Why is there a spurious component in this resampling process?

Started by fl in comp.dsp3 years ago 1 reply

Hi, Thank all of you for replying to my previous post on filter impulse response. One of the replies said to directly calculate the fir coef...

Hi, Thank all of you for replying to my previous post on filter impulse response. One of the replies said to directly calculate the fir coef as the original filter. I considered this method. The only problem is too time consuming. As it is not a general FIR design, I have to find the background information to write the code from the bottom. OK. I have solved the problem with all of your in...


Perceived aliasing in small ratio SRC

Started by somenoob in comp.dsp14 years ago 15 replies

I'm pretty new to the world of frequency domain as proven by this question: Is there any correlation between the quality of resampler required...

I'm pretty new to the world of frequency domain as proven by this question: Is there any correlation between the quality of resampler required to make aliasing "imperceptible" and the magnitude of sampling rate change? I'm currently playing with various resampling algorithms, running some 44.1kHz content through them (with lots of high and low frequencies) and I noticed increasing the samp...


audio sampling rate question

Started by Phil in comp.dsp12 years ago 11 replies

with the newer flash recorders I now have higher resolution and sample rate options then my older 44.1/16 PCM I realize I can record at lower...

with the newer flash recorders I now have higher resolution and sample rate options then my older 44.1/16 PCM I realize I can record at lower safer levels with the 24bit resolution but I was wondering if sampling at 96 or 88 and resampling to my final level of 44.1 after post processing has any advantage. -- Phil


Demodulator Timing Recovery Architecture Question

Started by Randy Yates in comp.dsp14 years ago 12 replies

Which would be a better architecture: using a hardware VCO to drive the ADC as part of the PLL for timing recovery, or using a fixed-crystal...

Which would be a better architecture: using a hardware VCO to drive the ADC as part of the PLL for timing recovery, or using a fixed-crystal oscillator for the ADC and resampling in the digital domain? Assume the latter is done on a processor and not in hardware. -- % Randy Yates % "Watching all the days go by... %% Fuquay-Varina, NC % Who are you and who ...


Resampling/interpolation (uniform and non-uniform case)

Started by Alfred Bovin in comp.dsp10 years ago 19 replies

Hi all. I'm doing some work on a commercial black box system with three sensors that are being sampled at 10 Hz. One of the sensors are...

Hi all. I'm doing some work on a commercial black box system with three sensors that are being sampled at 10 Hz. One of the sensors are running freely, while the two other sensors are polled by a linux computer. For one of the polled sensors there is a unpredictable delay from when I request data to it is being received. This means that I'm basically getting the information from the ...


Frecuency shift and resample

Started by JAlbertoDJ in comp.dsp11 years ago 12 replies

I have a audio signal from 1000Hz to 1250Hz with samplerate of 8000Hz. But i want translate the signal to the range: 0-250 Hz. Then i could...

I have a audio signal from 1000Hz to 1250Hz with samplerate of 8000Hz. But i want translate the signal to the range: 0-250 Hz. Then i could resampling to 1000Hz, for example, with a low pass filter (antialiasing) with Fc=500Hz. ┬┐How can i translate frecuency from 1000Hz to 0Hz?


Resampling

Started by Inaki Val in comp.dsp15 years ago 26 replies

Dear, I want to resample a real signal from for instance 1024 -> 1027 samples. I prefer not to do it in the temporal domain, in order to...

Dear, I want to resample a real signal from for instance 1024 -> 1027 samples. I prefer not to do it in the temporal domain, in order to avoid large upsampling and downsampling steps. I have read that one possibility could be zero padding in frequency, but for this example I must use a IDFT and not IFFT to transform the signal to time domain. I have heard something about chirp-z transform, but


Measuring Peak Value

Started by Wolfgang in comp.dsp16 years ago 21 replies

Dear all, I've a stream of sampled values which were filtered and downsampled. I'm interested in the (exact) maximum value of an incomming...

Dear all, I've a stream of sampled values which were filtered and downsampled. I'm interested in the (exact) maximum value of an incomming peak. Is there a simpler way of finding that peak value instead of upsampling and searching the greatest value ? (The frequency content of the peak reaches half of the sampling rate, hence i've not a lot values around the maximum without resampling wit...


Physical continuation of analog filter (physical resampling)

Started by Peter Mairhofer in comp.dsp3 years ago 1 reply

Hi, Suppose I have frequency domain measurements from a channel between 1 and 10 GHz - for example like (c) in this...

Hi, Suppose I have frequency domain measurements from a channel between 1 and 10 GHz - for example like (c) in this figure: https://www.researchgate.net/profile/Sergey_Rylov/publication/2983309/figure/fig1/AS:39471058680 6272@1471117745152/Fig-1-Backplane-channel-characteristics-a-Backplaneline-card-application-b.pp m What is the best way to upsample the impulse response to a higher rate?


Resampling for interference cancellation

Started by bendous in comp.dsp10 years ago 10 replies

Hi all, I am not a DSP expert so please go easy on me. I have a VLF radio signal centred at 3KHz. The signal is corrupted with the mains...

Hi all, I am not a DSP expert so please go easy on me. I have a VLF radio signal centred at 3KHz. The signal is corrupted with the mains harmonics, which seem to be over 20dB higher than the signal! I am trying to implement an algorithm that isolates an out-of-band harmonic using a band-pass filter, resamples it several times to obtain the in-band harmonics, and subtracts them from the corru...


