Low pass filter at half Nyquist

Started by VelociChicken in comp.dsp11 years ago 32 replies

Hello, I'm currently using an FFT to zereo out all the frequences above half Nyquist for my application. It makes perfect cutoff, but I have to...

Hello, I'm currently using an FFT to zereo out all the frequences above half Nyquist for my application. It makes perfect cutoff, but I have to use buffers and the old overlap/add to remove clicks. My question to you good folks - is there a clever 'trick' to cut frequencies off above this specific (SR / 4) point using time domain techniques? Or a more efficient frequency domain method? ...


How to create peak/shelf filters, that are symmetric close to nyquist?

Started by jungledmnc in comp.dsp7 years ago 23 replies

Hi, is it possible to modify biquad peak/shelf filters, so that they have approximately the same shape allover the spectrum. Normally they...

Hi, is it possible to modify biquad peak/shelf filters, so that they have approximately the same shape allover the spectrum. Normally they start to "narrow" the closer they are to nyquist. I have checked it can more or less be compensated by combining multiple filters, but that's pretty much an alchemy :), so I wonder if there isn't some regular approach. Thanks in advance. jungledmnc


Nyquist frequency in FFTs

Started by Shafik in comp.dsp14 years ago 15 replies

Hello all, I noticed that all FFT routines transform a time array into a frequency array of the following form: F[0], F[n/2], real:F[1],...

Hello all, I noticed that all FFT routines transform a time array into a frequency array of the following form: F[0], F[n/2], real:F[1], imag:F[1], real:F[2], imag:F[2], etc.. Now the question is, why does "F[n/2]" (the nyquist freq) appear as the second term, and more importantly, what is it's phase? Is it assumed to be fully real? Just extremely curious, --Shafik


Basic Sub-Nyquist Sampling

Started by Craig in comp.dsp16 years ago 2 replies

I am having some difficulty understanding the sub-nyquist sampling theory. For example: if you have an ADC that can only operate at 250MHz,...

I am having some difficulty understanding the sub-nyquist sampling theory. For example: if you have an ADC that can only operate at 250MHz, then the max bandwidth is 125MHz. However, if I wanted 1000MHz and I want to use the same ADC, the bandwidth would "fold" into 125MHz output band. So I could potentially have 8 solutions if I injected in a 40MHz signal. 40, 210, 290, 460, 540, 710...


Nyquist sampling theorem

Started by Anonymous in comp.dsp14 years ago 18 replies

Hello, I read in the text book ("Digital signal Processing Principles, Algorithm and APplications" J. G. Proakis % D. G. Manolakis, page...

Hello, I read in the text book ("Digital signal Processing Principles, Algorithm and APplications" J. G. Proakis % D. G. Manolakis, page 30) that the Nyquist frequency (rate) is a double of the highest frequency of the signal. In the web, I learn that it is the double of the bandwith. Moreover in the web they state that there is a mistake in many textbooks where authors teach that it is the...


Nyquist rate

Started by Sharan123 in comp.dsp3 years ago 20 replies

Hello All, if I know that the signal I am going to sample (coming from Analog domain) is a triangular wave which repeats itself every 1 second...

Hello All, if I know that the signal I am going to sample (coming from Analog domain) is a triangular wave which repeats itself every 1 second (1 Hz signal) then I am a bit confused if I have to sample this signal at slightly more than 2*1 Hz (to satisfy Nyquist rate) or something else? Looking at some worked out examples which show FFT transform of a triangular signal, it seems that 2*1. Th...


Nyquist Lth band digital filter... IPlease!! ... I need answers

Started by victor940 in comp.dsp10 years ago 2 replies

Hello everybody Before I ask my question I want to tell everybody that I'm new in the area of DSP. Now, here is my question: Can somebody...

