[Ards]

I have x2 AVAA's in the front corners behind my main speakers and subs. Whilst they help, they are expensive and not as good as I expected. Perhaps another x2 might work better but I thought I would try convolution instead. I use them now with the convolution filters as there were other issues with my speaker's upper range that needed "taming"! I don't have magic ears so can't detect any major loss in depth/transparency but the sound is now less fatiguing and more balanced.

My Utopias are still very enjoyable direct from DAVE when I can't run the main rig; as long as I remember to switch off the filters!

The AVAA’s do take careful positioning - you need to measure the room and identify the major nodes you want to target and place them in those locations. The before/after results are clearly measurable eg in REW and the listening result in the room is that they help tame bass boom and ringing caused by the standing waves. It’s one of those affects you most notice after living with them for a while and then turning them off: the sound is coloured without them.

 

[Vyyy]

Hi @Rob Watts
As heavy Hugo2 user and creator, Do you start thinking about how to improove and on what aspects you would like to improove Hugo2 sound signature? You could name your personal taste there too For example me personaly dreaming of slightly more power from low volumes like class A amps have control in lows and vocals and bit deeper/warmer.
Feel free to express
Thank you.

 

[CaptainFantastic]

@Rob Watts Hi Rob, I have a question about my TT2. I purchased it in November and it works great, I am really enjoying the performance, so much so that I added the M Scaler shortly thereafter.

My question - I noticed first with low impedance headphones (Owl, Hawk), that after I plug the headphone in there is a channel imbalance for 5 to 10 seconds. It's almost as if there is no sound in the right earcup. In fact there is, just much lower. Then within 10 seconds it comes up to normal and to the best of my abilities I tested and I think it is perfectly balanced after that. I noticed that in fact it does it with my Sennheiser and ZMF too (300 ohm), I guess I hadn't put them on right after plugging in. So this is not just when powering up, it's after every headphone change.

Is this normal? Are the capacitors charging after headphone changes and the ones dedicated for the right channel are slower? The unit seems flawless otherwise... should I be worried about issues down the road or even now is there a chance that in fact the channels are not balances and I just haven't evaluated enough?

Thanks!

 

[InstantSilence]

Is rca or xlr outs (when connected to an external amp) best used for most transparent sound, out of chord dacs?

 

[darkless]

Is rca or xlr outs (when connected to an external amp) best used for most transparent sound, out of chord dacs?

For Chord DACs the RCA outputs provide the most transparent sound as they have fewer parts in the signal path.

 

[Rob Watts]

@Rob Watts Hi Rob, I have a question about my TT2. I purchased it in November and it works great, I am really enjoying the performance, so much so that I added the M Scaler shortly thereafter.

My question - I noticed first with low impedance headphones (Owl, Hawk), that after I plug the headphone in there is a channel imbalance for 5 to 10 seconds. It's almost as if there is no sound in the right earcup. In fact there is, just much lower. Then within 10 seconds it comes up to normal and to the best of my abilities I tested and I think it is perfectly balanced after that. I noticed that in fact it does it with my Sennheiser and ZMF too (300 ohm), I guess I hadn't put them on right after plugging in. So this is not just when powering up, it's after every headphone change.

Is this normal? Are the capacitors charging after headphone changes and the ones dedicated for the right channel are slower? The unit seems flawless otherwise... should I be worried about issues down the road or even now is there a chance that in fact the channels are not balances and I just haven't evaluated enough?

Thanks!

I doubt that it has a permanent in-balance between channels - this would have been checked with Chord's QA when your TT2 was measured.

But I can't explain or even think of a possible mechanism - very bizare. Do you use the same headphone cables? If so, check them out.

 

[flyte3333]

Hi @Rob Watts

How many analogue input and output channels will DAVINA have?

And will it have built-in mic pre ?

And headphones output?

 

[Rob Watts]

The prototype had adjustable mic inputs with adjustable phantom power. To listen you need to connect a DAC via the dig outputs. The pro model has not been specified as yet...

 

[Christer]

The prototype had adjustable mic inputs with adjustable phantom power. To listen you need to connect a DAC via the dig outputs. The pro model has not been specified as yet...

Hmm,
Every time the subject "Davina" pops up after quite a few years of promises, it reminds me a bit of my still active days as a free-lance Travel/Wildlife
writer/photographer once asking a Magazine editor "when do you need the article?" And his response was "Yesterday!"
Can´t help feeling a bit the same about Davina.
Cheers CC

 

[Jawed]

The prototype had adjustable mic inputs with adjustable phantom power. To listen you need to connect a DAC via the dig outputs. The pro model has not been specified as yet...

So it sounds like you have a working ADC with all the bugs solved?

My understanding is that you are using a pulse array DAC in the feedback loop. If it's like DAVE, then it consists of a 2nd stage WTA filter up to 256FS followed by a IIR stage that goes up to the ~100MHz pulse array driver. You've never talked about the FIR tap length of the 2nd stage WTA as far as I can remember. But presumably the second stage adds a time lag.

