The Antelope Zodiac / Zodiac + / Zodiac Gold DAC Thread
Oct 17, 2009 at 8:49 PM Post #46 of 529
Quote:

This is not bad! Why? Because the EAR can only hear a certain dynamic range (from the loudest to the lowest levels. Each bit is around 6dB so at 20 bits we have 120dB dynamic range. That is huge. A CD has only 16 bits thus 96dB and that is already pretty quiet.


Hi Dan and all,

Good points - and a question:

What is the typical best dynamic range that one finds in digital recordings? I am not involved in the recording process - so I have been curious about this. Do the best digital recordings even come close to having 96dB of dynamics?

And what about Vinyl?
wink.gif


All the best,

Vinnie
 
Oct 18, 2009 at 4:41 AM Post #47 of 529
No, I'm not going to debate Dan Lavry. I am going to suggest people listen to the clocks and converters and draw their own conclusions.

Marcel

Quote:

Originally Posted by Dan Lavry /img/forum/go_quote.gif
First:
Jitter is important for digital audio but atomic clock is not less jitter! Atomic clock offers time accuracy such as nanosecond drift over a long time. This is important in many applications but useless for audio. Even a poor crystal, say 500 parts per million will alter the pitch of the music by less then 1 cent (which no human can hear).

Also, a 100ppm crystal (pretty poor) will make hour music longer or shorter by 0.36 second, and no one would care. Jitter is important, and atomic clocks do not offer particularly good jitter. They cost a lot but that does nit change the facts.

Second
32 bits? We use a lot of bits for audio work station. We need that for mixing and editing of audio tracks. But the outcome is 24 bits or less. In fact, the individual audio channels recorded by an AD NEVER have 24 bit accuracy. 20 bits is real good! The other bits may be there but they are just random noise, not at all connected to the music. Those lower bits are the noise floor.
We are restricted to such noise levels because of analog noise in components but with careful design, we keep that noise low, as low as say 21 bits.

This is not bad! Why? Because the EAR can only hear a certain dynamic range (from the loudest to the lowest levels. Each bit is around 6dB so at 20 bits we have 120dB dynamic range. That is huge. A CD has only 16 bits thus 96dB and that is already pretty quiet.

If you stand next to a 747 engine (running), and call it the loudest you want to hear, then go to a sound isolation room and call it the lowest sound level, you have less then 126dB which is 21 bits. So who needs 32 bits? When it comes to digital audio workstation, you need that and more. For listening, no one needs even 24 bits.

Third
384KHz, as well as 192KHz is marketing driven, and it is NOT GOOD for best sound. Too slow is not good, and too fast is not good either. There is an optimal sample rate, and it is somewhere in the 60-70KHz. Of course there is no such standard so 48 or 88.2KHz is reasonable, and even 96KHz is just a bit too fast. I explained all that in great detail in my paper "Sampling Theory". The paper is too technical for some folks, I tried to keep is comprehensible...

Regards
Dan Lavry
Lavry Engineering



 
Oct 18, 2009 at 12:33 PM Post #48 of 529
Quote:

Originally Posted by Dan Lavry /img/forum/go_quote.gif
...

This is not bad! Why? Because the EAR can only hear a certain dynamic range (from the loudest to the lowest levels. Each bit is around 6dB so at 20 bits we have 120dB dynamic range. That is huge. A CD has only 16 bits thus 96dB and that is already pretty quiet.

...



Can you explain further why a bit is 6dB? I just do not understand what does the 24db record? Is it the loudness of the signal or the voltage of the signal?

Thanks.
 
Oct 18, 2009 at 2:37 PM Post #50 of 529
Quote:

Can you explain further why a bit is 6dB?


Hi GreenLeo,

Digital audio is mathematical in nature, so for an explanation for this, you need to see the derivations. Here is a link that explains is pretty clearly:

Deriving the 1 bit = 6 dB rule of thumb « Harder, Better, Faster, Stronger

Keep in mind that this applies to PCM audio (not DSD, as used in SACD. But isn't SACD dead anyway
confused_face(1).gif
).

Years ago when I was an EE undergrad, I took a class in digital audio theory. We used this book, which is very informative (over 800 pages):
Amazon.com: Principles of Digital Audio (9780071441568): Ken Pohlmann: Books

You can learn A LOT from it - highly recommended!

