[LugBug1]

Enjoyed reading that thank you. Also very informative, especially since I'm always changing filter options on my Audiolab dac (and always avoiding the the optimal one... Because of the pre-ringing warning!)

I shall be going optimal tonight To be fair I think the warning is only that the sound may become fatiguing compared to the filtered sound.

 

[MRC001]

When listening to DACs with filter options, I tend to prefer "sharp" to "smooth". The "smooth" modes are supposed to have better transient response, but it comes at a price: a slight roll-off of the extreme high frequencies. With many recordings it makes no difference because they're so crappy, energy above 13 kHz is either non-existent or distorted. But with excellent recordings of natural acoustic music, the "smooth" modes affected the sense of "air" in subtle ways making it slightly dead or soft in comparison. And subjective listening never convinced me that the transient response was any better.

 

[halloweenman]

Hi Rob. I'm forever enjoying reading posts. Now that you are developing an AD convertor and already have several superb DACs have you ever tried cycling A to D then back again to A and compared the results? I guess with perfect convertors you should not hear any audible difference. Perhaps after converting from A to D and back to A again 10 times or more you might start to hear a difference from the original to the converted? Surely this is the ultimate test?

 

[Mython]

Hi Rob. I'm forever enjoying reading posts. Now that you are developing an AD convertor and already have several superb DACs have you ever tried cycling A to D then back again to A and compared the results? I guess with perfect convertors you should not hear any audible difference. Perhaps after converting from A to D and back to A again 10 times or more you might start to hear a difference from the original to the converted? Surely this is the ultimate test?

I'm fairly confident Rob has many 'ultimate tests' mere mortals can barely conceive of!

 

[Rob Watts]

Hi Rob. I'm forever enjoying reading posts. Now that you are developing an AD convertor and already have several superb DACs have you ever tried cycling A to D then back again to A and compared the results? I guess with perfect convertors you should not hear any audible difference. Perhaps after converting from A to D and back to A again 10 times or more you might start to hear a difference from the original to the converted? Surely this is the ultimate test?

The problem with current ADC's is that the performance is so poor, we can easily see substantial measurement problems, which will for sure affect SQ.
I do have a number of tests planned that follows in this vein though. I think the most interesting one will be the effect of bandwidth limiting, without decimation, or doing anything else. Actually, bandwidth limiting is normally a good thing, as it eliminates HF noise which then causes noise floor modulation in the DAC, thus making it sound brighter. The noise shaping employed in Davina though should have noise increasing above 500 kHz, (as opposed to conventional ADC's that increase just above 20 kHz) so this would mean that removing the residual noise should have minimal benefit, as the out of band noise should be very low. But the issue illustrates the problem; there could be tangible real benefits, and there could be real sound quality problems in bandwidth limiting - so any listening test will need extremely careful evaluation (both listening and measurement). It's a perennial problem with audio, as people usually condition how something sounds like with their expectations, or their theoretical thinking - xyz must sound bad because of abc, it's obvious innit?

 

[DaveRedRef-III]

Hi Rob, I wonder, have you started listening tests yet with Davina? Would love to hear any early impressions.

Edit: just seen above that it may be still too early for that.

 

[Rob Watts]

Yes way too early! Got other projects too, which is taking up all my time at the moment. Should be back on Davina later this month...

 

[DaveRedRef-III]

So are you working on a Red Reference V which will unify BluDave then Rob?

No need to reply......

 

[Deftone]

Got other projects too

 

[Babysnake]

A question for Rob, if I may, whom I very much enjoyed meeting and chatting to at a recent small scale show in Suffolk

My question is about room correction

I get that this may be anathema to some but I have always had great success with various DSP based DRC boxes, in that I much preferred the sound with the correction on, as distinct to off

In its simplest form my question is this: if one applies any sort of DSP such as room correction or electronic speaker crossovers post DAC, (be it Hugo, DAVE or whoever), does one lose the great benefit of all that clever FPGA based programming once you go through the requisite ADC-DAC process in the DSP box? And assuming the answer to that is "yes", how can you enjoy things like DRC and high end DACs concurrently?

