1. This site uses cookies to help personalise content, tailor your experience and to keep you logged in if you register.
    By continuing to use this site, you are consenting to our use of cookies.

    Dismiss Notice

CHORD ELECTRONICS DAVE

Discussion in 'High-end Audio Forum' started by magiccabbage, May 14, 2015.
Tags:
First
 
Back
1 2 3 4 5 6 7
9 10 11 12 13 14 15 16 17 18
Next
 
Last
  1. agooh
    I can't imagine Dave will replace my berkeley reference dac , if watts listen to my berkeley reference he will increase the taps to 400 K to beat it .
    Still I can't wait to hear it .
     
  2. AFWannabe

    Thanks for the link. It is actually from a countryman journalist!
     
    So basically, if I understood correctly, the DSD input is converted to PCM (multi-bit) but not decimated, so its original sampling rate is not reduced, which is good, but it is still submitted to the WTA filter... So I guess it is not pure DSD, but definitely a progress from the Hugo.
    I'm curious to hear it and assess if it sounds like DSD on ESS Sabre based DACs.
     
  3. Rob Watts
    Yes absolutely.
     
    When we get closer to full production, I will be talking more about Dave, as there is a lot more to it than 164k taps!
     
    Rob 
     
    Chord Electronics Stay updated on Chord Electronics at their sponsor page on Head-Fi.
     
    https://www.facebook.com/chordelectronics https://twitter.com/chordaudio http://www.chordelectronics.co.uk/
  4. goobicii
    how strong is that class d headphone amp/pre amp in DAVE?     I want to drive Hifiman HE-1000,can it handle it in way similiar to top SS amps like GS-x mk2... Bakoon etc
     
  5. Rob Watts
    That's pretty complicated to explain, but in simple terms its about resolution first, then less jitter sensitivity, lower distortion and noise.
     
    Subjectively the resolution gives better depth perception, and lower THD and noise gives smoother sound.
     
    I will be talking in more detail closer to the production date.
     
    Rob 
     
    Chord Electronics Stay updated on Chord Electronics at their sponsor page on Head-Fi.
     
    https://www.facebook.com/chordelectronics https://twitter.com/chordaudio http://www.chordelectronics.co.uk/
  6. Rob Watts
    Its absolutely not class D - like Hugo, it has a single stage amplifier giving the OP and headphone drive together. The output is 6v RMS, so a bit more voltage than Hugo, but the same 0.5A RMS current.
     
    Rob
     
    Chord Electronics Stay updated on Chord Electronics at their sponsor page on Head-Fi.
     
    https://www.facebook.com/chordelectronics https://twitter.com/chordaudio http://www.chordelectronics.co.uk/
  7. agooh
    will license this technology out will reduce the prices !! if so it will be amazing to enjoy chord products at fair price . As I can see it's very promising technology .
     
  8. Kakki
    Thank you Rob, I will try to wait patiently... the day you will reveal more details of the secrets in Dave!!
     
  9. CoLdAsSauLt
    I'm by far no engineer, rather in the well-known first stages where you try to wrap your head around the bulk of information and try not to drown.
    Reading this thread, it sparked my technical curiosity however, as I thought I was seeing some parallels in how the Chord DAVE is treating the digital information and how Schiit's Yggdrasil is approaching the matter. Most obvious is that both don't use a "classic DAC-chip".  Anyone here that has more knowledge than me who can chime in and shed some light on this comparison? I know that there might not be enough details on the DAVE yet for that purpose (except for Rob himself then), but am I right to think it's approach is built upon the Hugo's approach? So the principles laid out by DAVE's predecessors might well be comparable?
    Schiit makes a lot of noise for instance about other DACs throwing away the original samples. How will DAVE handle this? What does HUGO do actually?. 
    Boldly spoken, what does the extra cost of the DAVE (whatever that might be in the end) over the Yggdrasil buy me in technical advancements or "better ways of doing A or B"? (Alternatively,if this question is too premature: Why should I buy a Hugo TT instead of an Yggdrasil? No intentions to hijack this thread, I'm just trying to understand the different approaches of the "Chord way" vs the "Schiit way", and learning the basic principles already established by Chord in order to understand the potential improvements that DAVE brings upon these existing techniques)
    I know, in the end the sound is what matters... and that's something we can't compare just yet. I'm nevertheless venting my curious mind :)
     
    Thanks in advance for your insights!
     
    Stijn
     
  10. Whazzzup

    Who gives a schiit, about schiit.
     
  11. LFC_SL
    Have read favourable reports about the yggdrasil over the Hugo TT, so a comparison vs DAVE should interest any audio enthusiast
     
  12. Whazzzup
    Maybe if I get Dave I'll post a comparison in the schiit thread.:p
     
  13. dallan Contributor


    I liked the TT much better when I compared them moments from each other at Canjam. Not even close, and I was hoping it would be.
     
