Re: Envelope Detector using Hilbert Transform

>"w106pjs" <w106pjs@xxxxxxxxx> wrote in
>news:9fGdnTz2ELZZLnrfRVn-tQ@xxxxxxxxxxxx:
>
>> All
>>
>> I have been kind going through previous threads in this group on
>> similar concern and question that I have. Still I feel my feet is not
>> on the ground yet...with this issue..
>>
>> Background:
>> I have working with ultrasound signals (300 Khz) from a solid state
>> sensor to a target at 3". The received signals from the sensor (300
>> Khz) are digitized through a sampling card and imported into MATLAB,
>> to preserve the time information as precisely as possible, sampling
>> frequency is high Fs = 50Mhz.
>> To generate a envelope on the 300Khz received echo, conventional
>> magnitude of the analytic is used in MATLAB.
>>
>> Envelope = abs(hilbert(echo_signal));
>> It works as expected. and then rest of the processing continues on the
>> envelope signal.
>>
>> Question and Concern:
>> 1. What should be an alternative approach to acheive the same. ?
>> Say FIR hilbert transformer, All pass IIR design..
>> With sampling frequency this high, my fear is FIR may generate real
>> higher length filter.Correct me If I am wrong!!
>>
>> 2. Hilbert function in MATLAB works using the FFT and IFFT
>> approach.What approach should be more efficient in terms of
>> implementing it in real time on the DSP processor.
>>
>> thanks in advance.
>> Paul
>>
>
>I'm not sure why you need to oversample by so much, so I will only
>comment on the hilbert implementation.
>
>The easiest way to do a hilbert transform is to use an odd length anti
>symmetric FIR filter. I generally use a Parks McClelland (remez exchange)

>fit for the filter to make the passband ripple small.
>
>The fir filter is a bandpass filter. It doesn't need to be very long if
>the signal of interest is not too close to DC. If you are sampling at
>50MHz, and the signal is a 300k, you are close to DC, so the filter will

>be fairly long. Sample at a few MHz, and the filter will be very easy to

>implement on a DSP.
>
>You get the real output from the center tap of the same delay line that
>is used for the imaginary part.
>
>Rick Lyon's book: Understanding Digital Signal Processing ( 2nd edition)

>explains this approach. Marvin Frerking's book: Digital Signal Processing

>for Communication Systems is another excellent reference (but very hard
>to find).
>
>The envelope is the SQRT (r^2 + i^2). This is easy to implemt on a SHARC

>or similiar floating point DSP.
>
>We have boards that can do this application.
>
>
>--
>Al Clark
>Danville Signal Processing, Inc.
>--------------------------------------------------------------------
>Purveyors of Fine DSP Hardware and other Cool Stuff
>Available at http://www.danvillesignal.com
>

Al Clark

Thank you for your prompt reply.
1. The reason it is oversampled by a significant amount is to preserve
the time resolution in the application. The DSP algorithm is being used to
precisely measure the time of flight for the ultrasound signal through air
which demands high resolution. Sampling at a low rate and interpolating
for better resolution was experimented with very little sucess compared to
oversampled routine.

2. Since 300K is not what signal of interest @ Fs=50 Mhz, the lenght would
certainly be large, does it become really impractical to implement on the
real hardware.
Is IIR filters (All pass filters) would be of any help in this case. What
band edges would you recommend for FIR BP filter.

thanks again.
Paul


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