I know there's a few people out there who actually read the papers that I post on my web site. I also know that the papers have gotten a bit...

I know there's a few people out there who actually read the papers that I post on my web site. I also know that the papers have gotten a bit ragged, and that I haven't been maintaining them. So here: I've made a start. http://www.wescottdesign.com/articles/Sampling/sampling.pdf My intent (with apologies to all of you with dial-up), is to convert the ratty HTML documents to pdf as...

Hi, I'm new to OFDM. I read in several textbooks and papers that timing synchronization can be separated into symbol synchronization and...

Hi, I'm new to OFDM. I read in several textbooks and papers that timing synchronization can be separated into symbol synchronization and sampling clock synchronization. The latter can be further broken down into sampling phase offset and sampling frequency offset. There is also carrier frequency synchronization (i.e., carrier frequency recovery technique), which is utilized to eliminate the...

Hello; I wrote this basic demo to help me learn about DSP for a class (digital filtering) at my school that I am taking now, which...

Hello; I wrote this basic demo to help me learn about DSP for a class (digital filtering) at my school that I am taking now, which illustrates sampling a signal, then it generates the CTFT (cont. time Fourier transform) and then the DTFT (discrete time Fourier transform) from the samples. Both the magnitude and the phase are plotted. One can adjust the sampling frequency, signal dura...

I read somewhere that if sampling two signals with frequency f and a sampling rate of N. Then the uncertainty in phase shift between the...

I read somewhere that if sampling two signals with frequency f and a sampling rate of N. Then the uncertainty in phase shift between the two signals will be 360*f/N. Someone know under which conditions this is true and how this uncertainty can be minimized? Next question is, suppose the signals contains harmonics with order n, will then the uncertainty in phase angle between the harmonic be 360...

Hello! I need to implement a low pass filter with a flexible cut-off frequency of 10-50Hz for signals sampled at 40kHz. I believe that for...

Hello! I need to implement a low pass filter with a flexible cut-off frequency of 10-50Hz for signals sampled at 40kHz. I believe that for this approach a multirate and multistage filter is the best choose. So first the signal is down-sampled by a factor of approximately 100, then the signal is filtered with an IIR filter and finally the signal is up-sampling (may up-sampling is not required)...

I need to implement a particular filter that has 240 taps and it works/designed for a sampling rate of 125kHz. I want to change the sampling...

I need to implement a particular filter that has 240 taps and it works/designed for a sampling rate of 125kHz. I want to change the sampling rate of the filter from 125kS/s to 250kS/s to match the the hardware I am using, which operates at 250kS/s if i run the filter as is (at 250kS/s), the spectrum of the filter is stretched by 2. Is there an easy way to modify the filter taps to change ...

hi every body, i am using 6713DSK for my project.i use AIC23 for sampling a 1khz sin wave and collect them in a buffer.i configured MCBSP for 32...

hi every body, i am using 6713DSK for my project.i use AIC23 for sampling a 1khz sin wave and collect them in a buffer.i configured MCBSP for 32 bit one frame format.after sampling i shift 32 bits samples to right to just have 16 bits of left channel.then when i plot the collected samples in CCS it is not a 1khz sin wave.does anyone know what is my problem. i dont have mush time.please help me if...

Hi, it's well known that sampling a continuous function according to the sampling theorem requirements, you get all the information on...

Hi, it's well known that sampling a continuous function according to the sampling theorem requirements, you get all the information on the contunuous function just from the samples. now, if you have to get an accurate estimate of the integral of the continous function from samples satisfying the sampling theorem requirements, how can you get a good estimate of such an integral? I noticed that...

I am wondering if there is any criterion relating the ability to resolve the smallest time-duration echo to the sampling rate of an ADC used to...

I am wondering if there is any criterion relating the ability to resolve the smallest time-duration echo to the sampling rate of an ADC used to convert a detected analog sonar signal to a digital sequence. The Nyquist sampling theorem states that you must sample a signal at greater than 2*f_h, where f_h is the maximum frequency of interest in the analog signal. A common mistake is to a...

Given I have an input sequence [ 1 0 1 0 1 0 1 0 ], the DFS is x[n] = 0.5 + 0.5*exp(j*pi*n). The sampling (Ts) is 100 secs. Am I correct in...