  14. Rob Watts
     
    Hugo and Dave don't use any kind of DAC chip, the analogue conversion is discrete using pulse array. The key benefit of pulse array - something I have not seen any other DAC technology achieve at all - is an analogue type distortion characteristic. By this I mean, as the signal gets smaller, the distortion gets smaller too. Indeed, I have posted before about Hugo's small signal performance - once you get to below -20 dBFS distortion disappears - no enharmonic, no harmonic distortion, and no noise floor modulation as the signal gets smaller. With Dave, it has even more remarkable performance - a noise floor that is measured at -180dB and is completely unchanged from 2.5v RMS output to no signal at all. And the benefit of an analogue character? Much smoother and more natural sound quality, with much better instrument separation and focus. Of course, some people like the sound of digital hardness - the aggression gets superficially confused with detail resolution - but it quickly tires with listening fatigue, and poor timbre variation, as all instruments sound hard, etched and up front. But if you like that sound, then fine, but its not for me.
     
    On the digital filter front - original samples getting modified - actually the vast majority of FIR digital filters retain untouched the original samples, as they are known as half band filters. In this case, the coefficients are arranged so that one set is zero with one coefficient being 1, so the original sample is returned unchanged. The other set being used to create the new interpolated value. The key benefit of half band filters is that the computation is much easier, as nearly half the coefficients are zero, plus the filter can be folded so that the number of multiplications is a quarter of a non half band filter. When designing an audio DAC ASIC, the key part in terms of gate count is the multiplier, so reducing this gives a substantial improvement in die size, and hence cost. So traditional digital filters use a cascade of half band filters, each half band filter doubles up the oversampling - so a cascade of 3 half band filters will give you an 8 times over-sampled signal, with one sample being the unmodified original data. You can tell if the filter is like this as at FS/2 (22.05 kHz for CD) the attenuation is -6dB. The filters that are not like this are so called apodising filters, and my filter the WTA filter.
     
    Going back eighteen years ago to the late 90's I was developing my own FIR filter using FPGA's. Initially, I was interested in increasing the FIR filter tap length as I knew from the mathematics of sampling theory that timing errors were reduced with increasing tap length. So the first test was to use half band Kaiser filters - going from 256 taps to 2048 taps gave an enormous sound quality improvement, so I had confirmed that tap length was indeed important subjectively. But at this point I was stuck; I knew that an infinite tap length filter with a sinc impulse response would return the original un-sampled signal perfectly - but the sinc function using only 16 bit accurate coefficients needs 1M tap FIR filter - and that would never happen, certainly not with 90's technology. So was it possible to improve the timing accuracy without using impossible tap lengths? After a lot of thinking and research, I thought there was a way - but it meant using a non half band filter, which would mean that the original sampled data would be modified. This was a big intellectual stumbling block - how can changing the original data be a good thing? But the trouble with audio is that neat simplistic ideas or preconceptions get in the way. Reality is always different, and reality can only be evaluated by a careful AB listening test. So I went ahead on this idea, and listened to the first WTA filter algorithm - and indeed it made a massive improvement in SQ - a 256 tap WTA sounded much better than 2048 tap half band Kaiser, even though the data is being modified. Why is this? The job of a DAC is NOT to reproduce the data it is given, but to reproduce the analogue signal before it is sampled. The WTA filter reconstructs the timing of the original transients much more accurately than using half band filters or filters that preserve the original data and it is timing of transients that is the most important SQ aspect.
     
    So the moral of the tale? Don't let a simplistic technical story get in the way of enjoying music!                
      
     Rob
     
    Chord Electronics Stay updated on Chord Electronics at their sponsor page on Head-Fi.
     
    https://www.facebook.com/chordelectronics https://twitter.com/chordaudio http://www.chordelectronics.co.uk/
    JaZZ and rocky500 like this.
  15. CoLdAsSauLt

    Thank you Rob for your fast, lengthy and very clarifying response!

    Are there disadvantages to the less computation intensive half band filtering and the folding compared to what I'll call "traditional full-band filtering"?

    Am I right to assume your theoretical end-game DAC uses half band (full band?) filters with +1M taps? Or will your WTA filter still be a better approximation/reproduction of the analog wave due to the taps not being infinite? If I understand you well, this is where we walk into the ceiling of what is possible with 16bit coefficients - so more (or infinite) taps stop making a difference due to this limitation. Do these 16bits refer to the recording (so for 24bit high-rez recordings these numbers would be higher) or to the filter / processor architecture?

    I'm looking forward to your +1M taps DAC somewhere in the future :) As computational power keeps rising, we might actually be able to pull it off. Or am I dreaming your dream too loudly now?
     
First
 
Back
1 2 3 4 5 6 7
9 10 11 12 13 14 15 16 17 18
Next
 
Last

Share This Page