Now I know why their products are so expensive.
DEQX Pty Ltd Announces Expansion
DEQX is expanding with some more audio legends!
27TH NOVEMBER 2024, Sydney Australia
We are excited to announce the addition of several award-winning audio pioneers to our team.
Joseph Narai – Chief Operating Officer
Joseph started with us part time as R&D Manager in February this year and joined full time as COO in July. Joseph, a seasoned technology innovator with over 35 years of experience, has held prominent leadership roles having founded multiple audio companies over the last three decades. His achievements include multiple design and industry awards including the Innovation in Media Award from the National Association of Broadcasters in the USA.
Chris Alfred – Chief Technical Officer
Chris joined us as CTO in July this year. He is a technology innovator with extensive experience in embedded programming and audio device development. Starting his career with our CEO Kim Ryrie at Fairlight in 1990, Chris has significantly contributed to digital audio editing for motion picture post-production, earning an Academy of Motion Picture Arts and Sciences, Scientific and Engineering Award in 2004.
Research Science Partners: Dickins Audio and Aurisium
We are also thrilled to announce our partnership with Dickins Audio and Aurisium, leaders in the audio technology industry.
Christophe Chabanne, CEO of Aurisium, brings extensive experience from his 22-year tenure at Dolby, where he co-invented Dolby Atmos. Christophe's efforts led to the standardization of Atmos at the ITU and earned him prestigious accolades including a 2024 Academy of Motion Picture Arts and Sciences Scientific and Engineering Award and a 2021 Primetime Emmy Award.
Glenn Dickins, CTO of Dickins Audio, has a rich background in Spatial Sound with extensive credentials, including over 120 patent families. Previously at Dolby as Lead of Research, Glenn was a key contributor to groundbreaking projects such as Dolby Headphone, ATMOS and VOICE.
Welcome
We are absolutely thrilled to welcome these distinguished professionals to our dynamic team, which already includes the incredible talents of Kim Ryrie, Dean Cooper, Alan Langford, James Murchison, Larry Owens, Luke Dearnley, and Gabriel Santos. Together, with our combined expertise and dedication we are excited about taking DEQX to new heights of innovation and excellence.
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DEQX Premate 8 digital active crossover / DSP
- Thread starter Keith_W
- Start date
TheoSweden
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After some waiting I have been starting to use my Pre-8, very intensive last two weeks.
Some Trouble:
-A alot of "snapping" / crackling sounds, dont know the english word. Sometimes its not even possible to listen and sometimes if I start on low volume and keep it like that there are no such sounds for hours.
If I then play loud for a while the very irritating sounds starts and they dont go away even on low volume.
Anyone alse experience the same?
-Para-eq lives it own life, it often changes the q value. I havnt really figured out when and why yet.
I hope it goes well for Deqx so we get a finished product, this Beta stege is a little painful in many ways.
On the positive side.. now my sound is fantastic! I have been working with my many different homebuild active systems for soon 20 years and quite quickly with the Pre-8 it sounded better than ever before with a large margin
Im not good at describe how it sounds but im very very happy so far! I guess it can be even better.
It seems very important/have big impact how you do the "create a speaker" measurement.. I have tried a lot of different way but placing the mic in the listenig position seems best so far.
Anyone else that have found out good ways to do things?
//Theo
Some Trouble:
-A alot of "snapping" / crackling sounds, dont know the english word. Sometimes its not even possible to listen and sometimes if I start on low volume and keep it like that there are no such sounds for hours.
If I then play loud for a while the very irritating sounds starts and they dont go away even on low volume.
Anyone alse experience the same?
-Para-eq lives it own life, it often changes the q value. I havnt really figured out when and why yet.
I hope it goes well for Deqx so we get a finished product, this Beta stege is a little painful in many ways.
On the positive side.. now my sound is fantastic! I have been working with my many different homebuild active systems for soon 20 years and quite quickly with the Pre-8 it sounded better than ever before with a large margin
Im not good at describe how it sounds but im very very happy so far! I guess it can be even better.I dont know what it means in this case, have you tried anything?
It seems very important/have big impact how you do the "create a speaker" measurement.. I have tried a lot of different way but placing the mic in the listenig position seems best so far.
