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Interpolator/Decimator Question

Started by Joe in comp.dsp14 years ago 10 replies

I was asked to create a Decimator/Interpolator to go between Narrow Band (8 KHz Sampling) and Wide Band (16 KHz Sampling). I am not so...

I was asked to create a Decimator/Interpolator to go between Narrow Band (8 KHz Sampling) and Wide Band (16 KHz Sampling). I am not so familiar with english engineer lingo, so does the above mean that the samplerate of the input signal which is fed to the decimator/interpolator-algorithm will either be 8KHz or 16KHz ? The implementation will be done in ANSI C. It has to a be a fixed point...


FFT, sampling rates and noise bandwidth.

Started by kyle in comp.dsp18 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...


Enery conservation during Sampling

Started by lordsathish in comp.dsp16 years ago 8 replies

Hi all... I am pretty new to signal processing... Let say we sample a band limited signal... So the spectrum of the sampled signal...

Hi all... I am pretty new to signal processing... Let say we sample a band limited signal... So the spectrum of the sampled signal will have spectral replications of the band limited signal and multiples of the Sampling frequency... So in this case the resulting signal seems to have more energy... since its spectrum is large compared to the bandpass signal and the sa...


FIR Filters using frequency Sampling

Started by Anonymous in comp.dsp15 years ago 2 replies

Hello All Our application is channel embed/de-embedding. The frequency characteristics of the channel are specified in a ...

Hello All Our application is channel embed/de-embedding. The frequency characteristics of the channel are specified in a (S-parameters file). To generate an FIR filter, matching the specified frequency response, Frequecny sampling method using IFFT is used. We have been able to use this with limited success. I am looking for the limitation's of this IFFT method - ...


OFDM/DFT/Sampling question.

Started by m26k9 in comp.dsp15 years ago 2 replies

Hello, I am pretty confused with some DFT/Sampling techniques and how these apply to OFDM. Fundamentally, if a (baseband) signal has a...

Hello, I am pretty confused with some DFT/Sampling techniques and how these apply to OFDM. Fundamentally, if a (baseband) signal has a highest frequency component of f Hz, that signal needs to be sampled at 2f Hz for aliasing-free data reconstruction. My confusion begins with OFDM, which does the process in reverse. That is IDFT is performed first. So, in OFDM, the output of the IDFT blo...


Audio FFT problem - PLEASE HELP

Started by cppt...@yahoo.com in comp.dsp16 years ago 1 reply

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are...

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are as follows: encoding : PCM sampling frequency : 16000 Hz resolution : 16 bits channel : mono signed : true endianness : little (as this runs on an Intel processor) The raw bytes get collected in a byte array, and I take two bytes at a time (resolution is 1...


Audio FFT problem - PLEASE HELP

Started by cppt...@yahoo.com in comp.dsp16 years ago 5 replies

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are...

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are as follows: encoding : PCM sampling frequency : 16000 Hz resolution : 16 bits channel : mono signed : true endianness : little (as this runs on an Intel processor) The raw bytes get collected in a byte array, and I take two bytes at a time (resolution is 1...


frequency sampling method implementation

Started by Luca Notini in comp.dsp20 years ago 1 reply

Dear group, I have implemented an algorithm that calculates the coefficients of an FIR filter using the frequency sampling method. The result...

Dear group, I have implemented an algorithm that calculates the coefficients of an FIR filter using the frequency sampling method. The result I obtain using my algorithm and that obtained using the fir2 function in MATLAB differ in the following way. My ideal frequency response consists of peak in the gain that is aproximated with a high degreee of accuracy by the frequency response of t...


Resampling audio

Started by Nadav in comp.dsp15 years ago

Hi, I am trying to extract MFCC coefficients for variable sample rate audio feed 1.Input Sample rate is dynamic and is not known in front (...

Hi, I am trying to extract MFCC coefficients for variable sample rate audio feed 1.Input Sample rate is dynamic and is not known in front ( an arbitrary media file ) though it is fixed per session and doesn’t change over time. 2.All Input feeds are being down sampled to a predefined sampling-rate ( say 8Khz or 16Khz ) 3.Re-sampling is done in the following way: a.2nd order Lowpass


HiperMAN Standard/OFDM

Started by Neelabh in comp.dsp16 years ago 1 reply

I recently started simulating a HiperMAN/WiMAX PHY on MATLAB. However, in the standard available in the ETSI website, there is mention of a...

I recently started simulating a HiperMAN/WiMAX PHY on MATLAB. However, in the standard available in the ETSI website, there is mention of a "sampling frequency" and a term "bandwidth over sampling frequency". I haven't a clue as to what these terms mean and where they're used in the OFDM process. Could somebody shed some light on what these terms mean in the context of OFDM? Thanks Neelabh...


resampling using duplication of samples

Started by kareem in comp.dsp18 years ago 2 replies

Hello. Generally we do up sampling by inserting zeros and passing through Image rejection filter. Instead of inserting zeros we can duplicate...

Hello. Generally we do up sampling by inserting zeros and passing through Image rejection filter. Instead of inserting zeros we can duplicate the samples and pass through a filter. the advantage of going for second mathod is i can choose a less order filter compared to the first methode. similarly is it possible to do the same methode for sampling rate conversion by non integer factor. if so wh...


