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Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project

That audio weaver looms amazing. Couldn't see any info on how a speaker designer could provide a simple user interface to the end user to control the parameters the designer has set though without having access to the whole tool set.

I don't understand how to do that myself.
I think there is a solution:
http://www.dspconcepts.com/sites/default/files/documentation/Audio_Weaver_Overview.pdf

Looks like the Server can send and recieve commands without using Mathlab.
Se the "PC Script Execution (Without MATLAB)" slide.

You might need a developer license. (Maybe this is the business model?)

I don't know, but maybe this will help on command syntax:
http://www.dspconcepts.com/sites/default/files/documentation/Audio Weaver Server Command Syntax.pdf

I might be far off, thoug.
 
A little error in downsampling post. Should be "about 150dB down at 400Hz"
Tried to filter LP 2x60dB/oct -> Downsample 32x -> 1024 FIR filter (impulse only) -> Upsample 32X -> scale 32x -> filter LP 2x60dB/oct.
Inserts sin and HP filtered whitenoise and it looks OK.

Not code efficient. Chose some shortcuts with filters samplefrequencies and decimation factors.

Audioweaver AWD and MASM files enclosed in .zip

The filter runs on my Corei5 CPU.

Regards Torgeir
 

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Oh I see, so with the LCD interface could you map control on the screen to stuff that us set up in audio weaver?

I'm thinking of a situation where an active speaker designer (me) using either your board or any other method out there can provide to tools to be able to say control volume, some eq, room FIR curves etc via some sort of simple user interface maybe on the PC.

I'm thinking more of commercial designs where you would want to keep the user away from all this backend stuff. Like the Grimm Audio stuff.

Maybe i should start an audio weaver thread

There is a dedicated forum for it already ;)

Forums | DSP Concepts

cheers
 
Guess the sampling frequency question boils down to if you are willing to use sharp filters or not.
It would be interesting if people started to use low samplefrequency for lower mid 300-600Hz. Then we have SF at about 2000 Hz and one could start discussing if you can hear slow/sharp LP antialiasing filters etc.
I think one issue of romcorrection bassfiltering is often overlooked. This applies to applications where more than 1 sub is used.
Under the room boundary the bass is added as to or more signals with a phasedifference that is lower than one period at room boundary frequency.
This means that phase change in one speaker will give a amplitude change in the sum of the signals.
When using IIR filters to correct a room resonance you risk to change the phase so much that the other woofers are cancelled out in the correction frequency area, in worst case. Using FIR it is possible to adjust the amplitude only (or phase only) so it is much easyer to adjust one wofer for linearity in the room without cancelling ut the other woofers.
A sidenote is that this also applies to subs with different frequency (and phase) responce in the bottom.
This is really important if you go for a setup as recomended by Harman:
http://www.harman.com/EN-US/OurCompany/Innovation/Documents/White Papers/multsubs.pdf
 
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Which means that even if I could ship it straight away it would be missing the ADC section.

cheers

David, are you saying that ESS found a major bug in their shipping Si? Or is it a deviation from spec performance? Is there a workaround?
Can the current generation still ship with this chip at reduced performance?
As someone else said, i too belong to an all digital camp and dont mind the ADC :)