Generally speaking, a higher rez recording will sound better than redbook - but its by far not the most important factor - I have plenty of redbook recordings that sound better than DXD recordings for example.
 
I have 4TB of music, and often set it to play random with Classical as a genre. Every so often a recording pops up, it sounds immensely spacious and transparent - then I will look to see what it is, and its just as likely to be a redbook than a high rez. The recording quality with PCM I think is more important than the actual format.
 
Ask me the same question when the Davina project has completed test recordings, I will know for sure how much redbook loses.
 
Rob

A bit off topic I know, but it's not often I get to send a message to someone who has contributed to making my life on this bit of spinning rock a bit more bearable! lol
There are two pieces of audio equipment I have bought that have made music reproduction much more than just 'amplified noise'. My Linn LP12 is one of them, and now the Hugo can join it. The Hugo is completely revolutionary rather then just being evolutionary IMO, and thank-you so much for developing this masterpiece Rob!
Apologies again for being off-topic.

Sent from my E6653 using Tapatalk

rob watts said:

 
Ringing uses an illegal signal from sampling theory POV as it is not bandwidth limited, so you would not actually get a perfect impulse from a perfect legal bandwidth limited ADC. So why worry about a signal you will never get? So it is actually pointless talking about it.

Click to expand...

 
So are the signals out of the ADC perfectly bandwidth limited to prevent ringing from high tap filters in the DAC? How much attenuation should the signal have at FS/2 to be considered bandwidth limited and no ringing to occur? Also, won't long filters in the ADC that would give strong enough attenuation induce their own ringing which will be embedded in the recording?

sunya said:

   
So are the signals out of the ADC perfectly bandwidth limited to prevent ringing from high tap filters in the DAC? How much attenuation should the signal have at FS/2 to be considered bandwidth limited and no ringing to occur? Also, won't long filters in the ADC that would give strong enough attenuation induce their own ringing which will be embedded in the recording?

Click to expand...

This is a very important question, and something I hope Davina will answer.
 
Now for DAC reconstruction or interpolation filters, sampling theory is straightforward. If you use an infinite over sampled infinitely long filter with a sinc impulse response, then you will perfectly recover the un-sampled data - it will be exactly the same if the signal is bandwidth limited. So the closer we get to this ideal filter, the closer we get to the original, and the more transparent the system becomes. Note that such a filter will not add any ringing pre or post to a bandwidth limited impulse, so the whole issue of ringing for a DAC reconstruction filter is a red herring.
 
But sampling theory states it must be bandwidth limited, and that means that the output at and > FS/2 must be zero, otherwise the reconstructed signal will not be identical. But it says absolutely nothing about what type of bandwidth limiting it should be, as sampling theory is only concerned about sampling an existing bandwidth limited signal - its about preserving something that is already bandwidth limited.
 
Now I know aliasing is very important - some people actually use it to soften up a sound - but I am not trying to change it but to keep it transparent as possible. Now I know that 140 dB rejection is not good enough in that it changes the SQ and gives measured artifacts - but how much is good enough? And will the issue of ringing be important? These are questions I want Davina to answer, when I will be able to listen to 768 kHz recordings and hear and measure the effect of the decimation filters.
 
I actually suspect that pre-ringing is not an issue, that only aliasing is important, as after all, nobody can hear 22.05 kHz, and if there is no energy at 22.05 there will be no output either, and in practice the amount of HF energy in audio is very low. Another problem is that out of band noise creates more noise floor modulation (so filtering it out may actually sound better!), so it won't be easy to get to the bottom of this problem.
 
Rob

I have not heard the Hugo, but I now regularly recommend that people get their hands on a Chord Mojo to get a sense of just how good digital sound technology can be.  I feel like it could be part of an audiophile ear training course.  With the exception that I usually think of audiophile equipment as being so overpriced as to be non-attainable.  At $600 US the Mojo is, for those who want it, an achievable ownership target.

rob watts said:

  If you use an infinite over sampled infinitely long filter with a sinc impulse response, then you will perfectly recover the un-sampled data - it will be exactly the same if the signal is bandwidth limited. So the closer we get to this ideal filter, the closer we get to the original, and the more transparent the system becomes. Note that such a filter will not add any ringing pre or post to a bandwidth limited impulse, so the whole issue of ringing for a DAC reconstruction filter is a red herring.

