• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

DEQX Premate 8 digital active crossover / DSP

There simply is NO massive advantage of using linear phase digital filters in a practical multi-way loudspeaker environment.
How would you correct phase shifts that require advancing the signal in time i.e. introduced by crossover filters in a 3-way loudspeaker at the LP? Minimum phase filters cannot respond to future inputs (they are causal) and lack a negative time component unlike a reversed-in-time all-pass FIR filter. That's quite a massive advantage IMO.
 
All right, the usual arguments against linear phase FIR vs. minimum phase IIR are the following:

1. High computation requirements vs. high CPU efficiency.
2. High latency vs. low latency.
3. Pre and post ringing vs. post ringing alone.
4. No phase distortion vs. phase distortion.

I am not going to discuss 1 and 2 to keep this post shorter. I generated these illustrations myself.

View attachment 397765

Let's take a square wave and apply a linear phase low pass filter and high pass filter to it. If we look at the LPF and HPF, we see that both filters have pre and post ringing. However, when the LPF and HPF are summed, the pre and post ringing disappears. Thus, LP filters rely on proper summation of LPF and HPF, otherwise pre and post ringing will be exposed.

What would cause the LPF and HPF to sum improperly? Here are a few causes:

1. Asymmetric HPF/LPF slopes
2. Improper time alignment between drivers
3. Different directivity different drivers which means they sum on-axis but not necessarily off-axis. Linkwitz made this observation.
4. High Q aggressive phase corrections. Putzeys made this observation.

I should add that the mathematical causes for pre-ringing are now well understood and there are algorithms that can reduce pre-ringing to negligible levels. Since you bring that up, let's look at the on-axis step response of my speaker.

View attachment 397768

We can see a little bit of pre-ringing. Now let's go 30 deg off axis:

View attachment 397769

See much difference in pre-ringing? Even if there was, there is the phenomenon of pre-masking where any signal, provided it is low in amplitude and occurs less than 20ms before the main signal will be masked by the main signal.

Now let us look at what happens if we were to use a min phase filter. Let us take a square wave again and apply min phase LPF and HPF to it.

View attachment 397770

Look at the output. The phase is completely distorted, and the post ringing does not go away. Now, you could argue that phase distortion is not audible, and post-ringing is masked by the main signal. Ohm's Acoustic Law and all that. 99.99% of speakers in the world use minimum phase instead of linear phase, and all of them distort phase in the same way. Quite simply we are used to hearing phase distortion like this.

Now I will wander into more controversial territory. I have the capability to generate both min phase and lin phase filters. I have directly compared them. There is no contest, linear phase sounds so much cleaner. The last time we met (maybe you have forgotten) you mentioned Bohdan Raczynski. You might be interested in what he has to say about linear phase. He cites a number of sources that show that there is increasing research that shows that phase distortion is audible.

So, do we choose linear phase, with negligible and inaudible pre-ringing? Or minimum phase with unavoidable phase distortion and twice as much post-ringing which can never be removed?

To wrap up - we should ask ourselves why should we choose FIR rather than IIR. FIR allows us to use min phase or lin phase. IIR forces us to use min phase only. We can get halfway there with mixed phase, but with very limited FIR taps. The ONLY valid reason to avoid using FIR is to avoid latency which is critical in AVR's so there are no lip sync issues. The other reason of course is so that you can use underpowered SHARC DSP chips which can't do high tap count FIR, put it into products like MiniDSP and sell it for cheap.

I am not letting DEQX off the hook here. I think it is incredibly overpriced for what it delivers. I will be meeting Kim and Alan tomorrow and I will have a chat with them.


I know this is a bit of a different thing but here is my DEQX Pre-8 created 3-way speaker step response. Measured inroom, 300cm distance.
Yes, there is a bit of a pre ringing but so what. The sound is excellent.

1728577118129.png
 
I know this is a bit of a different thing but here is my DEQX Pre-8 created 3-way speaker step response. Measured inroom, 300cm distance.
Yes, there is a bit of a pre ringing but so what. The sound is excellent.

View attachment 397831
Up to -5ms, it's inaudible. -10ms starts pushing it.
 
So in this case it is inaudible.
Yes and even you could go up to 20% there safely in the 5ms region. And the nice thing is, within that range, you can fix all crossover phase shifts and even most port/box phase shifts easily. But try to correct a phase shift caused by a wall reflection and the pre-ringing will go easily beyond -20ms and there even 2-3% deviations become audible.