Anyone else that have found out good ways to do things?
//Theo
Last edited:
Yes I have !
I think thats importent that you meassuring the speakers there direct sound .
So not with too much room influence , but that depens of the room And how far the listening position is from the speakers.
In my case I meassure halfway .
I have a roommode on the listening spot wich I have meassure with REW And make a notch in the EQ page .
Sounds great And verre lifelike And spatial .
I think thats importent that you meassuring the speakers there direct sound .
So not with too much room influence , but that depens of the room And how far the listening position is from the speakers.
In my case I meassure halfway .
I have a roommode on the listening spot wich I have meassure with REW And make a notch in the EQ page .
Sounds great And verre lifelike And spatial .
TheoSweden
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Who is this guy?
He doesn't understand what reverberation is. What he calls reverberation seems to be comb filtering. Reverberation is something completely else and they shouldn't be mixed.
TheoSweden
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I dont know him, but DEQX uses him as a guide to learn the new pre8 anyway. (links to his videos in the pre8 manual).
Could you please explain the "trim" better than him? I have played around a little with it but cant say I fully understand it yet. Any help is great
OP
Keith_W
Major Contributor
- Thread Starter
- #308
I think their choice of terms "trim" and "reverberation" is unfortunate. If he said "unwanted reflections" instead of "reverberation", it would be more accurate.
In this context, "trim" refers to "cutting" or "cropping" or "snipping". Other software calls it "curtain" or "gating" or "windowing". I'll explain it like this: DEQX sends a sine wave sweep to your speakers, and then it waits and listens for the signal to emerge from the speaker. The first thing it hears is the direct sound from the speaker - this is because it has the shortest path from speaker to microphone. If it waits a bit longer, it will start to hear reflections. Reflections are always delayed with respect to the direct sound because they travel a longer path from speaker to microphone.
At some point, we want to remove the reflections. Hence all the different terms. "Gating" means we shut the gate after a specified time. "Window" means we leave the window open for that amount of time. "Trim" means we snip off any sound that arrives after a certain time. They all refer to the same thing. Henceforth I will refer to it as "window".
The important thing to realise is that the time window is frequency-dependent. One period of 20Hz is 1/20s = 50ms. One period of 20kHz is 1/20000 = 0.05ms. If you leave the time window open for 50ms, you will capture 1000 cycles of 20kHz and a heck of a lot of reflections. So most software allows you to do frequency-dependent windowing (FDW) - where you keep the window open longer for low frequencies, and shorter for high frequencies. We can think of window length in terms of time (milliseconds) or cycles. I prefer to think about it in cycles, it's easier.
I saw that video and found it a bit hard to believe that DEQX only lets you specify one window setting for the entire 20-20kHz frequency range. I hope I am wrong about this because it would be a major limitation otherwise.
In this context, "trim" refers to "cutting" or "cropping" or "snipping". Other software calls it "curtain" or "gating" or "windowing". I'll explain it like this: DEQX sends a sine wave sweep to your speakers, and then it waits and listens for the signal to emerge from the speaker. The first thing it hears is the direct sound from the speaker - this is because it has the shortest path from speaker to microphone. If it waits a bit longer, it will start to hear reflections. Reflections are always delayed with respect to the direct sound because they travel a longer path from speaker to microphone.
At some point, we want to remove the reflections. Hence all the different terms. "Gating" means we shut the gate after a specified time. "Window" means we leave the window open for that amount of time. "Trim" means we snip off any sound that arrives after a certain time. They all refer to the same thing. Henceforth I will refer to it as "window".
The important thing to realise is that the time window is frequency-dependent. One period of 20Hz is 1/20s = 50ms. One period of 20kHz is 1/20000 = 0.05ms. If you leave the time window open for 50ms, you will capture 1000 cycles of 20kHz and a heck of a lot of reflections. So most software allows you to do frequency-dependent windowing (FDW) - where you keep the window open longer for low frequencies, and shorter for high frequencies. We can think of window length in terms of time (milliseconds) or cycles. I prefer to think about it in cycles, it's easier.
I saw that video and found it a bit hard to believe that DEQX only lets you specify one window setting for the entire 20-20kHz frequency range. I hope I am wrong about this because it would be a major limitation otherwise.
