Started by in comp.dsp18 years ago 1 reply

I am trying to implement FM algorithms for computer music applications. The language I'm using (Nyquist) has a primitive FM oscillator called...

I am trying to implement FM algorithms for computer music applications. The language I'm using (Nyquist) has a primitive FM oscillator called fmosc. I want to use modulators in series which is a very common technique in some Yamaha synths. In pseudo code it is the composition fmosc(fmosc(osc())) where all oscillators output sine waves. When I have tried this there are strong non-harmonic co...

## Fundamental Sampling Theory and Real-time Constraints

Started by in comp.dsp9 years ago 18 replies

Hi everyone, I'm new to the arena of DSP and have arrived at a bit of a sampling-based conundrum. I'm wondering if anyone here can help me out...

Hi everyone, I'm new to the arena of DSP and have arrived at a bit of a sampling-based conundrum. I'm wondering if anyone here can help me out with possible solutions, or can simply set my understanding straight. I have an application where my signal of interest has a bandwidth of 500Hz. At Nyquist, I'd be sampling at 1 kHz. Of course, I could oversample by this by some factor and filter it ...

Started by in comp.dsp11 years ago 13 replies

Hello this is the first time that I approach to DSP so I'm really newbie. I have to write a program that realize a Low Pass Filter at 50...

Hello this is the first time that I approach to DSP so I'm really newbie. I have to write a program that realize a Low Pass Filter at 50 Hz. My processor (ATTINY) have 10bit ADC so for first time I have used Oversampling and Decimation technique for add 3 "Virtual Bit" First Question: Since my Oversampled Frequency is: 4^3 * F-nyquist = 64 * 100 = 6.4KHz ...

## Re: Blinded by math. Was "How does an inverter affect phase?"

Started by in comp.dsp15 years ago 1 reply

Andor wrote: > Jerry Avins wrote: > > ... > > > > > Such a definition can be via specification of the response for one > > > > > single...

Andor wrote: > Jerry Avins wrote: > > ... > > > > > Such a definition can be via specification of the response for one > > > > > single sinusoid, and requiring linearity and time-invariance. As soon > > > > > as you do that, the response at DC and Nyquist follows, and we can stop > > > > > quibbering and just look at the facts. > > You find a reasonable mathematical way to express an intuitively simple >

## Polar Diagram

Started by in comp.dsp12 years ago 3 replies

Hi to all! I'm studing Nyquist plot for assessing the stability of a system with feedback. I'd like to calculate point where the graph...

Hi to all! I'm studing Nyquist plot for assessing the stability of a system with feedback. I'd like to calculate point where the graph intersect Real and Imaginary axis. I've found a method witch is based on Routh table, using it it's possible to find the point of intesect. Does anyone know where I can find somo information about? Thaink you very much! David

Started by in comp.dsp10 years ago 5 replies

May be this is a stupid question, so pardon me. For QPSK what is typically the frequency between the I, Q carriers, and the base...

May be this is a stupid question, so pardon me. For QPSK what is typically the frequency between the I, Q carriers, and the base band frequencies ? I just want to verify if the Nyquist criteria is satisfied, that is the I, Q carrier frequencies are atleast > = 2x the baseband frequency.Thanks in advance for your help.

## min/max delay filters

Started by in comp.dsp13 years ago 1 reply

Hello, This is help for a homework question, and I would not like someone to do it for me, just help point me in the right direction. The...

Hello, This is help for a homework question, and I would not like someone to do it for me, just help point me in the right direction. The question is: Consider the two-element filter [1.0 0.6]. Construct a finely spaced dimensionless frequency vector that extends from -1 to 1. Calculate and plot the amplitude and phase lag spectra for the filter. Identify the Nyquist frequency on the...

## Resampling with minimum delay

Started by in comp.dsp10 years ago 12 replies

I had to resample a signal with requirement of minimal processing delay. So I made a Lagrange polynomial extrapolator to predict the signal on...

I had to resample a signal with requirement of minimal processing delay. So I made a Lagrange polynomial extrapolator to predict the signal on the duration of +1 sample into the future. That is straightforward and it works good enough for the job. However what could be the other options for extrapolation of the Nyquist bandlimited signal? What is an optimal solution for this case? ...

## Undersampling

Started by in comp.dsp14 years ago 7 replies

I am on a team developing a system that will record and process large (1 GHz) bandwidth pulses. Doing some research I have came across some one...

I am on a team developing a system that will record and process large (1 GHz) bandwidth pulses. Doing some research I have came across some one that suggested sampling in the 2nd nyquist zone, for two main reasons. Sampling the signal at a higher frequency makes our analog front end much better and simpler deign. It also reduces harmonics within our frequencies of interest allowing more in...

Started by in comp.dsp16 years ago 4 replies

Hi, I have read the chapter on "Quadrature Sampling" from Rick Lyons book. The point where it is said that the quadrature samplers can work...

Hi, I have read the chapter on "Quadrature Sampling" from Rick Lyons book. The point where it is said that the quadrature samplers can work at half the Nyquist rate (Fs/2) seems a little confusing to me. In SSB or VSB signals, where the spectrum when centred around zero after downconversion is not symmetric around zero Hz, this looks obvious. But for double sideband spectrum (say AM) w...

