Resampling

Started by Inaki Val in comp.dsp14 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


Different Resampling Rates for Segments of a Sound File

Started by jeffdod in comp.dsp14 years ago 17 replies

I am having a problem with an application I am developing and don't know where to look for help. I thought I would try posing some questions...

I am having a problem with an application I am developing and don't know where to look for help. I thought I would try posing some questions here. I have an application where an analog sound source is recorded at a *very* slow rate (about 1/4 of the intended playback speed). Also, the machine playing the audio does not run at a regulated speed, which introduces a certain amount of "wow" in...


Frecuency shift and resample

Started by JAlbertoDJ in comp.dsp10 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/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 ...


Measuring Peak Value

Started by Wolfgang in comp.dsp15 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...


Why is there a spurious component in this resampling process?

Started by fl in comp.dsp2 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...


Resampling one signal to match another

Started by Randy Yates in comp.dsp6 years ago 34 replies

I have two digital signals, x1[n] at approximately 370 samples/sec, and x2[n] at approximately 400 samples/sec. How can I resample x1[n] so...

I have two digital signals, x1[n] at approximately 370 samples/sec, and x2[n] at approximately 400 samples/sec. How can I resample x1[n] so that it matches x2[n] without knowing the precise sample rate of either? -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com


NewBie Query: Resampling a digital waveform

Started by tanh in comp.dsp16 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...


Resampling for interference cancellation

Started by bendous in comp.dsp9 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.dsp12 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...


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 ...


Stereo to Mono with resampling?

Started by Anonymous in comp.dsp14 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...


Fractional Decimation in DSP

Started by Shenzhi in comp.dsp10 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


Physical continuation of analog filter (physical resampling)

Started by Peter Mairhofer in comp.dsp2 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 algorithm for limited hardware

Started by bitrex in comp.dsp4 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...


Fractional Resampling WCDMA gives noise

Started by DSPWirelessGuy in comp.dsp8 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...


Resampling questions - from 44.1kHz to 48kHz

Started by Newbie in comp.dsp16 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...


Resampling Polyphase Fir Filter

Started by angeleye in comp.dsp4 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 audio

Started by Nadav in comp.dsp11 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


resampling using duplication of samples

Started by kareem in comp.dsp14 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...