Digital to Analog

Started by sauwen in comp.dsp16 years ago 2 replies

Hi, Does anyone know if there is a program out there that can change digital data in the form of binary files back into analog to resample it...

Hi, Does anyone know if there is a program out there that can change digital data in the form of binary files back into analog to resample it to a different sampling rate? I guess, the whole point of this is to get a different sampling rate. Is it possible? Seems pretty unheard of to change it back to analog.. :X Thanks!


sampling sound, how to capture the information of different "tones" ?

Started by Nimo in comp.dsp15 years ago 6 replies

Hi.., well, in frequency domain if we sample a "sound" signal, actually we are sampling "amplitude values" at discrete places of the...

Hi.., well, in frequency domain if we sample a "sound" signal, actually we are sampling "amplitude values" at discrete places of the wave. what my point is.., if we sample this simple sound wave shouted by 2 persons => " ‘cogito ergo sum’ – I think therefore I am " the sentences is same, but their "tones" are different how that information( tone difference ) is captur


why a 1 GHz oscilloscope has a 20 MS/s sampling rate

Started by alb in comp.dsp11 years ago 2 replies

Hi everyone, I must say that this question may sound pretty silly, but I honestly failed to find an answer. A couple of days ago I found an...

Hi everyone, I must say that this question may sound pretty silly, but I honestly failed to find an answer. A couple of days ago I found an 'old' oscilloscope in the lab which reported a 1 GHz bandwidth, but when we looked up the specs it reported a 20 MS/s sampling rate. Now, my background is probably not supporting me much here, but a 1 GHz bandwidth signal cannot be represented wit...


What is the advantage on high-sampling rate ?

Started by Arthur in comp.dsp20 years ago 66 replies

Hello all, Recently, there is a tendency to use high sampling rate 96 KHz, 192 KHz in the audio application. To my knowledge, about ~ 20 KHz,...

Hello all, Recently, there is a tendency to use high sampling rate 96 KHz, 192 KHz in the audio application. To my knowledge, about ~ 20 KHz, everything will be cut off from the loudspeaker, amplifier, etc.. So, why 96 KHz, 192 KHz instead of 48 KHz.. Regards Arthur


Arbitrary Digital Tone generation?

Started by Shafik in comp.dsp19 years ago 13 replies

Hello folks, Suppose Im sampling at 8khz. To generate a 4khz tone, its easy, just generate alternating 1,-1,1,-1, etc... To generate a 2khz...

Hello folks, Suppose Im sampling at 8khz. To generate a 4khz tone, its easy, just generate alternating 1,-1,1,-1, etc... To generate a 2khz tone, its also easy. How can one generate a rational fraction of the nyquist freq? I want to able to generate 3.333khz while sampling at 8khz. Any ideas? This method needs to be very computationally cheap, or else its useless in this context. ...


Re: Sampling Universe Theory

Started by Jerry Avins in comp.dsp16 years ago

HardySpicer wrote: > On Jan 3, 12:47 pm, curiosus_2...@yahoo.com wrote: > > Sampling Universe Theory > > > > * Foreword * > > > > As I...

HardySpicer wrote: > On Jan 3, 12:47 pm, curiosus_2...@yahoo.com wrote: > > Sampling Universe Theory > > > > * Foreword * > > > > As I am working in the field of digital signal processing, I noticed > > strong similarities between digital signal processing and quantum > > mechanics. > > > > I am aware that analogy does not mean identity, however I was curious > > to see how far that was possible


Re: Sampling Universe Theory

Started by HardySpicer in comp.dsp16 years ago

On Jan 3, 12:47 pm, curiosus_2...@yahoo.com wrote: > Sampling Universe Theory > > * Foreword * > > As I am working in the field of...

On Jan 3, 12:47 pm, curiosus_2...@yahoo.com wrote: > Sampling Universe Theory > > * Foreword * > > As I am working in the field of digital signal processing, I noticed > strong similarities between digital signal processing and quantum > mechanics. > > I am aware that analogy does not mean identity, however I was curious > to see how far that was possible to extend the analogy. > > It


An Interesting Historical Paper on Sampling

Started by Rick Lyons in comp.dsp15 years ago 5 replies

Hi guys, I recently ran across an interesting article on the origins of the Sampling Theorem. If such things interest you, the article is...

Hi guys, I recently ran across an interesting article on the origins of the Sampling Theorem. If such things interest you, the article is at both of the following two web sites: http://webee.technion.ac.il/courses/044130/00755459.pdf http://www.hit.bme.hu/~papay/edu/Conv/pdf/origins.pdf See Ya', [-Rick-]


nyquist sampling rate

Started by Anonymous in comp.dsp15 years ago 12 replies

hi all. i have a signal of bandwidth 2 MHz on a carrier of 5 MHz. That means the Nyquist sampling rate is (5+2/2)*2 = 12 MHz. But I have heard...

hi all. i have a signal of bandwidth 2 MHz on a carrier of 5 MHz. That means the Nyquist sampling rate is (5+2/2)*2 = 12 MHz. But I have heard that it is not a good idea to sample on or near nyquist rate because of the sinx/x roll of charactersistics of ADC. Please guide should i oversample it a little such as 16 or 20 MHz. I am working on a communication system so the system is performance ...