[514077]

Rob watts got his own thread to talk about more technical aspects and new projects coming up,I don't know anyone in hi-fi that has such commitment to Chords customers,what a lovely bloke

I was thinking about you. Have you thought of changing your user name to Qutest?

 

[paul2qute]

I was thinking about you. Have you thought of changing your user name to Qutest?

I was thinking about it but I've looked in the mirror and I'm 45 going on 90 so think I would be kidding myself calling myself cutest,sorry meant qutest

 

[Jawed]

The kind of test that you are talking about is under development, in that I will be able to put a measurement number on the transient error from the decimation > reconstruction process. This will be useful to explore the actual errors involved in reconstructing transients from sampling; and I hope it will provide correlation between measurements and observed sound quality.

Couldn't this test be done entirely in software on a computer? If you generate a guaranteed bandwidth-limited synthetic signal (e.g. sum of lots of sine waves each varying in amplitude) you can sample this "perfectly" at an arbitrary rate (e.g. 1/16th of the native sample rate of the synthetic) and arbitrary bit-depth.

Then you can upsample with your test WTAs of varying lengths so that you can measure the timing errors by comparing with the original synthetic signal.

Anyway, I made a breakthrough and discovered the source of the residual distortion and fixed it, so harmonics were all below -150dB, and higher order harmonics were some 30 dB better - a huge change in measured performance in one stroke. So then I listened to it and was very surprised - the difference was easily audible, and not what I expected - it sounded much more transparent with better detail resolution. Moreover, subsequently with other projects I have seen similar things - when higher order harmonics are reduced it sounds brighter with better transparency - and that is odd, as one would expect it to sound warmer with the removal of high order harmonics.

Could these perceived differences represent an unintended consequence? We've recently been learning (and hearing) about the subtle problems caused by RF noise, e.g. changes in transparency and soundstage depth.

Perhaps the design change resulted in lowered sensitivity to RF noise? Since it's very hard to keep RF noise out of a test, it might have been a problem with the test environment...

One could argue that higher order harmonics are like RF noise and might even be at the same kind of power level. Perhaps harmonics and RF noise are both creating the same sound quality loss through the same mechanism?

Now playing: Victoria Williams - Polish Those Shoes

 

[paul2qute]

Uelong hope your OK mate? I lost my dad, my side kick with hi-fi on Dec 7th and felt the music died when he passed,but here I am

 

[514077]

Uelong hope your OK mate? I lost my dad, my side kick with hi-fi on Dec 7th and felt the music died when he passed,but here I am

That's where you've been, then. I'm sorry to hear about your loss. I do envy your enjoyment of music with your father. It must have been great to have someone to share it with. Hope it'll bring you some comfort in remembering, when you listen.

 

[Rob Watts]

Couldn't this test be done entirely in software on a computer? If you generate a guaranteed bandwidth-limited synthetic signal (e.g. sum of lots of sine waves each varying in amplitude) you can sample this "perfectly" at an arbitrary rate (e.g. 1/16th of the native sample rate of the synthetic) and arbitrary bit-depth.

Then you can upsample with your test WTAs of varying lengths so that you can measure the timing errors by comparing with the original synthetic signal.


Could these perceived differences represent an unintended consequence? We've recently been learning (and hearing) about the subtle problems caused by RF noise, e.g. changes in transparency and soundstage depth.

Perhaps the design change resulted in lowered sensitivity to RF noise? Since it's very hard to keep RF noise out of a test, it might have been a problem with the test environment...

One could argue that higher order harmonics are like RF noise and might even be at the same kind of power level. Perhaps harmonics and RF noise are both creating the same sound quality loss through the same mechanism?

Now playing: Victoria Williams - Polish Those Shoes

Sure you can run the test via simulation; but it would take a huge amount of time. I have ran mixed signal simulations running for about a second using server farms with 100's of processors, and it takes a very long time - and this was a simple simulation (M scalers are not simple to simulate). Running it in real time with hardware is easier in this instance. Also people are more likely to believe a hardware test than a simulation!

It was not RF noise that changed the sound quality with higher order harmonics; but I for sure recognise the strong possibility that the change in SQ had nothing to do with the measured change but with something else that had changed; after all, I have made mistakes in the past when I was linking SQ changes to an actual mechanism; that's why I added the point that in subsequent other tests, with different actual changes, when higher order harmonics are reduced it is audible in terms of it sounding more transparent. So I am building up a body of evidence with different tests that this is something real.

 

[analogmusic]

I had a question about DAVINA

Would Davina be able to generate Mscaled files at normal 44.1 sample rates?

From what I understand no - it wouldn't but can output 705 sample WTA scales file to a laptop

Not sure what one would do with such a file though as it's massive

I guess M scaler would remain as a real time device, mostly?

