An Active loudspeaker UNIFICATION thread

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I'm no expert...!

Derek - Very excited to see a subject expert on the DSP side! As I get a chance I'll look over the links and see if I can get up to speed. Nice size chip! CC was a nice touch. 🙂

If only I were a DSP expert...I am just a speaker designer!

What I have done is assemble a team of experts who can all produce remarkable components within their respective areas of expertise.....

My task is to provide each one of the teams (DSP & WiSA / power amps / DAC's & A to D ) with the best design brief to ensure that all the components blend perfectly into one complete system.

From a commercial stance, using modular construction allows us to widen our market by being able to sell individual components into different sectors as well as the complete system.

Re the DSP products, the real expert is Professor of DSP at Manchester University, Patrick Gaydecki, although we are also fortunate to have on the DSP team a recent graduate who scored the highest ever results in her PhD....One to watch!

Cheers
Derek.
 
Overkill,
Do you have an idea of cost on that small dsp board and the added cost of adding the Wisa wireless system? I'm curious. You can send that information to my as a PM if you like.

Will PM you....
As the JV with the university is a commercial venture we are all under NDA so I am limited when it comes to revealing costs / estimated RRP's and a lot of the technical explanations....But we can still accomplish a lot on here.

Cheers
D.
 
Jan,
Yes the use of output transformer with a ss amp is not exactly the norm. That was why I asked the question about just changing the impedance of the voicecoil to optimize this without the need for the trannies.

Overkill, it seems like we are very much on the same page and are just on opposite sides of the world. Any information you can share is appreciated, I have no problem signing an NDA with you if need be. It seems that you can't get away from working with Europe in the end and at least we are all friendly.
 
In a block diagram the only time the "base" would have to be the sole volume attenuation is if you used a USB connected device Samba share or similar UPnP connected share where it is directly reading the file.

Of course what isn't said here is that if we incorporate RCA/3.5mm in then we're looking at A/D conversion and that whole thing. I have no idea how much work involved with attenuating HPJ/Line in to useful digital that the DSP in the individual speaker can use/process/feed to the amp.

My understanding is limited to class D on that side, where unless you are looking at a discrete solution capture is less than stellar and I don't know enough about the market of chipsets to know the right option.

Overkill, are your cohorts versed in this as well - I would expect so... Also, I know right now it is a settings plugin configuration model. I think the chip has some basic display capability which would be fine for basic presets, scrolling through a basic LCD panel. While I know you have developed for Windows, I wonder what is possible from a Raspbian standpoint. Likely a different development thread, but just thinking out loud.
 
Binely,
If someone is going to use an analog source such as a per-existing preamp or even a cell phone output the only reason I can see to convert to digital is the use of digital filtering or crossover. If the crossover was done in the analog section you could just go straight to that point with no other digital conversion. Just another variation to think about.
 
I Deqx appears to do all the processing centrally. While not written in stone anywhere, I think options are open right now to something that would look more... diffuse. Allow me to encourage you to think a little more broadly.

For example. The A/D conversion may happen at the "base" but the DSP function processing can happen at the speaker. So, while the signal is being distributed, the base wouldn't necessarily be doing any clocking, that's the speaker's job to set master/slave. Instead, that can be done components at the speaker level, if you don't want to.

I suppose you could use the base as the master clock and then configure delays from there, but I have no idea if that would play well at all with a discrete dsp solution.

One of the advantages of doing all that can be that the speakers now become part of the communications package, instead of completely reactive components, so instead of guessing - maybe we can make it so the speakers can do some of the legwork in the analysis so you have multiple sampling points and suggested configurations.

Imagine if you were going through the house, and you were able to set up a few speakers around it such that if you walk away from the television, the speaker in the other room (maybe a little one) is still perfectly in sync with the others, yet doesn't cause any "echo effect" throughout the house of the sound. No need to pause the show, your guest can continue with the program while you go get a snack.

Or maybe you bring one speaker outside, the others remain. Again, same possibility - collectively synced, yet able to work with each individual space as desired. Now in these ways, not really viable for any low latency applications like any really distant placements involving a tv show or video game.

For that, you would just go to a "game mode" and forget all that crazy stuff. And even for a television show I suspect the latency wouldn't be much of an issue if you had reasonable speaker placement at that time.

For that matter, with the right legwork you could even make it the most amazing speakerphone system ever implemented. But that's waaaay out there.
 
