How to detect signal from microwave sensor?

Started by gpezzella in comp.dsp10 years ago 4 replies

Dear Friends My application should detect signal that come from MicroWave cavity (the one used in house alarm). When people walk in front a...

Dear Friends My application should detect signal that come from MicroWave cavity (the one used in house alarm). When people walk in front a MW, a little signal of few mV and frequency between 10-50Hz is generated (beat signal). I should detect this! I'm developing on ATTiny micro family. Now: Since AtTiny have 20x amplifier buit in, which is better: a) add 3 new bit by Oversampling a...


Finding maximum of sinc in 0..1

Started by jungledmnc in comp.dsp3 years ago 11 replies

Hi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is...

Hi folks, the audio world is sort of obsessed with sort of "true peak level". Unfortunately it is rather tricky to calculate, so it is calculated by oversampling to about 192kHz and taking the normal peak level. Sadly that's really not true peak level. So how about the (nearly) correct value - if we take say 2 * N + 1 samples around each sample S[0] (hence S[-N] .. S[N]), we can calculate ...


Improving DAC Performance with Delta Sigma

Started by Randy Yates in comp.dsp7 years ago 13 replies

Hi Guys, Could we linearize a multi-bit DAC converter that may have nonlinearity in the lower bits by quantizing to a coarser resolution and...

Hi Guys, Could we linearize a multi-bit DAC converter that may have nonlinearity in the lower bits by quantizing to a coarser resolution and utilizing a delta sigma modulator to improve the SNR? For example, let's say our DAC is 18 bits but only has 16 "good" bits. If we design a delta sigma modulator with a 16 bit input, an oversampling ratio of four, and a modulator order of two, we co...


True Peak Measure

Started by Manuel Naudin in comp.dsp9 years ago 13 replies

Hello, I'm trying to set up a java program to make EBU R 128 measures on audio files. ( http://tech.ebu.ch/docs/r/r128.pdf ) I have an issue...

Hello, I'm trying to set up a java program to make EBU R 128 measures on audio files. ( http://tech.ebu.ch/docs/r/r128.pdf ) I have an issue with the True Peak measurement. It is basically described as four times oversampling the original signal by : padding the signal with zeros applying a low pass filter Here's a basic java source code : public class TestOversample { public stati...


ofdm & fft processing gain

Started by Chris Stratford in comp.dsp16 years ago 1 reply

For 802.11a there are 64 point ifft @ 20MHz transmit signal OFDM. The carriers are spaced 312500 Hz apart [20e6/64]. Demodulated baseband...

For 802.11a there are 64 point ifft @ 20MHz transmit signal OFDM. The carriers are spaced 312500 Hz apart [20e6/64]. Demodulated baseband signal is from -10MHz to +10MHz.[48 data carriers + 4 pilot carriers + some zero channels at the highest frequencies to make up the number to 64 carriers. Now at the receiver, if we have a 80MHz oversampling ADC sampling the 20MHz wide signal, followed b...


OFDM and oversampling

Started by 6.20 in comp.dsp14 years ago 8 replies

Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the...

Hi ! I would like to know if in practice, it is useful to oversample the signal at IDFT output. The sampling rate at the IDFT output is the Nyquist rate (2 times the frequency of the baseband signal) but I believe that it is necessary to oversample in order to perform the Root Raised Cosine Filtering. If I am right, what is the method to oversample : - Interpolation (using an IDFT size 2 times ...


oversampling

Started by CW in comp.dsp11 years ago 1 reply

Assume i receive a signal and perform direct conversion from passband to baseband with a front end analog filter with bandwidth +/- B for...

Assume i receive a signal and perform direct conversion from passband to baseband with a front end analog filter with bandwidth +/- B for my signal. In my ADC I sample at 2B and have a signal plus noise limited to the band +/-B with each noise sample assumed to be uncorrelated. If i instead decide to sample the signal at the output of the analog filter at 10B i will have correlated noise on ...


How to design the NCO in digital timing recovery loop?

Started by clari in comp.dsp12 years ago 2 replies

Hi, everyone. I have a problem about the NCO design and hope to get some help from this forum. I need to design a digital timing recovery...