Click to expand...

 
Does this apply to any implementation? In my understanding a low-pass filter with a quite round fall-off characteristic (thus with a low Q factor) will produce a rather large-band ringing, far from the ideal of a filter with infinite sharpness, which exclusively rings exactly at the filter frequency (in return infinitely).
 
How much, in your view, is the audio band affected by delayed decay from the finite filter sharpness in common DACs and CD players? Isn't exactly this discrepancy between ideal and real-world reconstruction filter (among the other mentioned issues) responsible for the effort you had to invest for Hugo and finally DAVE? («So the closer we get to this ideal filter, the closer we get to the original, and the more transparent the system becomes.»)
 
Like Sunya, I'm also concerned about the bandwidth limitation necessary before A/D conversion, as it introduces the same (still underestimated) flaws. Hopefully Davina will make for recordings that sound even more natural in the future.

jazz said:

   
Does this apply to any implementation? In my understanding a low-pass filter with a quite round fall-off characteristic (thus with a low Q factor) will produce a rather large-band ringing, far from the ideal of a filter with infinite sharpness, which exclusively rings exactly at the filter frequency (in return infinitely).
 
How much, in your view, is the audio band affected by delayed decay from the finite filter sharpness in common DACs and CD players? Isn't exactly this discrepancy between ideal and real-world reconstruction filter (among the other mentioned issues) responsible for the effort you had to invest for Hugo and finally DAVE? («So the closer we get to this ideal filter, the closer we get to the original, and the more transparent the system becomes.»)
 
Like Sunya, I'm also concerned about the bandwidth limitation necessary before A/D conversion, as it introduces the same (still underestimated) flaws. Hopefully Davina will make for recordings that sound even more natural in the future.

Click to expand...

 
I suppose the point I was getting across that for a DAC interpolation or reconstruction filter, I have never seen the ringing performance being a significant factor; by this I mean there were always other explanations for the differences in SQ I can hear whilst optimizing the SQ during listening tests on the WTA algorithm. In broad terms, getting the filter to be close to the ideal sinc response is what you are doing - but when optimizing the sound quality sometimes that will mean more ringing, sometimes less ringing - its just not a relevant factor. The reason for this is that an ideal filter will ring infinitely, but will recover the un-sampled bandwidth limited signal absolutely perfectly with no change to the signals ringing at all. If we want to evaluate a filters performance using ringing, then we need to use a bandwidth limited impulse response test signal. So the ringing performance using an illegal (from sampling theory) impulse response has no bearing on the SQ - except that the more ringing, then perhaps the closer it becomes to the ideal sinc filter. Certainly going for a filter that does not ring is a very bad idea - the ringing is absolutely essential as it is the multiple rings from all the past and future samples that when added up gives the correct transient values.
 
Just to give you an idea - I have designed a 8FS WTA filter, and a 16FS WTA filter with identical ringing performance - except the 16FS filter has of course double the output resolution - and when you see the two impulse responses after they have been further filtered to 2048 FS  they look identical - but the two filters actually sound very different.
 
So the filter for the interpolation filter has to be a certain way with lots of ringing, if you want to recover the signal perfectly. But for the ADC the only requirement is that it is perfectly bandwidth limited, theory does not care how it is done. So I plan to just try different filters to define the aliasing problem (which I know for certain from listening tests is a major SQ issue) and then try differing ringing responses to see which type of filter sounds best. My current thinking is that ringing in an ADC will turn out to be unimportant; we just need it to be bandwidth limited by enough to eliminate aliasing, but I too am worried about the effect of 22 kHz bandwidth limiting for redbook. 
 
But until you do the work, and do lots of careful listening tests, you know nothing...  
 
Rob 

rob watts said:

 

jazz said:

   
Does this apply to any implementation? In my understanding a low-pass filter with a quite round fall-off characteristic (thus with a low Q factor) will produce a rather large-band ringing, far from the ideal of a filter with infinite sharpness, which exclusively rings exactly at the filter frequency (in return infinitely).
 
