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.

 

Reviewer Robert E. Greene, is actually one of the few reviewers, worth listening to (in my ears )
Listen what he has to say about EQ.

"If you have lived only in the ultra-purist high-end world of no user-controllable EQ, you will certainly find the Loki an educational experience. You will make a significant step towards understanding how audio actually works and how humans actually perceive sound."

His review of the intro equalizer model from Shiit is mandatory reading for audio purist (and everybody else for that matter). Have fun.

Schiit Audio Loki Mini Equalizer - The Absolute Sound

Schiit Audio Loki Mini Equalizer - Robert E. Greene contends that every music lover should have an equalizer—especially at the Loki Mini’s $149 price.

www.theabsolutesound.com

Sorry, but how does any of this relate to your synergy horns ?

 

Hopkins, nothing I guess the thread has "evolved" into some kind of blog, where I can write my nonsense, without disturbing the serious audiophile/purist threads
The design idea and function of the synergy horn is pretty much covered, I guess. As it is based on science and facts, there is not much to second guess. But questions are of course welcome.

 

Toole's research points out that good frequency response in the direct sound from the speakers is the most important characteristic of good sound, which also sounds logical and that our ability to listen beyond the room at high frequencies is quite good. Thus, he is skeptical to base the sound on a one-point measurement carried out in the listening position without general audio knowledge or knowledge to how this affects the speaker's direct sound.

People with good rooms and knowledge to tune DSP can improve the direct sound at high frequencies, there is no doubt about that in my eyes/ears but it's also a bit beyond the point Toole is making. Much of Toole's work is about the broad lines and about trying to break the famous "circle of confusion" and raise the standard of the whole industry (studios, manufacturers, consumers). I think he has very little to say about what everyone does in their own living room. For example, he plays with multi-channel himself to improve the spatial illusion, which may work on some recordings, even if it was not necessarily the way it was intended in the studio. As a kuriosum he uses a Trinnov DSP

 

@schlager I know this thread is yours, and about Synergy horns, I hope you wont mind this observation:

I heard the Dutch&Dutch speaker this past weekend at an audio show. It's not a horn, but this comment is on dsp...

To a purist the D&D does a lot wrong. Delay the main output to align with rear mounted woofers for example.

Yet the presentation was quite good, better sounding than many rooms. I attribute this to the basic acoustic technology used and the dsp that allowed the sales team to work with a few room issues. So there we were two powered speakers and a laptop for music source.. One of the cheaper and simpler systems at the show and sound was, while not my favorite, very good and perhaps 1/10 the cost of the better.

Just an observation to share

 

In this tread the ceiling is high and the door far away, as long as it is audio related, so thanks for your comment Solypsa.

I haven't heard the the Dutch&Dutch speaker myself, but the design philosophy tells me that it is a good speaker and the reviews (mostly non-affiliated reviewers) seems to confirm it.

There is actually more similarities between a synergy horn and the D&D speaker, than meets the eye. Most synergy horns are not born with DSP, but DSP are often inserted by the end user and most synergy horns out there in the living rooms are DIY. Danley have some DSP based models in the portfolio.

Both designs are waveguide based but the most important similarity is that both speakers control their dispersion pattern, but in different ways. A synergy horns dispersion pattern is defined by the angles and size of the horn mouth, while the D&D is using cardioid cancellation technique via acoustic cancellation, to control the dispersion below 200 hz. Dipole speakers behave in the same manners.

Most monkey coffins becomes omnidirectional (non-cardioid ) below 300 hz as the wavelength increases and the speaker baffle becomes small, compared to the wavelength. This rises two problems, 1. the dispersion control is lost below the Schroeder frequency (if there ever was one, to begin with) and 2. you get a suck out in the bass, due to cancellation with the surrounding surfaces, often around 80-200 hz leaving the mid-bass fatigue and anemic.

 

For anyone who has heard what good acoustics can do for the sound, it is obvious that the acoustics are hugely important. Anyone who has had good experience with DSP knows that this is an incredibly simple, cheap and effective way to improve sound quality. Anyone who has heard how good it can be when both parts are in place, knows that you are far from optimal sound if one of these is missing.

