Why Synergy horns?

In another thread I was asked, if I would provide more details about my speakers, so I thought why not?

I have played on active 4 way horn systems since 2016. First iteration was front loaded bass horn, midbass horn, tractrix midrange horn and tractrix tweeter horn. I worked nicely, with all the attributes associated with well implemented horns. Clarity, dynamics, realistic live sound etc.

However some problems will arise, with such horns. First of all, the center to center distance between the different horns is big, compared to the crossover frequencies. We need to be within 1/4 wave in distance at x-over for a seamless transition. For instance if you x-over from the midrange horn to the tweeter horn at 3 KHz the c-to-c distance would have to be 340/3000/4= 2.83 cm (1.11 inch). This is virtually impossible with "normal" horn configurations. This problem rears its ugly head, at every x-over throughout the audio frequency range. As frequency decreases, the wavelengths gets bigger, but so does the horns in the specific bandpass and then c-t-c also increases. It is a linear problem, that can't be solved with the regular approach, aka stacking horns on top of each other. This creates interference problems and lobing in the vertical response curves, that will color the reflection from floor and ceiling. Secondly a large column of vertically stacked horns, will push the sweet spot (SS) further back, for the horns to be perceived as more coherent and integrated, with one another.

But the biggest problem is that almost all horns beam with increasing frequency, it's their way of nature so to speak. What that means, is that the off-axis FR will not be similar to the on-axis FR. This translate into a poor power response, which is not considered a good thing, in terms of best sound quality.

Luckily we can circumvent all these problems with clever engineering and have our cake and eat it too, so to speak. Enter the Synergy horn.

 

I also believe any driver should have as wide a frequency band as possible. My conical horn field coil driver extends from 600-8000 Hz, for example. I think a 1 octave driver doesn’t make much sense - you can achieve multiple goals with a much wider range driver. Every transition from one driver to another carries a ton of issues and challenges.

Do you even know what thread you are reading? That is actually what the synergy horn is all about, but maybe you missed that point, plus all the others.

 

Remember what a DSP potentially replaces rather than being an addition per se (more an that later).

If you're using an existing, passively configured speaker setup and want to implement digital room correction via a DSP, then yes you're adding a processing layer in the digital domain. Evaluating its influence as such should be done with actual listening, which goes without saying, and then assessing whether its corrections, ultimately, are a worthwhile takeaway on the whole, but also if the means itself to do so (i.e.: the DSP and its corrections) are suspected to introduce a negatively perceived "processing imprinting."

I believe these two areas of evaluation could easily get mixed up; habitual exposition to the sound of one's setup over time can easily lead one to believe what's heard is, flaws and all, still a relatively "correct" representation of the recorded material or what is otherwise deemed fairly authentic. Or, simply that you've grown to like it for what it is in a more general fashion. The DSP-corrected sound at first then may seem a bit "strange" or boring even, if nothing else simply because it sounds.. different, now that obvious peaks and dips and possible timing issues have been smoothed out. Giving it a bit more time however the realization may seep in that what's heard with the corrections isn't that bad after all, but the real eye-opener may arise from giving it a week or so and then returning to the sound sans corrections - that's when it may dawn on you that you've lived with a sound that was flawed in vital areas.

It may also be that you've managed to bring about a sound from your setup with no means of digital correction whatsoever that's fairly "flat" and smooth and with no obvious timing issues. In such a case adding a DSP for digital room correction is likely just an unnecessary layer of processing. And yet from what I can assess, achieving this in the analogue domain to a degree that equals what a DSP is capable of would be no small feat in itself, and depending on one's specific setup - the combination of speakers and listening room in particular - it is nigh on impossible even. That's not necessarily to say you'd ultimately favor the DSP-corrected sound, even over time, and whatever the reasons for this it must always come down to that in the end: whatever you prefer after having given each approach its due time from thorough evaluation, is that which you prefer.

If assessing DSP however mostly or even exclusively comes down to mere assumptions and/or conjecture - that is, with no real hands-on experience to speak of - then it's time to instead open up about its possibilities a la giving it the benefit of the doubt, or try it out for oneself to get to know what it's all about. I'm guessing lack of experience and varying degrees of conjecture with regard to the use of DSP, in whatever function, is a predominant factor here..

Lastly, a DSP can alone replace a passive crossover as a digital, active ditto. This way it's placed prior to amplification which then requires additional amps to power each driver section directly. Used as such it brings with it some obvious advantages, not least of which is seeing the passive crossover removed between the amp and drivers. I may receive some flak for stating this, but I'd go so far to claim that whoever believes a passive crossover is the lesser evil compared to using a DSP sans passive crossover (and additional amps) for direct amp-driver control, hasn't heard a properly DSP-implemented, active setup to know the difference. Whatever impediments may be introduced with conversion steps involved in a DSP to my ears clearly is the lesser evil compared to the negative effects of passive crossovers, but that's just me.

