Hi, I needed some help understanding a derivation in Mr. Oppenheim's book Discrete Time Signal Processing. In Edition 3, Chapter 4 Sampling of...

Hi, I needed some help understanding a derivation in Mr. Oppenheim's book Discrete Time Signal Processing. In Edition 3, Chapter 4 Sampling of Continuous Time Signals, 4.6.1 Sampling Rate Reduction, Page 180, Eq. 4.74, he replaced the variable of summation from r to i + kM. I didn't understand how he did it or why it is valid and I would appreciate it if someone could clarify it.

Dear all, I'm sampling the audio signal through mic jack on C6713 board. The code is based on the example project "dsk_app". The audio can...

Dear all, I'm sampling the audio signal through mic jack on C6713 board. The code is based on the example project "dsk_app". The audio can be played back from the speaker jack. However, when I tranform the signal in the buffer to double precision values, it seemed that (later) half of the data in the buffer has been truncked (all zeored). Is there any one can give me some advice to solve this...

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...

Hello, I am trying to capture the motion of a mechanism with a high speed camera. The motion is a transient impact that reverses...

Hello, I am trying to capture the motion of a mechanism with a high speed camera. The motion is a transient impact that reverses direction with about 100 g's of acceleration. I have the displacement curve as acquired from a laser doppler sensor and I'm reading it on an oscilloscope with very high sampling rate. However, the camera is a much slower rate (4000 fps). My question: How...

I am having a little trouble with getting a clear picture of extracting what frequencies are represented in an input stream from the...

I am having a little trouble with getting a clear picture of extracting what frequencies are represented in an input stream from the DFT. Here's my current picture -- 1) Given N=16384 samples at a sampling rate of 5 secs or 0.25 Hz 2) 16384 samples = 81920 secs = 1365.3 min = 22.76 hrs 3) Each frequency sample of the DFT, Xk = c_k / N, the spectral coefficient in the discrete time f...

I am currently trying to simulate the s-curve of Gardner Timing algorithm. My simulation set-up is as follows, I am generating a QPSK signal...

I am currently trying to simulate the s-curve of Gardner Timing algorithm. My simulation set-up is as follows, I am generating a QPSK signal oversampled to a factor of 32. I am passing a received signal through a matched filter and then down-sampling the signal by 16 to get 2 samples/symbol required for Gardner TED. While performing down-sampling each time I am selecting 2 samples out of 32...

Hi! I'm trying to design a digital preenphasis filter - so far without any luck. The other post about a preemphasis filter design I found...

Hi! I'm trying to design a digital preenphasis filter - so far without any luck. The other post about a preemphasis filter design I found couldn't help me. I'm working with LabView, where I read wav files (so my sampling frequency is 44,1kHz) which I want to filter. What I tried so far is to calculate coefficients for a FIR filter in MatLab using frequency sampling. My MatLab code looks ...

Hi, I've a few fundamental question. I was going through the LTE specs. It said the fundamental symbol duration was .67us 1. If 30.72Mhz...

Hi, I've a few fundamental question. I was going through the LTE specs. It said the fundamental symbol duration was .67us 1. If 30.72Mhz is the sampling rate (15khz subcarrier spacing*2048 sub carriers) , how does LTE support bandwidths greater than 15.36, the spec specifies bandwidths upto 20Mhz. Even if I assume only 100 RBs are used, it still leaves me with a bandwidth of 18 Mhz. Doesn...

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...

Hi, I have a quantized input audio signal file (.wav). I want to convert the sampling rate from 11025 Hz to 24000. I know that I have to...

Hi, I have a quantized input audio signal file (.wav). I want to convert the sampling rate from 11025 Hz to 24000. I know that I have to upsample the signal, lowpass filter and downsample it. But how can I find the converting ratio? Is it 11025/24000 or 24000/11025? And I am new to DSP and also Matlab. So, anyhelp to write this sys in Matlab?Please help if you can. thanks a lot in adva...

Hi! I am using a MicroChip Dspic30 for sampling data, the data is about 10 to 30 kb and is after sampling stored in a external flash. Now i...

Hi! I am using a MicroChip Dspic30 for sampling data, the data is about 10 to 30 kb and is after sampling stored in a external flash. Now i want load this data and perform FFT on this data to calculate the maxiumum frequency. The problem is that i only have about 1kb data memory left. So the question is how to perform FFT whis this small data memory, have anyone tried it and made i working ...

