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Not according to this article

https://www.headphonesty.com/2025/07/elitism-lazy-marketing-killing-audiophile-hobby/

Audio shows are everywhere now. During the first half of the year, there's one almost every month. But for at least one longtime dealer, they've become more of a drain than a strategy.

    "The amount of shows is stupid and the results from shows is even stupider," he said bluntly.

To be clear, he isn't against shows completely. He's done plenty of them. But in his experience, they rarely bring in new customers or deliver truly impressive demos. Most visitors aren't there to buy. They're just passing time.

He called it "audio tourism." People poke their heads in between errands or while waiting on their partners, with no serious interest in gear. And even when they do sit down to listen, the sound usually isn't anything special.
"Most of the sound at audio shows is just average," he said. "I don't think most people go to an audio show and get their doors blown off."

Some of the industry's frustration now comes from logistics. In 2025, several major events like AXPONA, SIAV in Shanghai, and Kaohsiung Hi-End Show all happened on the same weekend. This forced brands to pick sides on where to spend money and manpower.

And that money doesn't go far. Booth costs, labor, shipping, and lodging have all gotten more expensive, yet most companies still have no clear way to track return on investment.

Even shows that report big turnouts aren't necessarily bringing in the right crowd. AXPONA 2024 hit over 10,000 visitors, but the new Gen Z ticket tier, despite some growth, still made up a small slice of attendees.

    "It's the same thing over and over and over again," he said. "There's nothing unique about any of them."

He doesn't think shows need to disappear. But if the goal is to grow the customer base, this isn't the way.

IMO the only benefits appear to be for the organizers who make a killing from renting out pint-sized hotel rooms with poor acoustics, and for a few days only whilst all of the exhibitors have to pay extra staff to do the heavy lifting and grunt work !! It's literally money for old rope for the organizers

Using Audioweaver, the Ultimate Preamplifier, and rePhase, we were able to implement a linear-phase crossover on the onboard SHARC DSP running at 192 kHz without much effort at all. For this example, we chose a simple 2-way crossover centered at 1 kHz and a response that mimicked an 8th-order Linkwitz-Riley filter, in which both sections exhibited an in-phase response of -6 dB at the crossover point.

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Using Audioweaver, the Ultimate Preamplifier, and rePhase, we were able to implement a linear-phase crossover on the onboard SHARC DSP running at 192 kHz without much effort at all. For this example, we chose a simple 2-way crossover centered at 1 kHz and a response that mimicked an 8th-order Linkwitz-Riley filter, in which both sections exhibited an in-phase response of -6 dB at the crossover point.

Although both the non-minimum phase version and the linear-phase version of the Lintwitz-Riley crossover have identical summed flat magnitude responses and are also in-phase at the crossover frequency, the linear-phase crossover, as its name suggests, has a linear phase response. This means that once both the low and high pass sections are summed together, the total output is just a delayed version of what is being fed into it. This is in contrast to the non-minimum phase version, which distorts the signal passing through because of its non-constant group delay. This linear distortion should not be confused with non-linear distortion, such as harmonic or intermodulation distortion.

Linear phase filters are a special class of FIR filter that sums to a unity response with a fixed time delay. So when both high-pass and low-pass sections are added together, the response is a delayed version of what is being fed into the crossover. As I have discussed in another thread, this type of filter can have issues with proper impulse response cancellation due to imperfect matching between drivers, which can result in pre-ringing leakage among other issues. The proponents of these types of filters will never talk about this aspect of this type of filter topology, instead pretending that it doesn't exist or is inaudible. Whilst pre-ringing at ultra-sonic frequencies in DAC reconstruction or oversampling filters, etc, is essentially inaudible, pre-ringing at audible frequencies in loudspeaker crossovers can be audible and unwelcome!

As in the group-delay corrected filter, this time we use rePhase to create the linear-phase FIR filter coefficients as in the following setup:-



RePhase can create complementary high- and low-pass filters. However, we will use a simple trick of subtracting the high-pass filter output from a delayed version of the filter input to create the low-pass complementary filter output without requiring twice the computation resources! This is crucial when using lower-powered DSP devices such as the SHARC DSP used in our preamp.

We set optimization to "Moderate" and hit the "Generate" key, and then we create the following correction filter! Note that rePhase always creates a correction filter with a net 0-phase response! This results in a non-causal impulse response that occurs before the impulse, which, whilst mathematically succinct, is physically unrealizable. To overcome this, rePhase allows you to realign the impulse response by delaying it and windowing it so that no response occurs before the impulse. In this case, this was done by delaying and centering the impulse response at the halfway point (i.e., half tap count) and using a Hanning window to mask out anything before the impulse. The delay of exactly half the tap count of the filter is what we use to subtract from the output of the filter to create the complementary low-pass filter, and later, when we measure the outputs of the filters, you will see they are both complementary filters with identical slopes. When summed together, we get a unity output with a linear-phase response! Now we just have to enter the filter coefficients into Audioweaver using the following test bench we have created to illustrate one channel only.