Fun Fact: It is extremely rare to see people embark on acoustic measures because they do not like the acoustics in the room. They do it because they don't like the frequency response of the stereo - in the room. Virtually everyone who is in this situation can reach the goal with DSP EQ/FIR. Yes - they might even be able to get by with traditional EQ too.

If one goes for a system with traditional tone controls, with enough EQ bands, you can generally achieve a reasonably even frequency response in the listening position. But with more precise correction you can achieve a psychoacoustic "disappearing act" of the speakers. With FIR EQ you can dampen/cancel bass resonance sounds in the room, so there is a large degree of overlap with what can be done with acoustic treatment. In relation to the line between acoustic treatment and DSP, the level of precision in the DSP filtering is crucial for whether it is meaningful to discuss where one should end and the other should begin.

 

Without an even frequency response, there is little or no credibility in the reproduction of voices and instruments. As a general rule, most struggle with an uneven frequency response in the bass, and a lower midrange that sucks, often due to SBIR generated problems, which reside in the category of boundary-induced comb filtering.

There are also many people who struggle with too much or too little treble. Some of these can be found in the cable corner and in discussions about matching components, looking for the great SYNERGY. The energy level in the treble is very dependent on the room it is in and the dispersion pattern of the speakers and many of the best speaker manufacturers have therefore - wisely - equipped their best speakers with a bit of primitive room/speaker correction in the form of resistors that can be switched in and out, so that you can increase the treble by a couple dB or reduce it by a couple of dB. But to cut a long story short: People move the speakers like carrying the rice home one grain at a time and are still unable to get a satisfactory frequency response. Enter with rockwool, diffuser and helmholtz resonators. It often gets much better, but it rarely gets good.

Smooth frequency response is such an important factor for reliable sound and trustworthy music reproduction, that almost everything else is subordinate as long as this is not in place. Most hifi systems that run without DSP/EQ correction have serious deficiencies in this respect. It doesn't seem like the number of digits on the price tag of the system makes much of a difference either. I have yet to hear a single uncorrected system that does not have clear sound emphases and timbral distortions that makes you sit, listening to the system instead of the music and it is totally incomprehensible to me, that people who dig deep into their wallets in search of good sound have not discovered this. DSP is such an obvious, simple and effective solution to the overriding FREQUENCY RESPONSE PROBLEM that many struggle with.

 

Besides an even frequency response the dispersion characteristic of the speakers are absolutely crucial. That is what gives good horns a bigger, clearer and more defined sound image. With big horns one has a better starting point when it comes to acoustic treatments and the decay of the room. If the decay as a function of frequency is uneven, this will color the sound. Uneven decay is not desirable, this will take form of coloring the tonal balance, and this coloring of the sound will be present even if the frequency response in the LP is nice and even. So a steady state frequency response in LP does not tell the whole story. If what comes out of the speaker is smooth and flat and there is unevenness in the frequency response in LP, then we can conclude that it is a result of the speakers dispersion and room acoustics.

So what happens in the time domain is interesting and will have a big influence on the sound we hear and how the experience will be.

Let's take a look at a filtered impulse response measurement I did on my system.



At time 0 we see a strong impulse, then initially after that a big dip, which translate to a strong direct sound wave, then some reflections after 5 ms, which ad to the spatial envelopment and then an even low decay. Not so bad for a relatively small room.

 

To DSP/EQ or not to DSP/EQ. When I talk to audiophiles I often get the feeling that they really do not want to optimize their sound. They tend to prioritize certain parameters, components or solutions that directly stops further development of the sound in their system. It seems that they think it is more important to have a nice box, a certain type of amplifier or to play analog sources etc. than to work further with DSP/EQ tools to integrate the speakers to the room. In many ways, they have already decided that it sounds good enough and the uncertainty as to whether there is anything more to gain, in terms of sound quality, is too great. It is a shame and in my experience with DSP/EQ tools, there is much more to be gained.

Some argue against such a solution because they have experienced that DSP/EQ actually degrade the overall sound. DSP/EQ are only tools and all types of tools have their strengths and weaknesses. If you use FIR filter with an automated process, you must take that into account in how you set it up and how you use it. As with all tools, it's mostly about the person behind the levers, that is the stopping block not the tool itself. The best programs with FIR are powerful tools with quite a few more options, that allow the user to tailor the speakers to the room, than than standard IIR EQ. However, this also greatens the potential for errors.