That is pretty much a homerun post.

 

You are converting to and from digital, and each conversion degrades the sound considerably. Even a digital source will need to be converted twice, depending on where the DSP conversion is in the chain.
I have never seen a high end DSP solution, where AC to DC conversion is well thought through and noise is minimized.

Granted, a tiny bit of extra noise may be the penalty we pay to insert a DSP in the chain. A long as I can't hear that noise in my listening chair, I'm fine. Besides that I don't find that these conversion stages degrades the SQ when the DSP is bypassed, as in doing nothing.

One must distinguish between what a DSP does with the signal and what the person using the DSP does with the signal. An AD/DA conversion is in practice transparent, even run through multiple loops and a DSP that does nothing is also transparent. A DSP that actually does something is not transparent, that's the whole point of it.
My point is that it is entirely possible to implement DSP in your system without losing sound quality in practice. On the contrary, it will be much better, provided that it is done correctly of course.

The vast majority of systems/speakers in normal living rooms will give a strongly time distorted and uneven frequency response, that translate into a different sound of the instruments, than what is found on the recording. EQ is added to smooth out the frequency response to get much closer to the sound found on the record. Or in other words; we tend to like how the instruments really sound over the time and frequency distorted version that looks like a mountain range. I am surprised how many people that do not understand and accept this simple principle. DSP EQ from quality components adds vanishingly little artefacts in my experience, if any, beside maybe a tiny bit of noise.

I have a strong suspicion that this "conversion degrades the sound considerable" you talk about is not generated in the silicon processor, but is a faulty connection in the meat calculator that sits in the listener. Solution: To Hell with DSP

 

Room correction is best done by actually physically correcting the room, not adding DSP - it’s never going to be as good as actually fixing the problem(s) at the source.

I agree that fixing the problems at the root is the best solution. You just can't truly time align a speaker with acoustic treatments, you need a FIR filter to do that. Even a 1. order filter gives time distortion and puts a lot of stress on the drivers. As you move up in filter orders, these timing issues gets more severe.

Fixing frequency problems in the order of +/- 10 dB can't be done with room treatment, speaker placement, let alone changing speakers or upstream components, as much of these are room related, though some frequency deviations also are speaker related.

 

Just curious: given your stand on dsp, I wondered if you were to listen to any analog source ( vinyl or magnetic tape ) would you want to use dsp for the required source eq?

I would always use DSP EQ, no matter what source being used, because the biggest problems we have, are speaker and room related (an old but still very valid truth).

 

@schlager May I ask you another theoretical question?

If ( as it seems to be ) you are listening exclusively to digital files for music source, would you consider an audio system that is simplified down to a digital 'player' with volume control output as a digital signal to pro sound amps with onboard dsp and nothing in between ( except say a Dante interface if that's the chosen connection form ) to be a good option?

From reading your posts you use mini dsp, I can't remember the dac, and then various SS amps so this is more a question on 'packaging', and maybe on reducing digital transfer points versus anything new in signal path.

Yes, that is what I am actually doing. Using my laptop as streamer or tapping into my music collection on HD. From the USB, it goes into a USB/SPDIF converter that feeds my MiniDSP OpenDRC-DA8. It controls volume, DAC, x-over filter and some basic speaker nearfield EQ above 1000 Hz. So the signal is digital all the way from source to the output DAC's on the MiniDSP, before running into the amps.

 

Hello

There is no analog signal processing in a world class system??? So vinyl is out??? What do you think the RIAA curve is where the signal has EQ applied at the cutter head and restored in the preamp?????

That's funny on the whole as just about everything produced today is recorded digitally so essentially all modern source material has been corrupted through the use of DSP in production.

Rob

Thousands of LPs in my collection are testament that "just about everything produced today" is indeed not recorded digitally.
I would hate to live in a world where everything is recorded digitally!

DSP has it's place, for sure - like multichannel movie soundtracks. Or in more budget friendly systems. If a speaker costs under $10K let’s say, I get why DSP is important. Above a certain threshold, components and rooms can and should be designed to not need this crutch.

Digital music playback has its advantages, and can sound very very close to a great analog setup. But the act of converting the original analog -- and it will always be analog! -- source to digital and then back again, destroys so much of what makes the original soundtrack great.

Go and listen to a well setup world-class system that's purely analog. I am lucky enough to have both a world-class digital and analog sources and I can tell you anything converted to digital doesn't have the realism of pure analog.

RIAA is not a conversion from one medium/format to another.

EDIT: I have indeed measured the response in my room, courtesy of the Trinnov processor and its 3D mic (leagues ahead of anything from DSPMini, but that’s another matter). Unadjusted room response is +/- 5dB which is very good. I have compared the same track in pure analog without any digital manipulation and running it through the Trinnov, which corrects phase, time alignment, and frequency. Guess what - you think the Trinnov sounds great until you hear how much more realistic the original sounds without any digital correction.

