[musickid]

@Rob Watts With mojo being the only chord dac without a dedicated crossfeed function and punching well above it's weight in price and size have you included or taken into account crossfeed when designing mojo? How can mojo compete with bigger models especially as crossfeed is so important with headphones? Many thanks and happy festivities mk.

 

[jarnopp]

@Rob Watts With mojo being the only chord dac without a dedicated crossfeed function and punching well above it's weight in price and size have you included or taken into account crossfeed when designing mojo? How can mojo compete with bigger models especially as crossfeed is so important with headphones? Many thanks and happy festivities mk.

Good thought, mk. Perhaps Rob’s crossfeed algorithm could be incorporated into Poly, via a firmware update? Then, you could set crossfeed levels (0, 1, 2, 3) via the GoFigure app.

 

[ecwl]

Good thought, mk. Perhaps Rob’s crossfeed algorithm could be incorporated into Poly, via a firmware update? Then, you could set crossfeed levels (0, 1, 2, 3) via the GoFigure app.

I think the crossfeed in Hugo 2 and above are implemented after the 16fs upsampling at 24-56 bit depth so while Poly can implement crossfeed, it won’t be the same as what’s implemented in Hugo 2.

 

[Rob Watts]

So cross-feed wasn't included on Mojo for a number of reasons - number of buttons being one, and power dissipation being another, so limiting battery life. Also, way back when Mojo was designed, I wasn't as keen on cross-feed as I am today; Hugo 2 with the improved transparency and better depth perception meant that cross-feed really improved the experience and for me went from something nice (Hugo 1) to something essential (Hugo 2).

 

[flyte3333]

Hi @Rob Watts and Happy New Year.

We know you advise against software resampling before your DACs and this is understandable since you speak a lot about your DSP being a key differentiator.

For speaker listeners, what are your thoughts on using convolution via some of the state of the art digital room EQ . Offerings like Acourate, Audiolense, and the latest Dirac Live.

Do you still recommend no DSP before your DACs, even if digital room EQ does help improve things in one's room?

Cheers!

 

[Rob Watts]

I have never been attracted to the idea of DSP to compensate for room problems - partly because the ear/brain is adept at accommodating frequency response changes - the brain measures the environment acoustic and compensates for it. If you are walking around with a chatting friend, and go from a acoustically dead room, to a live room, to then the outside, you don't suddenly think your friend's voice has been degraded or sounds different - the brain just deals with radically different acoustics, and you do not lose image and depth definition, instrument separation and focus, nor timbre qualities; but DSP degrades all of these qualities. I would sooner tune my room using good old fashioned acoustic set-up - as this won't damage musicality.

 

[ecwl]

@flyte3333 while I 100% agree with Rob Watt's comments on DSP, I did want to provide a counter perspective as someone who introduced an Acourate designed convolution filter into my system for the past few months.

I think the argument that the ears adapt to room acoustics is very strong and the detectability of phase shifts for humans is poor based on available scientific evidence. However, the ability for the ears to adopt to pretty much any acoustic anomalies are extremely strong. Think of all the people who ignore noise floor modulation, poor transient accuracy, timbral inaccuracies or even high noise floors and claim that Chord DACs are nothing special. Their ears are also adapting to sonic anomalies.

And I understand that IMHO the world's best DAC designer would want us to hear his DAC in the best possible system, aka. no ground loop leakage noise into the m-scaler/DAC from your streamer/desktop/router, maybe listen through headphones, or maybe through the most transparent amp and speaker system with no major room acoustic issues. So if we have room acoustic issues, just move our seats and speakers and treat the room acoustically.

But as a consumer, we all make compromises. I would argue that I have only been to 2-3 audio stores where their speakers, seat placement and room treatment are optimal. Even my local Chord dealer has to sell so many different speakers and home theatre systems to survive that his best home theatre/stereo room has slightly suboptimal listening seat placement and his speakers are not always optimally placed. And his demo room acoustics is definitely still better than my living room. And of all the local stereo enthusiasts I know, none of their rooms are properly acoustically treated and setup as well as the top stereo store demo rooms. (In fact, they're often very poorly treated and setup)

Having first optimally placed my speakers and subwoofers and listening seat within the limits of not turning my apartment living room into a stereo store, I have used parametric EQ in JRiver and now Roon and convolution filters to primarily correct bass issues. If you listen hard enough, there is always a very subtle loss in transparency and soundstage depth with the DSP. However, untreated bass peaks also cause loss in transparency and soundstage depth. So to me, it becomes a matter of what compromises are you willing to make and then listening to see whether the parametric EQ/convolution filter gives you more gains than the harms they produce.

