[Mython]

  I don't use ADC. That would be crazy/idiotic as you pointed out. miniDSP nanoDigi 2x8 B takes a digital coaxial or Toslink signal, ASRC it to 24-bit/96kHz (suboptimal, I know, for DAVE) and then applies digital parametric EQ and then outputs it via coaxial.

Thankyou for clarifying. OK, that makes a lot more sense - evidently I was, indeed overlooking something

 
 

....you can send to 3 Chord DAVEs as your DACs and then you can hook up 3 amplifiers to the 3 Chord DAVEs to playback on the speakers.

 
Sheesh! As it is, I'm patiently sitting on the sidelines, until such time as I can get a decent income and afford one DAVE, and now you're egging me on to buy three?!! LOL

 

[ecwl]

Originally Posted by Rob Watts /img/forum/go_quote.gif
 
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

 
So I really don't understand this well as a consumer. But as you increase the tap length, as you get to an infinite amount of ringing, doesn't the the amplitude of the ringing get smaller and smaller so that the final waveform accurately reflect the impulse response because the ringing is so infinitesimally small that it is essentially a flat line?
 
Because to me, aren't we just talking about Zeno's paradox?
 
Now that I marginally understand this better, the question on my mind is, with the longer tap length and more ringing at ever smaller amplitudes, does the small amplitude ringing affect what we hear? If we are saying that small signal linearity and noise floor modulation matter, would these small amplitude ringing also matter? Since I love Chord DACs, I suspect the answer is no. And this is because as you pointed out the waveforms are approximating the original signal much better than any other DACs out there because of the improved small signal linearity and noise floor modulation and because the increased tap length captures the transients and timing much more accurately than other DACs limited filtering and tap length.

 

[Rob Watts]

   
Don't get me wrong: I'm not worried in the least, since DAVE sounds perfect to me. But does that (the windowing function?) mean a single pulse doesn't appear on the oscillogram? How about rectangles?

OK its pretty simple - I have built in an impulse detector - which is a very unusual signal - and when the FPGA sees the impulse, then the filter is switched out, so it simply returns what is incoming. Its impossible for music to trigger the impulse detector.
 
Any other signal passes through the WTA normally.
 
The windowing function is to do with the mathematics of creating the FIR filter coefficients. Once I have calculated those coefficients, they are fixed into the FPGA.
 
Rob

 

[JaZZ]

 

  Don't get me wrong: I'm not worried in the least, since DAVE sounds perfect to me. But does that (the windowing function?) mean a single pulse doesn't appear on the oscillogram? How about rectangles?

OK its pretty simple - I have built in an impulse detector - which is a very unusual signal - and when the FPGA sees the impulse, then the filter is switched out, so it simply returns what is incoming. Its impossible for music to trigger the impulse detector.
 
Any other signal passes through the WTA normally.
 
The windowing function is to do with the mathematics of creating the FIR filter coefficients. Once I have calculated those coefficients, they are fixed into the FPGA.
 
Rob

 
Thanks, Rob, that clears it up.

An impulse detector! What sort of ringing would DAVE show with a square wave, then? (Let's say at a sampling rate of 44.1 kHz.)

 

[ecwl]

Another thing I'm wondering about now is why other DAC designers did not aim for longer tap lengths in their filter but focused more on ringing. If the long tap length filters are going to generate some very small amplitude ringing, maybe the DAC design would have to be able to handle these small amplitude signals without messing with the rest of the signal. If a DAC fundamentally cannot handle small signal linearity and noise floor modulation, wouldn't having more and more small amplitude ringinging deteriorate the sound further? Meaning that if you're going to design ladder DACs or multi-bit sigma-delta modulator chip DACs, these small amplitude ringing is going to exacerbate your small signal non-linearity and mess up with jitter and increase noise floor modulation. So as a result, you would never choose any filters with a long tap length. But if you have a great DAC design like Pulse Array where you have outstanding low-level linearity and no noise floor modulation, you can keep on increasing the tap length to get as close to the original transients and timing as possible without worrying about the non-consequential ringing...

 

[JaZZ]

 

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 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

 
I think I understand it. Perfect low-pass filtering means infinite Q means infinite ringing. Infinite Q means that just the resonance frequency itself is affected by the ringing, every frequency below is perfectly reproduced in the time domain (although that's impossible to measure). Any (real-world) low-pass filter with a finite Q factor will introduce transient corruption below the filter frequency to some degree. Right?
 
So DAVE approaches an infinite Q factor? I've seen measuring data indicating –0.04 dB at 20 kHz, which is pretty good.
 
However, in the real (non-DAVE) world avoiding pre-ringing may be beneficial, at least the corresponding filter 3 on my Corda Symphony sounded special to my ears (sometimes I considered it the best).

