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Old 21st December 2012, 07:23 AM   #281
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Originally Posted by PHEONIX View Post
The AD797 has only 20V/us slew rate and as an integrator on the output of a Dac its the gold standard for THD and Noise performance in the audio bandwidth.
Yes, it gives... nice numbers...
THD (when not in excess) is not so important, IM and phase modulation distortion are more. And they are related with slew-rate. I definitivaly use current feedback OPAs for analog in my designs everywhere i can. If i cannot (DACs), i try to find fast voltage feedbacks units.
And i do not care as well for very low noise numbers as long as i do not hear-it. It can even arrange the things a little, making the sound-stage more natural..
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Old 21st December 2012, 07:50 AM   #282
Hennie is online now Hennie  South Africa
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Old 21st December 2012, 08:02 AM   #283
Omholt is offline Omholt  Norway
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I have used a Lynx Two-B soundcard with a digital volume control (32 bits I believe) for years. Never heard any loss of resolution. I've also had analogue attenuators.
What I haven't been too found of though is using the volume control on a PC.
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Old 21st December 2012, 08:15 AM   #284
Julf is offline Julf  Europe
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Originally Posted by AR2 View Post
to be honest, digital volume out of ESS chip is the first volume coming out of a silicone that sounds good.
Just have to pick some nits... Silicone usually looks better than it sounds - I assuming you meant the chemical element silicon (Si, atomic number 14) that most semiconductors are made of, and not silicone, the synthetic polymer that is used in sealants and implants.

I do remember an old cerwin vega horn tweeter from the 80's that probably used both - it was usually referred to as "Dolly Parton" due to it's distinctive shape...
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Old 21st December 2012, 02:26 PM   #285
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Well, may-be time now, for Hypex, to design a digital input Class D amplifier.?
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Old 21st December 2012, 02:29 PM   #286
ds23man is offline ds23man  Netherlands
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Originally Posted by Esperado View Post
Well, may-be time now, for Hypex, to design a digital input Class D amplifier.?

That means moving the dac to the amp...... Class D is not digital, but a pwm switching analog amp.



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Old 21st December 2012, 02:38 PM   #287
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That means moving the dac to the amp...... Class D is not digital, but a pwm switching analog amp.

Well I guess that would be true if it was entirely impossible to generate the PWM signal in the digital domain (as opposed to just difficult to do well?).
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Old 21st December 2012, 02:50 PM   #288
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Originally Posted by ds23man View Post
That means moving the dac to the amp...... Class D is not digital, but a pwm switching analog amp.
Look for Nad M2 or http://www.diodes.com/_files/audio/ze0448.pdf or MAX98356.
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Old 21st December 2012, 03:00 PM   #289
ChrisPa is offline ChrisPa  United Kingdom
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Originally Posted by Esperado View Post
Look for Nad M2 or http://www.diodes.com/_files/audio/ze0448.pdf or MAX98356.
That's a class Z
and IIRC uses an adc in the feedback loop, and so therefore also uses the analogue domain for the feedback loop

Have a look a Bruno's original papers on the ncore and follow his logic about the circuit topology of class D. This has been discussed many times in this thread. Class D uses an analogue feedback loop and is an analogue amplifier. Other topologies are other topologies
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Old 21st December 2012, 03:01 PM   #290
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with proper gain structure and 32bit+ volume, with 24bit dacs and proper gain structure, remind me of how any of that has any impact?
Because with 113dB DNR the DLCP is a not quite 19 bit DAC. It depends on how one chooses to do the budgeting but, in rough numbers that's 113dB DNR - 80dB headroom above noise floor - 10 to 20dB crest factor = 13-23dB of volume control range in an optimal gain structure. I generally use around 20dB of volume adjustment but tend mostly to listen to lower crest factor material so this works for me. But digital volume control in DAC to the power amp topologies easily runs into trouble if the dynamic range requirements are wider.

Every dB of excess power amp gain is a dB less of volume control range. Since there's basically no margin to start with I'm curious to see if Hypex's analog output board addresses the scaling issues for nCore customers. The nCore's gain is set, reasonably enough, to hit its 400W output but that means it's around 20dB too high for the majority of listening done with it.

Also, while the TAS3108 has a 48 bit data path and 76 bit MAC, it's limited to 28 bit coefficients. I'm not sure if that means it implements IIRs with 28 or 48 bit feedback but, either way, there will be some numerical noise in 24 bit output for filters operating at low normalized frequency---exactly what low means depends on DSP details and the filters being used but center frequencies below 1.5 to 2kHz is probably a good rule of thumb for the DLCP. How much this matters depends in part on how Hypex has chosen to partition volume control between the DSP and DAC and what the AK4396 does with the least significant bits, neither of which is specified (and if either's been mentioned on the thread I missed it), but there is some possibility of audible artifacts. I would tend to guess inaudible or pretty minor if audible, but this is really the sort of thing that requires ABX tests.

Quote:
Originally Posted by qusp View Post
well I guess people will first work out an I2S input mod to avoid that 2704
Post SRC4382 would be better as it'd avoid the impulse response smearing from the brickwall, linear phase interpolation filters in the SRC4382, though that's a significantly more involved mod. There'll also be some asymmetry in the impulse response due to the non-integer resampling ratio, which I've found tends to reduce sound quality as well (though in the mainline case of resampling 44.1 it won't be too bad). Given one can follow PLL based clock recovery with a jitter cleaner at a lower BOM than the 4382 Hypex's choice to do sample rate conversion is an interesting one. It'd also be interesting to know if the AK4396 is operating with sharp or slow rolloff antialiasing filters. Curiously, the group delay doesn't reduce in slow roll. Maybe that's a datasheet typo or maybe AKM shifts the filter's position along the linear phase-minimum phase continuum.

Last edited by twest820; 21st December 2012 at 03:04 PM.
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