I have uploaded my pre-convolution and post-convolution measurements of frequency response of my speaker system. When I watch movies, I go without DSP and let my ears adapt to the room. But when listening to stereo music, I now have the convolution filter on. You can see there is no way for me to remove these >6dB bass frequency peaks effectively without essentially buying a new home (which actually most of my audiophile friends have suggested). Moreover, the use of a convolution filter means I can add a subsonic filter to remove low frequency rumbling of the woofers and subwoofers to provide more transparency in the upper bass and midrange.

One warning I do have is that it is actually very hard to optimize the convolution filter. I found optimizing parametric EQ's very easy, you just run Room EQ Wizard, see what you get, and then throw the parameters into your software and tinker with it based on your preferences.
With the convolution filters, first, you have to sort of know your preferred frequency response, and then there are lots of options in terms of excessive phase compensations depending on the frequency window you decide to choose. Most of the people online who advocate using convolution filters are more into active crossover systems and focus very roughly on the step response result and use traditional DAC chip DACs. So I strongly suspect they don't actually fine tune their convolution filters for optimal transient response. I basically spent 2-3 months tweaking the filters and re-listening until I figured out the optimal filter where I don't lose transient dynamics/accuracy, because this is something I'm super used to from Chord DACs. I have a strong hunch this is not something automated systems like Dirac can actually do. It took me a long time to figure out how to do it in Acourate manually and I suspect if I were to switch over to Audiolense, I would have to re-learn and create a new optimization algorithm all over again.

That said, with my system, the convolution filter made a huge difference. It is interesting about how our ears adapt. Before the convolution filter, I used to think my parametric EQ for the stereo made my system really close to my Focal Utopia headphones. Now that I have an optimized convolution filter, I'm actually hearing more of my stereo's deficits even though it sounds tremendously better than the parametric EQ. And I actually hear that my Focal Utopia headphones are actually still quite a bit better than my stereo.

 

[darkless]

@Rob Watts Happy New Year!

As I'm contemplating going from passive to active speaker crossovers with the heart of my system being my lovely Chord gear, I'm in desperate need of advice.

I have a setup consisting of an HMS/DAVE combo and a Benchmark AHB2 amp feeding a pair of open baffle speakers, PureAudioProject Quintet15 Horn1. I have recently decided to convert my passive crossover to a DSP crossover by performing the channel splitting, crossover and driver time-alignment on my PC. The plan is to feed the horn channels into my DAVE+AHB2 and the two woofer channels into an HMS/Cutest combo plus another power amp. For digital volume control and display I'm thinking of going with a MiniDSP SHD Studio in front of the signal chain, as I don't trust Windows not to one day crank the volume to 100% and blow my speakers (and my ears!).

My main problem is that resampling will occur up to three times: First by the MiniDSP performing an async resampling to 96 KHz (internal sample rate), then Windows mixer, and finally the HMS before going into the Cutest. Obviously this is far from ideal and will likely ruin any possibility of the Chord upscaling working as intended, not to mention that the digital attenuation in the MiniDSP might not be the best.

Alternatively I could go with an oldschool electronic active crossover after my HMS/DAVE combo and use the DAVE volume control as I do with the passive crossover, but that would mean that I wouldn't be able to time-align the speakers and the crossovers would be no steeper than 24 dB/octave, not to mention the much worse SNR and THD due to the added electronics.

A better solution would be to have the HMS in front of the PC and feed it the full upsampled signal, then perform the DSP and feed that into the two Chord DACs, but I'm not aware of any dual-BNC interfaces for the PC. I would also still need a way to see and remote control the volume for all six channels (speakers plus two subs). This could be a home theater preamp, but that comes with its own set of limitations, although in this case it might be the lesser evil compared to having the MiniDSP wrecking havoc on the digital signal before the HMS has a chance to shine.