Quote:

Is it the loudness of the signal or the voltage of the signal?


When doing D to A, the output of the converter is either a voltage or current (in the case of an output current, it is then converted to a voltage that passes to your RCA, or balanced, outputs). You feed that into a headphone amp (in order to provide drive for the low impedance of heapdhones, and a volume control).

Here is an example of bit resolution:

16-bit offers 2^16 = 65,536 discrete levels of resolution.

For a visual analogy, imagine having a stack of 65,536 pages of paper. That is a stack that is approx. 25 feet tall. The finest level of resolution is 1 piece of paper. Do you think your eyes could see that one piece of paper missing on a stack of paper 25 feet tall? Now think of it in terms of output voltage (and we know standard output voltage is usually around 2Vrms), so using the analogy above, you will see that 16-bit is damn good, and so is 96dB dynamic range (especially when recordings don't even have this kind of dynamic range because so many times they are boosted in level to sound louder - but they are killing the dynamic range anyway).

If you search in google, I'm sure you can find a ton of links that do a much better job at explaining this - but hopefully this is a good start and has not resulted in more confusion. You need an understanding of the very basic theory of digital audio (e.g. sampling and quantization) before you can make sense of all the math. That book I link to above should be a good start (at least the first few chapters), but it does get "mathy" as they all do because you can't fully explain all the theory and numbers without the math...

NOTE: Perhaps this topic should be moved to another thread - I just realized that I am contributing to thread-jacking, and that is not cool of me. I apologize, guys...

Vinnie
 
Oct 19, 2009 at 5:29 PM Post #51 of 529
Quote:

Originally Posted by danne /img/forum/go_quote.gif
Looks like I might wanna wait abit with that lavry da11 purchase then, I might be better off with one off these. ...


I suspect the Lavry DA11 and the Zodiac DAC are in COMPLETELY different price categories!
 
Oct 19, 2009 at 5:30 PM Post #52 of 529
I am eying the Ayre QB-9. That things looks like a great deal also.
 
Oct 19, 2009 at 6:23 PM Post #53 of 529
Quote:

Originally Posted by tosehee /img/forum/go_quote.gif
I am eying the Ayre QB-9. That things looks like a great deal also.


It only has USB input though?
That might be a prob if you want more than the computer as a transport.
 
Oct 19, 2009 at 6:46 PM Post #54 of 529
Quote:

Originally Posted by danne /img/forum/go_quote.gif
It only has USB input though?
That might be a prob if you want more than the computer as a transport.



I worried about this a bit, but more and more I find myself using PC as a transport only. I already have a dedicated laptop for this duty, so this is a best solution for me. If it really works as advertised, I don't need the CD transport any more as I can always rip it to HD and use the laptop + this DAC.
 
Oct 19, 2009 at 7:30 PM Post #55 of 529
Quote:

Originally Posted by GreenLeo /img/forum/go_quote.gif
Can you explain further why a bit is 6dB? I just do not understand what does the 24db record? Is it the loudness of the signal or the voltage of the signal?

Thanks.



Why 6dB per bit:

In digital audio, you "take samples" of the sound, and then place a value on each sample. You can think of it as placing a transparent grid on a drawing on "going for the closest" point of the grid. The vertical lines of the grid are for sample time, and the horizontal lines are for sample value (sample amplitude).
We will focus on the horizontal lines.

In the real world, each sample is assigned the value of the closest horizontal line just below it.

Think of money. Say one buys an item for one $1.05, but the tax is 3%. That gives you a total of $111.1419048.... As you can see, we go for the closest penny and settle for $111.14

Similarly, we have to settle for some precision in the digital word, and only accommodate "good enough" number of bits". We can just "keep going forever", nor do we need to.

So say you have only 2 bits. That means the horizontal grid is made of only 4 lines. Now add a bit. You now have 3 bits. Adding that bit means adding a new horizontal line right smack at the middle, between the each "existing" lines of the 2 bit system. (With 3 bits you have 8 levels).

Now, add another bit, with 4 bits you have 16 levels. In fact, you take the 3 bits grid, and add a new line between each pair of "existing lines".

