FFT questions

Started by Greg Aagard in comp.dsp16 years ago 1 reply

I haven't done fourier stuff for a while, and I had a couple of questions. I'm trying to get the frequency spectrum of a time-sampled signal...

I haven't done fourier stuff for a while, and I had a couple of questions. I'm trying to get the frequency spectrum of a time-sampled signal by using the FFT function in Matlab. I get rid of the redundant half of the output and scale the frequency axis so that it goes from 0 to the Nyquist frequency (half of 1/T). I'm pretty sure I'm doing that part correctly. I next divide the output b...


conventional wisdom how to upsample very large arrays, accurately?

Started by all4dsp in comp.dsp9 years ago 31 replies

Hello all, I'd like to know what methods are available, if any, for upsampling a large array. I have an application where I need to sinc...

Hello all, I'd like to know what methods are available, if any, for upsampling a large array. I have an application where I need to sinc interpolate an array of 1E9 (billion) double precision data points output by an ADC by 4x. The array captures a clock waveform centered at zero volts and is oversampled (meets Nyquist). Don't know if it matters, but my goal is simply to very accurately find th...


Scaling FIR for samplerate

Started by jeff227 in comp.dsp12 years ago 17 replies

Sorry for the "DSP 101" questions but I'm a newbie... Is there a way to scale FIR filter coefficients so I can have the same cutoff and...

Sorry for the "DSP 101" questions but I'm a newbie... Is there a way to scale FIR filter coefficients so I can have the same cutoff and transition bandwidth regardless of sample rate (provided the sample rate is always within Nyquist limits)? I am designing an audio processor that needs to have the same filter response regardless of user-selected sample rate (44.1KHz, 48KHz, 96KHz, etc.). ...


Nyquist, quantization and windowing gotcha's

Started by Richard Owlett in comp.dsp11 years ago 4 replies

I've been experimenting with a 3D version of spectrograms [amplitude vs frequency vs time]. Instead of plotting the spectrum of each time slice...

I've been experimenting with a 3D version of spectrograms [amplitude vs frequency vs time]. Instead of plotting the spectrum of each time slice (cf waterfall displays), I plot contours of equal amplitude across time. Borrowing from traditional spectrograms, each contour's color also indicates amplitude allowing adjacent contours to be distinguished when close together. The observed ar...


Bandpass subsampling?

Started by Vladimir Vassilevsky in comp.dsp6 years ago 3 replies

There is a need to estimate power which falls into given wide (~30% of Nyquist) bandwidth. Power spectrum doesn't matter; just overall number....

There is a need to estimate power which falls into given wide (~30% of Nyquist) bandwidth. Power spectrum doesn't matter; just overall number. The power estimate is not needed often; so a decimation factor of ~1000 is preferred. Straightforward way would be make FIR filter to cut out the bandwidth, run this filter at every 1000th sample, then measure power. But if the input is period...


Non uniform sampling

Started by santosh nath in comp.dsp16 years ago 11 replies

Hi all, Suppose a signal is known a prior. Time domain as well frequency domain characteristic is pretty defined. Say fs is the...

Hi all, Suppose a signal is known a prior. Time domain as well frequency domain characteristic is pretty defined. Say fs is the sampling frequency as per Nyquist criteria. Number of samples in 1 sec is N. Let us think two cases 1. Uniform sampling i.e sample interval is constant, I guess this is the usual case in most of the applications. 2. We keep N same in I sec but a variabl...


obtaining continuous time signal bin energy

Started by Brian Zhang in comp.dsp12 years ago 32 replies

Hi, I need to find energy for several harmonics of a signal, but feel confused about the relationship between the energy and DFT...