Back in post 248 I recounted Larry Owens' and my problems with my Beta Pre 8. Larry finally got the circuit board replaced and did a great programming with two calibrated microphones and a combination of the Pre 8 and Room EQ Wizard running on my Windows 11 PC. My stereo sounds great now, but I am looking forward to the Pre 8 doing the entire programming.
Only stumbled upon this thread now, and am not willing to read through 16 pages, so please forgive if this has already been asked or commented upon:
What the hell is it with those "taps"? Who the fuck cares? All you, we, in short: the user should care about is two things: 1) What processing things can it do at once and 2) at what latency.
Ideally, the throughput latency is at or below 1ms, which is good enough for any time-critical real-time application like live sound, live performance, and oh I dunno, action gaming. In the computing world, 1ms is a LONG time.
The humble Behringer DCX2496 Ultradrive (speaker management system with 3x analog in and 6x analog out) does EQ, crossover, freely configurable routing, and more, at 24/96 with a latency of less than 1ms between analog in and analog out.
This is trivial DSP technology. The usual processor chips are so fast and have been for so long, for so cheap, that any talk about "taps" and limitations in processing and how "using too many EQs" somehow "raises latency" is laughable. 20 years ago, any desktop PC with a proper audio interface could run 24/96 multichannel EQ and compression and dozens of filters at once, at 1-2ms stable latency, in realtime. You have a DSP (or CPU used as such) running at a few megahertz or a few hundred, giving it time to execute thousands of instructions per sample cycle, which means a LOT of stuff at once because the math involved in audio calculations is piss easy and utterly efficient if coded correctly.
Am I missing something here? I have a feeling I do. Please educate me about these severe digital processing limitations that presumably exist till today, even though that was a solved problem long ago.
What the hell is it with those "taps"? Who the fuck cares? All you, we, in short: the user should care about is two things: 1) What processing things can it do at once and 2) at what latency.
Ideally, the throughput latency is at or below 1ms, which is good enough for any time-critical real-time application like live sound, live performance, and oh I dunno, action gaming. In the computing world, 1ms is a LONG time.
The humble Behringer DCX2496 Ultradrive (speaker management system with 3x analog in and 6x analog out) does EQ, crossover, freely configurable routing, and more, at 24/96 with a latency of less than 1ms between analog in and analog out.
This is trivial DSP technology. The usual processor chips are so fast and have been for so long, for so cheap, that any talk about "taps" and limitations in processing and how "using too many EQs" somehow "raises latency" is laughable. 20 years ago, any desktop PC with a proper audio interface could run 24/96 multichannel EQ and compression and dozens of filters at once, at 1-2ms stable latency, in realtime. You have a DSP (or CPU used as such) running at a few megahertz or a few hundred, giving it time to execute thousands of instructions per sample cycle, which means a LOT of stuff at once because the math involved in audio calculations is piss easy and utterly efficient if coded correctly.
Am I missing something here? I have a feeling I do. Please educate me about these severe digital processing limitations that presumably exist till today, even though that was a solved problem long ago.
I have a DEQX Pre-8 for a 4-way active setup, have been part of the beta test group, and although there have been a few very minor software bugs, and some features that were not implemented in the early builds (such as the trigger outputs, they initially were not enabled but now they work fine)- and I have to say that the system sounds absolutely fantastic. The integration between drivers is seamless, the time domain / phase and group delay correction along with frequency response correction has yielded great sound in my system. I understand that there are ways to achieve this using other complex DSP solutions, but in my case all the correction was applied after doing a set of simple measurements which were uploaded to the DEQX cloud where their algorithm did the analysis and then downloaded that to my DEQX hardware. The imaging, the depth, the overall sound is fantastic. I doubt if I could have achieved this with other DSP hardware solutions; I know it is POSSIBLE to do with other DSP gear, but I could not have gotten there, I don't have the time or inclination to figure out how to configure such a system.
And by the way, the Pre-8 has a very nice MM phono stage in it built by Dynavector, which works wonderfully with the Dynavector 10x5 pickup in my VPI Scout. I hardly listen to vinyl, really, but I do have a collection of LPs that never came out on digital media and it's nice to be able to play them.
