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...

Hi all... I am pretty new to signal processing... Let say we sample a band limited signal... So the spectrum of the sampled signal...

Hi all... I am pretty new to signal processing... Let say we sample a band limited signal... So the spectrum of the sampled signal will have spectral replications of the band limited signal and multiples of the Sampling frequency... So in this case the resulting signal seems to have more energy... since its spectrum is large compared to the bandpass signal and the sa...

I am sampling voltage values. I need to convert it to the frequency domain using FFT . As I understand the frequency steps after FFT will be the...

I am sampling voltage values. I need to convert it to the frequency domain using FFT . As I understand the frequency steps after FFT will be the integer multiples of Fs/N (where Fs is the data sample rate in Hz, and N is the FFT length). How can I have sub-Hz frequency steps? Do I use use long sampling times and long FFT lengths, for instance, Sampleing at 1 kHz and using 10000 FFT length will...

The sampling theorem tells us that we must sample at least twice the bandwidth of a signal. When the signal is centered around dc to some max...

The sampling theorem tells us that we must sample at least twice the bandwidth of a signal. When the signal is centered around dc to some max freq say Fmax than we can say Fs> =2Fmax. However, when the signal is centred around some other higher frequency (say Fc), how is bandwidth defined? is it from Fc (the center freq) to the max value or from Fc-minfreq to Fc + maxfreq ? If you apply te

Hello, I have an ENG signal, sampled at 25kHz that has high mains interference that I want to remove. The problem is I need the surrounding...

Hello, I have an ENG signal, sampled at 25kHz that has high mains interference that I want to remove. The problem is I need the surrounding frequencies, maybe not 50-70, but I can't afford to sacrifice much more than this. The filter designs I've been looking at don't seem to be able to do this at such a high sampling rate. Any suggestions? Would downsampling help? Thank you in adva...

Hi, Does anyone know if there is a program out there that can change digital data in the form of binary files back into analog to resample it...

Hi, Does anyone know if there is a program out there that can change digital data in the form of binary files back into analog to resample it to a different sampling rate? I guess, the whole point of this is to get a different sampling rate. Is it possible? Seems pretty unheard of to change it back to analog.. :X Thanks!

Dear all, I have to develop an application with the following basic requirements: - 40 analog channels of Voltage/Current - ADC with 16 bits...

Dear all, I have to develop an application with the following basic requirements: - 40 analog channels of Voltage/Current - ADC with 16 bits of resolution - at least 96 sample point per cycle (50 and 60Hz), which means 4800 and 5760Hz of sampling frequency - simultaneous sampling of all channels The main concern is data acquisition. It is a like a data logger. Since a lot of mat...

I recently started simulating a HiperMAN/WiMAX PHY on MATLAB. However, in the standard available in the ETSI website, there is mention of a...

I recently started simulating a HiperMAN/WiMAX PHY on MATLAB. However, in the standard available in the ETSI website, there is mention of a "sampling frequency" and a term "bandwidth over sampling frequency". I haven't a clue as to what these terms mean and where they're used in the OFDM process. Could somebody shed some light on what these terms mean in the context of OFDM? Thanks Neelabh...

Suppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60...

Suppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60 secs...so how is this possible? The values of the capacitors etc would be hugh. Of course I could over-sample1000s of times...is this the norm? K.

on what basis the fft length is calculated related to the sampling frequency?I want to search for a maximum peak by obtaining the...

on what basis the fft length is calculated related to the sampling frequency?I want to search for a maximum peak by obtaining the spectrum of the signal.I am working on demodulator in that my task is to acquire the carrier from the receive modulated signal.Modulation used is BPSK.It can be used for 300,600,1200,2400,10800,5400 bit rates.sampling frequency is 518400 for 10800 and 5400 bit rate...

on what basis the fft length is calculated related to the sampling frequency?I want to search for a maximum peak by obtaining the spectrum of the...

on what basis the fft length is calculated related to the sampling frequency?I want to search for a maximum peak by obtaining the spectrum of the signal.I am working on demodulator in that my task is to acquire the carrier from the receive modulated signal.Modulation used is BPSK.It can be used for 300,600,1200,2400,10800,5400 bit rates.sampling frequency is 518400 for 10800 and 5400 bit rates,for...

