SACD 2 channel vs Redbook 2 Channel

Man, I can't quit laughing at this entire thread, especially looking back on the nature of the original inquiry! I like to stir up poop as much as anyone but you guys have really let Ritteri drag himself through the mud and toss himself in the gutter here! It borders on mean, cruel and unusual to see a reputation destroyed like this; he may have to have to change his screenname or move to another board if the insanity continues! I'm shocked that his condescending personal attacks where posted on this censored forum, but not stunned to see the deserved response. Anyways, let's stop kicking the dead horse! :) I may have started this but am feeling bad for the guy and his sock puppet supporters now, seeing that reason and logical (yet often funny as heck) retorts have fallen on deaf ears (was that a funny?)! Seriously, the best way to end this is to stop feeding the monkey, to paraphrase a Big Lebowski line. It's painful, it is a train wreck. Agree to disagree and simply quit wasting time and energy on this....unless of course there's more opportunity for the funny stuff from following Ritteri posts after he reads this one :)...just kidding!(sorry!)

Information? You want information?

Click here, go to this page and click on their list of links...

http://www.daisy-laser.com/tech3.htm

Ritteri writes:

I believe it can reproduce a perfect signal up to 22khz.

That's incorrect.

What follows is a simplification, but you'll get the idea. Members, feel free to correct me as I'm only an enthusiastic amateur and am keen to learn.

In order to avoid "aliases" (byproduct of the sampling) when converting the original analogue signal to digital, no signal at half the sampling frequency must be present. Since the Redbook sampling frequency is 44.1kHz, this means that no signal must be present at 22.05kHz.

Let's say there was a signal at 24kHz. Sampling would produce an "alias" - an artifact - at 12kHz, which you can hear. Clearly we don't want this to happen. So the signal must be way down in level at 22.05kHz.

Yet, to have accurate reproduction to 20kHz (the nominal limit of human hearing), we want normal signal strength (whatever there is in the performance) at 20kHz.

So the signal must be passed through a very steep filter which is not affecting the signal at 20kHz, and is 90dB down at 22.05kHz. The famous "brick wall".

Oversampling attempts to overcome this problem. If the sampling frequency is (say) 88.2kHz, then we have to pass the signal through a filter that is flat at 20kHz and 90dB down at 44.1kHz. Still pretty steep and difficult to make without nonlinearities, but doable. Now we need a mathematical algorithm to choose (essentially) every second sample point and save the amplitude, thus making the digital recording.

Let's say the studio is recording digitally at 96kHz and 24-bit words. They make the recordings, mix it in the digital domain, and now they have to prepare it for the Redbook format. Lots of very funky mathematics to convert down to 16/44.1.

Consider now a studio recording in DSD. They make the recording, mix it (DSD mixers are more available now) and put that on the disc.

Similarly with DVD-Audio. The studio could record stereo at 192kHz, probably mix digitally at that resolution, and save this on the DVD using lossless compression.

In principle, both are superior to Redbook.

Regards,

Littlemilton: And what makes you think your any better than Rittori? You are the last person who can point a finger or make a comment, as your crude behavior and need to make an ass out of yourself on every post has shown myself and others what this community really doesn't need.Grow up child.