Hello everybody Before I ask my question I want to tell everybody that I'm new in the area of DSP. Now, here is my question: Can somebody tell me why the "Nyquist Lth band digital filter" is better than the "raised cosine one"?. Or perhaps, somebody can tell me what are the good and bad features of both filters? thanks for your help!! Victor


Nyquist constrain and IQ represented signal

Started by RobR in comp.dsp11 years ago 41 replies

Hello, there is a question that bugs me for quite a long time: You can read about Nyquist constrain online, that to reconstruct...

Hello, there is a question that bugs me for quite a long time: You can read about Nyquist constrain online, that to reconstruct all frequencies within a signal, it has to be sampled with at least twice the bandwidth _or_ maximum frequency. Maybe this _or_ is already the problem... Let's see, I have an IQ branched digital signal. So my maximum positive signal is not equal to the bandwid...


Need Phase-Lead filter.

Started by HelpRaptor in comp.dsp15 years ago 9 replies

I need a digital filter or sequence of digital filters with the following response: Amplitude flat across all frequencies up to the Nyquist...

I need a digital filter or sequence of digital filters with the following response: Amplitude flat across all frequencies up to the Nyquist (folding) frequency. Phase starts at 0 degrees at low fregs linearly increases with frequency until it is 180 at the Nyquist (folding) frequency. I was thinking about do this with FFT e.g. multiply the spectrum by exp(j*phi*omega). Are there ot...


Image-reject IF downmixing

Started by Kevin Neilson in comp.dsp15 years ago 6 replies

When digitally mixing an IF down to baseband, one is left with a spectrum that consists of the baseband (Fif - Fmix = 0Hz) and an image (Fif +...

When digitally mixing an IF down to baseband, one is left with a spectrum that consists of the baseband (Fif - Fmix = 0Hz) and an image (Fif + Fmix). If the IF is greater than the Nyquist freq, the image will wrap back into the first Nyquist zone (0 to Fn). Normally the next step in demodulation is decimation, which consists of lowpass filtering out the image (often with CICs) and then drop...


choosing a sampling rate lesser than nyquist rate(sub nyquist rate)

Started by praveen in comp.dsp16 years ago 17 replies

Hello, I have a signal consisting of 4 harmonics (200k,400k,600k,and 800k Hz) and dc component.The signal is very pure and SNR better than 60...

Hello, I have a signal consisting of 4 harmonics (200k,400k,600k,and 800k Hz) and dc component.The signal is very pure and SNR better than 60 dB. I have to sample it in sub Nyquisit rate(lesser than 1600k). What sampling rate should i choose so that there no alaising. I cannot chose higher sampling rate because of my hardware constraints. waiting for reply regards praveen


Low freq "analog" of Nyquist? ( possibly naive question )

Started by Richard Owlett in comp.dsp16 years ago 43 replies

I understand Nyquist specifying a minimum sampling rate to determine the high frequency component of a signal. What happens at at the other...

I understand Nyquist specifying a minimum sampling rate to determine the high frequency component of a signal. What happens at at the other end of the spectrum? I.E. Is there a minimum time window required? E.G. If the signal has a significant 1 Hz component and sample window was .1 sec with a sample rate of 10 kHz, would an FFT portray the 1 Hz component?


Orfanidis prescribed nyquist peak filter parameters

Started by Anonymous in comp.dsp2 years ago 12 replies

First post! Advance apologies for newb-ness. I've made a quick implementation of Orfanidis 'decramped' prescribed nyquist gain peaking filters...

First post! Advance apologies for newb-ness. I've made a quick implementation of Orfanidis 'decramped' prescribed nyquist gain peaking filters in C++. Seems okay but I would like to be able to specify... Peak gain in dB Peak frequency in Hz Peak Q say 0.1 to 4 rather than... (G0 = reference gain at DC = 1) G = boost/cut gain GB = bandwidth gain w0 = center frequency in rads/sam...


2D sinc

Started by alexryu in comp.dsp10 years ago 3 replies

I've always seen 2D sinc interpolation done in a separable fashion - interpolate along the x axis to the correct x-coordinate, and...

