wide band aliasing noise in the spectrum of non-uniformly sampled signal

Started by qaisar in comp.dsp16 years ago 2 replies

Hello, As you know that whenever we perform GDFT( General Discreat Fourier Transform), on a nonuniformly sampled data, we obtain...

Hello, As you know that whenever we perform GDFT( General Discreat Fourier Transform), on a nonuniformly sampled data, we obtain wideband alysing noise alongwith signal spectrum, which donot allow us to analyse the signal with apprepriate approximation. If any one have some idea how to solve this problem, please tell me. I shall be thankfull to you for your help. Qaisar ...


Anti-Aliasing filter

Started by naebad in comp.dsp15 years ago 21 replies

Ok I am sampling with an ordinary A/D (not sigma-delta) at 11025kHz which mean I need any noise to be attenuated at half sampling which is at...

Ok I am sampling with an ordinary A/D (not sigma-delta) at 11025kHz which mean I need any noise to be attenuated at half sampling which is at 5512.5Hz. I have read that the nosie level needs to be less that the R.M.S Quantisation Level of my A/D. Now I swing +or - 10 volts with 16 bits. So my Quantisation level is delta = dynamic range/2^16=20/65536=0.000305176 volts. (or 0.305mV) Now...


help with complex decimation and band shifting

Started by Kamil in comp.dsp12 years ago 5 replies

Hi there, I am working in the frequency domain and this must be done in the frequency domain. I have a signal that I would like to decimate...

Hi there, I am working in the frequency domain and this must be done in the frequency domain. I have a signal that I would like to decimate by a factor of 4. Do I simply keep every 4th bin and keep it at that? Would I have to apply a lowpass filter in order to prevent aliasing? If so, how to do I determine the correct lowpass filter to use? After my complex decimation, I need to do a b...


Interchanging of filtering and decimation operations

Started by vasindagi in comp.dsp12 years ago 28 replies

Hi, It is known that downsampling introduces aliasing if the signal to be downsampled has high frequency components. Hence it is passed through...

Hi, It is known that downsampling introduces aliasing if the signal to be downsampled has high frequency components. Hence it is passed through a low pass filter before it is downsampled. I want to know whether the filtering and the downsampling operations can be interchanged as they are essentially the same, or is it always necessary to filter the signal first and then downsample it. Thanks i...


How can a filter impulse response be interpolated?

Started by fl in comp.dsp3 years ago 19 replies

Hi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a...

Hi, I know how to interpolate a digital signal. It is first interpolated by inserting 0's. For example, one can add 4 0's to each data for a 5 times interpolation. Then, a low pass filtering to eliminate the aliasing frequency. Now, I have a low pass filter from 0 to 10 MHz pass band with a sampling rate of 40 MSPS. I want to get the same 0 to 10 MHz response (it is not a flat pass band...


Delta Sigma ADC + channel switching

Started by Moritz v. Buttlar in comp.dsp17 years ago 1 reply

Hi ! I have the following question: If I use an AD7738 (Analog Devices) or ADS1256 (TI) 24bit delta/sigma ADC with "fast channel switching" (I...

Hi ! I have the following question: If I use an AD7738 (Analog Devices) or ADS1256 (TI) 24bit delta/sigma ADC with "fast channel switching" (I think it's some mechanism to flush the digital filter so that you don't have to throw away 6+ samples before you get a valid one), what effect on the anti- aliasing filter requirements and transfer function does the channel switching have ? It wo...


Order of LPF at DAC output

Started by krishna_1105 in comp.dsp12 years ago 3 replies

Hi, In the case of an anti-aliasing filter used prior to an ADC, the filter order is derived from the cut-off frequency being twice the...

Hi, In the case of an anti-aliasing filter used prior to an ADC, the filter order is derived from the cut-off frequency being twice the maximum frequency of the signal, stop-band attenuation and pass-band ripple defined by the ADC resolution. In the case of DAC, where LPF is basically used to smooth out the quantisation steps, how do we decide the filter order. It does seem intuitive that the be...


How to find phase shift of signal through a system

Started by karanbanthia in comp.dsp8 years ago 2 replies

Hi, Signal from the sensor passes through the following blocks in my system - Ac Coupling circuit, Anti-Aliasing filter and then goes to DSP...

Hi, Signal from the sensor passes through the following blocks in my system - Ac Coupling circuit, Anti-Aliasing filter and then goes to DSP controller where it is low-pass filtered twice. I want to calculate the total phase shift introduced by the system at frequency 50Hz. Following are the required data : Phase Delay Group Delay ...


sinusoids and aliasing...

Started by fisico32 in comp.dsp10 years ago 5 replies

Hello forum, while a composite signal (made of many sinusoids), if sampled at a sampling frequency f_s at least twice the largest frequency in...

Hello forum, while a composite signal (made of many sinusoids), if sampled at a sampling frequency f_s at least twice the largest frequency in the signal, can be "uniquely" reconstructed from its samples, a continuous pure sinusoid of freq f instead, no matter if sampled at twice or more its frequency, will give samples that can be the samples of other sinuosids, all those with frequency f+-n...


Nyquist Condition question.

Started by A.E lover in comp.dsp13 years ago 1 reply

Hi all, (1) I know the famous Nyquist Condition, f(t) with bandwidth B is sampled without aliasing if Fs> 2B. Today I read another...

