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)...

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...

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...

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...

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...

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...

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...

**Re:** Why am I getting the same frequency response plots when one has aliasing and the other does not?

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....

**Re:** Why am I getting the same frequency response plots when one has aliasing and the other does not?

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...

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...

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...

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...

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...

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...

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...

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