Hi, I need to find energy for several harmonics of a signal, but feel confused about the relationship between the energy and DFT (FFT) values. 1. I wanted to prove that for a periodic signal $s(t)$ with $N$ harmonics we can get its energy through its $2N$ points DFT as long as the highest frequency is in Nyquist Range. Through instinct I think there will be a relation such that $$\fra...


sampling frequency for high pass filter

Started by karthick121 in comp.dsp12 years ago 2 replies

hi all, i am wondering about how to sample for a high pass digital filter which has a high bandwidth. Because according to Nyquist theorem it...

hi all, i am wondering about how to sample for a high pass digital filter which has a high bandwidth. Because according to Nyquist theorem it should be more than two times the bandwidth of the high pass filter. so how it can be achieved with the limited ADC resolution that practically we have? please reply thanks


Determining "effective" length of signal from frequency spectrum

Started by Andy365 in comp.dsp8 years ago 6 replies

Hello gurus, say you have an (almost) random real frequency spectrum: G(f). You sample this spectrum at df: Gi = G(i*df). Next you take the...

Hello gurus, say you have an (almost) random real frequency spectrum: G(f). You sample this spectrum at df: Gi = G(i*df). Next you take the IDFT of this spectrum: gi = g(i*dt). Nyquist tells us that the extent of this signal, tN is given by: tN = 1/(2*df) However our signals have the property that the signal is only nonzero up to tmax: |g(t)| > 0 for 0 < t


Gradual resampling

Started by jon222 in comp.dsp12 years ago 7 replies

Does anyone have Java or C code for a gradual resampling. I already have implementation of Nyquist?Shannon sampling theorem, where I can...

Does anyone have Java or C code for a gradual resampling. I already have implementation of Nyquist?Shannon sampling theorem, where I can resample to any ratio. I know to calculate the amplitude of the signal at any time. Now I need to know at which time to take the amplitude to achieve the gradual resampling e.g. from 100% - 200%. It means at the beginning it is the same signal and gradually to th...


Undersampling with inverted spectrum

Started by b2508 in comp.dsp3 years ago 26 replies

Hi, I undersample signal of bandwidth BW that is completely in second Nyquist zone. It ends up folded to some frequency between 0 and new fs/2....

Hi, I undersample signal of bandwidth BW that is completely in second Nyquist zone. It ends up folded to some frequency between 0 and new fs/2. This spectrum is inverted. If I now downconvert this signal to baseband by multiplying it with complex sine, I can obtain "right" spectrum by swapping I and Q axes afterwards. However, I would like to downconvert this signal by multiplying with real...


As "Nyquist" is to "sample rate" "????" is to "sample period/duration/width/?" ?

Started by Richard Owlett in comp.dsp15 years ago 16 replies

I'm interested in speech signals as input to speech recognition software. I get the impression that minimum acceptable sample rates begin at 8...

I'm interested in speech signals as input to speech recognition software. I get the impression that minimum acceptable sample rates begin at 8 kHz ( or above ). I assume this is based on which formants are considered "significant". I have somewhat arbitrally chosen 44.1 kHz. The data I have available is a studio quality CD. From another thread, I assume that some characteristic time o...


Anti Aliasing of Arbitrary Waveforms

Started by Scott Gravenhorst in comp.dsp11 years ago 209 replies

What are the common method(s) for preventing aliasing of arbitrary waveforms generated within a DSP application? I understand that waveforms...

What are the common method(s) for preventing aliasing of arbitrary waveforms generated within a DSP application? I understand that waveforms supplied as analog signals and presented to an ADC must first be lowpass filtered to remove harmonics above the Nyquist frequency, but what about generated waveforms? I know about BLIT and it won't be appropriate for completely arbitrary waveforms. I'...


Channel-induced ISI

Started by Anonymous in comp.dsp13 years ago 7 replies

Hi, guys: Please give me some help for the following question: What is the relationship between the channel-induced ISI (i.e. fast-fading...