What would be the best way forward?

 

[Rob Watts]

My guess is that old school actives would be best - the thought of multiple resampling with accumulated transient timing errors fills me with dread!

As to phase accuracy - I am not convinced this is important. For sure, phase amplitude non-linearity (where the phase shifts with amplitude) is extremely important - I have heard ultra small errors creating hardness and aggression and poorer depth perception. But I have no evidence that a simple phase shift that is amplitude linear is actually important - unless you are getting strong frequency response changes too.

As too driving drivers directly, I am convinced this will be very important. Quite how the best way to do this is unclear; but the groundwork is being undertaken now.

 

[InstantSilence]

My guess is that old school actives would be best - the thought of multiple resampling with accumulated transient timing errors fills me with dread!

As to phase accuracy - I am not convinced this is important. For sure, phase amplitude non-linearity (where the phase shifts with amplitude) is extremely important - I have heard ultra small errors creating hardness and aggression and poorer depth perception. But I have no evidence that a simple phase shift that is amplitude linear is actually important - unless you are getting strong frequency response changes too.

As too driving drivers directly, I am convinced this will be very important. Quite how the best way to do this is unclear; but the groundwork is being undertaken now.

Mr Watts, what is the most important part of a dac to yield most transparency?

 

[Rob Watts]

Mr Watts, what is the most important part of a dac to yield most transparency?

I don't think I can easily answer that, as so many things are ultra important. Moreover, small things make huge differences, and there are countless small errors inside a DAC.

Also, the era/brain's response to errors isn't linear or log or dB but more like log(log). So reducing some errors by 30 times has a similar impact subjectively to reducing it again by thirty times; this is very odd, as one would expect that the law of diminishing returns would apply. For example - going from 164k WTA taps to 1M WTA taps was a larger change subjectively than 26k WTA to 164k WTA. Or with depth perception - a change in perceived depth is the same for 200dB errors to 230dB - and then a change of 230 to 260 gives a similar improvement in depth perception. This just highlights that ultra small changes can have a big impact, and DAC design really is about the whole being greater than the sum of it's parts.

Having said that, I can say that there are three major factors:

1. Interpolation filter - reconstructing timing of transients without uncertainty
2. DAC design - the conversion from digital to an analogue current
3. Analogue design - taking that analogue current and converting it to a voltage that can drive OP devices.

Factors 1 and 2 are roughly equal in importance, and maybe 3 less important. 3 of course depends upon 2.

Unfortunately, each of these areas have a myriad way of upsetting performance - which is why it's so interesting to discover these errors and then mitigate them. I get intellectual stimulation in doing this, and after months (sometimes decades on some errors) I get a musicality pay-off too when reducing these errors in my audio systems.

Happy New Year everyone, and let's hope 2021 gets us back to normality.

 

[Ards]

I would sooner tune my room using good old fashioned acoustic set-up - as this won't damage musicality.

While laudable, unfortunately this is rarely practical in our rooms as correcting for low bass issues requires A LOT of acoustic treatment. Much simpler to have some simple DSP take care of it. Simple pragmatism.

There are some non-DSP technological solutions such as the AVAA active bass traps that provide tangible results.

AVAA C20 - Active Bass Trap - PSI Audio

 

[flyte3333]

I would sooner tune my room using good old fashioned acoustic set-up - as this won't damage musicality.

Interesting because some experts in this field (room acoustics) say that done wrong, this can be very damaging.

And done right can be very complex to achieve ! Costs $$$$

 

[HeeBroG]

While laudable, unfortunately this is rarely practical in our rooms as correcting for low bass issues requires A LOT of acoustic treatment. Much simpler to have some simple DSP take care of it. Simple pragmatism.

There are some non-DSP technological solutions such as the AVAA active bass traps that provide tangible results.

AVAA C20 - Active Bass Trap - PSI Audio

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!

 

[AndrewOld]

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!

Perhaps you should look to changing your speakers?