Since the ADC is sampling at 16FS, the typical WTA 1 delay (e.g. 0.1s or 0.7s) is not present. But it seems like there would still be some delay due to the tap-length of WTA2.

How do you compensate for this delay in the feedback?

 

[Rob Watts]

Yes it's a pulse array DAC in the ADC feedback loop - with a latency of just two or three master clock cycles - the whole ADC runs from 104MHz, so there is no significant delay to worry about. The problem has been getting from 104MHz down to 768k or 705k - and this has turned out to be a major headache.

The reason why it's a major problem is that studios run a common clock to sync up all their ADCs. Using an external clock is a crazy idea, as it adds to noise, jitter and aliasing. But in the case of studios they absolutely have to do this, there is no alternative.

Now to get the best analogue performance you must use a local clock - and one that is not being adjusted via a PLL, as this adds jitter and noise resulting in skirts and noise floor modulation. So to go from 104MHz (local clean clock) to 768k (synced to external clock) without any degradations requires using a very complex fractional interpolation filter. And that's exactly what I have been working on over the past 3 months. And I have gotten some interesting and perplexing results. I am not, at this stage going to publish my results, as I am still processing things - but the work suggests that aliasing, if it's observable on a digital domain measurement, is easily audible - even when doing single blind listening tests - as I couldn't believe what I was hearing.

I have still got a lot more work to do on this issue, but in principle the problem has been solved in that the output now no longer differs from the input with digital domain measurements.

 

[Christer]

Yes it's a pulse array DAC in the ADC feedback loop - with a latency of just two or three master clock cycles - the whole ADC runs from 104MHz, so there is no significant delay to worry about. The problem has been getting from 104MHz down to 768k or 705k - and this has turned out to be a major headache.

The reason why it's a major problem is that studios run a common clock to sync up all their ADCs. Using an external clock is a crazy idea, as it adds to noise, jitter and aliasing. But in the case of studios they absolutely have to do this, there is no alternative.

Now to get the best analogue performance you must use a local clock - and one that is not being adjusted via a PLL, as this adds jitter and noise resulting in skirts and noise floor modulation. So to go from 104MHz (local clean clock) to 768k (synced to external clock) without any degradations requires using a very complex fractional interpolation filter. And that's exactly what I have been working on over the past 3 months. And I have gotten some interesting and perplexing results. I am not, at this stage going to publish my results, as I am still processing things - but the work suggests that aliasing, if it's observable on a digital domain measurement, is easily audible - even when doing single blind listening tests - as I couldn't believe what I was hearing.

I have still got a lot more work to do on this issue, but in principle the problem has been solved in that the output now no longer differs from the input with digital domain measurements.

Good to hear you´re still working on it ,but I do have a question.
What do you mean and refer to when you say "studios" and all their ADCs and external clocks?
The big commercial pop studios with a "million sliders" to pull up or down as desired?
I´ve never been inside one of those.
And have no plans to do so either.
Why do they have to employ external clocks?
As far as I know, but I could be wrong, at the Classical Music digital recording session "studios" I have been to, there has only been ONE ADC used, either PCM or DSD and sometimes both in parallel, and that/those two same ADC/DACs also used at mastering.
Mostly only one though.
Not really studios as such, more often a basement room where the recording team set up their computer, an ADC/DAC and speakers and headphones and lots of cables to the mics in the hall, for monitoring.

Several recording engineers I´ve watched closely and interviewed at work recording classical music, aim at balancing as much has possible already at sessions and then need little or even no post production at all.
I´ve never heard anyone mention external clocks needed or used anywhere in their chain?
Have I missed something?
Cheers CC

 

[Rob Watts]

Pro interfaces have 8 ADC's and if you need to use more than 8 channels than you need to sync your interfaces together. So the pro model must have the ability to sync to external word clocks. Classical recording is after all a small percentage of pro audio.

 

[roskodan]

Hi @Rob Watts
As heavy Hugo2 user and creator, Do you start thinking about how to improove and on what aspects you would like to improove Hugo2 sound signature? You could name your personal taste there too For example me personaly dreaming of slightly more power from low volumes like class A amps have control in lows and vocals and bit deeper/warmer.
Feel free to express
Thank you.

Interesting, I guess that would be a Hugo 3. What I'm wondering is, what about a dedicated headphone amplifier designed by @Rob Watts ? I always felt the output stage of my Hugo to be worth as basis for a standalone solution. Basically, I was contemplating about a more powerful output buffer for which the Hugo line could act as a pre amplifier. But I guess that would cannibalize Chord existing product line. Anyhow, would be really interested in hearing what would @Rob Watts take on a dedicated headphone amplifier be like in terms of design philosophy.

Still being impressed on a daily basis by my little shiny magical device, must be all those fancy colored lights I guess.

 

[Rob Watts]

Adding an extra stage will always degrade transparency, so I am not interested in designing an analogue headphone amp. If you need more power for headphones go for a TT2.

More power on a Hugo 2 would need more batteries, or a charge pump to provide a boost voltage; but you have power dissipation problems too - Hugo 2 is a small package. For my needs, Hugo 2 has more than enough power.