PS I am talking from exhaustive tests and experience. I don't have a good explanation for it.
 
The benefit of FIR is very minor IMO. At certain speaker crossovers (and where FIR had its strength) it may not be an audible advantage at all.

I would worry more about the lack of minimum phase of the room and treat that first if the goal is really better and accurate sound.
 
How would you correct phase shifts that require advancing the signal in time i.e. introduced by crossover filters in a 3-way loudspeaker at the LP? Minimum phase filters cannot respond to future inputs (they are causal) and lack a negative time component unlike a reversed-in-time all-pass FIR filter. That's quite a massive advantage IMO.

You add additional delay so that you produce a net linear phase response which can only really be done using an FIR filter but of course the drawback is this will incur more overall delay.
 
All right, the usual arguments against linear phase FIR vs. minimum phase IIR are the following:

1. High computation requirements vs. high CPU efficiency.
2. High latency vs. low latency.
3. Pre and post ringing vs. post ringing alone.
4. No phase distortion vs. phase distortion.

I am not going to discuss 1 and 2 to keep this post shorter. I generated these illustrations myself.

View attachment 397765

Let's take a square wave and apply a linear phase low pass filter and high pass filter to it. If we look at the LPF and HPF, we see that both filters have pre and post ringing. However, when the LPF and HPF are summed, the pre and post ringing disappears. Thus, LP filters rely on proper summation of LPF and HPF, otherwise pre and post ringing will be exposed.

What would cause the LPF and HPF to sum improperly? Here are a few causes:

1. Asymmetric HPF/LPF slopes
2. Improper time alignment between drivers
3. Different directivity different drivers which means they sum on-axis but not necessarily off-axis. Linkwitz made this observation.
4. High Q aggressive phase corrections. Putzeys made this observation.

I should add that the mathematical causes for pre-ringing are now well understood and there are algorithms that can reduce pre-ringing to negligible levels. Since you bring that up, let's look at the on-axis step response of my speaker.

View attachment 397768

We can see a little bit of pre-ringing. Now let's go 30 deg off axis:

View attachment 397769

See much difference in pre-ringing? Even if there was, there is the phenomenon of pre-masking where any signal, provided it is low in amplitude and occurs less than 20ms before the main signal will be masked by the main signal.

Now let us look at what happens if we were to use a min phase filter. Let us take a square wave again and apply min phase LPF and HPF to it.

View attachment 397770

Look at the output. The phase is completely distorted, and the post ringing does not go away. Now, you could argue that phase distortion is not audible, and post-ringing is masked by the main signal. Ohm's Acoustic Law and all that. 99.99% of speakers in the world use minimum phase instead of linear phase, and all of them distort phase in the same way. Quite simply we are used to hearing phase distortion like this.

Now I will wander into more controversial territory. I have the capability to generate both min phase and lin phase filters. I have directly compared them. There is no contest, linear phase sounds so much cleaner. The last time we met (maybe you have forgotten) you mentioned Bohdan Raczynski. You might be interested in what he has to say about linear phase. He cites a number of sources that show that there is increasing research that shows that phase distortion is audible.

So, do we choose linear phase, with negligible and inaudible pre-ringing? Or minimum phase with unavoidable phase distortion and twice as much post-ringing which can never be removed?

To wrap up - we should ask ourselves why should we choose FIR rather than IIR. FIR allows us to use min phase or lin phase. IIR forces us to use min phase only. We can get halfway there with mixed phase, but with very limited FIR taps. The ONLY valid reason to avoid using FIR is to avoid latency which is critical in AVR's so there are no lip sync issues. The other reason of course is so that you can use underpowered SHARC DSP chips which can't do high tap count FIR, put it into products like MiniDSP and sell it for cheap.

I am not letting DEQX off the hook here. I think it is incredibly overpriced for what it delivers. I will be meeting Kim and Alan tomorrow and I will have a chat with them.

Well done with that analysis. Agree with you about the audibility of phase distortion not so much for continuous wave repetitive signals but more noticeable on transient signals with a long interval. Also most speakers with a flat frequency response usually have a non-minimum phase response ie they have transmission zeros in the right half plane ie LR4 filters etc. This can be observed in their step response. There are very few speakers that have an overall minimum phase response such as Dunlavy and Duntech. Their overall response approximates to a 2nd order high pass filter with a monotonically decreasing phase response which may explain the appeal of such speakers.

