Does DSP belong in State of the Art Systems?

[Mark Seaton]

In saying that you would be incorrect. No amount of DSP can remove a standing wave! And its standing waves that cause problems with bass, especially below 80Hz or so. Standing waves can cause cancellation at certain frequencies so a loss of certain bass notes (along with others being emphasized). You can put all the power you want to into that cancellation and it will still cancel... In a nutshell, it doesn't work. DSP will work for removing peaks (since they are the result of superposition) but you can't do with nulls.

Obviously Kal clarified his point already, and I would add that anyone who doesn't understand that DSP can't directly fix room acoustics isn't going to be assembling a truly state of the art system on their own.

It's interesting that you mention the Double Bass Array (pitch-catch approach). Most executions of this utilize a digital delay on the rear set of bass sources. While possible, the amount of delay commonly required isn't simple to find in analog components.

 

[Daverich4]

I will have to disagree with this. DSP.... (plug-ins) work at a specified sampling rate. "Most" plug-ins work at 96/24.... if your source is 192/352.8/DSD, then you are down sampling! The same can be applied to hardware boxes as well.

I use convolution filters made for me by Home Audio Fidelity and there is one for each sampling rate. I use them in Roon and I’m under the impression Roon uses the filter that’s appropriate for the sampling rate being played.

 

[Bruce B]

I use convolution filters made for me by Home Audio Fidelity and there is one for each sampling rate. I use them in Roon and I’m under the impression Roon uses the filter that’s appropriate for the sampling rate being played.

Correct... but you're still using the same chip

 

[Daverich4]

Correct... but you're still using the same chip

I’m not quite sure what chip you’re referring to.

 

[sbnx]

He is referring to the microprocessor chip that is running roon and calculating the corrections using the filter you made.

however, the server is likely using two different cores for this

 

[Atmasphere]

anyone who doesn't understand that DSP can't directly fix room acoustics isn't going to be assembling a truly state of the art system on their own.

It's interesting that you mention the Double Bass Array (pitch-catch approach). Most executions of this utilize a digital delay on the rear set of bass sources.

I've been talking about a Distributed Bass Array; I was part of a Double Bass Array for many years playing in various orchestras. Either way though, no digital delay is required

 

[Mark Seaton]

I've been talking about a Distributed Bass Array; I was part of a Double Bass Array for many years playing in various orchestras. Either way though, no digital delay is required

Funny.

I am glad I clarified, as I've always seen DBA referred to the double bass array, which uses a sufficiently dense array/spacing of bass sources on the front wall with similar at the rear, where the rear are operated in opposite polarity and delayed an amount equal to the time it takes sound to travel 1 length of the room. This does require a delay of some form, and can work quite well, albeit less efficient at the lowest frequencies. Unless you are entirely free to place the seating location front to back and complete freedom to place the various subwoofers, a simple delay on the closer bass sources (most often the rear), can often make for a dramatic improvement in the resulting combination for those cases that aren't so lucky to see very smooth results just by sending all of the subs the same signal.

 

[Atmasphere]

Funny.

I am glad I clarified, as I've always seen DBA referred to the double bass array, which uses a sufficiently dense array/spacing of bass sources on the front wall with similar at the rear, where the rear are operated in opposite polarity and delayed an amount equal to the time it takes sound to travel 1 length of the room. This does require a delay of some form, and can work quite well, albeit less efficient at the lowest frequencies. Unless you are entirely free to place the seating location front to back and complete freedom to place the various subwoofers, a simple delay on the closer bass sources (most often the rear), can often make for a dramatic improvement in the resulting combination for those cases that aren't so lucky to see very smooth results just by sending all of the subs the same signal.

This works great in public address where the rooms are significantly longer than the bass frequencies being reproduced- it increases intelligibility.

But in a normal room like most audiophiles have, this does not address the standing wave issue. So the peaks and valleys will still be there. A Distributed Bass Array won't result in flat bass; you get much smaller peaks and valleys instead, and more of them. But its great at breaking up the standing waves, no digital processing needed and the resulting bass sounds clearer and more resolved (as do the mids and highs).

 

[Kal Rubinson]

I've been talking about a Distributed Bass Array;

Same for me. I've read about the Double Bass Array but never got to try it.

 

[pjwd]

Same for me. I've read about the Double Bass Array but never got to try it.

I have used it in a 600 seat concert hall.
The acoustic engineer did all the calcs for delay etc. but when we tested it the best sound was achieved with bass units in phase and no time delay
It may have had something to do with the fact the bass units were dipole or the hall was well damped ( it was a multipurpose hall .. speech and music) but it was a surpring result .. we never had the time to do extensive meaurements which was a shame.
The sound reinforcement system was all dipole and worked very well for speech clarity
There are a lot of factors at play and I learnt you need to have flexibilty in the design

Phil

 

[Alrainbow]

Dipole bass in a hall for sound reinforcement ..! Not surprised you had phase issues ...

