All posts with more than 3000 Hits, prior to 2008

Tue May 20, 2003 4:07 am

When it comes to archiving, DSD makes complete sense. The masters are wasting away every day. Why not put it way up there, and thus not having to worry about theories and the 8 percent or so that can differintiate spacial separation. 22khz I feel is too low. It is too close to the error of human ear. Why not double it or triple it?

I think the problem is this, the cd takes the first 16 bits at a rate of 44.1 khz. It does not care where it sample as long as it is consisistent? WHy not give something more for it to grab? Better yet, with storage being almost a non-issue, why not have it the best it can sound? Even if most humans can't hear it. I think the human ear is very adaptable, and if one hears a tune after a while they get trained to the sound. That is why my link was important. Studies with humans and spacial separation are showing that there are people that CAN hear the difference!!

Tue May 20, 2003 4:12 am

Get all Elvis music masters in the best possible sound quality for safe keeping. Cds aren't going to be the last medium used to reproduce the music for us. Something better will come along that will benifit from the best sound available.

8)

Tue May 20, 2003 4:56 am

I agree with Genesim that in the interest of preserving music for posterity, they should sample it with an outrageous bit/sampling rate. If they have to play those tapes again each time a new type of media comes out, they won't last much longer.

Those of you that collect live rock music other than Elvis may have heard of the Clevaland Agora, a venue where they used to broadcast live shows on the radio each wee. The Agora amassed an amazing library of excellent live concerts on twin track reel to reel. A couple years ago, it was reported that they were archiving all their stuff to audio CD and discarding the tapes since they no longer needed them. I about cried. I have heard now that the tapes, or at least some of them, have been rescued from the scrapper, but tapes only last a finite amount of time and I really hope the music industry is trying to archive this stuff, but sadly I think they have other more important things to do like harass students downloading MP3s and such....

Greg

Tue May 20, 2003 10:26 am

genesim wrote:If you look closely...I mean study it Vinylman. I did not change the Frequency. Any sound peak can be hit, as long as it is in the 96 db range. This has NOTHING to do with the frequency of the sample. When sampled at a different point...it is again has the same SAMPLING RATE...but taking a more accurate part of the curve. In other words, the cd can hit any part of the sound curve, just not all of them at the same time as on a 100khz transfer. Get it! What it does sample is going to be more accurate!!! There is a limit and it comes from the medium and how it is programed.


Sorry, this is exactly what I mean and where you are wrong. The problem is that you define your own theory and examples. And at the same time you mix the important difference between dynamic range and frequency range. Your examples is only refering to the time domain. If you try to enter the frequency domian and try to understand the fundamentals of the fourier analysis you'll we se another way to approach the sampling theory. Using the Nhyquist theorem and fourier analysis you'll see that in theory your samples will be reproduced correctly regardless of where you hit the original waveform. This is a fact that all educated engineers do agree on.

Nhyquist says: A signal must be sampled at least twice as fast as the bandwidth of the signal to accurately reconstruct the waveform.

That leaves us with a problem for high frequency signals as frequencies above half the sampling rate wil alias at a frequency inside the spectrum. So an anti aliasing filter is needed to filter out the those rapid fluctuation in the wavforms that makes your samples different, gensim.

Maybe this made it any clearer. I had to go back to my old books from the university to get the correct english words for the theorem. But it is basically the same as I have been trying to teach you for a week now. but maybe my english is the problem, as I feel that you often jump to the less important parts of my postings.

From now on I will try to take one piece at a time and maybe we can agree on this part before we go further into details.

Tue May 20, 2003 4:19 pm

ok a piece at a time. Yeah this will help.

I agree on your theorem. Too bad it is not what I am talking about. Frequency has little to do with this.

Look, take a certain range that is possibly produced by a cd. Now cut the sample 16 times. Ok then cut it 64 times that. ALl is in the range of the cd and frequency or whatever. As long as the cd doesn't sample(16 times) at the same spot of each wave form then the higher quality tranfer will have the advantage.