This debate is more about budget and goals. DSP is great when budget or time don’t allow the root causes of problems to be addressed, and there’s nothing wrong with that. I don’t know what these horns cost, but I’m assuming they are budget friendly and DSP is an acceptable compromise.

 

Do you even know what thread you are reading? That is actually what the synergy horn is all about, but maybe you missed that point, plus all the others.

I saw the post about how you were trying to justify a 1 octave driver and replied to that.

 

I would always use DSP EQ, no matter what source being used, because the biggest problems we have, are speaker and room related (an old but still very valid truth).

You understand you are not fixing the room related and speaker problems you claim to have with DSP, I hope? Digital correlation is not a solution - take the time to fix the problems at the source and you will be rewarded with much better sound.
That's all I will say on this topic, as I doubt we will change each other's minds.

 

I saw the post about how you were trying to justify a 1 octave driver and replied to that.

Pls point to that post, because one of us made a mistake.

 

That is a pretty narrow effective frequency range for the horn ... 500 - 1000Hz. Only one octave. Is that a typical result for a Synergy horn? Why are the tweeters at such a low level?

Here it is.
Apologies if I seem to be coming on strong. As I said when I started, I applaud the innovation- we certainly need more of it- and I welcome any speaker that doesn’t look like a column .

I think it’s NB to “call a spade a spade” and not make it seem like a panacea.

 

You understand you are not fixing the room related and speaker problems you claim to have with DSP, I hope?

DSP is essential Digital Signal Processing, what we really are discussing is DRC Digital Room Correction, which is impulse response correction, does a bell ring? If not, you should really read up on the newer research done by John N. Mourjopolous and Angelo Farina.

 

Here it is.
Apologies if I seem to be coming on strong. As I said when I started, I applaud the innovation- we certainly need more of it- and I welcome any speaker that doesn’t look like a column .

I think it’s NB to “call a spade a spade” and not make it seem like a panacea.

Ah, thanks. Yes the midrange in a synergy horn function as a filler driver. Frequency span varies with different design, in my situation it covers 500-1200 hz and the tweeter cover from 1200-20000 over 4 octaves. But in reality it does not matter, what covers what, because the CD horn with all the drivers works, sounds and measures like a single wide band driver with no crossover.

 

 

EDIT: I have indeed measured the response in my room, courtesy of the Trinnov processor and its 3D mic (leagues ahead of anything from DSPMini, but that’s another matter). Unadjusted room response is +/- 5dB which is very good. I have compared the same track in pure analog without any digital manipulation and running it through the Trinnov, which corrects phase, time alignment, and frequency. Guess what - you think the Trinnov sounds great until you hear how much more realistic the original sounds without any digital correction.

Firstly, you most likely over did the correction, so artifact started to show up. Secondly, you probably haven't given yourself time to re-adjust to the new sound, and the infamous "skewed preferences" still lingers on.

 

Firstly, you most likely over did the correction, so artifact started to show up. Secondly, you probably haven't given yourself time to re-adjust to the new sound, and the infamous "skewed preferences" still lingers on.

No and no.
Can’t “overdo” the correction with a Trinnov. And there is no bias as I own all the components. Why would I be biased against something I own and can listen to anytime?
Try not to judge how others think, just a tip.

 

I also believe any driver should have as wide a frequency band as possible. My conical horn field coil driver extends from 600-8000 Hz, for example. I think a 1 octave driver doesn’t make much sense - you can achieve multiple goals with a much wider range driver. Every transition from one driver to another carries a ton of issues and challenges.

@schlager already explained the rationale behind the "filler driver," so won't get into that, but the Synergy horn is way smart in that it sums the output of its different drivers into a single point source with no driver-transitional issues.

Myself I plan on going the Synergy route down the road eventually, but until then I'm more than OK with the single large format mids/tweeter horn per channel of my main speakers that span ~600 to 17-18kHz (it controls directivity down to 500Hz, but sounds better crossed a bit higher).

What do you use below your field coil-loaded conical horns - direct radiating, large diameter woofer/mids, or horn-loaded dittos? Subs?

 

What do you use below your field coil-loaded conical horns - direct radiating, large diameter woofer/mids, or horn-loaded dittos? Subs?

I use a transmission line active sub, which is good down to 25 Hz (-5dB).

 

I use a transmission line active sub, which is good down to 25 Hz (-5dB).

Serving sub-duties only, or is there a midbass section as well?

 

Serving sub-duties only, or is there a midbass section as well?

Just up to 80Hz. The upper bass horns run from 80-600, midrange from 600-8000, super tweeter to over 20k.
BTW, the passive crossover has just 8 components.