Hi, guys: Please give me some help for the following question: What is the relationship between the channel-induced ISI (i.e. fast-fading and frequency selective fading) and Nyquist criterion of zero ISI? I know both of them, but I can't find any link between them. Channel-induced ISI is produced because the coherent bandwidth f0 is narrowed than the signal symbol rate. But I reme...


Bandwidth of complex IQ signals

Started by billykao in comp.dsp12 years ago 11 replies

Hello, I have a DDC system. The input signal is sampled at 100 MHz. The carrier is 70 MHz. Signal bandwidth is 30 MHz.(from 55 ~ 85 MHz) The...

Hello, I have a DDC system. The input signal is sampled at 100 MHz. The carrier is 70 MHz. Signal bandwidth is 30 MHz.(from 55 ~ 85 MHz) The output of the DDC is I and Q of 30 MHz. Can I represent 30 MHz bandwidth simultaneously? How can I do it? How to explain it using Nyquist theory? Thank you. Billy _____________________________________ Do you know a company who employs DSP engin...


matlab help needed for Digital signal pulse shaping and baseband transmission

Started by pmm in comp.dsp13 years ago 1 reply

Hi all, I am a student and quite new to dsp. I need some help with matlab. I am trying to write the matlab code that will sample the input data...

Hi all, I am a student and quite new to dsp. I need some help with matlab. I am trying to write the matlab code that will sample the input data stream u = [+1 -1 +1 +1 -1] and make pusle shaping to nyquist raised cosine pulse before being transmitted to the channel. Is it that the PAM block performs the sampling function but only selects one pulse (middle pulse) to trigger the Tx Filter? The ny...


Fast measurement of amplitude and phase

Started by Vladimir Vassilevsky in comp.dsp8 years ago 10 replies

There is a control loop which matches the amplitude and the phase of generated analog real sine wave to a digital reference. The frequency of...

There is a control loop which matches the amplitude and the phase of generated analog real sine wave to a digital reference. The frequency of the sine wave is known; it could be set anywhere from 0.001 to 0.999 of Nyquist. The system is very linear by itself, the noise could be neglected, and the waveform could be considered to be a pure sine. So, the system has to compensate the ampli...


Frequency multipliers: A way around Nyquist limitations?

Started by Joel Kolstad in comp.dsp14 years ago 16 replies

I was wondering... Say I happen to have a 100MHz DAC around... if I had a perfect anti-imaging filter, I could get 50MHz of bandwidth from it,...

I was wondering... Say I happen to have a 100MHz DAC around... if I had a perfect anti-imaging filter, I could get 50MHz of bandwidth from it, and with realistic filters I might get 40MHz. Say I take the filtered (analog) output from the DAC and feed it to a 4x frequency multiplier. Now my top frequency is something under 200MHz (perfect filters), or say 160MHz (realistic filters). I...


FFT, sampling rates and noise bandwidth.

Started by kyle in comp.dsp14 years ago 7 replies

Ok :) I'm trying to simulate a DAQ system. We have a fixed number of samples to make from a signal source and we're trying to measure the...

Ok :) I'm trying to simulate a DAQ system. We have a fixed number of samples to make from a signal source and we're trying to measure the amplitude of a synchronously sampled sinusoidal signal. I'm doing this with an FFT. In my simulation I am assuming a noise power per hertz (N0) and scaling by half the sampling rate, which is assuming a perfect rectangular filter at the Nyquist frequency...


Common practice - Aliasing in transition region?

Started by Anonymous in comp.dsp10 years ago 9 replies

I'm designing a low pass FIR and want to minimise the number of taps - no surprises there. The FIR is an antialiasing filter in a DDC The...

I'm designing a low pass FIR and want to minimise the number of taps - no surprises there. The FIR is an antialiasing filter in a DDC The target bandwidth is 80% of the available bandwidth eg Fs = 125MHz Passband is 50MHz Nyquist = 62.5MHz When designing the filter is it reasonable to have the stop band start at 75MHz? My rationale for this is that any signals in the range 62.5 to 75MHz ...