Digital audio Processor

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I'm building digital audio processor witch process signal in frequency domain (filtering & equalization) tweeter,mid & bass have their own amplifier. what i am asking is what must, nice or cool features you wish to have in your audio system?
 
I'd say, off the cuff:

1. high-pass filters
2. low-pass filters
3. shelving filters
4. Notch filters
5. Attenuators
6. Arbitrary slopes for all filters
7. Time delays
8. Dithering
9. Equalisation, with selectable preset/programmed filters
10. Digital Room Correction (also with selectable configurations)
11. Analogue volume controls
12. Master/slave control of volume controls so that master loudspeaker controls system's overall volume
13. Remote control
14. Selectable Digital and analogue inputs into master loudspeaker
15. Digital or dot matrix display of system status (volume levels, inputs and muted channels).
16. Control outputs to turn on/off/mute power amplifiers.
17. Control panel for in case remote control fails (battery low or fault).

Will it be ready for Christmas?
 
Hei sory for a late reply. just moved to new aparment.

since it process signal in frequenzy domain using convolution it covers filters you mentioned. time delays can realy hear them? (except those caused by speaker placement) and analogue is just wrong except for live performance. And what comes to controlling i asked my local dealer hdmi transmitter and he said that i need need license for that because some copyright. any recoments how to control cut off frequenzys, bass , etc. since it is stereo 48K band equalizer?

it's working just fine (manual filter configuration and automatic flat response) but unfortenately i can't puplish any documents before i get my grade cos its my graduation work.
 
fzaad

It seems that you need to read up on loudspeaker design in order to understand the requirements for making a good crossover. But before I make any further comments, can you please clarify what it is that you have developed? So far it seems like it is an equaliser...?
 
well... if you feed signal to it it tracks down frequenzy components of that signal. then you make desired frequenzy response you want your speaker to have. then it filter original signal and send it to speaker. am i making any sense to you ?
 
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