The highlighted green connection lines show the signal path for the high and low-pass filter sections, consisting of the FIR high-pass filter section and the low-pass section, which subtracts the high-pass from a delayed version of the input signal to create the low-pass filter. And we also include the delayed filter input and summed outputs routed to the other unused preamplifier outputs, which facilitate measuring the filter response. This helps us do differential phase tests; otherwise, the net phase response contains so much delay that it is very hard to visualize the shape of the phase response curve. For those playing along at home, the resource usage for this 2048 tap FIR filter is quite modest at only 14% ! And for a stereo version using identical coefficients for both channels as you would normally use, the SHARC DSP supports SIMD instructions, which means that it can process two multiply-accumulates (MACS) in one clock cycle so we would expect there would be no increase in resources for a stereo version of this filter !

The magnitude response of both the low and high pass filters from the dScope analyzer is shown below. The blue trace is the low-pass filter response, and the red trace is the high-pass filter response, whilst the grey trace is the summed response, and the orange trace is the relative phase difference between the summed output and the delayed input:-



Now, here are some test results from the scope at various frequencies. The yellow trace is the delayed 1 kHz square-wave input signal. The blue trace is the low-pass filter output. The magenta trace is the high-pass filter output. As expected, the bottom green trace is the summation of both the low and high-pass sections, which, of course, is identical to the delayed input signal!

One thing that should be abundantly clear in the traces above is the amount of pre-ringing or the response before the leading edge of the square wave that these filters exhibit, and which is not present in the IIR version, which is also shown below for comparison! In theory, this may not be an issue on-axis where both speakers crossover with essentially a uniform frequency response, and the pre-ringing should essentially cancel out, but this cannot be guaranteed off-axis, so we would expect imperfect impulse response cancellation to occur and some pre-ringing to leak through. However, maintaining a perfect uniform on-axis response would also be a significant challenge, especially as the speakers' characteristics drift over time with no mechanisms to correct for this. Whilst pre-ringing at ultrasonic frequencies in a DAC reconstruction filter is inaudible, pre-ringing in the audible bandwidth is not! Now let's compare this with the non-minimum phase Linkwitz-Riley 8th-order filter!

Shown below the top yellow trace is the input signal, which is the leading edge of a square wave. The blue trace is the low-pass output of the linear phase filter, while the magenta waveform is the low-pass filter output of an 8th-order non-minimum phase LR crossover (LR-8). Note how the LR-8 filter has no output before the leading edge, while the linear phase variant exhibits pre-ringing energy before the leading edge! This is because a linear-phase filter always has a symmetrical impulse response. In contrast, as used in the non-minimum phase variant, a causal recursive filter cannot have a symmetrical impulse response, which is why it produces nothing before the impulse.

Now we compare the high-pass filter sections of the same crossover. Again, the top yellow trace is the input signal, which is the leading edge of a square wave. The blue trace is the high-pass output of the linear phase filter, while the magenta waveform is the high-pass filter output of an 8th-order LR crossover (LR-8) implemented using IIR filters. Note how the LR-8 filter has no output before the leading edge, while the linear phase variant exhibits pre-ringing energy before the leading edge!

The issue here is that unless two speakers are perfect both on and off-axis, you will always get imperfect impulse response cancellation and thus have issues with pre-ringing artifacts, which are totally unnatural and can be audible !! The reason you use a crossover in the first place is that drivers aren't perfect devices; you need to attenuate their out-of-band characteristics as much as possible, and the last thing you need is the crossover introducing potential artifacts at the crossover frequencies, which the speaker drivers can reveal !

Whilst pre-ringing at ultrasonic frequencies, such as that from a DAC reconstruction filter, is essentially inaudible, pre-ringing at audio frequencies is not! This is why FIR filters with cutoff frequencies selected to fall within the audible band could be problematic! Note the absence of this issue from protagonists of these types of filters. They will argue that pre-ringing is not audible below a certain threshold, which they don't quantify, let alone admit even occurs. To put that into perspective, let's have a look at the frequency response of a typical premium quality midrange driver shown below. This driver and its variants have been utilized in numerous high-end speaker systems. Even if we applied high-order linear phase filters to filter out the undesirable out-of-band anomalies, the in-band response still shows quite a bit of response irregularity, thus bringing into question its ability to suppress pre-ringing entirely.

Now to answer a long-lost question that was posted on our www.diyaudio.com thread at the time of our pre-order for the UP2 and UPP some years back and by what appears to be a protagonist from the DIQX camp that obviously had no intention of buying anything from us other than a desperate attempt to dissuade people from it then yes it is not true as stated by this individual that it is hard to implement a linear phase crossover on our preamp using Audioweaver. As the preceding demonstration illustrates, it is quite simple, but of course this type of filter comes with some caveats of its own, which have already been highlighted by a number of esteemed audio engineers, so with the Ultimate Preamp, you get to choose your poison.