 

[STR-1]

Hi all. I would be grateful for some advice on the advantages of using a server (if that's the right word) with the Dave. My system is currently AK380/Dave/HE1000 and most of my files are 16/44 ALAC rips, although I also have a few higher resolution PCM files (I've nothing against DSD and happy to build up a collection of those files as well if I can make full use of them). I'm not sure the AK380 is sending through the best quality signal (each firmware upgrade seems to alter the sound) and it's certainly not the most convenient way of choosing albums and tracks on the fly to play over a long listening session. I'd like a permanently plugged in storage and play option that I can control easily from my iPad, and most importantly an option that sends the highest quality signal from my music files to the Dave. I don't really know this part of the market but I have read the odd review on one or other of the Aurender servers and the CAD CAT, which seemed very complimentary but those were very expensive bits of kit. I am hoping that in addition to the extra convenience of getting a server I can also achieve a noticeable hike in the quality of sound coming out of the Dave. Thanks in advance for any advice or recommendations.

 

[Crashem]

Rob watts seems feels Dave is well isolated from source and therefore unclear what improvement you will see. However in my experience a good source (i.e. Server) makes a difference. I think you have two choices:

1) network storage and use something like auralic aries/au render: advantage is have specialized audio device attached to dac and leave computer related noise elsewhere. Also typically get more flexibility in terms of location. Negative is that you lose capability of computer for things like up sampling, capabilities of media server, and more complications around setup.

2) direct connect of computer: in this case, I would look at CAPS server which is build you can find on computeraudiophile.com. If you can't build, I can recommend some stores who can build one to order (pm me). Get an audiophile usb card though. Go linear power supply if you like as well.

 

[yellowblue]

As Romaz namned earlier you can use the Sonic Orbiter SE with Roon. Works really well even for me. If you don´t mind to have an extra core-computer running that could be a good solution - as good sounding as Aurender etc. You can choose a good LPS (or Anker-battery) and good cables if you want to try out if that affects the sound.

 

[lovethatsound]

Hi all. I would be grateful for some advice on the advantages of using a server (if that's the right word) with the Dave. My system is currently AK380/Dave/HE1000 and most of my files are 16/44 ALAC rips, although I also have a few higher resolution PCM files (I've nothing against DSD and happy to build up a collection of those files as well if I can make full use of them). I'm not sure the AK380 is sending through the best quality signal (each firmware upgrade seems to alter the sound) and it's certainly not the most convenient way of choosing albums and tracks on the fly to play over a long listening session. I'd like a permanently plugged in storage and play option that I can control easily from my iPad, and most importantly an option that sends the highest quality signal from my music files to the Dave. I don't really know this part of the market but I have read the odd review on one or other of the Aurender servers and the CAD CAT, which seemed very complimentary but those were very expensive bits of kit. I am hoping that in addition to the extra convenience of getting a server I can also achieve a noticeable hike in the quality of sound coming out of the Dave. Thanks in advance for any advice or recommendations.

Hi STR-1
Have a look at the bluesound range,the bluesound vault 2 might be what your looking for,it's got a 2tb hard drive,it can rip cds,download hi res,and stream,you can do it all from your i pad,its fairly cheap, about £1000

 

[ecwl]

Hi STR-1
Have a look at the bluesound range,the bluesound vault 2 might be what your looking for,it's got a 2tb hard drive,it can rip cds,download hi res,and stream,you can do it all from your i pad,its fairly cheap, about £1000

 
Except you can't connect the Bluesound Vault 2 via USB to Chord DAVE. But then you can just use Toslink. I looked into this before because my friend who will inherit my QBD76HDSD already owns a Bluesound Vault 2. He was also going to inherit a Toslink cable from me but I'll have to give him a digital coaxial RCA-BNC because I end up using that Toslink cable to isolate my video system mostly from my audio system. For those who cared, I'm using the Toslink between the Oppo and the miniDSP. I'm amazed by the increased depth and smoothness and timbre I get from watching movies. Force Awakens Lightsaber fight scene had these subtle sounds coming from the lightsabers that I never heard before. I'm actually surprised Chord has not built an A/V processor based on Mojo/DAVE technology. 

 

[Christer]

Meanwhile over at computer audiophile the plot  seems to thicken a bit comparing DAVE and  the T+A DSD8.
The guy behind HQ player claims that most of what rob Watts has said about DSD is  either "********" or  "complete ********".He also says that Mojo benefits from external upsampling.
Claims and counterclaims!
 
All I know without having auditioned the T+A is that DAVE  is the best DAC I have heard with DXD.
 
