Hello, I am developing a software modulator and am currently working on the qpsk portion of it. This problem will apply to pretty much...

Hello, I am developing a software modulator and am currently working on the qpsk portion of it. This problem will apply to pretty much all modulation schemes however. Let me first explain my algorithm: This program needs to be very fast (capable 100 mhz sampling rates) so I am pregenerating all of my samples. The user inputs the center frequency and the sampling rate and I fill a seri...

Is there a Nyquist sampling theory for spatial dimensions? Suppose point A to point B is about 1 mile, how many sensors I should place to get...

Is there a Nyquist sampling theory for spatial dimensions? Suppose point A to point B is about 1 mile, how many sensors I should place to get a fair estimation of the property of the whole range?

Hello! This is my first post on the site. I would like to ask a question if I may. I am trying to decode Mode S reply signals for my diploma...

Hello! This is my first post on the site. I would like to ask a question if I may. I am trying to decode Mode S reply signals for my diploma thesis. The first thing I must get up and going is preamble detection. My teacher suggested that the analog processing before sampling would remove any DC component of the signal. He also suggested using a 16 MHz sampling frequency. Seeing as how the...

Hello, I have an ENG signal, sampled at 25kHz that has high mains interference that I want to remove. The problem is I need the surrounding...

Hello, I have an ENG signal, sampled at 25kHz that has high mains interference that I want to remove. The problem is I need the surrounding frequencies, maybe not 50-70, but I can't afford to sacrifice much more than this. The filter designs I've been looking at don't seem to be able to do this at such a high sampling rate. Any suggestions? Would downsampling help? Thank you in adva...

Hello, I'm trying to add 2 different digital waveforms together, but the 2 are sampled at different rates.. Logically, first I must...

Hello, I'm trying to add 2 different digital waveforms together, but the 2 are sampled at different rates.. Logically, first I must resample them to another common sampling rate, but.. how should I go about it? Can I reconstruct the analog waveform and then resample at the desired sampling rate? If my priority is to preserve the frequency content, can I just simply use some sort of c...

Sorry if these are newbie questions, but can anyone give me good descriptions of these? The bin splitting is obviously related to the samples...

Sorry if these are newbie questions, but can anyone give me good descriptions of these? The bin splitting is obviously related to the samples per second, the sampling time, and the frequency resolution. It seems related to the number of samples that you fit into the sampling period, but the concept is not totally clear to me. Thanks for any advice.

Hi all I have a board with 6 daisy chained AD7687 pulsar ADCs. I have these ADC connected via a FPGA to an ADSP2186 running at 40MHz. ...

Hi all I have a board with 6 daisy chained AD7687 pulsar ADCs. I have these ADC connected via a FPGA to an ADSP2186 running at 40MHz. Signals I'm measuring are in the 45 to 65Hz range. I employ a synchronous sampling mechanism where the sampling rate is always a 2^N multiple of the incoming base frequency. In this specific case e.g. for 50Hz the sample rate is 50x128x8=51200Hz. ...

I am planning to use a Lagrange interpolator in the context of fractional delay filter to change the sampling rate by a factor between 1 and 1.5...

I am planning to use a Lagrange interpolator in the context of fractional delay filter to change the sampling rate by a factor between 1 and 1.5 (lower). My signal bandwidth is about 0.23 when sampling frequency fs is normalized to 1. The signal has 12 bits resolution and I would like to maintain this resolution at the output of the interpolator. How can I determine the degree of the Lag...

Hi, Many papers argue that todays bottleneck is signal processing/communications is the ADC (because DSP is no problem because of Moore's...

Hi, Many papers argue that todays bottleneck is signal processing/communications is the ADC (because DSP is no problem because of Moore's law). There are many approaches which promise to reduce the sampling rate such as compressed sensing or finite rate of innovation sampling (which do not (yet) work in practice). On the other hand, there are cheap high resolution ADC available at...

Very simple task. We have a sensor that puts out a frequency proportional signal to its input. We have a freq2volt converter but would like to...

Very simple task. We have a sensor that puts out a frequency proportional signal to its input. We have a freq2volt converter but would like to take it out, and just sample the waveform to extract the frequency. The range is from 5KHz to 8KHz and we'd like to have accuracy in the range of 6-12Hz sampling at 8bits accuracy. Probably 24KHz should do it. Can someone just go over the basic algorithm ...