Second, what I really like about my DIQX crossover at the moment is the implementation of linear-phase crossovers (FIR). The software treats it just the same as Linkwitz or Butterworth crossovers. Just choose it from the list, that simple. All other DSP solutions so far expect you to be a DSP expert to be able to implement linear phase crossovers. I was looking for this in audioweaver but so far have not found it yet. Maybe I'm missing something. Now before anyone likes to discuss the downsides of linear phase crossovers and why I would want them but there is only one simple answer: Because it sounds better in my system and except from linear phase crossover I can't think of anything why I would need the amount of DSP power that is on offer in the Ultimate Preamp

Sorry, but you are misinformed. You don't have to be a DSP expert to implement a linear phase crossover. In actual fact, you don't have to write a single line of DSP code as the above example demonstrates. Applications such as Audioweaver, as used by the Ultimate-Preamp, encapsulate all of the low-level DSP code for you, but if you still want to whine about that, then perhaps you should not get involved with DSP and active speakers in the first place

And from another thread about DSP.

REW and Acourate are at the opposite end of the spectrum. Both are very manual tools, and the tools need to be deployed in the correct order and in the correct situation. When I say "manual", I mean that you have to inspect the measurements yourself and decide what you want to do. Acourate has a few more "luxury" features compared to REW, which is why I prefer it. REW can not be used on its own, you need RePhase. Both are extremely flexible, but also close to impossible to use if you do not know what you are doing.

Sorry, but our example above disproves this statement as an exaggeration of the truth. As I have demonstrated, a little bit of elbow grease gets you a simple linear-phase two-way crossover straight off the bat without much effort at all. Not only that, you do not need any additional convolver software, as Audioweaver provides all of the necessary modules to begin with. Similarly, it is easy to compare one filter topology with another and run different filters concurrently as we did above when we directly compared the linear-phase LR-8 FIR-based crossover to the non-minimum phase LR-8 IIR-based counterpart. Alternatively, you could easily construct a multi-way crossover with non-minimum phase IIR filters and then apply global phase correction to linearize the phase as we did in our example on another thread. In each case, it is not difficult to do using Audioweaver and free software such as rePhase. Anyone telling you otherwise either doesn't understand or has ulterior motives to push people towards a particular brand of software or hardware which is probably why this person never mentions Audioweaver, or if they do, they then dismiss it as being overly complicated and aimed at engineers and as shown above nothing could be further from the truth. Have a nice day

There is a school of thought that says the only way to accomplish a linear phase crossover is to use linear phase filters which are a special class of FIR filter that sums to a unity response with a fixed time delay. So when both high pass and low pass sections are added together the response is a delayed version of what is being fed into the crossover. As I have discussed in another thread this type of filter has issues with proper impulse response cancellation due to imperfect matching between drivers which can result in severe pre-ringing leakage among other topics. The proponents of these filters will never talk about this aspect of this type of filter topology, instead pretending that it doesn't exist or can easily be minimized so that it is never a problem.

The other way of linearizing the crossover is to take an existing non-minimum phase crossover and apply a global group-delay correction filter which is just an FIR filter with a large number of taps. The advantage here is there are no FIR filters in the actual crossover to generate any pre-ringing artifacts. This is the approach taken by a few commercial designs as well as here, although the correction filter usually requires a large number of taps and has a large delay depending on the sampling rate and tap count. So this should be taken into account if you want to use these speakers with other speakers or in conjunction with a video display in say a home theater system in order to avoid lip-syncing issues.

We took a four-way 8th-order Linkwitz-Riley crossover from one of our clients who own one of our Preamps and applied a global group-delay correction filter to linearize the phase response. The LR crossover was already flat but the phase response was anything but linear so a square wave fed into it did not look like a square wave coming out of it. A linear phase response means that there is just a time delay in the signal path whilst still maintaining a flat magnitude response but there is no waveform distortion.

To measure the phase distortion of the crossover in isolation we needed to sum the outputs of the crossover before they were level adjusted and time delayed so we could focus on the crossover alone without the effects of the speaker and the room etc which we know adds its own artifacts to the response. Later on, we will look at compensating for those as well but for now we are just focussing on fixing up the crossover and making sure the group-delay correction filter is doing its job properly.

To evaluate the crossover we just looked at one channel and summed the outputs to measure the net frequency response. We also fed the input to the crossover to one of the outputs of the preamp as a reference for the analyzer to create the frequency response. The summed outputs of the crossover are fed to another channel on the preamp. We also added a few multiplexers or selector switches to enable us to switch in and out various filters quickly so we can quickly evaluate the effects of different filter configurations. Both the input to the crossover and the summed outputs are then connected to our dScope III audio analyzer. The dScope III can measure both amplitude and interchannel phase. The following Audioweaver test bench is what we used to evaluate the effectiveness of the global correction filters.