How much, in your view, is the audio band affected by delayed decay from the finite filter sharpness in common DACs and CD players? Isn't exactly this discrepancy between ideal and real-world reconstruction filter (among the other mentioned issues) responsible for the effort you had to invest for Hugo and finally DAVE? («So the closer we get to this ideal filter, the closer we get to the original, and the more transparent the system becomes.»)
 
Like Sunya, I'm also concerned about the bandwidth limitation necessary before A/D conversion, as it introduces the same (still underestimated) flaws. Hopefully Davina will make for recordings that sound even more natural in the future.

Click to expand...

 
I suppose the point I was getting across that for a DAC interpolation or reconstruction filter, I have never seen the ringing performance being a significant factor; by this I mean there were always other explanations for the differences in SQ I can hear whilst optimizing the SQ during listening tests on the WTA algorithm. In broad terms, getting the filter to be close to the ideal sinc response is what you are doing - but when optimizing the sound quality sometimes that will mean more ringing, sometimes less ringing - its just not a relevant factor. The reason for this is that an ideal filter will ring infinitely, but will recover the un-sampled bandwidth limited signal absolutely perfectly with no change to the signals ringing at all. If we want to evaluate a filters performance using ringing, then we need to use a bandwidth limited impulse response test signal. So the ringing performance using an illegal (from sampling theory) impulse response has no bearing on the SQ - except that the more ringing, then perhaps the closer it becomes to the ideal sinc filter. Certainly going for a filter that does not ring is a very bad idea - the ringing is absolutely essential as it is the multiple rings from all the past and future samples that when added up gives the correct transient values.
 
Just to give you an idea - I have designed a 8FS WTA filter, and a 16FS WTA filter with identical ringing performance - except the 16FS filter has of course double the output resolution - and when you see the two impulse responses after they have been further filtered to 2048 FS  they look identical - but the two filters actually sound very different.
 
So the filter for the interpolation filter has to be a certain way with lots of ringing, if you want to recover the signal perfectly. But for the ADC the only requirement is that it is perfectly bandwidth limited, theory does not care how it is done. So I plan to just try different filters to define the aliasing problem (which I know for certain from listening tests is a major SQ issue) and then try differing ringing responses to see which type of filter sounds best. My current thinking is that ringing in an ADC will turn out to be unimportant; we just need it to be bandwidth limited by enough to eliminate aliasing, but I too am worried about the effect of 22 kHz bandwidth limiting for redbook. 
 
But until you do the work, and do lots of careful listening tests, you know nothing...  
 
Rob 

Click to expand...

 
Thank you for the explanations! I'm aware that measuring «ringing» effects reaching down into the audio band is problematic due to the fact that any low-pass filtering of the test signal logically prevents sharp start and end points in any event, so to discover particularly strong or audibly relevant delay of decay from a specific low-pass filter is difficult or impossible. Maybe it can be mathematically deduced? Or talking of ringing effects in this context sounds a bit absurd, as the issue is a perfect reconstruction specifically also addressing the time domain, which only an ideal, infinitely sharp filter provides. I'm just interested in the exact mechanism that's responsible for the achieved sonic perfection – and isn't this effectively the better transient behavior in the audio band? So one could say you concentrate the whole filter ringing to the filter frequency (where it rings infinitely then) and take it away from audible frequencies.

just curious , did chord get an offer for dave for making vinyl master of modern native digital recording s ?

I was thinking about analog preamplifiers and the degree of transparency and distortion. I am beginning to wonder if the Davina and DAVE combo as a digital preamplifier would in most cases best most (if not all) analog preamplifiers. Theoretically it doesn't seem to make sense as you're going through ADC to DAC and then digital volume control. However, I was thinking that in reality, you're more likely to get more signal degradation and distortion through analog gain and analog volume control. I am guessing this is particularly true in terms of small-signal linearity and distortion.
 