Hi all, I have the following question related from sampling. Suppose I have the original band pass signal with spectrum in [-250, -150]...

Hi all, I have the following question related from sampling. Suppose I have the original band pass signal with spectrum in [-250, -150] and [150, 250]Hz. The negative and positive spectrum are antisymmetrical themselves, which means the spectrum is not locally symmetrical with respect to the axis -200 Hz and +200Hz... For simplicity, you can take two non-equallateral triangles at -2...

Hello, everybody. I'm just a beginner. I use ADSP-21065L SHARC EZ-KIT Lite where is codec AD1819A. The default sampling rate is 48 kHz and I need...

Hello, everybody. I'm just a beginner. I use ADSP-21065L SHARC EZ-KIT Lite where is codec AD1819A. The default sampling rate is 48 kHz and I need to change it. But it's so difficult for me to do it. I would like to change it for example from 48 kHz to 8 kHz. Please help me if you can. Thank you very much Marek The code: /*---------------------------------------------------------------------...

The sampling theorem tells us that we must sample at least twice the bandwidth of a signal. When the signal is centered around dc to some max...

The sampling theorem tells us that we must sample at least twice the bandwidth of a signal. When the signal is centered around dc to some max freq say Fmax than we can say Fs> =2Fmax. However, when the signal is centred around some other higher frequency (say Fc), how is bandwidth defined? is it from Fc (the center freq) to the max value or from Fc-minfreq to Fc + maxfreq ? If you apply te

Hi all We are designing a 128TCM QAM receiver. Our idea is to take advantadge from the Digital Downconverters (DDC as GC4016 or similar)...

Hi all We are designing a 128TCM QAM receiver. Our idea is to take advantadge from the Digital Downconverters (DDC as GC4016 or similar) to convert from HF to baseband directly ,let's say sampling at 1MHz, then DDC (CIC + decimation filters), and then a DSP system at 50KHz sampling time where the complete receiver will be implmented. The problem we are guessing is about timing recove...

Hello I must sample 4 analog signals, and I must ideally sample them at the same time. And I'm going to down sample the data rate later....

Hello I must sample 4 analog signals, and I must ideally sample them at the same time. And I'm going to down sample the data rate later. The sampling frequency at the ADC is 10 kHz and afterwards the signal gets LP filtered to 50 Hz bandwidth before down sampling to 200 Hz rate. But what happens if I use a muxed ADC. The last channel is always sampled later than the first cha...

Hi, Sorry if this question is simple. I am trying to get this confusion out of my head. I have a hard time visualizing and understanding this...

Hi, Sorry if this question is simple. I am trying to get this confusion out of my head. I have a hard time visualizing and understanding this simpling question. Let's say that we have only one tone signal (eg. 100MHz). So, when you look in the frequecny domain you will see a tone at 100MHz. Now, when we do sample it let's say at 400MHz sampling frequency, why do we see tones at (400-100...

I was once asked over a telephonic interview what the sampling rate of a square wave.. What would be the best answer for this.

I was once asked over a telephonic interview what the sampling rate of a square wave.. What would be the best answer for this.

Hello everyone, I am trying to design a low pass decimator filter in MATLAB. I am supposed to decimate a signal sampled at 200MHz down to...

Hello everyone, I am trying to design a low pass decimator filter in MATLAB. I am supposed to decimate a signal sampled at 200MHz down to 10MHz. The signal bandwidth is 8 MHz and the signal spectra is centred at the sampling frequency. I began with the following code: --------------------------------------------------------------- Fs_adc = 200e6; % ADC Sampling Frequency Fpass1 =...

A problem I had while doing graphics, a short time ago, is that while doing repeated up/down-sampling cycles, there was continual degradation....

A problem I had while doing graphics, a short time ago, is that while doing repeated up/down-sampling cycles, there was continual degradation. This of course leads to the idea: a general framework for resampling that avoids this problem. Specifically, let M -> N denote the operation (and its matrix) for sampling from positive sizes M to N. The ideal situations are that: * M -> N s

Somebody asked me this question and I was unsure of the answer. It is well known that if you quadruple the sampling rate that you reduce the...

Somebody asked me this question and I was unsure of the answer. It is well known that if you quadruple the sampling rate that you reduce the quentisation noise by 6dB. Normally this means you can drop off one bit.So if you have 12 bits you can make do with 11 bits by sampling 4 times faster and maintain the same quantisation noise level. Taking this to its logical conclusion if we have 16 bi...