All of the DSP is done on a PC instead of the Preamp itself because the group-delay correction filters required are beyond the resources of the onboard SHARC DSP. The Ultimate Preamplifier Plus (UPP) has a unique feature in that it allows one to run DSP on an external device such as a PC as though it was running inside the Preamp. In fact, in another thread, we proved that the audio path was transparent and the performance of the Preamp was unaffected by the noisy environment of a PC ! The Ultimate Preamplifier Plus can also share DSP between its own onboard DSP as well as an external PC so for example you could build a non-minimum phase LR crossover using the onboard SHARC DSP whilst running the correction filter on an external PC and then switch the correction filter in or out from the Preamp menu. This is the approach our client is taking but for proof of concept and convenience, we ran everything on an external PC which allowed us to quickly evaluate and test everything in the one environment. The main aim was to validate the effectiveness of the global group-delay correction filters and prove that you don't need to use dedicated linear-phase filters to achieve this.

First, we measured the frequency and phase response of the crossover filter alone using a log frequency axis. We note the flatness of the filter as expected but the phase is anything but flat ! The top blue trace is the sweep of the input which is ruler flat as expected and the bottom red trace is the crossover summation of all drivers which is also ruler flat as expected.



To evaluate the phase response we reran the frequency response test using a linear frequency axis instead of the usual log axis. If the phase is linear then the phase response should look straight and not curved. Since the analyzer cannot distinguish phase greater or less than 180 and -180 degrees respectively it wraps or folds the phase so the response always is contained within a 360 degree envelope which makes it look like a saw-tooth. From the phase plot below we can see the phase is anything but linear which was expected !



And now for the money shot. What does this crossover filter do to a square wave? The square wave is important because it is not just a single tone but rather a series of odd harmonically related tones with a monotonically decreasing magnitude. If the phase is not linear, it will manifest itself as distortion in the waveform even though the magnitudes of the harmonics are still correct. So, let's look at the test results on a scope.

Firstly at 1kHz and then at 100 Hz !! This is what a non-minimum phase crossover does to a square wave !! Can we fix this ??

The following Jitter tests illustrate the benefit of using the built-in Inter-sample Overs Guard in preventing internal numerical overflow inside the asynchronous sample rate converter of the SHARC DSP. We used the same J-test methodology that we used for our other tests.

Firstly the following shows the scope screen capture of one channel of the output of the Preamp with the J-test signal from the analyzer applied directly to the USB port of the Preamp. For all intents and purposes, it should look like a 12kHz sinewave but because of Intersample Overs, there is internal overflow in the sample rate converter which manifests itself as waveform distortion similar to clipping which is a side effect of upsampling or interpolating to a higher sampling rate.



The following shows the resultant FFT spectrum and notable artifacts can be observed.



Expanding the frequency scale of the spectrum above shows in much more detail the problems caused by the inter-sample overs.



With the Inter-samples Guard switched on we noted how many times the sample rate converter was overloading in the statistics collecting mode ! Once activated the Guard reduced the signal input level to avoid any more internal arithmetic overflows and reported that the signal had to be reduced by 3dB to achieve this.



And the signal from the output of the Preamp was reduced in amplitude by 3dB but is no longer visually distorted!



Now let's see the effect on the J-test spectrum from the analyzer. A vastly much improved and cleaner spectrum is observed below when the sample rate converter is not overloading and essentially proves how transparent the SHARC sample rate convert can be !



The expanded frequency scale of the above FFT spectrum is shown below !

Someone emailed me recently after I posted the results of routing digital audio through an external PC for the purposes of expanding the DSP capability of the Ultimate-Preamp and asked whether or not using a PC adds any additional noise, distortion or jitter to the audio stream ? Well I said in theory it shouldn't due to the isolation and transparency provided by this interface but there is only one way to find out and that is to measure it.

So using the same J-test methodolgy that we used for the normal jitter tests on the various inputs to the preamp we can now apply the same test methodology whilst routing the audio through an external Dell PC used in our lab and see how it affects the jitter and noise. So we setup Audioweaver on a PC to process audio from the Preamp at 192kHz in a direct input-to-output connection as in the following:-



Then we measured the jitter from the output of the Preamp using the same J-test methodology as we used on earlier tests of jitter.



The screenshot below is the same as above but with an expanded frequency scale. As can be seen the spectrum is as clean as it is with internal DSP processing and any artifacts are way below the threshold of hearing !



We did the same for THD+Noise.



As can be seen in the test results above the PC is essentially transparent to the audio stream and does not add any noise, distortion or jitter to the incoming audio stream thanks to the hardware of the Ultimate Preamplifier Plus. This is contrary to the reputation of PC's as a noisy and hostile environment due to high-speed digital design and a switching power supply not optimized for low noise. However, the UPP mitigates these issues with ease without the use of re-clockers, network switches, fancy USB cables, add-on FIFO's etc, which would not improve anything or could indeed make things worse simply because our hardware is designed right in the first place and the above measurements are proof of that



.

Please click on the following link for the latest firmware updates for the UP1,UP2 and UPP. (You need to be a registered member of this forum to access this page.)

https://analog-precision.com/forum/firmware-updates/firmware-update-24/

Please click on the following link for the latest firmware updates for the UP1,UP2 and UPP. (You need to be a registered member of this forum to access this page.)

https://analog-precision.com/forum/firmware-updates/current-firmware-update/

[DSP loudspeaker crossovers done right]

Don't be fooled by the blurb on the DEQX website regarding group-delay correction or compensation of loudspeakers. It's a way of oversimplification of reality and they have been pushing the same barrow for decades which is probably why they have never commercialized their own speaker systems using it, instead pushing the gimmicks onto other naive loudspeaker designers. Trying to fix up a loudspeaker on a single axis was never going to work. Loudspeakers are far more complex than this and radiate into free space in all directions and not just on one axis alone! It's easy to apply an inverse convolution filter to some point in space and say "can you hear the difference ?" but is it an improvement or a step backward towards ultimate accuracy in a practical listening environment ?