Given I have an input sequence [ 1 0 1 0 1 0 1 0 ], the DFS is x[n] = 0.5 + 0.5*exp(j*pi*n). The sampling (Ts) is 100 secs. Am I correct in saying that the spectral coefficient 0.5 at pi is equivalent to saying the DFS has a 0.5 peak every 200 secs? My reasoning is between adjacent x[n] samples a time Ts has passed which corresponds to 2*pi of the sampling period. Therefore, half, i.e....

I'd like to ask a couple very basic questions. Being hands-on rather than mathematically-focused I'm trying to visualize sampling in a project...

I'd like to ask a couple very basic questions. Being hands-on rather than mathematically-focused I'm trying to visualize sampling in a project I'm planning to start. 1. By sampling say a 3.1 kHz band-limited voice channel at the Nyquist rate am I guaranteed to capture *all* the information in the channel? Is nothing lost (ignoring ADC resolution)? More to the point: To know what the ...

Gents, I know that the amount of system phase lag, introduced by a DSP alone, decreases with an increase in its sampling...

Gents, I know that the amount of system phase lag, introduced by a DSP alone, decreases with an increase in its sampling frequency. Assuming the ADC read + DAC write both happen within a sample period and assuming the only work the DSP does is setting its output=input, does anyone know a formula that describes the relationship between phase lag and sampling is linear? I set up a small...

Hello, I use the PADK (Professional Audio Development Board) with a TI TMS320C6727 DSP and I want to reduce the sampling rate to 8kHz......

Hello, I use the PADK (Professional Audio Development Board) with a TI TMS320C6727 DSP and I want to reduce the sampling rate to 8kHz... The problem is, that the minimum possible sampling rate of the board is 32 kHz, so I have tried to solve the problem with a simple counter... I have an interrupt service routine (ISR), which is called with 32 kHz. In this routine I have a counter, counting to...

Hi all, I would like to know if there is an easy method to do the following: I have a signal of, say, 10ms, sampled at 32kHz (320...

Hi all, I would like to know if there is an easy method to do the following: I have a signal of, say, 10ms, sampled at 32kHz (320 samples). I would like to convert this to a signal sampled with a varying sampling rate, for example, starting at 16 kHz, going down to 4 kHz at the end. This would involve a time-variant anti-alias-filter and non-uniform sub-sampling. Is there a standard...

"Fourier Transform" transform the time domain data into frequency domain information on the premise that the sampling rate is constant or in...

"Fourier Transform" transform the time domain data into frequency domain information on the premise that the sampling rate is constant or in some predictable manner. If the sampling clock jitter, it introduces noises into the frequency domain information. I would like to know, is there is formula whereby I can calculate the amount of noise introduced, based on how much the clock jitter ?

Hello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon...

Hello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon reciever. Now the problem is the sampling frequency of the signal is stuck to 48KHz on Denon screen. I tried changing the sampling frequency of signal using sound card's control panel (Sound cards tried: X-Mystique and M-Audio) but it didnt help . Can any...

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it....

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it. What is the maximum sampling rate I could achieve. I have used sound card in Linux for data acquisition(voice) maximum sampling speed with my laptop was 140000 samples/sec which means i could analyze frequencies up to 140000/2. now I want to use c671...

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it....

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it. What is the maximum sampling rate I could achieve. I have used sound card in Linux for data acquisition(voice) maximum sampling speed with my laptop was 140000 samples/sec which means i could analyze frequencies up to 140000/2. now I want to use c671...

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver,...

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver, SFO=(T'-T)/T At receiver, after CP removal, the (m,n)-th sample of the time-domain received signal is r'(m,n),(include SFO). How can we use Interpolation/decimation technique to correct the r'(m,n) (to get desied signal r(m.n) without SFO) Who can help m...

I'm trying to understand G.729. The only compression algorithm I've coded before is ADPCM, and it wasn't keyed to a certain sampling rate - that...

I'm trying to understand G.729. The only compression algorithm I've coded before is ADPCM, and it wasn't keyed to a certain sampling rate - that is, it would be just as happy with 8100 samples per second or 7900 samples per second as it would have been with 8000, just the quality would be slightly higher or lower. With G.729, all the documentation refers to a sampling rate of 8 KHz. Is it ...