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Binely,
If someone is going to use an analog source such as a per-existing preamp or even a cell phone output the only reason I can see to convert to digital is the use of digital filtering or crossover. If the crossover was done in the analog section you could just go straight to that point with no other digital conversion. Just another variation to think about.

I must be confused, then. Sorry. Or I'm being confusing. :crazy:

I think if we use analog in, to send over BT from a "base", I think it is already has to be in digital format to transmit via A2DP to the reciever which then goes to the discrete DSP via I2S, then to the gain stage.

I guess if there is no "base" then I'm off base lol. I guess you could have the "master" speaker combine the same function as the "base but it might make it a bit unwieldy/confusing". Just a thought.
 
Binely,
Sorry for adding confusion here. I was thinking that there are going to be some people who would not use the wireless at all, those speakers would just be connected as normal with a wire. So in a simple installation as a typical audiophile would use of two speakers wired you could have the direct analog input. If the speaker is only connected wirelessly then you are right, you go to the digital conversion to get that done.
 
Jan,
Yes the use of output transformer with a ss amp is not exactly the norm. That was why I asked the question about just changing the impedance of the voicecoil to optimize this without the need for the trannies.

Of course, that's also an option. My very first (tube) amp ever I build was a cyclotron with 807's feeding directly into speakers with 800 ohms voice coils!

And I recently acquired a McIntosh 2105 which also has an (auto) transformer to couple the amp to the load. That way you can get 100W into 8 ohms with only 35V supply.

Jan
 
Overview

Hi Guys,

I am sure this thread will open up discussions on a huge variety of different system implementations....Great stuff!

I am not going to be able to spend a lot of time contributing to systems suggestions which are too far removed from our own ideas, so it might be helpful if I give you a broad overview of the direction our products are going....

(1) The speakers will be grouped into Silver, Gold & Diamond type packages with various options.
The passive speakers as designed to be bundled with an AV amp capable of performing all DSP / amplification and room Eq functions.
Cabled or WiSA.

A key feature is that no component is redundant if upgrading, ie one can start with the entry level cabled solution and then add on WiSA via a central transmission hub with matching receiver boxes for each loudspeaker.
Remote firmware upgrades combined with modular plug in PCB cards ensure longevity and cutting edge performance.

(2) The active speakers will have both analog and digital inputs.
Both WiSA and cabled options.
Different models will have different electronic package options.
The DSP's inside each speaker can be combined with WiSA to allow free communication between all loudspeakers in a multi channel rig as well as communication with central media hub.

(3) The electronics packs will be modular so one can have a stand alone active amplification pack or go all the way to a "six pack" and include A to D, WiSA, DSP, DAC's & power amplification (class D or class A/B).

The commercial products will all be very high performance and priced accordingly.

Re this thread and what I can contribute / how members can benefit:

I think a key outcome of this thread would be to establish a "blue sky" specification for the kind of flexible systems and products I have outlined above that will appeal to a broad range of custom installers and the "serious" DIY market.
This will increase sales volume and allow us to lower the costs through economies of scale.

I hope the above is of interest and look forward to some good old brainstorming!

All the best
Derek.
 
Derek,
I am going to go and get my aluminum hat and put it on, you have read my mind. Two like minds here, I just don't have a team of people like you to pull it off. I have had an idea for a modular electronics center for some time. I won't go into my industrial design idea on that but I think we are so well matched in our thinking it is truly scarey! As you like the look of my speaker enclosure I have had one of those types of concepts for a entertainment center system, all modular and inter-mixable.

Now as someone on the other thread had to put in a contribution to the thread after the rest of us talk about supporting this site, I'm going to have to put together a team to build my dream. I've been working towards this point since 1976, there was just not the technology yet to pull this off. I designed my first horn loaded speaker in Junior high school before I even had an idea of how a speaker really worked, that was so long ago.

ps. If it is possible I may be very interested in packaging some of your development into my package. It could be very synergistic. But that is for the future.

pps. Derek, you wrote that so clearly and concisely that I actually copied it into a word document to use as the basis for a vision statement. Couldn't have described the concept better.

Now I just need to meet my new neighbor who just moved in down the street, He only goes by one name, Pharell. Haven't seen what kind of car he drives yet.
 
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X men....Tin hat and mind reader!

Hi Steven (It is Steven?)

I think a lot of music lovers / audio geeks our age (still young at heart!) share the dream of a high end modular system that sounds great, looks great and is not beyond the ££ reach of all but the lucky few....

I am sure good things will come out of this thread and who knows what collaborations might evolve?!

Cheers
Derek.
 