What would fix up a signal chain on an electrical connection simply won't work with a distributed device such as a loudspeaker. This is why no serious high-end audio manufacturer would ever use a DEQX! Yes, sure our own  UPP can do long-length high sample rate, ultra-deep convolution, and FIR filtering and do a much higher resolution correction than a DEQX could ever do but do you really want to go down this path?  When it comes to loudspeakers there simply is no free lunch here and no easy fix and throwing lots of DSP at the problem can often make the problems a lot worse. It certainly is not a panacea for a bad speaker design to begin with!

Anyway don't take our word for it but we suggest you read these three important documents below before outlaying your hard-earned money on such a device or anything that is just a fancy inverse convolution filter ! These are documents written by other well-established audio engineers and echo the same sentiment as us. They tell a completley different story to what DEQX does. This is what DEQX will never tell you and why they never sell loudspeaker systems themselves using their own DSP hardware and even if they did it was always going to be a band-aid solution to a complex problem ! It all boils down to who are you going to believe ? These well-respected eminent audio and electrical engineers or some marketing hacks trying to jump on the DSP bandwagon whilst telling everyone how they should be designing speakers !

https://www.grimmaudio.com/wordpress/wp-content/uploads/speakers.pdf

And the take-home message from this document is:-

DSP loudspeaker crossovers done right

From the above we've learned that:-

• Heavy-handed correction exacerbates acoustical problems
• Sharp, linear-phase filters cause pre-ringing
• Targeting an exact linear-phase sum can cause pre-echos.

In short, brute-force correction sounds grainy and smudgy. When you hear cymbals go "splash" instead of
"crash", it's naive DSP at work. So:-

• Do not shave off the hair, a nasty stubble will grow back.
• Do not correct beyond the very beginning of the impulse response.
• The gentler you correct, the wider the angle over which the correction still improves things.
• Target a minimum phase sum.

For the time being I would strongly recommend designing the correction manually. This rules out FIR as the
main workhorse. For each bump or dip one corrects, one should know exactly where it comes from, and make
sure that it isn't better corrected for acoustically. Unfortunately, designing DSP filters does not relieve one from having to know one's acoustics.

Also:-

https://linea-research.co.uk/wp-content/uploads/LR%20Download%20Assets/Tech%20Docs/CrossoverFilters%20White%20Paper%20-C.pdf

Also shown in Figure 8 is the impulse responses for a complementary high-pass brick-wall FIR. If we construct a crossover filter bank with such a complementary pair of filters, the Gibbs ripples are also complementary (since we will expect the filters to sum to a flat response with linear phase, producing a perfect impulse). The summed result will thus be free of any Gibbs ripple, so what's the problem? The problem is off-axis. The complementary ripples will only cancel if the delay suffered by the signal from each driver is identical. Off-axis, where the path lengths differ, the ripples will not cancel, leading to the possibility that Gibbs ripple might become audible (just like a high-Q ringing filter).

Such summing errors will be more pronounced at higher crossover frequencies because the ripples are more closely spaced. Lower frequency
crossovers will have wider ripples which will more easily cancel in the presence of off-axis induced delays.

It is evident that steeper cut-off slopes give rise to Gibbs ripples of greater duration. It makes sense therefore to restrict the cut-off slope to be no more than is necessary for the application.

And from the Linkwitz Lab website pretty much the same cautionary tale

https://www.linkwitzlab.com/frontiers.htm

I - Digital crossovers

Some people think that digital crossovers will replace analog ones, because digital filters can be designed with desirable characteristics that are impossible to realize with analog circuitry. In particular, lowpass and highpass filters with extremely steep slopes and linear phase shift are possible. Steep slopes reduce the overlap region between drivers. Linear phase shift eliminates waveform distortion and merely causes a delay of the signal.  Such characteristics can be obtained from the digital equivalent of tapped delay line filters, which have a finite impulse response (FIR) duration that depends upon the number of taps used. Digital FIR filters can have almost any desired frequency response, if the number of weighted taps is made sufficiently high. [1]

The linear phase shift comes at a price. The impulse response rings. The more so, the steeper the filter slopes. Both lowpass and highpass sections of the crossover ring, but when the outputs are combined, as for a crossover, then the two impulse responses add to a non-ringing, delayed pulse.

