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Old 31st July 2001, 09:38 PM   #1
Ignite is offline Ignite  Canada
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I was fooling around with a lunky 12 band EQ when it donned on me how cool a completely software-based very-high resolution equalizer would be. Perhaps this exists. I don't know. Still, it intruiged me. I came to the idea of designing a program build to implement minute alterations to all DirectSound programs. It wouldn't be all that hard to do create an EQ that had a specific band for each hertz in the audible range. It might slow an older computer signifigantly, but perhaps not. My knowledge of computer audio stiffly drops off past making relatively simple things. However, I'm a quick study when it comes to DirectX. Does anybody think this sort of stuff would be actually useful? Or does it already exist?
Also, it would only be a step away from implementing all sorts of fun algorithms designed to counted distortions caused by amplification and speaker imperfections. I have almost know nowledge of those sorts of maths - except basic things - so it would be a "learning" process should I attempt to code such software.
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Old 31st July 2001, 09:58 PM   #2
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Default This exists, but ....

Hi there,

This exists today. Steinberg VST24 and Nuendo, probably others.

It would still be useful since I don't have this type of system presently (and I DO want it badly) so there is more to it than meets the eye!

What would be very useful is if it could upsample to 24/96 and work in full resolution. Most bands need to be low frequency -- need a shitload of bands under 1000Hz, not all that many over. Phase neutral. Also need phase filters. Low latency would be nice, also digital crossover (split 2 channels into left/right tweeter/bass). Also an option for adding digital gain to the tweeter (less energy in treble) to make full use of dedicated tweeter DAC and reduce gain in associated amplifier.

Do you have all the software you need?

Petter
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Old 31st July 2001, 10:11 PM   #3
Ignite is offline Ignite  Canada
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Good ideas. I don't know how one would implement a software crossover, though. The only way I can think of doing that is using multiple stereo sound cards or using surround cards like the SB Live! in some sort of hybridized stereo setup. It would be relatively easy should someone purchase two sound cards to create digital crossovers. One could design a very complex active crossover network with almost limitless options in that way. In fact, that is such a cool idea I think I will buy a second SB Live! and do just that. The only forseeable obstacle would be getting two sound cards to work simultaneously, but I doubt that's a real problem. DOS-mode compatability drivers would probably mess up, but that's not a problem at all.

First things first though: I need to read up a bit on how I would integrate the program(s). It's easy enough to design DirectSound plugins, or global effects for a specific program, but I haven't the foggiest idea of how to do it on a windows-wide level. I assume it's reasonably easy, though.

Oh, and should anybody know something about this stuff or think they could learn and can code in C/C++ With Windows API / MFC (whichever, MFC is generally easier but sort of annoying sometimes.. It's pretty much a must for DirectX though) and would like to help me, I would welcome it.


The other thing I was thinking about was the feasability of creating a digital profile of your speaker linearity (or lack thereof) and use it to create (near) linear response via software filtering.

[Edited by Ignite on 07-31-2001 at 05:15 PM]
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Old 31st July 2001, 10:51 PM   #4
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I have a digital profiler (frequency and phase for in-room response) + unfinished speaker (missing crossover + port.

The problem with profiling is the calibrated microphone. Sound-cards + CPU's can do the job. My system is Clio-lite from Audiomatica (still in DOS, so I have an old Compaq colour portable in docking without hard-drive -- less noise that way!)

Maybe I could beta-test? Soon i will have access to a 16 channel digital output card.

Petter

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Old 6th August 2001, 01:37 PM   #5
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Default Listening room optimiser

In an issue of Hi-fi world (a UK magazine)some while back there was an article on a listening room optimiser.

The digital output from the CD player was passed though a DSP chip and then out again to whatever DAC you were using.

The LRO box came with a calibrated microphone - you set the thing up by defining a "Sweet spot" in the listening room by placing the mic at each corner of an imaginery 1 yard cube around your head. The box than worked out the freq response, effect of standing waves, etc and automatically calcualted a digital filter for the DSP to correct the signal with.

