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Old 20th December 2012, 06:58 PM   #261
ChrisPa is offline ChrisPa  United Kingdom
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Possibly chosen to ensure it's always deliberately asynchronous
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Old 20th December 2012, 07:32 PM   #262
Julf is offline Julf  Europe
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Possibly chosen to ensure it's always deliberately asynchronous
Precisely.
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Old 20th December 2012, 08:17 PM   #263
Davey is offline Davey  United States
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Not yet - right now I have a Behringer DCX2496 driving the ncores, but aiming to replace it with the DCLP - just got shipping acknowledgement.

Not sure what's so excellent about it, as 16 or 36 dB is just as good as 26, as long as the overall gain structure of your system makes sense.

Easier, yes, but not better. It is never a very good idea to over-amplify and then have to attenuate. Ideally you have just enough amplification in your whole chain to give full output with full-scale input. The nice thing about a digital source is that you know for sure what the highest possible input amplitude will be.
Where is your system volume control when using the DCX2496?

It sounds like you will now have to un-modify your Ncores for use with the DLCP. Some simple, fixed attenuators installed between your DCX2496 could have (now) easily been removed to facilitate your DLCP integration.

We don't always have the option to avoid an amplify/attenuate scenario when integrating various commercial gear. A fixed attenuator is a sonically transparent option that can optimize gain structure in most situations.

Cheers,

Dave.
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Old 21st December 2012, 02:15 AM   #264
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what I mean is; if you have to turn the digital volume down to a point where it is noticeably effecting the sound quality, your gain structure is wrong, because you shouldnt be needing to turn it down that much, if you do have to, you have too much gain.
Our ears works on a logarithmic way.
Reducing the volume (level control) in the digital world reduce the numbers of available bits. While analog volume control reduce the levels of each step, but keep their number (definition) the same. We have to remember too that our ears are able to ear signals UNDER the noise floor.
If 24 bits are preferred for audio reproduction, it is not for the signal/noise ratio (96dB is more than enough) but for definition of the lowest signals.
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Last edited by Esperado; 21st December 2012 at 02:17 AM.
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Old 21st December 2012, 02:23 AM   #265
qusp is offline qusp  Australia
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edit: sorry for snippyness, need to think, just opened eyes, but I really at this point do not see the relevance

with proper gain structure and 32bit+ volume, with 24bit dacs and proper gain structure, remind me of how any of that has any impact?

you are using the very same mechanisms used in the production process to mix that music

Last edited by qusp; 21st December 2012 at 02:34 AM.
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Old 21st December 2012, 02:34 AM   #266
OllBoll is online now OllBoll  Sweden
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Originally Posted by Esperado View Post
Our ears works on a logarithmic way.
Reducing the volume (level control) in the digital world reduce the numbers of available bits. While analog volume control reduce the levels of each step, but keep their number (definition) the same. We have to remember too that our ears are able to ear signals UNDER the noise floor.
If 24 bits are preferred for audio reproduction, it is not for the signal/noise ratio (96dB is more than enough) but for definition of the lowest signals.
Isn't extra noise exactly what happens when you reduce the number of bits, thus the signal / noise ratio is lowered?

And 16 bits is enough for music, so in practice with a 24 bit dac you have 8 bits of padding you can shave away without losing anything at all. And when you are shaving off more than 8 bits then the signal is too quiet for you to notice the increased noise anyway.

This of course as some others here already have said requires that your gain structure is ok.
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Old 21st December 2012, 02:37 AM   #267
qusp is offline qusp  Australia
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it also ignores the difficulties of building a multichannel high grade analogue balanced control and the effect it has, with 12 decks minimum, with lower CMRR due to resistor matching (thus higher noise and higher distortion), OR a chip based digitally controlled analogue attenuator, which adds more active elements and lowers the performance to its performance.

man this is a hard nut to crack, the audiophile stubbornness is so strong with this issue, people that normally understand much more complex mechanisms somehow manage to miss the point with digital volume.

its made even more ironic by the venue here ie. a DSP/Crossover that uses these mechanisms at its very heart, if you cant palette digital volume, perhaps you should look at other devices....

Last edited by qusp; 21st December 2012 at 02:46 AM.
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Old 21st December 2012, 03:31 AM   #268
HFGuy is offline HFGuy  Canada
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Am I understanding this correctly, if using the AES input you can only do stereo ? But if you use the analog inputs you get 8ch in ?
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Old 21st December 2012, 04:11 AM   #269
ds23man is offline ds23man  Netherlands
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If you apply dither you can shave of more than 8 bits with digital volumecontrol!


Ok a part list:

Dac: AK4396
Dac intergrator: NE5532
Balanced output buffers: OPA1632
DSP: TAS3108
USB audio: PCM2704C
ADC: PCM4202
SRC: SRC4382
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Old 21st December 2012, 04:57 AM   #270
qusp is offline qusp  Australia
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well I guess people will first work out an I2S input mod to avoid that 2704

the opa1632 is a great choice though, love that chip!
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