Hi Guys; We are now designing a digital commnication system. Signal's sample rate is 2.56MHz, and signal bandwidth is 120kHz. I would like to...

Hi Guys; We are now designing a digital commnication system. Signal's sample rate is 2.56MHz, and signal bandwidth is 120kHz. I would like to run low pass filtering and decimation by 8 (320kHz) continiously. In this case, can I use frequnecy sampling filter(FSF) which is described in Richard Lyon's book, "Undestanding Digital Signal Processing". This book says that FSF is effective when t...

Hye , I am using 2 DSPs TMS320F2812, and i am trying to configure them one as master , the other as slave for a full duplex communication. ...

Hye , I am using 2 DSPs TMS320F2812, and i am trying to configure them one as master , the other as slave for a full duplex communication. IS the following configuration correct ? MAster -CLKX configured as output and using sampling rate generator -CLR configured as input and connected to CLKX of master -FSX configured as output and using Frame syncrhoniqtion provided bye sampling r...

I have a rather preliminary question about filter design.. Am I right in saying that an implicit assumption in filter design is that...

I have a rather preliminary question about filter design.. Am I right in saying that an implicit assumption in filter design is that the sampling is uniform ? Just like in FFT, where data has to be uniformly sampled. I have data which is non-uniformly sampled, and I have to develop a second-order HPF that can work on the data to reject low-freq. components. Is there a specific way to proceed...

Hi, Consider a sigma delta adc which first oversamples, does noise-shaping and then filters and decimates to give final output at...

Hi, Consider a sigma delta adc which first oversamples, does noise-shaping and then filters and decimates to give final output at Nyquist rate and with 16 bits per sample. For this final output, note that quantisation stepsize is uniform. Now, is it possible to get this same output using a normal adc for some sampling phase and uniform sampling of the analog signal ? Or will i...

I am new to the wonderful world of digital signal processing and have questions about down sampling. We have a biological signal originally...

I am new to the wonderful world of digital signal processing and have questions about down sampling. We have a biological signal originally sampled at 200 samples/sec which is producing huge amounts of data. We are only interested in frequencies 20Hz and below. So, I know *theoretically* I can low-pass filter the data with a cut-off frequency of 20Hz and then down sample at 40 sample/sec, with...

Hi, In smart antenna domain, the sampling rate normally is the same at the symbol rate. In TDMA systems, there are not many sync symbols which...

Hi, In smart antenna domain, the sampling rate normally is the same at the symbol rate. In TDMA systems, there are not many sync symbols which can be used as reference signals. In order to accormodate to the convergence of the adaptive algorithm, some longer sync symbol information is required. In my view, the symbol has been shaped to preserve spectrum efficiency. Why can't we use more samp...

Hi, I'm an hold analog engineer , actually developing a DSP based application and I need to change of 90 degrees the phase of a signal on the...

Hi, I'm an hold analog engineer , actually developing a DSP based application and I need to change of 90 degrees the phase of a signal on the range of 50 - 250 Hz aving the system a 200uSec sampling rate. I have considered the Hilbert FIR filter but for the moment I cannot get it to work not even reducing the sampling rate. In particular I nedd to phase shift by 90 degrees a signal (voltage)...

Could some DSP guru please clarify some ideas that I have ? Suppose that I am sampling some signal at some frequency fs. The sampled bytes get...

Could some DSP guru please clarify some ideas that I have ? Suppose that I am sampling some signal at some frequency fs. The sampled bytes get collected in a buffer (max buffer size is some power of 2). Let us suppose that I sample for some time and collect enough samples to completely fill up the buffer. As the sampling frequency is fs, I collect f samples per second, and so the time requ...

Pardon this intrusion from rec.audio.pro, but some said I might find an answer here. Here's what I understand: Fact: DSD has a sampling...

Pardon this intrusion from rec.audio.pro, but some said I might find an answer here. Here's what I understand: Fact: DSD has a sampling rate of 1 bit * 64 * 44100 per second. Assumption: Each sample raises or lowers the volume one "bit", one unit. Conclusion: A 22.05 kHz tone gets 128 samples and has a peak to peak height of -32 to +31 to -32 again (6 bits) and a 44.1 kHz tone only -1...