 

[MRC001]

Do as much room correction as you can the old-fashioned way: tube traps, bass traps, diffusers, acoustic foam, whatever it takes to get the sound as good as you can get it. Then use DSP as a last resort to correct things you can't correct otherwise. Two main reasons for this. First, keep the musical signal as pure as possible, minimize the amount of signal processing. Second, room treatment can fix some issues at their root, while DSP might only stick a band-aid over it. Put differently, using DSP to correct room effects may only shift the distortion around: reduce amplitude problems while creating new time problems, etc.

In short, I'm not saying room correction DSP is bad. It's a useful tool that can make an improvement, but may not be as clean as actual room treatment.

 

[rayl]

I've always pondered the same questions on EQ for room correction and have concluded, based on principle and not necessarily from empirical research, that one either does it before it hits the DAC or stay in analog after the DAC as additional ADC/DAC will inevitably use inferior reconstruction filters.

Would love to hear more informed answers.

I also agree that room treatment is to be preferred, but there will be inevitable room modes. If they are peaks, then EQ to correct them at the listening position is fine for me. (For nulls -- those cannot be EQ'd away.) For my own setup, I chose the analog after DAC route to mitigate a nasty peak at 120 Hz. I've already stuffed around 90 Sabins of absorption (measured @ 125 Hz), in most of the loud locations by walls. For a medium size room, not much more I can do w/o making the room look like a studio!.

I am less worried about phase effects bec both room effects and analog EQ, are minimum phase -- so any phase shifting from EQ just offsets the opposite phase shifting from the room if one is flattening the frequency response. (If you are correcting for effects from the speakers, then that's a different story, but we're talking about room correction here -- not correcting for off-axis speaker performance, etc.

 

[Rob Watts]

I had a friend who was building a set of loudspeakers, and was using a ADC/DSP/DAC to solve EQ issues with the speaker. He was very happy with it - until I demoed a Hugo. With the Hugo in place it became very apparent that the losses in the DSP vastly outweighed the benefits; so he scrapped the DSP EQ and fixed the loudspeaker issues using the old fashioned way, of getting the loudspeaker design right.

So my experience suggests that I agree with the other posters; fix room problems with room treatment.

 

[halloweenman]

It's not always possible to fix room problems with room treatment in a typical hifi listening room. Take my room for example, I have a low frequency peak at around 45hz of around 12db at the listening position due to a room mode and standing waves. It's nigh impossible to treat low frequencies such as this without massive, like fill literally over 12m2 of the room with mineral wool to have any meaningful effect at 45hz. I'm only aware of one commercial bass trap, RPG modex plate, that is effective at such low frequencies, even then this is not aesthetically acceptable in a domestic environment and costs £££ and are extremely difficult to install and get right.

I found the best solution was to reposition speakers and listening position so that listening was acceptable. The next best solution was a small cut using 64 bit floating point roon eq high pass or peak/dip filter before DAC.

If you are listening to Robs crossfeed using headphones doesn't that use dsp eq anyway?

 

[Babysnake]

So my experience suggests that I agree with the other posters; fix room problems with room treatment.

I primarily listen to my system through a Blu2/DAVE and a pair of LCD-3s , so I am now frantically chasing that sort of absolutely stunning performance for a wider audience. Yet the more I think about this the bigger it appears to my simple (non-technical) mind the problem is

Leaving aside the aesthetics and WAG acceptability or otherwise of treating a room, the problem extends to any DSP that occurs after the DAC including, for example, active electronic crossovers for loudspeakers. At the HiFi meet where I was privileged enough to chat with Rob, a company (Kudos?) was demonstrating active loudspeakers (with NAIM gear I believe) which sounded very good indeed, and so in that case there is an ADC-DSP-DAC process occurring in the crossover units, downstream of the primary DAC

So am I right to say that in that case you are listening primarily to the DAC in the crossover?? And if that is true, then if (like me, Sanders Model 10) you have active loudspeakers it makes no difference the 'quality' or nature of your upstream components (endgame DAC, turntable or whatever) you are always listening to the DAC in the crossover

That doesn't seem to be intuitively right to me somehow, so have I got it wrong at some technical level?

Triodes and horns anyone ?