All would be fine, if we listened only in anechoic spaces or to speakers with coincident drivers. In reality we use speakers in rooms with reflections and reverberation and the the drivers are separated from each other due to their sizes. As a consequence the off-axis response of the speaker matters and contributes to what we hear. With the drivers non-coincident, the lowpass and highpass outputs are delayed different amounts at points off-axis, and the ringing is no longer canceled in the addition. In the best case the drivers might be coaxial, but this has another set of problems. Very steep crossovers can also cause a very abrupt change in the polar pattern of the speaker, when transitioning from a large diameter driver to a small one. Under reverberant conditions and/or listening off-axis this may have audible consequences.

And the proof is in the eating of the pudding. If linear-phase filters were all they were cracked up to be then Linkwitz would have incorporated them into his LX series open baffle designs and yet he settled on a standard crossover design implemented using either analog or digital IIR filters. There are no FIR filters in his designs

Please feel free to add any technical material you may have on these speakers.

https://www.audioasylum.com/messages/speakers/48322/review-dunlavy-audio-labs-millennium-speakers-review-by-brian-a-at-audio-asylum

After many months of research I went with these speakers for the following reasons.
Extreme tonal accuracy, +/- 1dB from 20Hz to 20KHz.
Non-ported enclosure for clean bass.
Fabric dome tweeter for smooth treble.
Phase/time aligned for accurate transient reproduction.
Non-parallel cabinet walls to minimize internal standing waves.
Quick signal drop off in waterfall plots.
Clean pulse and step response.
Auditioned them and enjoyed the best sound I have ever heard!

It is hard to describe these speakers for I believe they are pretty much fully accurate and transparent. They totally disappear when playing despite their 76 inch height and 300 lbs. each weight. When you slip in a SACD/CD the wall dissapears and becomes a music stage for the musicians. Others have noticed this too and are quite dissapointed when they get home only to find their speakers don't do the disappering act, you can hear where each speaker is. Dunlavy's are matched to withen 0.5dB of each other to ensure this.

These speakers have a certain cleanness and accuracy that is very hard to describe. When I was at the factory in Colorado springs and did A/B comparisons with other Dunalvy speakers these just sounded more open. It was like the band was in the open, on a hill outside or something. With the other speakers, though they sounded great, the music was slightly veiled. I believe this is born out by the waterfall plots of the Millenniums, very clean and quick decay.

The soundstage of these is excellent. With some source material you can hear sounds coming from beyond the left and right of the speakers. Instruments are accurately placed across the area between, and beyond, the speakers. The speakers totally disappear, when the singer and drums are between the speakers, that is where the sound is, you don't hear any sound from the speakers themselves. The sweet spot is only wide enough for one though. However, anywhere in the room the sound is tonally balanced and enjoyable, you just don't get that nice balanced imaging you do in the middle. This is what totally turned me off of planer speakers, their horrible off axis tonal balance, and their laughable bass for that matter.

Perhaps the most noteable aspect of the Millenniums is there stupendous bass. It is so clean you sometimes think its lacking. There is absolutely no boom, overhang, thump, or distortion, even at high levels. I must admit I am used to hearing the 100Hz hump on most cheaper speakers that simulate a good bass response. I notice songs sound different. The bass is very clean and deep too. You can feel deep bass that you couldn't with lesser speakers. Lest you think these speakers are lacking in bass, if you play a song that truely has bass, these will reproduce it in spades. It is just amazing to listen to songs and for the first time hear and feel the bass/drum line of the song very cleanly. There is lots more detail and nuances to be found in the deeper few octives of music than I was aware of before.

The mids are flawless too. Voices are clean and undistorted. Dianna Krall sounds fabulous. Here music with the bass and piano sounds simply stunning. Celine Dion sounds excellent too. The harshness and forwardness of her voice is subdued to some extent, more enjoyable. Though with some of her songs, let's face it, she's really hollering. Men's voices also are nice, never nasally or boomy, always clean.

The treble is so clean and accurate, it captures the metalic sheen of cymbols and other metalic instruments but you need SACD to really enjoy this. There is no brightness, exageration, or harshness to be found, unless it is in the recording. When I compare the treble of these to that of my Paradigm Studio 100's in the home theater room, there is no comparison. Fabric dome tweeters are some much cleaner than metal domes, more natural and enjoyable.

The speakers themselves are a joy to look at. Their looks really grow on you. You start to appreciate their unique hourglass shape. It is so gracefull. My wife, who about flipped when she first say them, way too big and ugly as sin. Now she likes their looks. When we have visitors the women are usually amazed my wife allows me to have them. All the guys drool. You can tell there is some prime sound able to be produced by these massive transducers. The speakers are actually larger than I am, I'm 6-3 and 250 lbs. The speakers have an inch and 50 lbs on me. My son calls them the speaker gods.

About the only thing I can say bad about them is how they reveal poor recordings. Imagine my dismay when two of favorite groups as a teenager have some bad recordings, Fleetwood Mac and Abba. I love listening to SACD music with these. Horns and violins are richer and not strident as they can be with normal CD's. Cymbols are more metalic. The soundstage is enhanced.

Is this the perfect speaker? Probably not, but I am very happy with it. To me it sounded even better than Dunlavy's top of the line SC-V and SC-VI's. For someone with some money to burn, fly out to Colorado and listen to them for yourself. I don't think any dealer carries these for demos.