Why? because of mathematics, in reference to numbers representating a sound curve. I gave a example of a sound curve. True it is over simplified..but you have to zoom in to get what I mean(I think of using a graphing calculator for this). You are foccusing on the frequency(speed) and I am actually referring to VOLUME!!

Tue May 20, 2003 5:03 pm

genesim wrote:ok a piece at a time. Yeah this will help.

I agree on your theorem. Too bad it is not what I am talking about. Frequency has little to do with this.

Look, take a certain range that is possibly produced by a cd. Now cut the sample 16 times. Ok then cut it 64 times that. ALl is in the range of the cd and frequency or whatever. As long as the cd doesn't sample(16 times) at the same spot of each wave form then the higher quality tranfer will have the advantage.

Why? because of mathematics, in reference to numbers representating a sound curve. I gave a example of a sound curve. True it is over simplified..but you have to zoom in to get what I mean(I think of using a graphing calculator for this). You are foccusing on the frequency(speed) and I am actually referring to VOLUME!!


We speak about two different things, that is for sure. But you are now talking about a limitation in the dynamic range and not the frequency range. And that is certainly true. But 16 bit is good enough to recreate about 96dB. It doesn't sample 16 times but it calculates 16 times if needed to make the 16 bit data word. So it can actually recreate the curveform refering to what you call volume too but not above 96dB.

So now we agree about the facts that a DSD master doesn't have a better frequency range than a pure PCM master when both are transfered to CD-DA. And we also do agree that the frequency reproduction is the same.

The only thing left then is the dynamic range. So when one is assuming that the CD-DA has a limit of 96dB from 0 to maximum. And the Elvis matertape source is maximum say 80dB, isn't it obvious that a direct PCM transfer can reproduce as accurate as a DSD master transfered to CD-DA format.

Tue May 20, 2003 5:53 pm

Uh no! because of the soundcurve representation that are going to be (millions?) several intances where the sound is going to be rounded off where the DSD representation will be rounded off in a more accurate way!

I don't think I ever argued beyond the range of a cd! Then again, I admit, I get confused with terminology. By the way, doesn't this go against your whole HDCD arguement?

Again, you are arguing the same thing. Just because it CAN produce it, doesn't mean it does. YOu have to look at EVERY aspect of the sound curve(this is my theory-NOT what is actually heard).

These are the facts. Certain representations of the sound curve are GAINED when sampling from a superior master. How much well probably millions when you consider how much it is sampled. Every deviation will be a more accurate representation. My theory holds only if the cd does not sample in the same place as the lower transfer. Any slight (like starting at a different place, like the next bit or the bit before), will be every sampled bit as more accurate(taking out the rounding and the technology actually guessing right on the final process). Thanks to Greg for actually pointing this out to me. I knew it came down to the physics of the actual process, but I just wanted to make sure.

Your arguement way back there was 16 bit vs 24 bit(the gain from HDCD). Which again goes against your gain. 96db to what over 100db? The bits may have changed but the sampling resolution stayed the same! Now if we are talking 24 bit tranfers, this is a whole nother story, then we are down to 24 bit masters vs DSD masters. Most sound critics go with DSD because of the sampling capablity being superior.

The original argument was DSD Technology vs HDCD. Well I have "proven" my part, you have actually disproved yours by saying that Elvis transfers are "80db"(where did you get this number?).

Tue May 20, 2003 6:19 pm

Ok just got the figure. SO HDCD boasts that it can get up to 144db(only when decoded!!!). Again, you disproved yourself Vinylman on the DSD post. I don't really agree with that though. I think eliminating distortion in the high end range is a good thing, EVEN if we can't hear it.

My main problem with HDCD is this..it is obsolete now. It is inferior to both DVDA and SACD. WHy would anyone want to encode that way? There is really no benefits on a standard cd and they really have no sales.

Wed May 21, 2003 10:12 am

bump

Wed May 21, 2003 6:06 pm

I am awaiting a response Vinylman!