As the whole thing worked in the digital domain there was no analogue signal degradation.

The unit was a prototype and didn't go into production, the DSP used in the design has since been superceded by much faster versions, so whatever the previous cutoff frequency was it could be significantly improved upon now.
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Old 6th August 2001, 04:48 PM   #6
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Default DSP based in Audio Electronics

In a recent Audio Electronics (www.audioxpress.com) there was such a beast where the guy built the whole thing from scratch. It seems to me there are better ways to do this (in order of priority):

1. PC based
2. Buy an eval DSP board with necessary ports (and software libraries).

It would likely be so much easier and cheaper to do it on a PC.

Petter
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Old 7th August 2001, 12:22 PM   #7
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Lightbulb DSP Audio cards for PC's

Perhaps using a PC card with a sufficiently powerfull DSP card in it would be a way forward - i believe such cards exist but they are expensive.

Not sure that what kind of processing power would be required from a PC to appropriately condition the digital audio stream in real time, hence the use of a DSP card.

I'm interested in doing this if anyone's got any ideas about software.
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Old 7th August 2001, 12:39 PM   #8
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Default I suspect a PC is good enough

but if you want to go the DSP way, get an evaluation board from http://www.ti.com or http://www.analog.com or http://www.cirrus.com which has all the inputs and outputs you need.

They also have software ...

But the PC method is where I would go. Check out the specks of nuendo or VST24 to see what can be done in real time.

An alternative is a good soundcard with a programmable DSP. These things are becoming more and more powerful by the minute.

Petter
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Old 6th September 2001, 05:09 AM   #9
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I've been studying digital audio and digital signal processing extensively for the past few years. Right now I'm working on building a high-powered audio DSP processors for just the type of tasks mentioned above...

The basic design revolves around an Analog Devices SHARC processor. You need this kind of power if you want to implement the really kinky algorithms (eg big FIR filters or convolutions). The trouble is, these computations require hundreds or even thousands of MAC (Multiply and Accumulate) operations *per sample*!!! So, you can see that 300 MFLOPS is barely enough to handle even some of the simpler algorithms in real time. Today's PC's are not really up to the task, because although they can have fast processors, they're general purpose processors which are not optimized to handle real-time signal processing, and a PC also has the overhead of a klunky OS like windoze. Then, of course, there's the fan noise, or that you can't use your 1GHz Athlon while your music's on...

I did look at using eval boards, but was put off by the price and the limited capabilities and i/o options you have with them... perhaps some new ones have come along lately, but I'm already well on the way to finishing this monster.

Anyway, when I'm finished, the DSP box I'm building should be able to perform some smokin cool tricks, like speaker impulse response correction (which removes all phase and frequency response errors from the speaker) as well as linear-phase crossovers for subwoofers (where phase errors are most apparent) and so on... whatever you're up to programming!

My preliminary design was finished as my graduating university project in May of this year, but I'm doing some rework with my project partner to improve the design. The final design will include:
- multiple digital SPDIF inputs
- AD1896 ASRCs to remove jitter and resample everything to either 48 or 96kHz, 24 bit (selectable)
- ADSP-21065L processor (32 bit floating point processor)
- microcontroller for LCD panel display and host control with spare ports for controlling whatever goodies someone wants to add on.
- good output DACs (currently under review, but probably the CS43122), with I2S outputs for alternate DACs.

So, right now the design is sidetracked somewhat while I build some more test DACs to evaluate. Hopefully I'll get my website back up again soon, and can post updates as more progress is made...
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Old 6th September 2001, 05:25 AM   #10
hifiZen is offline hifiZen  Canada
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oh, forgot to mention.. I designed this thing with my student budget in mind, but the expensive part is going to be the 4-layer circuit board.

However, if enough people are interested, a larger production run would bring the PCB cost down dramatically. If anyone's interested in the possibility of helping review the design, programming software, or just building one for themselves, please drop me an email and let me know what kind of target price range you would consider...
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