This problem comes from my simulation of Single Carrier Frequency Domain Equalization(SC-FDE) system. Suppose the symbol rate is Ts, for reason...

This problem comes from my simulation of Single Carrier Frequency Domain Equalization(SC-FDE) system. Suppose the symbol rate is Ts, for reason of pulse shaping filtering I perform fractional space sampling with sample rate Ts/2. A FFT block length is 512 and the pilot(used for channel estimation) length is 64. Channel estimation is done as follows: (1) use length 64*2(sampling rate Ts/2) pi...

Hello People I have been scratching my head on this doppler problem: Because of the doppler effect, the received signal will either be...

Hello People I have been scratching my head on this doppler problem: Because of the doppler effect, the received signal will either be expanded or compressed on time axis. Now, the expansion or compression is sampling rate conversion. So, I tried to use 'resample' function of matlab. The actual length of the signal is 30720 and sampling freq is 48000Hz. Now, when I use y=resample(x,100...

Hi, I have a conceptual question for the group, which I have been struggling with. It can be summarized as "what is the difference between...

Hi, I have a conceptual question for the group, which I have been struggling with. It can be summarized as "what is the difference between sampling offsets and carrier offsets on a received constellation?". In more detail... Suppose there is a demodulated BPSK signal at the output of an ideal bandpass channel and the receiver is perfect. The received constellation will be at +1, -1. No...

Hi all, I am another guy getting stuck with the same old stuff! The relationship between point in a frequency response and the sampling...

Hi all, I am another guy getting stuck with the same old stuff! The relationship between point in a frequency response and the sampling frequency/time characteristics of the impulse response. I have created a custom frequency array, representing a frequency response from an algorithm, between 0 and 300Hz, with 0.2Hz resolution, ie 1500 points. I then need to create the impulse response from ...

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 ...

Assume there is a real function f and its spectrum F. If f is discrete sampled, we have samples at some t = st * k, where st is sampling period...

Assume there is a real function f and its spectrum F. If f is discrete sampled, we have samples at some t = st * k, where st is sampling period and k = 0, 1, 2, ..., N - 1. Also assume every frequency in f is within [-fc..fc] range, fc = Nyqist frequency. If f is a simple sine function whose minimums, zeros and maximums are at the sampling points, F will be very "good", without leakage and ju...

Sorry if these are newbie questions, but can anyone give me good descriptions of these? The bin splitting is obviously related to the samples...

Sorry if these are newbie questions, but can anyone give me good descriptions of these? The bin splitting is obviously related to the samples per second, the sampling time, and the frequency resolution. It seems related to the number of samples that you fit into the sampling period, but the concept is not totally clear to me. Thanks for any advice.

Suppose we have an ADC with three inputs (multiplexed). The ADC takes each input in turn. Suppose the frequency that the ADC scans ...

Suppose we have an ADC with three inputs (multiplexed). The ADC takes each input in turn. Suppose the frequency that the ADC scans bewteen inputs is fd. How should fd be related to the sampling rate per channel fs? 10 times bigger or 100 times bigger? and if the scan rate is too slow what effect does this have. Obviously the inputs cannot be matched in time because of the scan rate. Hard...

It is necessary to solve a following problem. Frequency of sampling ? 120kH. It is necessary to calculate and realize the low-frequency...

It is necessary to solve a following problem. Frequency of sampling ? 120kH. It is necessary to calculate and realize the low-frequency filter with a passband 29 kH, frequency of stopband 30 kH, ripple in a passband no more 0,1dB, suppression not less 96dB, the phase should be linear. After a filtration a signal decimated twice, up to frequency of sampling 60 kH. What approaches can be for t