Read up on the manufacturer's website for the theory behind these speakers. The time alignment of the drivers, the hourglass shape etc. It is www.dunlavyaudio.com. Good reading.

This forum was created in remembrance of the late John Dunlavy and his contribution to both the art and science of audio reproduction which has lived on in many of the loudspeaker systems that he created over the decades and are still being used today. Not only that many of the original concepts developed by John Dunlavy such as baffle felt have been used by many other speaker designers with little or no credit given back to John.

Click here to view the all new Dunlavy/Duntech Archive Forum

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Sovereign 2001 Brochure

hfe_duntech_sovereign_2001_brochure_en.pdf

https://classic-hifi.net/product/duntech-sovereign-2001/

https://web.archive.org/web/20040906211613/http://www.apogeespeakers.com/reviews/a_search_for_the_ultimate_speaker_bestofaudio.htm

[Sovereign 2001]

Sovereign 2001

Click here to download brochure hfe_duntech_sovereign_2001_brochure_en.pdf

[(from left to right: Z6, ULC Reference RCA, ULC RCA, ULC Balanced Reference)]

    (from left to right: Z6, ULC Reference RCA, ULC RCA, ULC Balanced Reference)
     
     
     
      Most audio system components, such as pre-amps, DAC’s, etc., are far from distortionless, regardless of their cost or level of design sophistication. In many instances, audible distortion can be traced to non-linear distortion products caused by load capacitance that adversely affects the transfer characteristics of the circuit. This problem is often encountered when the interconnect cable connecting a pre-amplifier to a power amplifier exhibits excessive capacitance (usually greater than 100 pico-Farads). Unfortunately, it tends to be a frequent occurrence with very high quality pre-amps that exhibit a high slew rate (a desirable property), but have a tendency to become unstable when feeding a capacitive load. But most interconnect cables exhibit nominal capacitance values that range from about 20 to more than 50 pico-Farads per foot. When this capacitance is considered in parallel with the input capacitance of typical power amplifiers, the total can easily exceed 100 pico-Farads for an interconnect cable only 1 meter in length; enough to become the source of audible distortion in many systems.
     
      In addition to causing the production of non-linear distortion products (harmonic, I.M., and transient types), such excess capacitance can also result in a roll off of the amplitude response at high frequencies, resulting in a loss of definition and/or the blurring of complex musical transients.
     
      Dunlavy Audio Labs ULC (Ultra Low Capacitance) Interconnect cables, with a nominal capacitance of a truly remarkable 8 pico-Farads per foot, ensures distortion free transmission for virtually all applications. This means that the signal remains virtually unchanged from one component to the next allowing the listener to hear the music with unparalleled clarity and resolution, even in extremely long runs.
     
      With the amount of components in today’s high end audio/video systems, one shouldn’t have to worry about cables negatively affecting the signal. With Dunlavy Audio Labs ULC Interconnects, you don’t have to.
     
      DAL ULC interconnect are available with standard RCA terminations, reference RCA terminations, and balanced (XLR) terminations.
     
     
       

Electrical:
Impedance:	186 Ohms nominal
Resistance per foot                 (D.C.):	Less than 180 milli-Ohms for center conductor and less than 4 milli-Ohms for shield
Total loss for 3m length:	Immeasurable over the 20 Hz to 20 kHz range
Capacitance per foot:	8 pico-Farads nominal
Inductance per foot:	290 nano-Henrys nominal
Amplitude and Phase response:	Into an 10 K Ohm load, an 8 foot length of DAL ULC cable exhibits a drop of only 0.5 dB and a phase lag of only 30 degrees at 5 MHz (immeasurably flat from D.C. to 250 kHz)
Pulse-coherence factor:	Terminated in 10 K Ohms, an 3 meter length of DAL ULC interconnect will reproduce 1 MHz square waves and 1 micro-second wide pulses with an accuracy indistinguishable from the output of the signal generator
Velocity of Propagation:	Greater than 90%, relative to light in free space
Physical:
Size:	0.265 inches in diameter
Color:	Neutral tan
Lengths:	Standard: 1 meter; 2 meter; 3 meter; 7 meter
Optional Lengths:	Lengths from .5 meter to 30 meters +
Terminations:	ULC Standard: RCA                 ULC Reference: Neutrik RCA                 ULC Balanced Reference: Neutrik XLR
- All terminations are heavily gold plated.
Inner and Outer Conductors:	Patented multi-strand ultra pure silver plated copper wiring

     

[Guaranteed Electrical SpecificationsSeries Resistance:]

Truly unique "LCR ULTRA" satisfies the need for an affordable loudspeaker cable with nearly ideal electrical properties, i.e., ultra-low loss, maximum bandwidth, pulse-perfect performance, etc.

Used within high quality audiophile systems, "LCR ULTRA" loudspeaker cables possess all the electrical properties required to ensure that no audible degradation of complex musical waveforms and transients can occur between the output terminals of a power-amp, and the input terminals of a loudspeaker.

