OFDM/DFT/Sampling question.

Started by m26k9 in comp.dsp10 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...


Chroma Decimation in Digital Video Signals

Started by Randy Yates in comp.dsp13 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...


How to find phase shift of signal through a system

Started by karanbanthia in comp.dsp6 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 ...


A tricky problem

Started by Elnaz in comp.dsp7 years ago 1 reply

Hi all, I am convolving a vector of bits (zero-padded in between to avoid aliasing) with the erf function. On the paper, erf function goes...

Hi all, I am convolving a vector of bits (zero-padded in between to avoid aliasing) with the erf function. On the paper, erf function goes from - infinity to infinity which will cause erf functions in my convolutoin to cancel each other's tails out. However the problem is that in MATLAB no matter how long I define the function I will continue to have problem with the tails not canceling ea...


Anti-aliasing filtering for interleaved sampling approach

Started by Alexz in comp.dsp11 years ago 3 replies

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to...

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to built a sampling system to achieve 500 MHz sampling rate by means of two 250 Msps ADCs while driving them with samlping closk featured by 180 deg. phase shift. So that in time domain would be sampling by two channels in interleaved manner. Then the enti...


Beating Nyquist?

Started by Andor in comp.dsp12 years ago 48 replies

Friends, just now I stumbled upon this webpage: http://www.edi.lv/dasp-web/ Specifically, in this...

Friends, just now I stumbled upon this webpage: http://www.edi.lv/dasp-web/ Specifically, in this chapter http://www.edi.lv/dasp-web/sec-5.htm they state that they can sample a 1.2GHz signal using a pseudo-random sampling instants with an average rate of 80MHz (in the last line of section "5.2 Aliasing, how to avoid it"). I know that for nonuniform sampling, a generalization of...


FFTW speed !!

Started by m.baldasseroni in comp.dsp12 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.dsp13 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...


Delta Sigma ADC + channel switching

Started by Moritz v. Buttlar in comp.dsp15 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...


Band limited signal into soundcard with sample & hold?

Started by Ben Jackson in comp.dsp13 years ago 13 replies

Let's say I have a signal of

Let's say I have a signal of


CAUTION! was "What is the advantage on high-sampling rate ?"

Started by Rick Lyons in comp.dsp15 years ago 150 replies

Hi Guys, in a recent thread, mention was made of a "Sampling" paper by Dan Lavry. At the following web...

Hi Guys, in a recent thread, mention was made of a "Sampling" paper by Dan Lavry. At the following web site http://www.lavryengineering.com/pdfs/sample.pdf you can see Dan's 1997 paper: "Sampling, Oversampling, Imaging and Aliasing - a basic tutorial". I recommend caution if you decide to read that paper. In the second paragraph Dan wrote: *** Sampling theory *** ...


software anti-aliasing filter.

Started by Anonymous in comp.dsp5 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


How can a filter impulse response be interpolated?

Started by fl in comp.dsp2 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...


Decimation design

Started by seb in comp.dsp15 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...


Higher upsampling with minimum phase downsampling produces more aliasing

Started by jungledmnc in comp.dsp5 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 ...


Nyquist rate for sampling complex-valued data?

Started by kiki in comp.dsp14 years ago 14 replies

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than...

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than Nyquist rate and it still can avoid aliasing and reconstruct perfectly... Is that true? Any theory behind it? Thanks a lot


Shannon limit question

Started by Richard Owlett in comp.dsp14 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? ...


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

Started by qhe in comp.dsp13 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...


FM carrier offset and aliasing effects

Started by jacobfenton in comp.dsp8 years ago 5 replies

I am doing digital FM demodulation using the common ATAN2 and first difference method. When the carrier is offset from center, there is...

I am doing digital FM demodulation using the common ATAN2 and first difference method. When the carrier is offset from center, there is a non-linear region that appears and seems to be tied to the FM demodulation sample rate. Without be able to change the FM demodulation sample rate, are there any work arounds from this non-linear region? This non-linear region is just outside my bandwidth of ...


Order of LPF at DAC output

Started by krishna_1105 in comp.dsp10 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...