I think nowadays, most people who need a preamplifier are ones who still uses turntables with their phono preamplifier. I wonder if going through Davina's ADC and then handling gain digitally into DAVE would sound better in most cases than an analog preamplifier. Of course, that's assuming people actually playback their turntable live. If they're converting their LPs in Davina's ADC first (without actually playing the music out loud) and then later playing back the album off a computer on DAVE later, the ADC recording of the LP would actually also eliminate any vibration-associated distortions associated with turntable playback, making the LP sound even better. Of course, even if someone wants to play the LP live, and if you move the turntable into another room to avoid vibration-associated distortions, running a long digital cable between Davina to the DAVE is probably going to have lower distortion than running a long analog stereo interconnect from the turntable to the preamplifier in the listening room. Obviously, this is just a thought experiment. I guess we will have to wait till Davina to be released as a product to find out...

ecwl said:

  I was thinking about analog preamplifiers and the degree of transparency and distortion. I am beginning to wonder if the Davina and DAVE combo as a digital preamplifier would in most cases best most (if not all) analog preamplifiers. Theoretically it doesn't seem to make sense as you're going through ADC to DAC and then digital volume control. However, I was thinking that in reality, you're more likely to get more signal degradation and distortion through analog gain and analog volume control. I am guessing this is particularly true in terms of small-signal linearity and distortion.
 
I think nowadays, most people who need a preamplifier are ones who still uses turntables with their phono preamplifier. I wonder if going through Davina's ADC and then handling gain digitally into DAVE would sound better in most cases than an analog preamplifier. Of course, that's assuming people actually playback their turntable live. If they're converting their LPs in Davina's ADC first (without actually playing the music out loud) and then later playing back the album off a computer on DAVE later, the ADC recording of the LP would actually also eliminate any vibration-associated distortions associated with turntable playback, making the LP sound even better. Of course, even if someone wants to play the LP live, and if you move the turntable into another room to avoid vibration-associated distortions, running a long digital cable between Davina to the DAVE is probably going to have lower distortion than running a long analog stereo interconnect from the turntable to the preamplifier in the listening room. Obviously, this is just a thought experiment. I guess we will have to wait till Davina to be released as a product to find out...

Click to expand...

 
My first reaction to this was wow, no way. But then I thought a bit more, and sure THD and noise plus noise floor modulation will be better than a conventional pre-amp - plus we will not have the small signal non-linearity of the analogue volume control, as you rightly say, which will improve depth. In 768 kHz mode, I think it could be more transparent than an analogue pre-amp - its actually one of the tests I have in mind. If I can get 44.1 kHz to be as transparent (via Dave) as a high-end pre-amp, then that would indeed represent an enormous improvement. Most recording engineers accept that putting a signal through an ADC than back to DAC represents a big loss in transparency, so doing that without a big degradation would represent a major advance.
 
One of the headaches I am wrestling with today is the potential loss in going from 104 MHz (that's the actual rate the ADC works at) to 768 kHz. I have fitted a massive FPGA to the prototype, so I can actually do this decimation with an FIR WTA filter - not the usual moving average filter, so I have been designing that filter. I plan to actually build two prototypes, then feed each one with the same microphone signal, then make two recordings at 768 kHz, so I can experiment with differing settings to see where the issues are. It will be possible to record at 6.144 MHz 32 bit too (or perhaps at an adjusted direct ADC rate), then compare that to 768 kHz. I strongly suspect that 768 kHz can be made perfectly transparent, but unless you compare it to something better, you don't know for certain.
 
Anyway, the schematics will be finished tomorrow, so one milestone passed.
    
Rob

Really longing for your ADC, Rob.

mtoc said:

  Really longing for your ADC, Rob.

Click to expand...

 
 

rob watts said:

  Soon I will be talking about the Davina ADC - schematics are almost finished, and board layout is about 75% done now. I plan to do a blog post once Gerber files are released to Chord.
 
Here is a screen shot of the layout so far:
 

 
The three DIP switches (bottom left) will allow me to switch on/off all 11 analogue integrators.
 
Rob

Click to expand...

My dream is using Davina as an audio analyzer....  

Mr Watts

Have you thinking about the possibility to include in the Davina the necessary pre-amp to work with a Dynetic Phono Cartridges or a moving coïl?
For me, it would be a hit in the hifi world.
I am dreaming to use my 30 years old cartridge with my Davina and my Dave :
http://cdn.shure.com/user_guide/upload/1817/us_pro_v15iv_ug.pdf
 
Jean