There is also Volumio streaming and render node in the Pre-8 but I haven't used that yet. I have a Lyrion music server with many thousands of FLAC files in it, which I feed to the Pre-8 through a Squeezebox Touch via SPDIF. I probably could play those FLAC files directly through Volumio but I am used to using the Squeezebox.
And by the way, the Pre-8 has a very nice MM phono stage in it built by Dynavector, which works wonderfully with the Dynavector 10x5 pickup in my VPI Scout. I hardly listen to vinyl, really, but I do have a collection of LPs that never came out on digital media and it's nice to be able to play them.
There is also Volumio streaming and render node in the Pre-8 but I haven't used that yet. I have a Lyrion music server with many thousands of FLAC files in it, which I feed to the Pre-8 through a Squeezebox Touch via SPDIF. I probably could play those FLAC files directly through Volumio but I am used to using the Squeezebox.
OP
Keith_W
Major Contributor
- Thread Starter
- #312
Are you missing something? Yes, you are missing a lot. But with an attitude like that, I am not inclined to explain. But before I embark on answering your questions, I have to ask you: are you here to troll, or do you genuinely want to understand? All the items I highlighted suggest to me that you have made up your mind, and further discussion with you is going to be a complete waste of time.
I own the Pre-8 beta preamp and it’s terrific. I’m sorry but I agree with you that this person is a problem with this language. Punt
I forked out for a Premate 4.
My friend, a professional installer of high-end systems, tried many years ago to get the first ones working well. He never could (to his ears) and even visited Kim to see if he was doing anything wrong (he is a qualified electrical engineer who also modifies and builds equipment). He found it was fantastic to integrate subwoofers into systems and used it for that. He still uses an old one for that in one of his systems.
I am having it delivered to him before he installs it in my system.
Many years ago, I was at GTG with a guy who had a system built entirely around one. What an ear-bleeding experience—not my cup of tea at all. Talk about an etched, brittle metal sound that he played far too loudly for my taste. I thought the SS amps he used were not the right match—he should have had some nice low-power valve amps. Surprisingly, everyone thought so as well, but the guy just loved it.
I will use it with my MSCALER and TT2 DAC/amp, directly connected to my speakers, and to integrate a subwoofer into the new high-sensitivity speakers I am getting, which are built around the SEAS exotic drivers.
Will see how it goes.
Thanks
Bill
My friend, a professional installer of high-end systems, tried many years ago to get the first ones working well. He never could (to his ears) and even visited Kim to see if he was doing anything wrong (he is a qualified electrical engineer who also modifies and builds equipment). He found it was fantastic to integrate subwoofers into systems and used it for that. He still uses an old one for that in one of his systems.
I am having it delivered to him before he installs it in my system.
Many years ago, I was at GTG with a guy who had a system built entirely around one. What an ear-bleeding experience—not my cup of tea at all. Talk about an etched, brittle metal sound that he played far too loudly for my taste. I thought the SS amps he used were not the right match—he should have had some nice low-power valve amps. Surprisingly, everyone thought so as well, but the guy just loved it.
I will use it with my MSCALER and TT2 DAC/amp, directly connected to my speakers, and to integrate a subwoofer into the new high-sensitivity speakers I am getting, which are built around the SEAS exotic drivers.
Will see how it goes.
Thanks
Bill
Hi, im trying to understand the function each measurement does. I’m thinking like this:
1 measure each driver independently
2 measure drivers collectively at same mic position.(this seems to time align)
3 measure collectively to apply room correction. ( From LP or other position)
Does seem correct? Has anybody played with mic position to create better FR or just better sound? I know this seems basic, but I’m coming from manual DSP and DEQX has little manual control so it seems like mic position might be a main point of control.
Ideas on how influence results for better effect?
Thanks, Ted
1 measure each driver independently
2 measure drivers collectively at same mic position.(this seems to time align)
3 measure collectively to apply room correction. ( From LP or other position)
Does seem correct? Has anybody played with mic position to create better FR or just better sound? I know this seems basic, but I’m coming from manual DSP and DEQX has little manual control so it seems like mic position might be a main point of control.
Ideas on how influence results for better effect?
Thanks, Ted
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