Best of all, perhaps, its simple design and relatively low manufacturing cost result in a retail price that is well below that of most popular high-end audiophile loudspeaker cables.

And, unlike so many of today's expensive loudspeaker cables that exhibit questionable electrical attributes, DAL's "LCR ULTRA" cable was designed by a staff of highly competent engineers working in the exceptionally well-equipped laboratory facilities of Dunlavy Audio Labs.

Guaranteed Electrical Specifications

Series Resistance: less than 0.001 Ohms per foot.

Series Inductance: less than 0. 17 micro-Henrys per foot.

Parallel Capacitance: less than 20 pico-Farads per foot.

Attenuation: Virtually immeasurable at frequencies below 20 kHz, for a 20-ft. length terminated by a 4-Ohm resistive load.

Velocity of Propagation Factor: Greater than about 70% of the velocity of light in a vacuum.

Standard Lengths: 8, 12, 16, & 24 ft. (Other lengths are available by special order.)

End Connectors: Heavily gold-plated, large spades

Size: Approx. 0.75 in. wide by approx. 0.35 in. thick.

"LCR Ultra" Loudspeaker Cable is a Trademark of Dunlavy Audio Labs, Inc. (Patent Applied For)

[(from left to right: Z6, ULC Reference RCA, ULC RCA, ULC Balanced Reference)]

    (from left to right: Z6, ULC Reference RCA, ULC RCA, ULC Balanced Reference)
     
     
     
      For many years, the design and performance of loudspeaker cables has been a subject shrouded in mystery. Comparison of cables, it seems, has been limited almost entirely to the vagaries of subjective evaluation. The extravagant and undocumented claims touted in almost all advertisements for cables has led sensible audiophiles to suspect that a substantial amount of snake-oil and buzzard-salve have been the key components of their "high-tech" designs. What has been lacking, but sorely needed, has been a sober approach based upon the application of sound engineering considerations and well known scientific principles.
     
      Now, with the Z6, a cable exists that was designed according to sound engineering and scientific principles. It’s accurately documented performance properties are truly relevant to how the cable interacts with the loudspeaker and the power amplifier - essential to obtaining accurate, unaltered reproduction of the most complex musical transients. It is a cable with a difference that can be measured by credible objective comparisons with other cables, offering the potential for audible improvements in music and movie sountrack reproduction. The Z6 represents a real engineering advance at an affordable price. Compared to other loudspeaker cables selling for much more, the DAL Z6 is truly without peer in every performance category.
     
      For example, most audiophiles are familiar with the need for carefully selecting a low-loss cable with an impedance that matches that of their television sets as a means of reducing picture blurring and ghosting. Unlike other loudspeaker cables, which exhibit impedances that range from 20 to over 100 Ohms, DAL Z6 exhibits an impedance of only 6 Ohms, insuring and excellent match to well-designed loudspeakers with average impedances ranging from 4 to 12 Ohms. As a consequence, Z6 loudspeaker cable eliminates almost all blurring and ringing of complex musical transients that other cables with much higher impedances cannot accurately reproduce.
     
      Further, the extremely linear amplitude and phase response of the DAL Z6 insures perfect reproduction of even the most delicate and detailed nuances found in music, compared to even the most expensive cables which suffer typical losses exceeding 2-3 dB at only 20 kHz.
     
      So forget all the subjective and fanciful claims of the competition, and recognize the distinct and real advantages that sound physics and proven scientific principles used in the design and manufacturing of the Z6 allow. It’s better because it is.
     
      Dunlavy Audio Labs Z6 loudspeaker cable is a Stereophile "Recommended Component."
     
      The DAL Z6 cable’s design is covered under United States patent #5,510,578 (April 23, 1996).
     
     
       

Electrical:
Impedance:	6 Ohms, nominal.
Resistance per foot (D.C.):	Less than 2.5 milli-Ohms
Total loss for 8ft. length:	Immeasurable over the 20 Hz to 20 kHz range into an 8 Ohm resistive load
Capacitance per foot:	340 pico-Farads
Inductance per foot:	21.8 nano-Henrys
Amplitude and Phase response:	Into an 8 Ohm load, an 8 foot length of Z6 cable exhibits a drop of only 0.5 dB and a phase lag of only 30 degrees at 5 MHz (immeasurably flat from D.C. to 250 kHz)
Pulse-coherence factor:	Terminated in 8 Ohms, an 8 foot length of DAL Z6 cable will reproduce square waves up to 1 MHz with an accuracy indistinguishable from the output of the signal generator
Physical:
Size:	0.5 inches in diameter
Color:	Neutral tan
Lengths:	Standard: 8 foot; 12 foot; 16 foot; 24 foot singles and pairs
Optional Lengths:	Lengths from 6 inches to 100 feet
Terminations:	BN4: Banana pin                 GS4: 1/4 inch spade                 GGS4: 3/8 inch spade
- All terminations are heavily gold plated in six inch lengths. Terminations employ unique threaded bolt application system for easy end user installation.
Internal Wiring:	Patented multi-strand high purity silver plated copper wiring. (US Patent # - 5,510,578; 4/23/96)