Phase Shifter

Sounds like copyright protected material. You should probably get permission from the copyright holder first.

Thanks, but I don't need it.

Wire it along, send by post, pack it and use your favourite disposal agent, pack it and use some delivery service, give it to the next pizza delivery service that comes my way.

BTW - I dont need it.

The last I heard some drunk wanted to send money via Skype. Send me 180 bucks which he did not send to me and I take your Vulcan.

PS How much does he eat? I mean 5Volts or 10 Volts or do I have to switch the light on for him? Let me know!

First things first.

I am not Anonymous Poster#1. Mr. Anon, you just got yourself reported.

audiyo-b, you're willing to part with your magazine? I can't pay you for it if you want to do it that way. What do you propose?

regards,

Vulcan

Vulcan: I have the circuit in front of me, since I'm designing a cross-over / compensation schematic for new speakers. I'm not entirely familiar with the requirements of forums such as this, but the code of practice would appear to prevent direct contact, apart from questions of authorization. (Payment to me is impracticable and absolutely unsolicited). I guess the reference is the best I can do.

If it's of any help, you might try Rod Elliott's website : ESP Sound, and look at Project 103, which I thought might be an alternative to the fairly complex design referred to above. However, since the sub in the new design is not critically distant from the main speakers, there seems no need for phase adjustment. I think Rod deals with the issues quite comprehensively, especially if you follow his internal links....

Cheers......audiyo-b....

The author says the desired phase shift only works at one frequency. I need a circuit that will work for all frequencies (audio, that is). Thanks anyway.

regards,

Vulcan

You cannot get any circuit to phase shift the same angle for all frequencies. The math just doesn't work that way. You can get an all pass filter to change the phase angle at a specific frequency to a specific angle or delay.

redfred, yes I kinda figured that. However, it's possible to do it digitally (using a microprocessor). How that's done, I still have to find out. I love a challenge.

regards,

Vulcan

Strictly speaking a microprocessor cannot directly alter an analog signal. You have to convert the analog signal to a digital signal. This happens with an A/D converter. To get an accurate measurement at the A/D converter one should somewhere have something acting as a Nyquist filter to attenuate out of band signals that will create erroneous aliasing. Then and only then can a microprocessor manipulate data. The massaged data then must get converted back to an analog signal with a D/A converter. Then just like the input side a filter network should reside somewhere to remove out of band signal created by the D/A process. Very often in today's snootier audiophile hardware packages this last filter is the speakers themselves because they are not designed to move air at frequencies significantly higher than human hearing.

All that I just described happens in the dedicated processor board known as a sound card. Many different complicated and convoluted electrical designs exist on a sound card.

If you do wish to delve into this realm, TI makes some of the best stand-alone Digital Signal Processing products around. I warn you. Do not be fooled by the relatively low cost of these development boards. This is a realm of many twisty little corridors with mysterious rooms filled with cryptic text and terminology. Many a mind have wandered into this rabbit hole never to come back the same.

You've only succeeded in getting me interested.

I'm aware of the need for filters before the A/D converter and after the D/A converter and the programming is not really a problem. It's the algorithm for creating the phase shift that's going to take some doing. I can do brute-force programming and achieve it but I don't know how well it will work. If DSP has circuits, however, I'll check them out.

I also know about those rabbit holes. Alice told me all about them.

regards,

Vulcan

Oh look, a bunny and another bunny.

Are you familiar with a LaPlace transform and his digital cousin the Z transform?

No, I'm not familiar with LaPlace. He died before I was born . I took a two year course to learn radio and TV servicing (couldn't afford college). I continued my studying on my own after that.

This is a brief description of what I'm planning to do:

The black waveform is the original signal and it's increasing in frequency. I take the half-wave, A-B, and make a copy, 1-2. I put Point 1 in the middle of Points A and B, which is the approximate location of the 90° part of the cycle. Point 2 gets inserted into the middle of Points B and C. I do this for all cycles and will end up with a 90° phase-shifted replica of the original signal. It won't be perfect but should be good enough. This example is a constant amplitude waveform but the technique should work for variable amplitude as well.

All I really have to detect is the point where the zero-crossings occur and the middle points of each crossing.

I'm doing it this way because using a formula to get the phase shift is going to take longer to execute in terms of programming entry as well as program execution. Further, I can vary the phase shift to any value I want.

'Hope you get my points .

regards,

Vulcan