 

[Rob Watts]

Absolutely 44.1 will be supported. The intent is to record at 705 kHz, then later decimate it down to 44.1 - but you could equally just record at 44.1. The decimation will be uniquely aliasing free - from 104.25 MHz right down to 44.1 kHz.

 

[jayz]

Couldn't this test be done entirely in software on a computer?

Sure you can run the test via simulation; but it would take a huge amount of time.

I imagine one could obtain/create high-level models (abstract/mathematical/behavioural descriptions) of the filters and achieve far higher performance than clocked mixed-signal simulations but whether the accuracy of results compared to hardware is sufficient, will depend on what the goal is. As Rob says, hardware is usually considered THE reference.

 

[Deftone]

(ADC)

I hope a lot of bands will use it, would probably help a lot with that part of the recording process

Yeah but then there’s the mix and mastering professionals that could still intentionally limit the output recording quality. Unless Rob develops a chip that can be implanted in to the mastering engineer’s brain (somehow secretly) this would then give them a small electric jolt every time they consider using heavy compression tools...

 

[Mython]

(ADC)



Yeah but then there’s the mix and mastering professionals that could still intentionally limit the output recording quality. Unless Rob develops a chip that can be implanted in to the mastering engineer’s brain (somehow secretly) this would then give them a small electric jolt every time they consider using heavy compression tools...

I reckon a taser function might be more efficient... (silver-plated OFC shock leads, obviously)


On a serious note, I think I recall Rob mentioning being approached by a Russian mastering engineer very keen to see improvements in ADC (I can't find the post, right now), so there is reason to be optimistic that not all present-day mastering engineers are chimpanzees playing with a boxful of compression tools in a mastering studio wendy-house.


EDIT - Found it:

I was fortunate enough to hear Doug Sax working at the Mastering Lab in the early 90's. At that time, it was the best reproduced audio I had heard.

Talking of pro-audio, whilst in Moscow last week I was doing a technical talk on Dave and Mojo, and afterwards a young recording engineer talked to me. Naturally, I chatted about Davina, and he said that he would like a vari-speed option on it, as a lot of ADC/DAC's don't allow it. So I thought about it, and reckoned it was possible, so we will add it as a feature on the pro version of Davina. Now he needs it so that he can match pitch from different tracks. So I said why didn't he use digital effects to change the pitch? He said they were awful, and completely changed the character of the voice, making it sound unnatural.

We then talked about removing DC offsets - and he said that using low pass filters sounded worse than taking a file, calculating the DC and subtracting to offset. I explained why it would make a difference - but the point I am making is that there are very good mastering and recording engineers who really do have good ears and are prepared to make great efforts to get more musical sound.

But they need the tools to do the job properly - and that's why I am so keen on creating the tools for them.

Rob

 

[Babysnake]

A quick question for Rob

Does the M-Scaler upscale and output to the DAVE on the BNC input as well as the USB? (the 'manual' is a little vague on this)

And if one has a choice, purely from a technical perspective, which is likely to give an improved SQ?

 

[Rob Watts]

Yes the BNC input is just another input and treated by the M scaler in the same way. As to which input sounds the best - that will depend upon your system and how noisy the sources are, and how sensitive the rest of your system is to the sources... In my system, I use all inputs, and I have never felt that one is better than another, but YMWV.

 

[Mython]

Yes the BNC input is just another input and treated by the M scaler in the same way. As to which input sounds the best - that will depend upon your system and how noisy the sources are, and how sensitive the rest of your system is to the sources... In my system, I use all inputs, and I have never felt that one is better than another, but YMWV.

Thanks for your input, Rob

 

[JWahl]

Hello again, Rob.

Lately I've been thinking more about the effects of environmental RFI on the audio signal. I have this ferrite choke that I bought awhile back, originally to try on the Mojo's USB input. I've found a positive effect on longer USB cables, though the need for it seems to be negated when using a small (around 7 to 8cm) USB cable. I also suspect the galvanic isolation on the Qutest (which I hope to buy when available) will negate the need for that RF filtering.

I recently had the wild idea to try putting that choke on my headphone cable near the Y-split with one extra loop. The theory being to reduce low level RF intermodulation picked up by the longish headphone cable. This subjectively seemed to reveal more low level detail and subtleties, but I can't help but wonder if it negatively affects the frequency/phase response, though I don't necessarily perceive it. I may buy some more in different sizes to do more experimentation.

I bring this up here, because I wonder how much of the extreme measured potential of Chord DACs gets masked by low level environmental RF picked up from the headphone or line cables. More specifically the headphone cables, since there's usually nothing filtering noise from the cable to the drivers. Of course, this would depend on the condition of the users RF environmental pollution. I'd be curious to get one of those handheld RF spectrum analyzers to see where the problem areas might be.

I also think this could account for some of the perceived benefits of balanced headphone cables, since they should theoretically cancel RF noise picked up by the headphone cable. Also noting that the heavy ferrite choke hanging from the headphone cable is hardly ideal for practical use.