Derek,
Yes it is Steven, but my mom just says, Steveeeen when I still do something she doesn't like! She's 82 and still scares me. 😀

Yes, I truly believe we are kindred spirits in this quest. Like you I want to bring this to the masses, and if the audiophiles can get past the fact that the price isn't ridiculous, they can join the party.

My claim to fame if not for my industrial designs will be the cone material that I have developed. Perhaps that will be something you are interested in. I have toyed with patenting the material but then I have to divulge what I am doing. It is truly a unique development in that area.
 
Interesting read. I'm not nearly as technically savvy as most here, but I'll throw in what I'm doing/planning along these lines. Hopefully anjump123 and others can benefit from it.
This is only for my personal setup, though - not a commercial venture.

My aim is for digital room correction, time-aligned drivers, actively amped speaker system.

My only source is a highly optimized PC with no rotating parts, linear power supply, etc. It's a "headless" computer, meaning without a screen, operated from an iPad. The computer runs Win Server 2012 (in core mode), JRiver media center, and Acourate. Acourate is a software package by Dr Uli Brueggemann that enables digital crossovers, driver linearization, driver time-alignment, and room correction. Runs 24 bit / 96 kHz - beyond that the CPU load is very significant. It's a lot of work initially; then is set it and forget it.

The computer connects through USB to a multichannel DAC/ADC (such as Lynx Hilo, Prism Titan, etc). You need the ADC portion for taking measurements. The DAC already receives the 6 or 8 digital channels through USB, so it acts simply as DAC, and to drive the amps. Each channel drives an individual amp, which in turn drives a speaker driver. In my case, I'm starting with existing commercial 3-way speakers + subs, but in the future I should have DIY speakers + subs. Subs and midbass with class-D amps inside or attached to the box. Midrange and tweeter with tube amps.

Lot's of work to get it going, then it's as simple as something my 10-yer olds can use.
 
Lewinski,
What are you using for a microphone and where is it placed? I would assume with a dsp and dac you could actually use the speakers themselves as the microphones with an inverse function sending back to your computer to calculate the room correction needed.
 
Overkill,
For the exact reasons you are stating I decided to change this to a sealed enclosure. I will use eq to bring the -3db point back up to where it could be with a ported enclosure. I'm working with Pete on the Slewmaster thread to make some killer discrete amplifiers that can work with this speaker, a bi-amp system with enough power to work with these speakers. I did the original speaker development a round cone, I make the actual cone myself and just changed the shape to the elliptical shape to be able to narrow the face of the enclosure and still have the same sound. The boxes I did the original development work with were larger and were ported. The fs is 35hz and it is strong bass. the excursion on that speaker is designed to max at 1 1/4" p to p. they are serious speakers with my own motor design, lots of expensive Neo magnet material in the motor and a very long gap design with a short coil. The dome tweeter will be a Be design with again my own motor design. The original speakers I tested are loud enough that most would go running from the room at full output.

Perhaps I am attempting to pull off to much but I want to do something exceptional. I spent a long time getting the surfaces of that enclosure to look right and all the reflective surfaces to look as good as any Apple product or industrial design. I won't tell you how many hours of cad work went into that design, it was a real learning experience having to make surfaces work to that level. I also had to design for manufacturing and the design is done so it can actually be molded as you see and has internal ribbing and an integral frame for the cone driver so the motor goes in from the back and the cone assembly is dropped in from the front. The 4 screws on the outer edges of the dome tweeter won't be there, I redesigned it to have it attached from the inside so only the four center screws will be there, no way with the motor design to remove those last four screws.

Let's give this a go:

http://www.diyaudio.com/forums/multi-way/275476-tc9fd18-dipole-tweeter-crossed-1kh.html#post4350905
 
Lewinski,
What are you using for a microphone and where is it placed? I would assume with a dsp and dac you could actually use the speakers themselves as the microphones with an inverse function sending back to your computer to calculate the room correction needed.

Hi Steven.

I use a Dayton EMM-6 measurement mic. It's not super fancy, but works well for now for my purposes and includes a calibration file.

Acourate is prepared for the mic to be at a single point in the listening position. well, that is true for room correction. Of course for driver linearization the mic is close to the driver, and the driver (speaker) should ideally be placed in an anechoic environment (outdoors would do for us mere mortals, or in the middle of the room if outdoors is not feasible).

Using the speakers themselves as mics is beyond me. It is being quite a learning curve using Acourate as it is intended. Couldn't get away with innovative measurement approaches on top of it.
 
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