Not to question your subjective opinion, but did you compare (ABX) it to no preamp? What source did you use?
Of course - and yes this is my subjective opinion of course.
I have tried many different configurations an sources - let me name a few:
TT via PS Audio GCPH phono direct into Ncore via balanced outputs
Plinius CD-LAD preamp via balanced
Digital sources:
NAD M51 dac direct via balanced
Teddydac direct single-ended and also via Plinius
Audio-GD Ref5 Dac direct via balanced and also via Plinius pre
Lenehan PDX dac direct via single ended and also via Plinius pre
Bel Canto Dac3 direct via balanced also via Plinius pre
Jungson moon CDP via Plinius
That's all I can think of so far, but is not the total list - fit example a friend brought around the latest Emotiva preamp and we played around with that as well.
I've had the benefit of hearing a lot of gear using the Ncores - and in my humble opinion the best sound that I have attained was the TT direct connected via the PS Audio - and now with the B1 buffer based pre that I've recently made that has a TDA1543 based nos dac.
Audio is a very subjective thing so I'm not making a factual assertation - just what I think. Take it how you like.
AJ
And by "clarity" I'm talking about the sound appearing "pristine" - totally uncoloured and being the audio equivalent of a fresh and pure autumn morning - chilled still air and a cloudless sky with the promise of a nice day.
Both the Plinius and Emotiva, and now that I remember it a Quad Elite pre all coloured the sound somewhat. The Plinius have it some extra weight and slight euphonics much like I expect a good tube pre may, the Emotiva was relatively transparent but a little uninvolving, while the Quad was natural but a little muddy / accentuated in the base.
It's all about synergy I've learnt if nothing else.
Both the Plinius and Emotiva, and now that I remember it a Quad Elite pre all coloured the sound somewhat. The Plinius have it some extra weight and slight euphonics much like I expect a good tube pre may, the Emotiva was relatively transparent but a little uninvolving, while the Quad was natural but a little muddy / accentuated in the base.
It's all about synergy I've learnt if nothing else.
may be 2xSMPS600
? Each for 1 NC400.About NC400 all ok. But how about balanced input? My DAC has unbalanced output, so may be some adapter needed? And what about volume control? Is it ok to use GOLDPOINT Goldpoint Stereo Mini V Stepped Attenuator 100K | eBay or digital volume control in foobar2000 (on computer) will be enough? My music source played from: computer -> Flamenco (USB to I2S converter) -> DAC -> amplifier -> STAX 507
may be 2xSMPS600
If you want to. Not necessary.
No, just correct wiring.
I thought you specifically wanted digital control. That stepped attenuator is definitely analog, but in pretty big steps. I would use the digital volume control provided by your computer and DAC. What DAC are you using, and what kind of volume control does it implement? As it is USB/I2S-based, I assume it does understand USB audio volume control messages.
Thanks for answer! Volume control is big problem for me. If I will use digital gain in foobar2000 then it will be loose 1 bit from sound quality (this bit is needed for volume). So sound will be less quality after digital control. Am I write?
I use DIY DAC; it's based on delta-sigma SM5865 and digital filter SM5847AF (24 bit 192 KHz).
Why would you loose 1 bit?
Anyway, even if you lost more than 1 bit, it is no issue. Your DAC has a dynamic range of less than 20 bits, so even if you sacrifice 4 bits of the 24 bit input range, it won't make any difference.
Does the input system of your DAC handle USB audio volume messages? I couldn't find any information on your USB to I2S converter.
No it has not. Only computer has volume control.
Though I think digital control will lose to high quality passive attenuator. If you change source signal, you miss quality, you involve some distortion to original source.
OK. Is that a software limitation in your DAC controller?
Isn't that precisely what a passive attenuator does?
Again, you have at least 4 bits of your 24 bit input to "use up" (giving 24 dB of attenuation) before you even get into the 20-bit dynamic range of your DAC. Most music has a maximum dynamic range of 12 bits on a good day, and even if you attenuate down to -48 dB, you still have a full 16 bits left. Remember that additional bits don't give any more "precision", they only give you increased snignal-to-noise ratio when the signal is properly dithered.
Are you aware that your digital filter already processes and dithers the signal anyway?
no
well, it is in theory. I must check it
. May be you are right.it one of the best digital filter. It is reduce digital trash and does not worsens sound.
Indeed. That was precisely my point - the processing and dithering (that actually adds noise on purpose) does not harm the sound.
In theory yes but not in practice. You don't need all 24 bits, 16 bits are enough. So you could think of the 24 bits as 16 bits of nice sound and then 8 bits of padding.
You can then attenuate -48 dB without loosing any SQ at all, which should be enough for everyone unless your gain structure is very wrong and in that case making an passive fixed attenuator would solve the problem.
Well, that theory is the Nyquist–Shannon sampling theorem. If you don't believe in that theory, all digital sound should be suspect. Either the theory works, and we can represent an analog wave in digital form, or it doesn't, in which case we all are just imagining we are hearing music from our speakers
Well, that theory is the Nyquist–Shannon sampling theorem. If you don't believe in that theory, all digital sound should be suspect. Either the theory works, and we can represent an analog wave in digital form, or it doesn't, in which case we all are just imagining we are hearing music from our speakers
I assume you are joking? -I'd say that is to stretch the explanatory power of a theory into something that no theory can do. Is it only because of that theory digital audio works or is the theory just the best approximation we have to explain a given type of phenomenon... Definitely not to disregard the usefulness of the theory, but just to state the "universal" problem relating to how any theory map onto the world.
Well, if we disagree, then we do so on a fundamental ontological basis, and no argument will probably get us closer to agreement, so that would be a pointless exercise.
However, now that some of you guys seem to know quite a bit (more than me) about digital audio, I would very much like to understand the assumptions behind your assessments. You say that 16 or 20 bits are sufficient and with signal processing offering more bits than this, we can adjust volume with absolutely no loss in (information) or (perceived sound quality), no?
Without trying to prove the opposite, just to state some of my own observations, I perceive a quite big improvement when listening to HDCD (20bit) over CD (16bit). I have also experienced how much the digital volume control on a sonos streeming system destroyed the sound compared to not using the digital volumen control and instead using the passive attenuator on a rather cheap Chinese tube amp. Day and night difference to me (in that case of course). How would these observations relate to your understandings of audibility of digital sound processing?
No need to go into metaphysics. I am not saying the theory is flawless or perfect. It is just that the issues discussed here are such fundamental parts of the Nyquist–Shannon sampling theorem, that if they weren't true, none of our current digital audio systems would work.
I would absolutely not use the word "absolutely". Doing a linear scaling operation (that is what a volume/gain adjustment is) will have a very small effect due to rounding errors from the finite (24-bit) precision. What we are saying is that that effect is lower than the noise contribution of both the DAC (that has a dynamic range of 20 bits) or the source material (that probably has a dynamic range of less than 12 bits). That effect will probably also be much smaller than the effect of an analog attenuator.
Just a simple reminder, the dynamic range / SNR of an undithered digital system is:
16 bits - 96 dB
20 bits - 120 dB
24 bits - 144 dB
I assume you have done an ABX with two versions of exactly the same material, and rules out things like different gain or EQ between the HDCD and CD versions?
It is of course possible for a digital volume control to be badly implemented, just as it is possible for an analog control to be badly implemented.
One of my nCores stopped working this morning. Now it powers up for a split second before shutting down again. I understood that the nCoress had short circuit & overheat protection but whatever made it call FATAL has not cleared by itself. Is there anything I can do to the module to reset it?
- Status
- Not open for further replies.