I've always seen 2D sinc interpolation done in a separable fashion - interpolate along the x axis to the correct x-coordinate, and then interpolate the new values down the y axis to the correct y-coordinate. So if you have data defined on a grid, D(m,n), and you want to interpolate D(m1,n1+eps), you only need to do one sinc interpolation. This is well and good (Nyquist's theorem is Nyquist's t...


sampling a perfect sinusoid at Nyquist rate?

Started by jjmai in comp.dsp12 years ago 16 replies

Let's say you have a perfect sine wave at frequency w. According to Nyquist, in order to be able to recover the sine wave, you need to have a...

Let's say you have a perfect sine wave at frequency w. According to Nyquist, in order to be able to recover the sine wave, you need to have a sampling rate of at least 2w. So if you decide to sample at 2w, you end up with 2 samples for each cycle of this sine wave. If you sample at the peaks and troughs (90 and 270 degrees) of the sine wave in time (or spatial) domain, you indeed preserve t...


Sampling Theorem history

Started by Tom in comp.dsp16 years ago 10 replies

It is generally credited that the Sampling Theorem is due to fistly the Mathematician Whittaker and Shannon and the Russian Kotelnikov. I have...

It is generally credited that the Sampling Theorem is due to fistly the Mathematician Whittaker and Shannon and the Russian Kotelnikov. I have no doubt that Whittaker was first but was Shannon aware of Whittakers work? Also where does the Russian engineer fit in? What role did Nyquist play. We talk of the Nyquist frequency (half sampling) but why name it this if the work is due to Shannon? Or...


Interpolation and the additive white gaussian noise

Started by Prasanna in comp.dsp13 years ago 20 replies

Dear all, The following problem recently came up in one of my simulations. I have a PSK modulated RRC filtered signal that I have to transmit...

Dear all, The following problem recently came up in one of my simulations. I have a PSK modulated RRC filtered signal that I have to transmit over the AWGN channel. Before RRC filtering, I interpolate the IQ signal to twice the Nyquist rate. Now consider the two setups: The first one transmits the RRC filtered signal at twice the Nyquist rate, and adds white Gaussian noise to it. The secon...


FIR gain

Started by Philip Newman in comp.dsp15 years ago 11 replies

I have an FIR filter with the system equation y[n] = 0.25x[n]+0.5x[n-1]+0.25x[n-2] which gives an impulse response of h = [0.25 0.5...

I have an FIR filter with the system equation y[n] = 0.25x[n]+0.5x[n-1]+0.25x[n-2] which gives an impulse response of h = [0.25 0.5 0.25] without resorting to Z-transform analysis, how can I work out the gain of the filter at DC and at the half nyquist frequency? I think the gain of the filter at DC is the sum of the impulse response, in this case 1. but what about the half-nyquist? ...


QPSK baseband shaping ....

Started by Kappa in comp.dsp11 years ago 1 reply

I have built a QPSK modulator, but I have some doubts about the Baseband Shaping Filter. The Baseband Filter is a "SRRC Digital Filter" with...

I have built a QPSK modulator, but I have some doubts about the Baseband Shaping Filter. The Baseband Filter is a "SRRC Digital Filter" with cutoff at Nyquist / 2 with selectable roll-off. In my project roll-off is 0.35. Example (SAT): 27500 Msymb/s ---> Nyquist Frequency is 55000 Mhz (for I and Q) ---> cutoff Frequency 27500 Mhz. May need to interpolate at least 2 ? Is right ? It i


ADC Sampling with variable frequency and Nyquist?

Started by Anonymous in comp.dsp2 years ago 7 replies

I wonder if I can sample a signal using an ADC with variable sampling frequency. Lets say the signal bandwidth is 10 MHz at baseband. In order...

I wonder if I can sample a signal using an ADC with variable sampling frequency. Lets say the signal bandwidth is 10 MHz at baseband. In order to reconstruct the signal, the ADC sampling frequency must be > 20 MHz. What if I use a sampling frequency that varies between 19.9 MHz and 20.1 MHz? What does the Nyquist sampling theorem tells us about this case? Are there applications that actua