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Old 23rd April 2008, 01:18 PM   #511
NeoY2k is offline NeoY2k  France
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Quote:
Originally posted by Spartacus
Previously to using the Eval board, I was using a WM8741 based DAC. The Wolfson part has several (3 in hardware mode) filters with different ripple and roll-off characteristics. I found it hard to pinpoint any significant differences between them and wouldn't fancy my chances of saying which was which in a blind test. Kind of like swapping cables.

The Wolfson is a fine part, but changing to the Sabre made an instant improvement - knackered old cliche, but I really could hear information I'd never heard before, and it was quite obvious I was listening to a higher resolution DAC.
Ah! The answer at a question I asked some time ago, before being absolutely convinced that, if people were gettint that mad about this Sabre, that because it should really overperform everything existing.

Hum, just to know: does any of you have listened to it in 8 ch mode? How does it performs then?
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Old 23rd April 2008, 02:54 PM   #512
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Default Sabre and HDCD

For at least one the posters here, moi, the HDCD ratio frequently listened to is probably closer to 5 or 10%...and you know about the rewards of that last 10% of the performance curve. When it's good, it's really good. One way up that curve is "and" rather than "or"; I plan on both straight Sabre and HDCD fed Sabre.

Funny, the inventors of HDCD had expected it to be an interim format in the transition to a high resolution media. Thanks in part to a good old fashioned format war, it is still here a decade beyond. The PM model 2 is still regarded as one of the best, if not the best studio A/Ds for mastering; so HDCD CDs are still being released.

Cheers,

WMS
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Old 23rd April 2008, 03:01 PM   #513
4real is offline 4real  Netherlands
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Default Re: Sabre and HDCD

Quote:
Originally posted by wildmonkeysects
[B]For at least one the posters here, moi, the HDCD ratio frequently listened to is probably closer to 5 or 10%...and you know about the rewards of that last 10% of the performance curve.
With only about 5000 available titles I don't think that you can account for 5% to 10%. Maybe that people having a HDCD capable player will get to these percentages....

If you like it, you should use a HDCD decoder.. if not, don't
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Old 23rd April 2008, 05:30 PM   #514
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Quote:
I have looked at most commonly available DF's and even the
highly regarded Anagram Q5 still is half band and has 0.004
passband ripple.
Uhm, and also the Sabre is still a half band filter. For zero alias this is the only one that I know of at the moment.

http://audiopraise.com/vanity/overview.php

Ciao, George
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Old 23rd April 2008, 05:30 PM   #515
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Default Lost in Translation?

"only" 5,000? Collect them all!
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Old 23rd April 2008, 05:40 PM   #516
4real is offline 4real  Netherlands
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I just contacted the Dutch distributor of ESS, and he told me that I could only buy in quantities of +100K

I only want one or two
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Old 23rd April 2008, 05:53 PM   #517
peufeu is offline peufeu  France
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Quote:
Uhm, and also the Sabre is still a half band filter. For zero alias this is the only one that I know of at the moment.

http://audiopraise.com/vanity/overview.php

Ciao, George [/B]

Sure about the Sabre being half band ?
What is a zero alias filter ?
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Old 23rd April 2008, 06:09 PM   #518
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Peufeu,

A halfband filter does not fulfill the Nyquist criteria: it does not eliminate all components below Fs/2, but it is instead ~symmetrical around Fs/2. All this for eliminating half of the coefficients in the FIR filter.

Why halfband is a problem, according to Bruno Putzeys:

Quote:
I'm referring to the fact that for economical reasons upsampling and decimation filters usually don't fulfil the Nyquist criterium, but are -6dB at fs/2 in order to make all even coefficients trivial (halfband).
They also don't ring out smoothly at the ends but stop in a small "spike" that's the result of numerically optimising the filter to fit a certain spec with the least number of coefficients (equiripple). I wouldn't consider this a problem in filters with very low in-band ripple (in halfband filters this corresponds to a large stop-band rejection), because then the tails are so deep down that any irregularity there can't possibly be audible.
A filter with very low inband ripple is the npc SM5842 /47.

This is the thread:

http://recforums.prosoundweb.com/ind...6/0/#msg_14586

The Sabre is half band, according to the data sheet: it starts to filter at 0.454 Fs and reaches full attenuation at 0.546 Fs.

Ciao, George
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Old 23rd April 2008, 06:18 PM   #519
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Zero alias filter is an expression coined by forum member Glassman.., and I suppose that wants to describe a full Nyquist -compatible digital filter, with a lot of cells "wasted" for precision. If I'm right, it means that in it's datasheet somewhere should appear:
Passband: ~ 0.4 Fs
Stopband: ~ 0.5 Fs

Ciao, George
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Old 23rd April 2008, 07:07 PM   #520
peufeu is offline peufeu  France
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Yeah, I know what a halfband filter is. I was doubting because there's a typo in the datasheet with "dB" instead of "fs", page 16 of the datasheet, and also because I was wondering if the "slow roll off" was halfband or not (seems likely), and in this case would the "sharp roll off" be halfband too, etc.

Anyway I googled "zero alias filter" and found no reply except just one for-pay IEEE article so I wondered if this was some super secret technology or a name someone picked up for "a filter that is optimized to alias as little as possible", from what you say it is the second option. It sounds a bit marketspeak though.

I wonder what filter lengths this uses.

The XC3S200 has 12 18-bit multipliers, running at less than 100 MHz, let's guess 88.2M. That's anout 1 GMAC/s which is really good for a 10$ chip. Multiply both price and performance by between 100 and 500 for a V5SX, lol.

The specs of the Vanity mention 33-37 bit arithmetic and 47-67 bit accumulator. I wonder what that means wrt the number of bits used to encode the coefficients. If he went simple, he would have used 2 18bx18b mults to make a 18bx36b mult. Or 4 to make a 36bx36b mult. But he could also have extended a mult with gates to make a 24bx18b mults (using less mults). I wonder where the 33 bit arithmetic comes from.

For 24 bit data and 18 bit coeffs it would give about 6000 MACs per sample per channel. You could do some filtering with that.

Oversampling 44.1 at 4 times...there is 216 kbits of BRAM which must store the sample fifo (48 bits per sample, 2x24) and the coeffs (18 bits, 4 coeffs per sample, but the coeffs are symmetrical around 0 and identical left and right, so 18 bits per sample)...

Allowing extra circuitry for encoding SPDIF (if he used the SPDIF core from opencores, this one uses lots of resources) I would guess the FIR would apply to about 2.5-3K samples of 44.1 audio, max (which means either 3K taps or 12K taps depending on how you count). Which is a lot more than your usual hard filter chip. If he had used a XC3S100 he would have had about 1/4 of this, but he forked the extra $5 to go for the 200, so it is probably more taps than the 10 could handle. So, it gives an estimate.

I wonder why he didn't use a XC3S250E which has the same number of mults but costs less and can configure itself from a SPI flash. I would guess the uc is there to download several configurations from the SPI flash depending on the user settings, perhaps even depending if DSD or PCM is used.

I can't resist the call of reverse-engineering, lol.
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