In the types of systems we are discussing here, we must think holistically in order to get the most out of the potential.
Let's take two seconds and think that you would have actually gotten better sound with a FIR-DSP/EQ solution and pro sound card.
You don't want this because:
- the box is perhaps uglier.
- you must use a PC.
- you use analog sources.
- you have not heard good DSP/EQ implementations.
- it is too complicated.

This is a choice only you can make, but in my mind it is a poor prioritization, if optimal sound quality is important to you.

 

in my experience with DSP/EQ tools, there is much more to be gained.

Until you compare with other approaches, you cannot really claim (and neither can I) that there is "much more to be gained", don't you think ? There certainly is a lot to be gained from correct acoustic behavior, but then there is the old "shit in / shit out" argument... I think we have been over this before

 

I have quite a lot of experience with different system approaches, both with and without EQ and those with EQ correctly implemented always sounds best to me. I started using DSP EQ in 2006, but that was IIR EQ. The last 5 years or so I have used FIR filter, so I know quite well what can be gained. In the end, we want to clean up the impulse response in the room, that is what the best modern software solutions can bring to the table. For example with the same EQ in place, it is very easy to demonstrate how more corrections in the time domain effects the clarity of the sound. The bass becomes tighter and the overall sound is more focused, dry and clear. The best DRC programs have group delay cancellation and by applying a greater time window, the program get more information of the impulse response of the room and can do a better job of cleaning it up. It's not to say "then just apply max correction", at some point you get artifacts and you have to back up a bit. How much correction is needed, has much to do with speaker directivity, room size, listening distance, acoustic treatment ect.

How DRC works in details is an extensive task to explain, but if you have never heard good implemented DRC, I can only recommend you to try it or find someone to demonstrate it for you.

Or read here for a more in depth guide. https://audiophilestyle.com/ca/ca-a...nd-room-correction-software-walkthrough-r682/

 

Neutrality is an ever recurring theme in the High Fidelity world. What is neutral? Why is it important?

First of all, I want a definition that goes as follows:
Neutral audio reproduction is a reproduction without signature/coloration.

What do I mean by signature?
In this context, a signature can be many different things. One example is jitter in the DAC. Another is deviation in the frequency response. An example of a signature could be harmonic distortion, another could be noise. There are an incredible number of things that can give a sound signature/coloration. Different signatures can be experienced as both positive and negative, but what they have in common is that there is something that repeats itself from recording to recording, which makes it more difficult to hear the particular signature in the recording in question.

How can different products provide a different signature?
Audio signatures varies greatly from hifi product to product. A product often gives several different types of signature at the same time, and only some of them can be compensated for. Forget to correct a DAC that does not suppress jitter. Intermodulation distortion is also impossible to correct for and if you try anyway, you will introduce new errors. With speakers, the picture looks a bit different.


So what about speakers and rooms? And what about the famous flat frequency curve, why doesn't it automatically sound good?
Loudspeakers must relate to a room in 3 dimensions. This gives a completely different type of error to which we react completely differently. For example if the speaker has an uneven sound dispersion and widens a lot at a certain frequency, there will be a lot of energy in the room at that frequency and you've got a very clear signature.

Measuring the sound energy in the room at each individual frequency is not straight forward either. There are lots of filtering and interpretation methods available, that gives different results. Our ears are quite interested in what is happening in the time domain, but at the same time they have an incredible ability to pick up the direct sound. In other words, everything we correct in the room will also become part of the direct sound, to which our ears are sensitive. Here the best DRC audio software comes into play.

So what is the correct sound?
There is only one correct point of reference in the audio world and that is the recording. Some recordings are bad others are good, but often it is not the ones people pick out as examples of good and bad, that actually are the good and bad. Often, recordings that are easy to reproduce are highlighted as good, and recordings that present challenges are often highlighted as bad. Some highlight certain recordings as good because they suit one's own setup, while others highlight other recordings again, because they have a setup that sounds different. Then you have to ask yourself what the value of frequency curves and basic references are. That is perhaps one of the easiest things to answer. There aren't really any types of music or recordings that benefit from intermodulation distortion. Nor does anyone enjoy peaks and dips in the frequency response. There are no types of music or recordings that benefit from strong resonances, or from speaker compression. No music types and recordings benefit from a lot of noise, lack of bass, poor tonality, etc. All kinds of basic errors drag down the total SQ. It is these errors that we must get rid of.