## Pitch detection algorithm

Started by in comp.dsp16 years ago 3 replies

Hello! I'm trying to develop an application whose primary purpose is to detect which tone the user is singing (real time) using frequency...

Hello! I'm trying to develop an application whose primary purpose is to detect which tone the user is singing (real time) using frequency analysis, and I wonder which algorithm to use. I analyze samples of approx 1/10 sec. I first tried using fourier transform, but since the frequencies are spread out evenly up to the Nyquist frequency, the resolution is way too bad (approx 3 Hz). I n...

## Equation for Phase Response of a SOS

Started by in comp.dsp13 years ago 3 replies

I'm close to an answer on this topic, but there is something I'm missing. I have found the following equation in a textbook for determining the...

I'm close to an answer on this topic, but there is something I'm missing. I have found the following equation in a textbook for determining the phase response of a second-order circuit: PHI(w) = Angle(K) - atan(((w/wc) - (wc/w))/(1/Q)); where wc: Filter cutoff in radians w : range from 0:pi (nyquist) Q : Quality/Damping factor But when I tried this exact code in Matlab and...

## Filtering before changing sample rate

Started by in comp.dsp16 years ago 6 replies

I think I know the answers, but then again ;) I have a "hobby" project to evaluate intelligibility of speech with varying parameters (...

I think I know the answers, but then again ;) I have a "hobby" project to evaluate intelligibility of speech with varying parameters ( primarily sample rate and bandwidth ). [ for perspective, in a previous incantation I "discovered" formants ;] My sample universe is a reading of the Bible recorded under studio conditions. As it is a commercial recording, I'm assuming Nyquist is the ...

## sampling ...

Started by in comp.dsp8 years ago 6 replies

Sirs, I have some questions related to sampling and subsequent re-construction of signals. I am hsitoping to get answers ... 1) Assume a...

Sirs, I have some questions related to sampling and subsequent re-construction of signals. I am hsitoping to get answers ... 1) Assume a signal consists of two signals of frequencies f1 and f2 such that frequncy f1 < f2. Sampling is done to ensure fs > = 2xf1 and fs < 2xf2. Since the sampling rate does not meet the nyquist rate for f2, it would lead to f2 aliasing into f1. When the discre

## Convert a complex stream to a real one at twice the rate

Started by in comp.dsp13 years ago 1 reply

Could someone pass mea pointer to how the GC1012 converts a complex output stream to a real one at twice the rate? Similarly how it offsets...

Could someone pass mea pointer to how the GC1012 converts a complex output stream to a real one at twice the rate? Similarly how it offsets the spectrum by one-fourth the Nyquist rate? TIA!

## Variable rate interpolation

Started by in comp.dsp12 years ago 6 replies

I'm primarily an analog and logic designer, so please bear with my lack of DSP chops. I've got an application where I'm looking to feed a DAC...

I'm primarily an analog and logic designer, so please bear with my lack of DSP chops. I've got an application where I'm looking to feed a DAC at a constant 128 MSPS from a data stream that will be a power of 2 fraction of that (64 MSPS down to 250 kSPS). I'd like to be able to get pretty close to the Nyquist limit (say 0.4 or so) of whatever the original data rate is. It seems so far li...

## Question about a Bandlimited Process

Started by in comp.dsp8 years ago 17 replies

Is it possible to even simulate a true bandlimited process? That is we would like the spectrum to be zero at some frequency less that nyquist....

Is it possible to even simulate a true bandlimited process? That is we would like the spectrum to be zero at some frequency less that nyquist. Surely passing gaussian noise through some FIR filter would shape the noise to be 'small' in the band of non-interest but is not exactly zero. But seeting up a FIR is probably a quick and easy way out. Would a better means of simulating a bandlimit...

## Oversampling 8KHz to 44.1KHz

Started by in comp.dsp17 years ago 16 replies

There's a 8KHz audio sample rate I want to convert up to 44.1KHz sample rate. What's the right approach to keep Mr. Nyquist happy?

There's a 8KHz audio sample rate I want to convert up to 44.1KHz sample rate. What's the right approach to keep Mr. Nyquist happy?

## DSP Implementation Woes

Started by in comp.dsp14 years ago 13 replies

http://s188.photobucket.com/albums/z...t=envelope.jpg I am using a real-time DSP chip and I get a modulated continous output which I am unsure...

http://s188.photobucket.com/albums/z...t=envelope.jpg I am using a real-time DSP chip and I get a modulated continous output which I am unsure if it a biproduct of the DAC process. If I sample at just fractionally above the Nyquist rate, I obtain a discrete sequence that is described (without loss of any generality) that is similar to the linked image. When this sequence is resynthesised...

## Re: Sampling: What Nyquist Didn't Say, and What to Do About It

Started by in comp.dsp11 years ago

Wow, you got quite an audience for this paper so far... Using Acrobat 6.0 the fonts look fine to me. I will comment on the initial state of...

Wow, you got quite an audience for this paper so far... Using Acrobat 6.0 the fonts look fine to me. I will comment on the initial state of the doc on opening. The bookmarks panel is open even though there is nothing in it and the document is sized to fit the remaining screen space. I suggest that you turn off the bookmarks panel in the initial view since it is empty. Also, I