1984 - 1999: Bell Labs: circuit design, DSP algorithm design & implementation. 1999 - 2005: Hardware manager at Globespan (then a leading DSL chip vendor). 2005 - present: The MITRE Corporation: Communications Systems Engineering, Modelling & Simulation, Electronic Warfare signal processing algorithm & development.

It is not a problem at all. Had you also plotted the imag(y), and then abs(y), you would notice similar behavior in the imag (though shifted in phase), and abs(y)...

Re: FIR filter shortening technique

Reply posted 2 years ago (07/11/2019)
Assuming your filter is a purely real filter (i.e. all the taps values are purely real numbers):An N-tap FIR filter is a filter with (N-1) zeroes (the roots of the...

Re: Possible Bug - Need your Help

Reply posted 3 years ago (06/04/2019)
Yep... Working now.-Aaron Chesir

Re: Possible Bug - Need your Help

Reply posted 3 years ago (06/04/2019)
My test results:1) Correctly redirected to the blogs when NOT logged in.2) Redirected to https://www.dsprelated.com/blogs.php but it is blank indeed when logged...

Re: Negative Latency DSP Methods

Reply posted 3 years ago (01/16/2019)
In broadcast television production, the audio is handled separately from the video. This is evident in cases in which the delays of the audio filtering are not exactly...

Re: fixing sample rate error/mismatch

Reply posted 3 years ago (01/15/2019)
If the source with the timing error has an *average* rate which is equal to that of the non-error source, then buffering the error source as well as the non-error...

Re: Time domain signal slikes

Reply posted 3 years ago (01/14/2019)
Whenever I see such phenomena, the first thing I check is if I am reading the ADC bits correctly: Is it possible that you are reading a 16-bit word using the wrong...

Re: raised-cosine filter paramiters.

Reply posted 3 years ago (01/07/2019)
The MATLAB on-line documentation for the rcosdesign function states that sps means Samples Per Second:sps — Samples per symbolpositive integer scalarNumber...
In re-reading what I had written, I need to clarify:Second: Pass the "Known Signal" into the adaptive filter input, and declare the "error" signal to be the difference...
I assume that you are trying to build a model of the impulse response of the microphone output that results from the sound input.If so, I recommend that you use...
To give you a geometric way of viewing this... In the Z-plane, all purely-real sinusoids are represented by a complex-conjugate pair of poles on the unit circle....
If your sampling frequency is 10 Hz, then all input frequencies from 0 to 5 Hz will show up properly. As you advance the input frequency from 5 toward 10, the aliased...

Re: Down sampling an iq signal

Reply posted 3 years ago (10/04/2018)
Please check your statement "I [downsample] by 2 by dropping every second sample from the iq stream. The resulting spectrum [has] of course half the bandwidth but...

Re: Correlation of complex sinusoid

Reply posted 3 years ago (06/11/2018)
Your software is behaving properly.In auto-correlation, you are multiplying waveform #1 with itself, but shifted in time, and the magnitude of the auto-correlation...

Re: Inverse filter

Reply posted 3 years ago (06/08/2018)
Rahul stated that he does not know his input signal, but he presumably knows the design of the HPF. Therefore it should be straightforward to determine the transfer...

Re: Inverse filter

Reply posted 3 years ago (06/08/2018)
Design a filter whose transfer function is the reciprocal of the filter you have. Feed the output of your current filter to the input of this second filter. The...
When one uses the term "reliability" with regard to the output of a bandpass filter, one is not typically referring to the the constancy with which a fixed mathematical...
Yes, the term "reliable" is appropriate in this case for the following reason: Since your sample sequence spans only 60 seconds, the band of 0.20 - 0.25 Hz has...

Re: Fourier Transform for a Chirp Signal

Reply posted 4 years ago (05/24/2018)
The Fourier Transform of a *single* chirp from f1 to f2 is (in frequency space) a pulse beginning at f1 and running to f2, with a rise portion (and overshoot) at...

Re: Interpolated FIR filter group delay

Reply posted 4 years ago (05/18/2018)
With a 25 KHz sampling rate, if you were to try do filter in one stage you are asking for a passband that is 1/2 of 1% of the Nyquist frequency!! Failure is guaranteed.With...

Re: Online interpolation of temporal signal

Reply posted 4 years ago (05/15/2018)
If you expect that the accelerometer readings should be linear with time, at least over the time interval being considered, then use linear regression: Use the...

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