So matching components doesn't matter?
To some degree but matching shouldn't be about a light sounding X that is paired with a dark sounding Y. In a passive system we can have an amplifier that measures perfectly in isolation, but has an output impedance that does not match the speaker, which also in principle measures perfectly, but the two do not match. You can have the same problem between preamplifier and power stage, etc.

Conclusion:
Neutral sound is not about boring sound, it's not about bass-poor sound, or cold sound, or sound that lacks life. It's about sound that isn't dragged down by single factors. In the search for good sound, one should always look for the biggest mistakes, often found in the speaker/room interaction. For every error you manage to minimize, you will get more musical enjoyment, more life, more dynamics, more space, more details and not least, sound that gives you an insight into the differences that prevails between different recordings. Insight into what they wanted to achieve in the studio, the little thing they were looking for when the artist finally nodded in recognition, smiled and said "Now it sits!", that's how they wanted us to hear them.

 

Conclusion:
For every error you manage to minimize, you will get more musical enjoyment, more life, more dynamics, more space, more details and not least, sound that gives you an insight into the differences that prevails between different recordings. Insight into what they wanted to achieve in the studio, the little thing they were looking for when the artist finally nodded in recognition, smiled and said "Now it sits!", that's how they wanted us to hear them.

Very well put. +1

 

Very well put. +1

Agree.

 

I have quite a lot of experience with different system approaches, both with and without EQ and those with EQ correctly implemented always sounds best to me. I started using DSP EQ in 2006, but that was IIR EQ. The last 5 years or so I have used FIR filter, so I know quite well what can be gained. In the end, we want to clean up the impulse response in the room, that is what the best modern software solutions can bring to the table. For example with the same EQ in place, it is very easy to demonstrate how more corrections in the time domain effects the clarity of the sound. The bass becomes tighter and the overall sound is more focused, dry and clear. The best DRC programs have group delay cancellation and by applying a greater time window, the program get more information of the impulse response of the room and can do a better job of cleaning it up. It's not to say "then just apply max correction", at some point you get artifacts and you have to back up a bit. How much correction is needed, has much to do with speaker directivity, room size, listening distance, acoustic treatment ect.

How DRC works in details is an extensive task to explain, but if you have never heard good implemented DRC, I can only recommend you to try it or find someone to demonstrate it for you.

Or read here for a more in depth guide. https://audiophilestyle.com/ca/ca-a...nd-room-correction-software-walkthrough-r682/

Hi, thanks for putting down detailed thoughts. Do you have videos of your EQed synergy horns playing classical?

 

Do you have videos of your EQed synergy horns playing classical?

No, but I can make one, but it will be on my Iphone 7, so don't expect high SQ. It will be on System 2, which is the one that is hooked up at the moment.

As a kuriosum I had some middle aged longtime audiophiles visiting the other day. We played both System 1 and 2. System 1 is my original big synergy horns. Old ears listening carefully to it

You don't see any bling in any of my systems, all cables are DIY. MiniDSP, china chip amps, Behringer PA amps for sub duty ect. They were quite shocked at the SQ. First response was "it just sounds so effortlessly".
Later I had a chat with one of the guys and he said that he had never heard a bass that dry and precise combined with power and slam.

System 2 under "review".

 

No, but I can make one, but it will be on my Iphone 7, so don't expect high SQ. It will be on System 2, which is the one that is hooked up at the moment.

A lot of people do phone videos here, and understand the limitations. So don't hesitate!

 

Neutrality is an ever recurring theme in the High Fidelity world. What is neutral? Why is it important?

First of all, I want a definition that goes as follows:
Neutral audio reproduction is a reproduction without signature/coloration.

What do I mean by signature?
In this context, a signature can be many different things. One example is jitter in the DAC. Another is deviation in the frequency response. An example of a signature could be harmonic distortion, another could be noise. There are an incredible number of things that can give a sound signature/coloration. Different signatures can be experienced as both positive and negative, but what they have in common is that there is something that repeats itself from recording to recording, which makes it more difficult to hear the particular signature in the recording in question.