Also the lower the order the filter slopes the less errors summing between drivers so automatically going for the highest slope possible doesn’t necessarily give you a free lunch. In the end what Putzeys proposed was to design the crossover using say an LR4 IIR crossover with linear phase correction using an FIR filter and that’s where a PC will come in handy especially at the higher sampling rates. However, the cost is increased latency and more processing power.
 
Anyway, your ASR reporter is back. I saw Alan Langford from DEQX at a Melbourne audio show today. So here is some DEQX news.

1728638562029.png


This is Joseph, Chief Engineer for DEQX explaining how linear phase XO's work.

1728638585584.png


This is their new 3 channel Class D amplifier based on Purifi modules - the Ampy. It is so new that it is not on their website yet. I forgot to ask them about more details about this amp. I will be back there tomorrow.

I again reiterated my concerns about DEQX pricing to Alan. I thought that the reason why DEQX was so expensive was because they are writing their own software (see earlier in this thread) and therefore had to amortize their cost over fewer units. Alan said no, the reason why it is so expensive is because of the components. I was a bit surprised to hear this. He said that each component costs a couple of dollars instead of a couple of cents. So I asked the obvious question - "why did you do this? Nobody can hear the difference between a SINAD of -100dB and -118dB (which is what DEQX is claiming). The difference will be swamped by DSP". He said that they wanted to make a flagship product that can challenge the best DAC's on the market.

While I was talking to him, we were interrupted by another show goer who asked "why does it cost A$23,000?". His reply was "it performs as well as a DCS, does more than a DCS, and costs less". I watched the other showgoer for a reaction and I saw his furrowed brow. Exactly right, I would have furrowed my own brow if I wasn't being careful about remaining deadpan. I don't think they will convince the high end crowd by quoting a SINAD of -118dB. To reach that market, you need to sell romance. Tell stories. A DCS Ring DAC and their history of making DSP units for the Royal Air Force tells a story. Using off-the-shelf ESS DAC's does not. To be clear, I am not sceptical about their choice of ESS DAC's. It is a fine DAC, and as far as I am concerned it will be great for the job. I even give them the benefit of the doubt that it has -118dB SINAD. What I AM sceptical about is the wisdom of marketing a DSP unit to the high end crowd who are still wedded to turntables and priced like a high end DAC.

So I asked the next obvious question. "Why don't you bring out a cheaper DEQX using less expensive components?". He said that a cheaper DEQX is under development and will be released once the software is ironed out.

Which brings me to the software. Last year, it had alpha status. Now it is in beta. They actually had a working DEQX Premate 8 on display. Last year, it was a dummy unit and they were using an older DEQX HDP-5. I expressed my disappointment that development was going so slowly.

I was also told that they have a new guy working with them - Glenn Dickins. For those who don't know, Glenn Dickins wrote the first paper describing linear phase in the AES in 1996. I did not know he was an Australian!

Anyway, I will be back at the show tomorrow. I did not chat to Kim today, I could not find him. Happy to field any questions from ASR to DEQX.
 
Anyway, your ASR reporter is back. I saw Alan Langford from DEQX at a Melbourne audio show today. So here is some DEQX news.

View attachment 397977

This is Joseph, Chief Engineer for DEQX explaining how linear phase XO's work.

View attachment 397979

This is their new 3 channel Class D amplifier based on Purifi modules - the Ampy. It is so new that it is not on their website yet. I forgot to ask them about more details about this amp. I will be back there tomorrow.

I again reiterated my concerns about DEQX pricing to Alan. I thought that the reason why DEQX was so expensive was because they are writing their own software (see earlier in this thread) and therefore had to amortize their cost over fewer units. Alan said no, the reason why it is so expensive is because of the components. I was a bit surprised to hear this. He said that each component costs a couple of dollars instead of a couple of cents. So I asked the obvious question - "why did you do this? Nobody can hear the difference between a SINAD of -100dB and -118dB (which is what DEQX is claiming). The difference will be swamped by DSP". He said that they wanted to make a flagship product that can challenge the best DAC's on the market.

While I was talking to him, we were interrupted by another show goer who asked "why does it cost A$23,000?". His reply was "it performs as well as a DCS, does more than a DCS, and costs less". I watched the other showgoer for a reaction and I saw his furrowed brow. Exactly right, I would have furrowed my own brow if I wasn't being careful about remaining deadpan. I don't think they will convince the high end crowd by quoting a SINAD of -118dB. To reach that market, you need to sell romance. Tell stories. A DCS Ring DAC and their history of making DSP units for the Royal Air Force tells a story. Using off-the-shelf ESS DAC's does not. To be clear, I am not sceptical about their choice of ESS DAC's. It is a fine DAC, and as far as I am concerned it will be great for the job. I even give them the benefit of the doubt that it has -118dB SINAD. What I AM sceptical about is the wisdom of marketing a DSP unit to the high end crowd who are still wedded to turntables and priced like a high end DAC.