So each additional bit DOUBLES the number of quantization levels (horizontal lines). In fact, the space between the lines (quantization error) is halved for each added bit.

With a voltage linear system, twice the voltage, would mean twice the loudness. With a power linear system, twice the power, would mean twice the loudness. But the ear is not at all linear.

Now we want to go with that above information and apply it to the ear. The ear reacts to loudness with a scale that is nearly logarithmic. So what is a factor of 2 mean?

The dB scale for voltage is dB=20*log(X) so we plug in X=2 and get
dB=6.02
In the "logarithmic world", multiplication is turned into addition, division becomes subtraction. So double the number of lines means adding 6dB. Half the error means 6dB more dynamic range. Half the error again means 6+6=12dB... and so on

This all may sound strange. Why is ear logarithmic? For that you have to address mother nature. One way to "reason it" is protection of our hearing. As things get louder, the ear response becomes lesser and lesser... That way we are protected from some huge acoustic power damaging our hearing.

If one is not into the math and engineering, think of loudness being perceived accordion to a "curve" that translates voltage (or power) into loudness in some way that can be figured mathematically or graphically, and it is not a "straight" conversion.

So how many quantization lines do we need? That does depend on the ear, but with 20 bits, that is plenty good, we can not hear better then 20 bits in the most extreme cases!!!

Note that while each additional bit (new horizontal line), you get closer to the sample values, at some point, with enough grid lines, the spacing is so close that there is not much to be gained from more quantization lines. At 20 bits, you have a round 1 million lines! That is 6dB X 20 = 120dB.

So why 24 bits? Well in the computer world, much is being handled in "multiples of 8 bits", which is a byte. As long as we had a 16 bit format, it was 2 bytes. The minute you went for 17 bits, the hardware and software required one more byte, and 2 X 8bits = 24 bits. The last bits are just there, they do not help the music, because the signal never gets to be 24 bit accurate. We can not do it, because of analog noise, and we do not need it because the ear does not hear 144dB.

Regards
Dan Lavry
Lavry Engineering
 
Oct 19, 2009 at 7:46 PM Post #56 of 529
I was just going to post how pointless most of the statistics on that DAC are (32bits, 384khz? LOL!) but I see Mr. Lavry already beat me to it.

This thing is simply for people who have too much money and not enough knowledge.
 
Oct 19, 2009 at 8:13 PM Post #57 of 529
the antelopes look like really nice kit.
but i only want a dac. pity they dont make a dedicated dac!
 
Oct 19, 2009 at 8:22 PM Post #58 of 529
Oct 19, 2009 at 8:23 PM Post #59 of 529
Quote:

Originally Posted by Dan Lavry /img/forum/go_quote.gif
The last bits are just there, they do not help the music, because the signal never gets to be 24 bit accurate. We can not do it, because of analog noise, and we do not need it because the ear does not hear 144dB.

Regards
Dan Lavry
Lavry Engineering



There we have it guys. There is no need in paying for expensive DACs, converters or transports that don't do the trick.
I'm sticking with the 16bit/44Hz format until something provenly better comes up.
 
Oct 19, 2009 at 9:58 PM Post #60 of 529
Quote:

Originally Posted by punk_guy182 /img/forum/go_quote.gif
There we have it guys. There is no need in paying for expensive DACs, converters or transports that don't do the trick.
I'm sticking with the 16bit/44Hz format until something provenly better comes up.



You can hear better then 16 bits, 18 "real bits" is better then 16 bits, I would not mind 20 bits. And a little faster sample rate then 44.1KHz is better also. 48KHz is better then 44.1KHz. 88.2/96KHz is also pretty good. Any faster is not a good idea.

I heard some real beautiful 96KHz 24 bits (probably 18-20 real bits). Some of the high definition audio (DVD Audio) is fantastic to listen to. It is unfortunate that it did not gain commercial acceptance.

At the end of the day, I too do most of my listening to CD format (16 bit at 44.1KHz). I would like to have a few more bits, and somewhat faster sample rate. I would love to have say 18-20 real bits (that means 24 bits format) at 60-70KHz. But given that most of the music comes on CD's, I listen to CD's... And there are some fantastic CD's out there.

Regards
Dan Lavry
 

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