Quadrature tracking filter

[Oli Oli Oxin Free]

I'm trying to implement a demultiplixer. I have a recording device that is going to be connected to the computer. There are differennt parts in the demultipliexer and I need some help. I'm stuck on Implementing one of the parts and that is the quadrature tracking filter which is composed of 2 low-pass filter and a bunch of multiplier. It's multiplying though function waves. I'm trying to multiply sin(30*pi (t)) to sin[30*pi(t)) + angle]. and then i'm doing other stuff and then i'm adding the functions later on. The main things that I need to know is how to multiply sine and cos wave functions together. I appreciate any input or ideas to how to start something like that.

Thanks a lot

Moe

[SMeserve]

Hi Moe,

I see that you posted a similar question here in the forums.  I would continue the conversation there if their answers are not sufficient.  Thanks!

[Oli Oli Oxin Free]

Thanks a lot for replying

I'm completely lost. I'm have a DIFAR composite audio signal and i'm trying to demultiplex it to get the the components out of it. The components are the omni, N-S direction, E-W direction. How do i go about in Labview creating a demultiplexer. The audio signal is coming from a recorder connected to the computer through the mic jack. Any inputs to how to start a project like that would be really helpful. Thanks a lot for the help

[SMeserve]

Hey Moe,

I am unfamiliar with the DIFAR format for audio signals.  From what I see here, you will be able to pull in the waveform from the sound card using LabVIEW.  However, I am unclear of the analysis methods that you can use to pull the signals out of that waveform.  Will you only be receiving one waveform into your computer? Did you receive any documentation with your sensor that indicates the methodology of separating the signals?  Thanks!

Message Edited by StarsHockey on 01-29-2009 11:08 AM

[Oli Oli Oxin Free]

Hey Stephen,

The analysis method is the demultiplexer method. To implement it, it has a 3 parts to it. The audio signal is going to go through three paths for frequency pilot recovery, phase pilot recovery and broadband signal level adjustment. 

For frequency pilot recovery, the signal passes through a bandpass filter prior to being applied to a phase-locked loop. The bandpass filter is used to remove harmonic terms that might be present in a distorted waveform. I have the bandpass filter, how would you implement a phase locked loop and a 2nd order phase locked loop in labview?

The phase pilot recovery is composed of quadrature tracking filter and I have a way of implementing that part, but I need to know how I would be able to multiply for example sin(2*pi*15*t) by cos[(2*pi*15*t)+ Ø? That is the angle shift(phase shift).

The DIFAR audio signal has three components that I have to demultiplex from the waveform, the omni part and the directional components (N-S and E-W) and output them and play.

Thanks for replying..I know it's a harder question than most but I really need it

Thanks again in advance

Moe

[SMeserve]

Hi Moe,

To bring all three waveforms to the same phase, I would use the Align Waveforms VI in the Signal Processing palette.  As far as multiplying waveforms, you should use the methodology discussed in the other forum post that you started here.

Hope this helps.  Thanks!