So I asked the next obvious question. "Why don't you bring out a cheaper DEQX using less expensive components?". He said that a cheaper DEQX is under development and will be released once the software is ironed out.

Which brings me to the software. Last year, it had alpha status. Now it is in beta. They actually had a working DEQX Premate 8 on display. Last year, it was a dummy unit and they were using an older DEQX HDP-5. I expressed my disappointment that development was going so slowly.

I was also told that they have a new guy working with them - Glenn Dickins. For those who don't know, Glenn Dickins wrote the first paper describing linear phase in the AES in 1996. I did not know he was an Australian!

Anyway, I will be back at the show tomorrow. I did not chat to Kim today, I could not find him. Happy to field any questions from ASR to DEQX.


Any details about the new amplifiers? Price, power, availability...?

When is pre-8 software ready for launch?
 
Just one man's opinion but I would never again buy an expensive piece of gear dependent on a small company's proprietary software when much cheaper options are available which let the user choose the software from a large variety of sources. The struggles they seem to be having getting this product to market in a finished form is a red flag. I will stick with an Okto dac8 and computer based software, thanks.
 
Any details about the new amplifiers? Price, power, availability...?

When is pre-8 software ready for launch?

It was my intention to ask about the amplifiers but I ran out of time. Too many people to chat to. I will be back in the show in a few hours so I will ask them.

They said that the software will be out of beta in Q1 2025.

Just one man's opinion but I would never again buy an expensive piece of gear dependent on a small company's proprietary software when much cheaper options are available which let the user choose the software from a large variety of sources. The struggles they seem to be having getting this product to market in a finished form is a red flag. I will stick with an Okto dac8 and computer based software, thanks.

I agree. I actually asked Alan whether they would consider opening up the DEQX to accept standard .WAV files so that you can use Acourate to generate corrections. He didn't answer the question, he went off on a tangent and said that it is an excellent product and how much he liked it. I will press him for an answer today.
 
Re: the amplifiers.

"How much power do they make?"
DEQX: "400W into 4 Ohms."

"How much will it cost?"
DEQX: "I don't know, I haven't priced it up yet."

"When will it be available?"
DEQX: "We are having trouble getting the metalwork manufactured. Probably Q1 2025".

Re: why the software is still in beta.

"We had trouble with one of our programmers who wrote messy code. We parted ways with him and we have a new guy cleaning up the code. It should be out of beta before the end of the year".
 
Ah !
Thats Nice as a Bèta user I’am glad to hear ;-)
Thanx for asking !
Sound is great but it lives a life of its own sometimes .
But i knew that so no complanes !
 
Re: the amplifiers.

"How much power do they make?"
DEQX: "400W into 4 Ohms."

"How much will it cost?"
DEQX: "I don't know, I haven't priced it up yet."

"When will it be available?"
DEQX: "We are having trouble getting the metalwork manufactured. Probably Q1 2025".

Re: why the software is still in beta.

"We had trouble with one of our programmers who wrote messy code. We parted ways with him and we have a new guy cleaning up the code. It should be out of beta before the end of the year".

400w x 3?
 
Yes. 400W into each channel.

Oh, and I did press them for an answer about opening up the DEQX to accept .WAV files generated by Acourate/Audiolense. The answer: "We have no philosophical objection to it. But we need to sort out our problems first as a priority before we think about creating a pathway to host corrections generated by third party software".
 
Given the abundance of Class D amplifiers on the market, I struggle to understand why they are launching new amplifier models instead of focusing their efforts on finally completing and stabilizing their DEQX PreMate 8/4. I was particularly interested in the Pre-4, but I have no desire to be a guinea pig at that price.
 
Yes. 400W into each channel.

Oh, and I did press them for an answer about opening up the DEQX to accept .WAV files generated by Acourate/Audiolense. The answer: "We have no philosophical objection to it. But we need to sort out our problems first as a priority before we think about creating a pathway to host corrections generated by third party software".t
Thank you for your effort Keith !
Any impression of the sound ?
sinds you have been there I am bit curious
 
Back
Top Bottom