JaZZ
Headphoneus Supremus
Watts has also programmed his WTA filter to ignore test impulse patterns so it’s impossible to ‘see’ its pre/post ringing behaviour in the time domain...
@Rob Watts: What's the reason for this?
Watts has also programmed his WTA filter to ignore test impulse patterns so it’s impossible to ‘see’ its pre/post ringing behaviour in the time domain...
I realize sometimes there's understanding something and then there's really understanding something. I've always known from what Rob Watts has said in this column that in order to allow the S/PDIF coaxial/BNC input of DAVE to accept 384kHz/24-bit signals, they are not galvanically isolated. I have a USB source and Toslink source. But I also needed room correction/parametric EQ for my video system so I use a miniDSP product to implement the DSP and connect it's coaxial output to Chord DAVE. I said to myself, even if there is some RF/ground noise getting into the DAVE when I'm playing music, I should just ignore it because I'm not going to hook and unhook the BNC cable every time I want to listen to music. Besides the miniDSP product already has isolation transformer on the output.
My dealer and I were talking about cables and so I changed some in my video system but I wasn't ready to test them yet so I went back to listening to music. I couldn't understand why there's improvement to the sound. So I put back the original cables and then just plug in and unplug the BNC connection to the DAVE, something I told myself I would never bother to do for convenience. Yup, that was what made the difference.
I thought I would share in case people are still using suboptimal BNC inputs into DAVE. If you can, switch over to Toslink. I'm guessing if you have good sources like Chord CD players, it probably doesn't matter. But if like me, your other sources aren't the greatest, you may want to not use the BNC inputs. Or unplug them for optimal musical enjoyment.
Can you use USB input to Dave within the realm of your video/audio setup? Shouldn't that be better sounding than optical?
DAVE HiFi News Review! http://bit.ly/1U0mG1d
"Especially with archive recordings, the switch from PCM+ to DSD+ mode pays dividends in the sharpening-up of the sonic picture and instrumental timbres alike, and freeing just a shade more of that considerable dynamic ability."
Hmmm... well, that's controversial, if nothing else!
Watts has also programmed his WTA filter to ignore test impulse patterns so it’s impossible to ‘see’ its pre/post ringing behaviour in the time domain...
@Rob Watts: What's the reason for this?
Quote:
Watts has also programmed his WTA filter to ignore test impulse patterns so it’s impossible to ‘see’ its pre/post ringing behaviour in the time domain...
@Rob Watts: What's the reason for this?
The WTA algorithm actually uses my own windowing function of the ideal sinc impulse response. I took an awful lot of time (man years) with this, both in listening tests and in trying to understand what was going on - the understanding being used to allow me to try listening to different things. At the end of the day, the algorithm is fine tuned by listening tests, but you need understanding in order to change the critical parameters - in short knowing what those parameters are. Anyway, with the blu CD player, using an impulse response you will get all the coefficients to a 24 bit accuracy, so it would be easy to reverse engineer the WTA algorithm.
Frankly I should not have worried. This industry has its collective head in the sand. I have been publically talking about the importance of very long tap length filters for 17 years and still nobody else bothers about it.
Rob
The WTA algorithm actually uses my own windowing function of the ideal sinc impulse response. I took an awful lot of time (man years) with this, both in listening tests and in trying to understand what was going on - the understanding being used to allow me to try listening to different things. At the end of the day, the algorithm is fine tuned by listening tests, but you need understanding in order to change the critical parameters - in short knowing what those parameters are. Anyway, with the blu CD player, using an impulse response you will get all the coefficients to a 24 bit accuracy, so it would be easy to reverse engineer the WTA algorithm.
Frankly I should not have worried. This industry has its collective head in the sand. I have been publically talking about the importance of very long tap length filters for 17 years and still nobody else bothers about it.
Rob
I realize sometimes there's understanding something and then there's really understanding something. I've always known from what Rob Watts has said in this column that in order to allow the S/PDIF coaxial/BNC input of DAVE to accept 384kHz/24-bit signals, they are not galvanically isolated. I have a USB source and Toslink source. But I also needed room correction/parametric EQ for my video system so I use a miniDSP product to implement the DSP and connect it's coaxial output to Chord DAVE. I said to myself, even if there is some RF/ground noise getting into the DAVE when I'm playing music, I should just ignore it because I'm not going to hook and unhook the BNC cable every time I want to listen to music. Besides the miniDSP product already has isolation transformer on the output.
My dealer and I were talking about cables and so I changed some in my video system but I wasn't ready to test them yet so I went back to listening to music. I couldn't understand why there's improvement to the sound. So I put back the original cables and then just plug in and unplug the BNC connection to the DAVE, something I told myself I would never bother to do for convenience. Yup, that was what made the difference.
I thought I would share in case people are still using suboptimal BNC inputs into DAVE. If you can, switch over to Toslink. I'm guessing if you have good sources like Chord CD players, it probably doesn't matter. But if like me, your other sources aren't the greatest, you may want to not use the BNC inputs. Or unplug them for optimal musical enjoyment.
I don't know much but does that mean that there is no pre or post ringing in Dave due to high tap length and wta algorithm ?
I am well aware of the miniDSP product (and similar products), although I admit I have not personally heard them.
My reason for being aware of the miniDSP is because of my interest in Siegfried Linkwitz's LX521.3 dipole loudspeaker (now LX521.4 in digital-only X-over form)
However, as you will note from my earlier post (linked above), I would prefer to build the LX521.3 with 'analogue active crossover', rather than employing miniDSP for crossover purposes.
My concern is that the miniDSP, and others of the same ilk, introduce an ADC->DAC stage into the playback chain, over and above the existing ADC (recording of the original performance) and hi-fi DAC.
I can't imagine shelling-out for a high-end DAC (particularly one as accomplished as DAVE) and then introducing an ADC->DAC stage, with vanilla Cirrus DAC chips, or similar, (generally running at 24/96 in this application). On a theoretical level, at least, it seems to me to be unnecessarily undermining the purity of the end-result.
Am I overlooking something?
I actually posed this issue as a question on another thread, but did not get an answer.
The ideal response is a sinc impulse response, this means it will have an infinite amount of pre and post ringing. But a filter that has this response will perfectly reconstruct a bandwidth limited signal absolutely perfectly with no difference whatsoever, its just displaced in time.
So we have a paradox - the filter that has the most ringing will re-create the signal perfectly with no added ringing whatsoever - but clearly having an infinite amount of ringing is not reproducing the input signal (an impulse) perfectly. How do we explain this contradiction?
This is an incredibly important question as virtually the whole audio industry talks about the importance of no pre-ringing - but they have all got it completely and utterly wrong.
The answer to the conundrum is that an impulse response is an illegal signal - it is not bandwidth limited as it has the same energy at FS/2 as at DC, being a completely flat frequency response - that's why the signal is used for frequency response measurements. But sampling theory absolutely requires bandwidth limited signals - that means at exactly FS/2 the signal level is zero. Indeed, in a properly designed ADC, there will be negligible output at FS/2, so an impulse will never be presented to a DAC using a music file.
So if you use an conventional illegal impulse response signal then the best filter will have the worst ringing; but using music, or a bandwidth limited impulse response, it will actually have the least possible difference from the original continuous analogue signal that was in the ADC - and of course will sound very much closer to the original analogue signal before it was sampled.
Rob