But more voices over at CA seem to be of the opinion that what DAVE actually  does so well could be done even better and a lot cheaper with a powerful i 7 computer.
And over here Rob Watts claims the industry has got its head in the sand.
 
Could anybody with more technical knowledge than me comment?
I have simple taste: I want the best! But if I can save some money and still get the best it would not hurt me at all.

 

[Crashem]

Honestly, I am itching to hear the DAVE's DSD. Best DSD I have heard is out of Lampizator which is completely different approach. Rob Watts has never been much of DSD proponent (understatement there). With my HUGO, it showed. DSD was actually quite bad. Basically looks like DSD is reworked for DAVE so I am sooo curious.

Btw, just saw post above. Doesn't surprise me hqplayer guy saying what he is saying about rob watts. Rob watts could be seen as solidly on the PCM side in DSD/PCM debate and miska is definitely a DSD proponent.

 

[ecwl]

Yeah. It was getting a little heated over at Computer Audiophile T+A DAC8 DSD forum. Miska, creator of HQPlayer is obviously very passionate about his work, just like Rob Watts is about his own. Miska's take is that the PC can do most of the work with HQPlayer, including the upsampling/filtering and of course with much longer tap length. He also thinks the PC can do the noise shaping better. All of this because the PC might have more processing power. And his take seems to be that he can output a DSD256/DSD512 to a good DAC that supports native DSD and he would get the best sound.
 
And you know people love the PCM vs DSD debate. So they dragged some of Rob Watts said about DSD into it. Moreover, they also commented on the Chord DACs. Of course, what Rob Watts said about DSD as a recording format is that the noise shaping is baked into the file and most of the time, the noise shaping is done at up to -120dB. I think this is when Miska got really involved and upset over the whole thing. Miska in turn completely disagrees because of course, if he were taking a PCM file and then converting it into DSD64/DSD128/DSD256/DSD512 he can potentially noise shape the file to a much lower noise floor with a computer CPU/GPU.
 
The other issue was that Miska acknowledges that he is not familiar with the Pulse Array design. But it sounds like his take is that since HQPlayer can upsample/filter with long tap lengths and noise shape with lots of computing power, he can feed the upsampled signal to any DAC and as long as the DAC is good at handling the upsampled signal, you're going to get great sound. And then his personal view is that because simpler is better, he thinks that the simplest highest quality 1-bit sigma delta modulation DACs that can handle DSD256/DSD512 paired with his HQPlayer upsampling would offer the best sound.
 
Miska and another user also commented that with the Chord Mojo, they prefer sending a HQPlayer upsampled 16/44 to 24/704 file via USB into the Chord Mojo, rather than sending the original 16/44 to Chord Mojo. Although the other user later acknowledged that DAVE sounds better with the native file compared to HQPlayer upsampling. I wonder if the issue there might be related to the Chord Mojo USB not being galvanic isolated or if it's true that HQPlayer upsampling is truly superior than Chord Mojo's.
 
My personal take is this and it obviously would not necessarily apply to Christer. First, I have a low-powered headless computer built for my former DAC and a music server system, running Windows 10, JRiver, with JRiver parametric EQ and remote controlled by my iPad via JRemote. Moreover, I can capture Tidal to play in JRiver with the parametric EQ. I would not give that up for HQPlayer. I'm sure others with Aurender/Auralic/Meridian/Bluesound or other music servers are happy with their user interface that they wouldn't want to mess with HQPlayer. Sometimes pragmatism takes over.
 
The other issue is that Miska's upsampling/filtering may provide more CPU/GPU/computational power than Rob Watts's FPGA but there is also the proprietary algorithm of the WTA filter. Rob Watts has said in the past when WTA filter was first developed, he found that WTA filter with 256 taps sounds better than a different more standard filter with much longer tap length. So it's unclear if Miska's filters are always better sounding even with more computational power.
 
More importantly, Rob Watts has discussed the advantages of Pulse Array DACs which standard 1-bit sigma-delta modulation, multi-bit sigma-delta modulation chips and ladder DACs don't have. So even if Miska is right about using the PC and HQPlayer being better than FPGA, you still have to contend with the fact that you'd still want the best Pulse Array DAC which is in Chord DAVE (unless you don't think Pulse Array DAC matters).
 
But as with everything in audio, there are always lots of strong opinions. For me, I'm happy with Chord DAVE and Mojo. I'm happy with Rob Watts's explanations why I'm super happy with Chord DAVE and Mojo so I'm sticking with them.

 

[rkt31]

@ecwl, Hq player is a bit difficult to set. can you please give the link for the best sq setting in Hq player ? i want to compare through mojo , an unaltered CD quality stream via foobar and upsampled stream from Hq player !