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Old 13 October 2016, 23:18   #55
meynaf
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Join Date: Nov 2007
Location: Lyon / France
Age: 44
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Quote:
Originally Posted by pandy71 View Post
It can be made and it is used in real life circuits - i already provided you multiple examples like PCM63 where there are two 19 bit DAC connected in a way to allow them process 20 bit samples (and you think that Burr Brown guys was so stupid that they use two 19 bit converters i.e. 28 bits to convert only 20? why they not used two 10 bit converters stacked as you propose?)
Of course two 8-bit together can't do 14 bit "as is". I didn't pretend they could (remember a thing called PWM).


Quote:
Originally Posted by pandy71 View Post
That's why i asked you as experienced software coder for explanation, i can understand if you don't know or if you not wish to share such knowledge but then say this openly.
I know what the calibration program does. What i don't know is exactly how the table is made - the code is unclear.


Quote:
Originally Posted by pandy71 View Post
IMHO this depends from you - as you don't care too much about facts if they not fit to your opinion.
You didn't show any fact - just theoretical stuff.


Quote:
Originally Posted by pandy71 View Post
This is quite obvious as in combined two 8 bit Paula DAC's at best we may have 16384 different values and if DAC is not perfect then it means that less than 16384 values can be available.
But we get more (see below).


Quote:
Originally Posted by pandy71 View Post
Please correct me but HRM provide this kind of values -36dB is exactly 6 bits.
Try this example in reality, in phase opposition so that they cancel each other. In theory (or from the HRM) you get only silence, but in real life you'll get a small beep.
(And this is exactly how the calibration program does it !)


Quote:
Originally Posted by pandy71 View Post
As i said - i've tried long time ago 14 bit on Amiga and IMHO it was not 80dB+ SNR but less - i would locate his somewhere around 60 - 70dB.
With what software ?


Quote:
Originally Posted by pandy71 View Post
Something can be fact for you and opinion for someone else.
If you write 'checked' you refereeing to subjective or objective method?
You can just check by yourself, choose your method.


Quote:
Originally Posted by pandy71 View Post
Well this almost direct value from HRM.
The HRM doesn't have to be especially precise.


Quote:
Originally Posted by pandy71 View Post
So somehow splitting 16 bit on upper and lower part leading us to 14 bit sample resolution... why not 10, 12 or 16 but 14? And why this is 2 bit difference?
Once again, it's not about splitting 16 bit on upper and lower part.


Quote:
Originally Posted by pandy71 View Post
Nope as common audio DAC in PC has usually 16 bit and additionally it is usually single bit converter.
You can just cut off 2 bits by software.


Quote:
Originally Posted by pandy71 View Post
If you run calibration on emulated in PC Amiga what you expect to get?
On emulators you get a flat response, as if it were perfectly linear.


Quote:
Originally Posted by pandy71 View Post
I expect to see measurement results not 15500 or 18000.
And yes i will be surprised to see for example SNR like 98dB but trust me if this will be for example 82dB (as this is very close to 14 bit) it will be more then very good for such circuitry like in Amiga and i will be very happy.
Then be happy


Quote:
Originally Posted by pandy71 View Post
Did you ever have opportunity to hear 10 bit DAC?
No, but it's easy to simulate. Just cut off the last 6 bits.
I know how 8, 14, 16 bits sound, so it's a reasonable estimate to locate 10 bit somewhere between 8 and 14...


Quote:
Originally Posted by pandy71 View Post
It is not clear to me - are referring to temporal dithering?
Seems yes. Isn't it (by its principle) very similar to PWM ?


Quote:
Originally Posted by pandy71 View Post
Well - if they are not used directly then it means that DAC resolution suffer and overall SNR will be worse.
Remember, values are irregularly spaced. The LSB can fix that.


Quote:
Originally Posted by pandy71 View Post
You refereeing to subjective or objective method?

If in real life you will feed analog signal from 8 bit DAC with 8 bit accuracy (assumption decent +-0.5LSB) then same samples played by 16 bit DAC with same accuracy (i.e. +-0.5LSB) will be objectively better - not sure if this can be hear but this is subjective area and everyone may have own opinion.
I won't go in the objective vs subjective land with you, as you seem to consider yourself objective and the others subjective (especially me).

Note : your "objective" data forgets about any possibility of having a good hardware lowpass filter. You concentrate just too much on the DAC.


Quote:
Originally Posted by pandy71 View Post
The one you didn't care...
Why would i care ?


Quote:
Originally Posted by pandy71 View Post
I've realized - you don't care about objective data.
Did you present objective data ?


Quote:
Originally Posted by pandy71 View Post
Well... not sure - what do you mean?
I mean that you didn't answer the question. Never mind, seems i've got that answer now (quoted below).


Quote:
Originally Posted by pandy71 View Post
Well if LUT has 65356 entries and they need to be mapped to 16384 16 bit words then i expect at best 16384 unique values (i assume that LUT is to deal with slow processing capabilities of Amiga - sample value is address for LUT and value read from LUT is feed to DAC's)
So, you confirm that you expect at best 16384 unique values.
But the 65536 are not mapped to 16384.

I ran a program to count them in actual data. My A1200's calibration file gave 17400 values. The one from Christian Buchner (available in the Play16 package) gave even more.
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