Hi all, (1) I know the famous Nyquist Condition, f(t) with bandwidth B is sampled without aliasing if Fs> 2B. Today I read another thing called Nyquist Condition which says: for a continuous time signal x(t), take x(t) convolute with itself and then sample the obtained signal at Fs, if the sampled signal =delta[n], then Fs satisfies the Nyquist condition i.e: g(t)= x(t) (*) x


FFTW speed !!

Started by m.baldasseroni in comp.dsp14 years ago 13 replies

Good morning, I'm using FFTW library for my project. I'm working on a PowerPC 7447 processor with vxWorks operating system. I have generated...

Good morning, I'm using FFTW library for my project. I'm working on a PowerPC 7447 processor with vxWorks operating system. I have generated fftw library through gcc 2.95 compiler using the sequent options for obj files: -O3 -fomit-frame-pointer -fstrict-aliasing -fvec-eabi -mcpu=7450 I'm working in single precision. My problem is the speed of fft elaboration. I'm using the BASIC interfa...


Continuous-time DSP with no sampling

Started by Yannis in comp.dsp15 years ago 72 replies

In principle, sampling is not necessary in order to do filtering digitally. This is discussed in the following paper: Y. Tsividis, "Digital...

In principle, sampling is not necessary in order to do filtering digitally. This is discussed in the following paper: Y. Tsividis, "Digital signal processing in continuous time: a possibility for avoiding aliasing and reducing quantization error", Proc. 2004 IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, vol. II, pp. 589-592, Montreal, May 2004. (If you are interested but can...


Chroma Decimation in Digital Video Signals

Started by Randy Yates in comp.dsp15 years ago 31 replies

Hi, I've seen several cases where the chroma decimation, e.g., when going from YCrCb 4:4:4 to 4:2:2, is done by simply throwing away every...

Hi, I've seen several cases where the chroma decimation, e.g., when going from YCrCb 4:4:4 to 4:2:2, is done by simply throwing away every other sample. I have two questions regarding this operation: 1. If there were a 1-D signal, simply throwing away every other sample without first filtering would cause (potentially) aliasing. Why doesn't this happen in a chroma signal, or wh...


Decimation design

Started by seb in comp.dsp17 years ago 9 replies

Hello, I am looking for good advices about my design. I am working with an ADC which work beetween 1kHz and 2kHz but the application need...

Hello, I am looking for good advices about my design. I am working with an ADC which work beetween 1kHz and 2kHz but the application need data sampled beetween 10 Hz and 2 kHz. So i decide to make decimation in order to get the sample frequence beetween 10Hz and 1 kHz. But in order to do it in the rigth way i make pre-filtering in order to avoid aliasing. So the decimation factor coul...


[ANN] HICE-218X: EZ-218x compatibal emulator released

Started by qhe in comp.dsp14 years ago

Dear 218x users: HICE-218x is another emulator for ADSP-218x family fixed-point DSPs. It's compatible with EZ-ICE-218x, but with a few...

Dear 218x users: HICE-218x is another emulator for ADSP-218x family fixed-point DSPs. It's compatible with EZ-ICE-218x, but with a few improvements: a.. USB2.0 interface, powered by USB, no AC adapter needed b.. Working at 460800 bps by virtual serial port and baud rate aliasing c.. Thoroughly optimized timing makes it much faster than EZ-ICE-218X d.. Hardware stepping, no limit...


Shannon limit question

Started by Richard Owlett in comp.dsp16 years ago 2 replies

I read that band limiting the input to an A/D eliminates aliasing. I also read that using a very high sampling rate makes that filter very...

I read that band limiting the input to an A/D eliminates aliasing. I also read that using a very high sampling rate makes that filter very simple. Suppose I: 1. filter with a single stage low pass RC filter with 3db pint at .5 MHz 2. sample at 44.1 MHz 3. do simple sample rate conversion by saving every thousandth sample Would I have a nice artifact free 44.1 kHz digitized signal? ...


Band limited signal into soundcard with sample & hold?

Started by Ben Jackson in comp.dsp14 years ago 13 replies

Let's say I have a signal of

Let's say I have a signal of


Higher upsampling with minimum phase downsampling produces more aliasing

Started by jungledmnc in comp.dsp6 years ago 20 replies

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by...

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by taking DFT, zeroing high octave(s) and IDFT. For generating particular pitch a choose a wavetable, which has all harmonics until 20k. Sound good so far, way better than just upsampling the original non-band-limited wavetable. The harmonics that exceed 22k ...


software anti-aliasing filter.

Started by Anonymous in comp.dsp6 years ago 27 replies

Hey guys, I designed a system to acquire certain physiological signals and recorded these signals using a data acquisition card (agilent u2531a...

Hey guys, I designed a system to acquire certain physiological signals and recorded these signals using a data acquisition card (agilent u2531a specifically). the problem is when designing the circuit i was unable to place an anti-aliasing filter before feeding the signal to the data acquisition device. The card does not have a software selectable filter at the input stage either. i


using IIR or FIR to implement the lowpass filter for downsampling?

Started by Nasser M. Abbasi in comp.dsp9 years ago 12 replies

For downsampling (decimation) one normally uses low pass filter before downsampling (anti-aliasing filter). I have thought that FIR is the...

For downsampling (decimation) one normally uses low pass filter before downsampling (anti-aliasing filter). I have thought that FIR is the best choice here for the low pass filter implementation for this case. But I was reading this page: http://www.dspguru.com/dsp/faqs/multirate/decimation and it said in 2.3.1 under "How do I implement decimation?" "To implement the filtering part...