 

[pjwd]

Dipole bass in a hall for sound reinforcement ..! Not surprised you had phase issues ...

We actually didnt have any issues .. it was the most non intrusive, natural sound reinforcement i have ever heard .. all i was noting that the classic theory of null and peak cancelling bass units did not sound optimal in this situation.
As long as you are applying dsp below the dipole peak there are no inherent phase problems with dipole bass
Phil

 

[Mark Seaton]

I have used it in a 600 seat concert hall.
The acoustic engineer did all the calcs for delay etc. but when we tested it the best sound was achieved with bass units in phase and no time delay
It may have had something to do with the fact the bass units were dipole or the hall was well damped ( it was a multipurpose hall .. speech and music) but it was a surpring result .. we never had the time to do extensive meaurements which was a shame.
The sound reinforcement system was all dipole and worked very well for speech clarity
There are a lot of factors at play and I learnt you need to have flexibilty in the design

Phil

Are you sure we are talking about the same thing? You had a set of dipole subs at both the front and rear of the 600 seat concert hall?

 

[pjwd]

Are you sure we are talking about the same thing? You had a set of dipole subs at both the front and rear of the 600 seat concert hall?

Correct

 

[schlager]

If one wants to get their feet wet with digital room correction, there is a freeware program called DRC, it works like Acourate, Audiolense, Dirac and the like, as correcting the impulse response. You just need to run it from a convolution engine, Jriver, ROON og EqualizerAPO.

DRC: Digital Room Correction

DRC: Digital Room Correction filter generator

drc-fir.sourceforge.net

No question that multi subwoofer systems works, it's simple physics. Take it to another level and use the Multi- Sub Optimizer freeware program, (you will need a DSP) that will integrate all the subs you can muster in your room, to your main speakers and give a big improvement in the spatial distribution of acoustic pressure and particle velocity. For all practical purposes bass problems can now be solved very easy. Enjoy.

Multi-Sub Optimizer Tutorial

The tutorial for Multi-Sub Optimizer, containing step-by-step instructions for using almost all of its features.

www.andyc.diy-audio-engineering.org

 

[Kingrex]

Everyone is talking dsp for sub woofers. I was hoping to use a dsp to gain parameter of frequency, slope and gain to then have a passive crossover built to allow biamping of my speakers. Question for me is. Can one biamp a speaker and get performance that exceeds a single amp. Consider my horn probably crossess with the woofers in the 2400 hz regions. My woofers actually handle a fair amount of midrange information.

 

[Steve Rowan]

Hi,
I use DSP in my system all of the time. In fact my Legacy Audio V speaker system uses the Wavelet to provide DSP, preamp, crossover, and DAC duties. One can choose which features to use. I use Wavelet for DSP and crossover duties. Compared to many systems its DSP is very easy to use. Set up the microphone, engage the software, one hears a couple of chirps. two for each channel, and the algorithms are calculated and the DSP is activated. One can turn it off and on easily and hear the difference it makes. I would not want to live with out it. Bill Duddleston of Legacy Audio can program Wavelet to work with almost any speaker system.

I also have a six pack of JL Audio F 113 Mk 2 subwoofers. They use JL Audio‘s DARO DSP software to help integrate the subs into your room and integrate seamlessly into your stereo system.

With either system it’s easy to turn off the DSP and play music with the DSP engaged and without. It really only takes a few moments to appreciate what these two DSP systems can do. In short using these two DSP systems really makes my system come to life. If there are compromises to the purity of the digital output I can‘t tell. It’s very easy to hear the difference between the several DACs I’ve owned and friends have brought over. I’ve owned 5 different DACs including the last the Lampizator Pacific and I’ve never had a problem hearing all the power and specialness this DAC possesses come shining through in spades. I can easily tell the differences between any cable changes, footers, weights, rack postion, preamps, turntables and phono stages, linear power supplies, clocks, reclockers, racks, CD versions, tubes, and anything else I can’t remember I’ve experimented with through time. And I’m not the only one that can hear these differences. One other thing I should mention. DSP works hand in hand with any room treatments you may be using. The better the room sounds before DSP the better it will sound after the DSP. There are many guest listeners who visit and also don’t have any trouble discerning the differences a change in even the most minor component can bring.
My conclusion, DSP is an important tool that the modern audiophile can use to improve theKR system dramatically.

 

[Brucemck2]

DSP is not easy to do right, any more than it is easy to do analog right. Dialing in a cartridge and the upstream phono preamp settings isn’t straightforward and takes specialized measurement gear to nail, but is clearly worth the effort, and the same is true for dialing in DSP. In the vast majority of systems judicious DSP, particularly below 100hz, makes an enormous difference. In most systems correcting for time and phase alignment issues, including well above 100hz, also materially improves things.