How can different products provide a different signature?
Audio signatures varies greatly from hifi product to product. A product often gives several different types of signature at the same time, and only some of them can be compensated for. Forget to correct a DAC that does not suppress jitter. Intermodulation distortion is also impossible to correct for and if you try anyway, you will introduce new errors. With speakers, the picture looks a bit different.


So what about speakers and rooms? And what about the famous flat frequency curve, why doesn't it automatically sound good?
Loudspeakers must relate to a room in 3 dimensions. This gives a completely different type of error to which we react completely differently. For example if the speaker has an uneven sound dispersion and widens a lot at a certain frequency, there will be a lot of energy in the room at that frequency and you've got a very clear signature.

Measuring the sound energy in the room at each individual frequency is not straight forward either. There are lots of filtering and interpretation methods available, that gives different results. Our ears are quite interested in what is happening in the time domain, but at the same time they have an incredible ability to pick up the direct sound. In other words, everything we correct in the room will also become part of the direct sound, to which our ears are sensitive. Here the best DRC audio software comes into play.

I think this all is an interesting theoretical exposition.

So what is the correct sound?
There is only one correct point of reference in the audio world and that is the recording.

What does this mean in practice, given that we cannot decode the "correct point of reference" of the recording without playing it back on a stereo system? Once we are back to the stereo replay doesn't neutrality become wholly subjective?

Conclusion:
Neutral sound is not about boring sound, it's not about bass-poor sound, or cold sound, or sound that lacks life. It's about sound that isn't dragged down by single factors.

Do you consider the "dryness" some people subjectively hear from some solid-state electronics to be a "single factor"? To be a "coloration"?

 

To DSP/EQ or not to DSP/EQ. When I talk to audiophiles I often get the feeling that they really do not want to optimize their sound. They tend to prioritize certain parameters, components or solutions that directly stops further development of the sound in their system. It seems that they think it is more important to have a nice box, a certain type of amplifier or to play analog sources etc. than to work further with DSP/EQ tools to integrate the speakers to the room. In many ways, they have already decided that it sounds good enough and the uncertainty as to whether there is anything more to gain, in terms of sound quality, is too great. It is a shame and in my experience with DSP/EQ tools, there is much more to be gained.

Some argue against such a solution because they have experienced that DSP/EQ actually degrade the overall sound. DSP/EQ are only tools and all types of tools have their strengths and weaknesses. If you use FIR filter with an automated process, you must take that into account in how you set it up and how you use it. As with all tools, it's mostly about the person behind the levers, that is the stopping block not the tool itself. The best programs with FIR are powerful tools with quite a few more options, that allow the user to tailor the speakers to the room, than than standard IIR EQ. However, this also greatens the potential for errors.

In the types of systems we are discussing here, we must think holistically in order to get the most out of the potential.
Let's take two seconds and think that you would have actually gotten better sound with a FIR-DSP/EQ solution and pro sound card.
You don't want this because:
- the box is perhaps uglier.
- you must use a PC.
- you use analog sources.
- you have not heard good DSP/EQ implementations.
- it is too complicated.

This is a choice only you can make, but in my mind it is a poor prioritization, if optimal sound quality is important to you.

Respectfully, don’t agree at all.
DSP is a “Band-Aid” and shortcut to achieve goals that can be achieved without DSP. The question becomes how much time, talent, and resources, does one have to achieve these goals without DSP? In a budget-friendly system, DSP is fine and a welcome improvement.
But for a product here in the “What’s Best” Forum, DSP is a not a solution I would ever buy into.
Why?
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.

It’s refreshing to see different paths and ideas, and I don’t mean to criticize the fresh ideas.
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.

 

Respectfully, don’t agree at all.
DSP is a “Band-Aid” and shortcut to achieve goals that can be achieved without DSP. The question becomes how much time, talent, and resources, does one have to achieve these goals without DSP? In a budget-friendly system, DSP is fine and a welcome improvement.
But for a product here in the “What’s Best” Forum, DSP is a not a solution I would ever buy into.
Why?
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.

I had similar questions concerning DSP in the previous pages of this thread. But passive filtering also degrades the sound, and filterless systems have their limitations as well. Each configuration has advantages and disadvantages.