Farrow Filter implementation in Matlab

Started by Imke Huismann in comp.dsp13 years ago 9 replies

Hi, for my diploma thesis I am studying different resampling techniques. Just now I wrote a Farrow resampler in Matlab. The structure of...

Hi, for my diploma thesis I am studying different resampling techniques. Just now I wrote a Farrow resampler in Matlab. The structure of the resampler is quite easy to implement, but I struggle with the polynomial coefficients. I tried to derive them from FIR filters, but although the filter approximations look good in time domain, the attenuation is quite low (20dB-40dB for an original fi...


Fractional Resampling WCDMA gives noise

Started by DSPWirelessGuy in comp.dsp9 years ago 15 replies

Dear all, I will be grateful if I can get some light on this. I have Tx Data upsampled from 3.84MHz to 12 MHz (@3.125X. On the Tx side I...

Dear all, I will be grateful if I can get some light on this. I have Tx Data upsampled from 3.84MHz to 12 MHz (@3.125X. On the Tx side I use a nice RRC filter to perform interpolation and upsample using Matlab inbuild polyphase filter to generate Test vectors. Problem is , on the receiver side, I do not use polyphase filter. I use RRc filter at the begining then downsample using 3,3,3...


NewBie Query: Resampling a digital waveform

Started by tanh in comp.dsp17 years ago 3 replies

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...


Fractional Decimation in DSP

Started by Shenzhi in comp.dsp11 years ago

Hi, Friends! Recently, I have some questions about Fractional Decimation. I think about them for a lot of time, but can't find a good...

Hi, Friends! Recently, I have some questions about Fractional Decimation. I think about them for a lot of time, but can't find a good solution. It comes from my reading Douglas W.Barker's paper "Efficient Resampling Implementations " presented on "Tips&Tricks" IEEE Signal Processing Magazine,2008. I found some interrelated materials in Mr. Fred Harris's book < < Multirate signal proces


Stereo to Mono with resampling?

Started by Anonymous in comp.dsp15 years ago 8 replies

Hello, This is my first post here. I don't know much on DSP but I think I found something interesting. Suppose we have a 44.1Khz stereo...

Hello, This is my first post here. I don't know much on DSP but I think I found something interesting. Suppose we have a 44.1Khz stereo wave file. We can treat it like 88.2Khz mono since stereo pairs are interleaved. If we downsample this to 44.1Khz (using a lowpass filter first), the result is a mono version of the original file, but not the same if we just added the two channels. It surp...


Resampling algorithm for limited hardware

Started by bitrex in comp.dsp5 years ago 1 reply

Suppose I have a waveform stored in flash memory on a fairly limited microprocessor, say an AVR 8 bit or something, and I'm sending it out to...

Suppose I have a waveform stored in flash memory on a fairly limited microprocessor, say an AVR 8 bit or something, and I'm sending it out to analog via one of the high-speed PWM channels. Is there a interpolation algorithm of some type that will allow bandlimited frequency transposition of this waveform in real time over a significant range on such limited hardware? Links to any examp...


Resampling Polyphase Fir Filter

Started by angeleye in comp.dsp5 years ago 19 replies

hello, i am trying to do 12 to 1 downsampling and 1 to 12 upsampling with polyphase fir filter, I wanted to know how to design the inner...

hello, i am trying to do 12 to 1 downsampling and 1 to 12 upsampling with polyphase fir filter, I wanted to know how to design the inner filter. suppose passband frquency is 0.5, stopband freq is 0.75. input sample rate is 24 ,inband ripple is 0.1 and outband attenuation is 80db.and it is 12 path filter with filter length is equal to 348 .each path is followed by 29 taps.how to design the inner ...


Resampling questions - from 44.1kHz to 48kHz

Started by Newbie in comp.dsp17 years ago 43 replies

I've had a look at dspguru.com and am trying to implement a program that will resample an audio signal from 44.1kHz to 48kHz. I'm running into...

I've had a look at dspguru.com and am trying to implement a program that will resample an audio signal from 44.1kHz to 48kHz. I'm running into some problems - namely too many zeros in my converted file. My algorithm is basically: Take 44.1kHz signal and interpolate by padding the signal with L-1 = 159 zeros in order to oversample. Apply a low pass filter with a stop band freq. of 24k...


Setting sampling rate of Tektronix VSA? Resampling non-integer rates?

Started by Peter Mairhofer in comp.dsp6 years ago 2 replies

Hi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g....

Hi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g. using SENSe:ACQuisition:BANDwidth or SENSe:IQVTime:FREQuency:SPAN), the sampling rate is always 150 MHz! Frustrating... With an Agilent MSG I am transmitting an LTE signal which is sampled at 30.72MHz and to be received by the Tektronix VSA. (But ...


Resampling by multistage decimation (Richard Lyons book)

Started by MRR in comp.dsp9 years ago 7 replies

Hello everybody, I am trying to understand the multistage decimation problem stated by Richard Lyons in his book "understanding digital signal...

Hello everybody, I am trying to understand the multistage decimation problem stated by Richard Lyons in his book "understanding digital signal processing" 1st edition, page 306. It is as follows: - We have an input data arriving at 400 kHz and we want to decimate by 100, but we are going to do this in two stage: 50 and 2. - The original signal bandwidth is something greater than 100 kHz, ...