That being said, in all cases you “lose something” with DSP, and for me the question has been “do the obvious large gains offset the obvious losses?”

FWIW I first got into DSP processing when using a Meridian 861 in a purpose built two channel room with big Infinity ribbons, big Krell LAT speakers, and AR/VTL/Aesthetix top of the line upstream gear. I later used heavily modified DEQX gear upstream of custom line arrays with each section driven by custom built 300B amps and then later BAT monoblocks. I currently switch between heavily tweaked convolution filters running through HQ Player and Fab Filter Pro running through J River, both running on purpose built servers. I occasionally use Trinnov or Dirac processing instead. In all cases I find the results more “different” than one being superior to another. In all cases I have found that the experimentation journey made me a lot more knowledgeable about what drives my listening joy in my room given my hearing and my preferences. The same is true for room treatments, but that’s for another thread.

 

[audiobomber]

Can one biamp a speaker and get performance that exceeds a single amp.

I experimented with horizontal biamping using a Meitner STR-50, Bryston 3B-ST and Classe 70, all solid-state, 50-100wpc. The speakers were two-way Energy Reference Connoisseur, 84dB/w/m @ 8 ohms, passive crossover at 1500Hz, with a severe impedance dip in the bass and a reputation for being difficult to drive.

While I could easily achieve better bass and treble with these amp combos, none were as satisfactory as using any of the amps by itself to power both channels. In all cases, the horizontal configuration resulted in a loss of coherence. The sound was more hi-fi, less musical and quite disappointing. I did not experiment with tube amps, but I find it very hard to believe that a SS/tube amp combo would make me happy, due to the different sound signatures.

I currently own two Meitner STR-50 Plus amps and a pair of Meitner MTR-101 Plus monoblocks. The stereo amps are rated for 50wpc and 10 amps. The monoblocks are rated for 100W and 35 amps. Topology of the amps is virtually identical. My speakers are Bamberg Engineering Sound Labs (4-ohms, 91dB@2.83V/m, crossover at 2000Hz). Using a single 50W Meitner stereo amp sounds OK, but vertical biamping with one 50W stereo amp per channel is a big improvement. Best sound is with a single 100W monoblock per channel. The extra current really grips those mid-woofers.

My conclusion is that vertical biamping with identical amps is an advantage, but you are better off using a single, better, power amp. On a related note, my experience with bridging a pair of stereo amps is that monoblocking is best left to sound reinforcement applications, and not for home audio, because it halves the load impedance, a major disadvantage.

There's more info on biamping in this article I co-wrote with John C Aussie for Audio Asylum:

Biamplification basics - an article - John C.-Aussie - General Asylum

Biamplification basics - an article - John C.-Aussie - General Asylum

www.audioasylum.com

 

[Kingrex]

I experimented with horizontal biamping using a Meitner STR-50, Bryston 3B-ST and Classe 70, all solid-state, 50-100wpc. The speakers were two-way Energy Reference Connoisseur, 84dB/w/m @ 8 ohms, passive crossover at 1500Hz, with a severe impedance dip in the bass and a reputation for being difficult to drive.

While I could easily achieve better bass and treble with these amp combos, none were as satisfactory as using any of the amps by itself to power both channels. In all cases, the horizontal configuration resulted in a loss of coherence. The sound was more hi-fi, less musical and quite disappointing. I did not experiment with tube amps, but I find it very hard to believe that a SS/tube amp combo would make me happy, due to the different sound signatures.

I currently own two Meitner STR-50 Plus amps and a pair of Meitner MTR-101 Plus monoblocks. The stereo amps are rated for 50wpc and 10 amps. The monoblocks are rated for 100W and 35 amps. Topology of the amps is virtually identical. My speakers are Bamberg Engineering Sound Labs (4-ohms, 91dB@2.83V/m, crossover at 2000Hz). Using a single 50W Meitner stereo amp sounds OK, but vertical biamping with one 50W stereo amp per channel is a big improvement. Best sound is with a single 100W monoblock per channel. The extra current really grips those mid-woofers.

My conclusion is that vertical biamping with identical amps is an advantage, but you are better off using a single, better, power amp. On a related note, my experience with bridging a pair of stereo amps is that monoblocking is best left to sound reinforcement applications, and not for home audio, because it halves the load impedance, a major disadvantage.

There's more info on biamping in this article I co-wrote with John C Aussie for Audio Asylum:

Biamplification basics - an article - John C.-Aussie - General Asylum

Biamplification basics - an article - John C.-Aussie - General Asylum

www.audioasylum.com

Great info. Thanks