect Odd Harmonic Light Frequencies to Reduce Nono-Linear Photon Radiation Interference
- Time Warp Compression/Expansion to Synchronize Here/There Time Coordinates

What we really need is standardized low voltage supplies, target devices and connectors. Then the "outlet" strip could have a single, high efficiency converter with multiple outputs.

The double-headed approach makes sense for several reasons. DJs and audiophiles will always want the top end of quality, so they will buy physical media, but for convenience you can't beat a digital file.

I got my Vinyl here and my mp3's to play on the vinyl here.

As a turntablist I have struggled with the possibility of vinyl analog records someday not being made anymore. I mean, it is not just a hobby for me but a profession. Many audiophiles will say that there is nothing like the warm analog sound of the "needle and grove" combo. Some folks 1 2 3 have created a way to control playback of digital audio with turntables using timecoded records. I have played around with them, but they seem to have a "metalic" sound when the program time stretches/compresses the digital audio.

Is it possible to emulate analog properties?

(hope that made sense)

Or (much) better yet, Rane Scratch Live.

Mixing with an iPod? You're serious right? Wow. No thanks. It's a much better idea to use something designed for the job. Check out Rane/Serato Scratch Live which allows you to play AIFF, WAV, MP3 (soon Vorbis) directly on your regular vinyl or CD decks. Feels exactly like vinyl, right down to scratching, juggling, backspins, poweroffs, etc. It even integrates with iTunes for all you Apple junkies.

There's far more than Nyquist to digital sampling.

For fun, look up quantization and sigma delta filters on google.

Most digital audio products use one at a bare minumum.

Here's a link or two.

Basically, an A/D convertor records all harmonics within it's pickup range. All sounds can be represented as a set of harmonics. A square wave, for example, is a mix of all harmonics possible. So, when it is sampled, all harmonics relating to that square wave below 22.5 kHz are recorded. Since humans can only hear harmonics in that range, it sounds just like a square wave would. As the square wave reaches a higher pitch, it is likely to be recorded closer and closer to a sine wave, since most of the audible harmonics are below the threshold of hearing, and the threshold of detection by the DAC. One would say it is deformed, but guess what! Our ears hear a 20 kHz square wave as a sine wave due to the same "problems" inherent in the DAC, so everything is A-OK!

Or so I was told by the telecomm students...

This might explain it in more detail...

There's far more than Nyquist to digital sampling.

For fun, look up quantization and sigma delta filters on google.

Most digital audio products use one at a bare minumum.

Here's a link or two.

Basically, an A/D convertor records all harmonics within it's pickup range. All sounds can be represented as a set of harmonics. A square wave, for example, is a mix of all harmonics possible. So, when it is sampled, all harmonics relating to that square wave below 22.5 kHz are recorded. Since humans can only hear harmonics in that range, it sounds just like a square wave would. As the square wave reaches a higher pitch, it is likely to be recorded closer and closer to a sine wave, since most of the audible harmonics are below the threshold of hearing, and the threshold of detection by the DAC. One would say it is deformed, but guess what! Our ears hear a 20 kHz square wave as a sine wave due to the same "problems" inherent in the DAC, so everything is A-OK!

Or so I was told by the telecomm students...

This might explain it in more detail...

The Pseudoacoustic Infector from Rane allows you to add a little bit of "This" and "That" to your audio. It also has BOTH "Power" and "Glory" switches, and a complement of other interesting features, plus a number of in jokes. :)

Another (newer) classic. I'm sure this WOM could be used in creating a digital version of the same.

"
Advertising, if one applies the proper transformations, is actually product information dispersal online.
"

A good example of this is ,

Rane Audio : www.rane.com

They have a reasonable list of technical data sheets on their site including a helpful "how to wire up odd cables for dodgy pieces of kit" sheet. As someone who runs a small venue we link to their website as a technical resource and we have printouts of their pages in the cupboard to aid people in wiring.

We've now started buying their kit too because we know it's made by competent people.