Hi, everyone. I have a problem about the NCO design and hope to get some help from this forum. I need to design a digital timing recovery loop. The architecture consists of a Gardner timing error detector, loop filter and NCO. (To my best knowledge, such structure is quite classical.) All three building blocks shall be implemented in digital domain. The A/D oversampling rate is 32. I have no p...


interpolation accuracy, oversampling and fractional interpolation

Started by renaudin in comp.dsp13 years ago 30 replies

Hi all, Interpolation of a sampled signal x(n) to generate an up-sampled signal y(n) can be represented mathematically as: y(n) = x(n/L)...

Hi all, Interpolation of a sampled signal x(n) to generate an up-sampled signal y(n) can be represented mathematically as: y(n) = x(n/L) /* here L is the intrpolation factor. Regardless of the type of interpolation, if we increase the value of 'L' weather it will increase the accuracy of interpolation process? Whats about the fractional Interpolation/Decimation factors how to deal wi...


Obtaining better Fourier coefficients by oversampling

Started by Peter Mairhofer in comp.dsp9 years ago 1 reply

Hi, Suppose I have a (hypothetic, real) periodic, sign alternating function: The alternations take place at rate f_nyq ("Nyquist rate") and...

Hi, Suppose I have a (hypothetic, real) periodic, sign alternating function: The alternations take place at rate f_nyq ("Nyquist rate") and the pattern repeats every, say, 200 alternations. So the function is periodic with f_nyq/200. If I want to obtain the Fourier coefficients (magnitude) I can sample the function with an oscilloscope and measure the peaks (which are spaced by f_n...


Difference between CDMA pilot channel acquistion and tracking

Started by Laron in comp.dsp10 years ago 10 replies

Hi all, Did acquisition and tracking with pilot channel sharing the same process in CDMA2000? the acquisition using xcorr find the...

Hi all, Did acquisition and tracking with pilot channel sharing the same process in CDMA2000? the acquisition using xcorr find the coarse delay,but how does tracking action? Plan to using 8x oversampling,then fed the stream to a digital matched filter to find best one from the eight. But really have no idea with the tracking process, do i need to using xcorr again when tracking? or ...


Difference between CDMA pilot channel acquistion and tracking

Started by Laron in comp.dsp10 years ago

Hi all, Did acquisition and tracking with pilot channel sharing the same process in CDMA2000? the acquisition using xcorr find the...

Hi all, Did acquisition and tracking with pilot channel sharing the same process in CDMA2000? the acquisition using xcorr find the coarse delay,but how does tracking action? Plan to using 8x oversampling,then fed the stream to a digital matched filter to find best one from the eight. But really have no idea with the tracking process, do i need to using xcorr again when tracking? or ...


Solving DC offset problem in DSP

Started by seankuay in comp.dsp13 years ago 7 replies

Hi Experts, I am using a zero-IF system now, where the analog mixer output has a dynamic/variable DC offset. The analog mixer output is fed...

Hi Experts, I am using a zero-IF system now, where the analog mixer output has a dynamic/variable DC offset. The analog mixer output is fed into an op-amp for amplification(low amplification, no saturation) and finally into ADC. I am currently oversampling the incoming signals. My problem now is due to the variable/dynamic DC offset. This is because i am not able to find a neat way of gettin...


Does time-domain interpolation help Manchester decoding?

Started by in comp.dsp11 years ago 4 replies

My receiver comprises the following parts: 1) a 10-bit ADC running at fs (M samples/symbol), 2) a time-domain "interpolation filter" based on...

My receiver comprises the following parts: 1) a 10-bit ADC running at fs (M samples/symbol), 2) a time-domain "interpolation filter" based on a comb-integrator stage with leaky integrator (sampling rate increase by N) does "interpolation" and acts as a dc-blocking filter as well, 3) a Manchester decoder based on a oversampling & counting algorithm decoding the interpolated signal and runni...


Help me to understant sampling theory, decimation by integer factor

Started by hyjeon_0_o in comp.dsp12 years ago 7 replies

Hi, everyone! I read "decimation by integer" and "interpolation by integer factors" in a DSP book. I'm just wondering why integer...

Hi, everyone! I read "decimation by integer" and "interpolation by integer factors" in a DSP book. I'm just wondering why integer factor to downsampling... Sometimes, to resample signal by rational factor, people use conversion oversampling(integer factor) -> low-passfilter -> downsampling(also integer factor). (This means "Sampling rate conversion by non-integer factors") Howe