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3
votes
1answer
31 views

Proof from wave equation of upper limit of active noise cancellation

Anybody know of a proof of the hard upper frequency limit of active noise cancellation using the ear canal volume as a boundary condition to solve for the upper frequency limit? I keep on encountering ...
0
votes
0answers
17 views

what is magnitude responce at ω = π/2M for freqz function [duplicate]

what is magnitude responce at ω = π/2M.Code Given is ...
1
vote
1answer
30 views

Difference b/w freqz and fft

Since both fft ,freqz are used for finding the frequency response of digital filters. the frequency response using freqz function is given below and frequency response using fft function is it ...
0
votes
0answers
38 views

Audio block-processing discontinuities

I have an algorithm that looks at an audio frame, calculates a threshold in db-rms, and decides to filter the frame whether or not the threshold is crossed. However, filtering each block is producing ...
1
vote
0answers
10 views

Why do I have to duplicate the labels used for the classification process?

I'm using the BCI Competition dataset IV 2a. I'm working on Left & right hand movement. I extracted data of channels (Fz,Cz,Pz). I filtered data [8-30], then normalized it. I calculated the ...
0
votes
1answer
20 views

adding phase noise in signal thorugh freuquency

I want to implement a simple signal for phase noise. According to my supervisor, i shoud be adding phase noise in the frquency. This is my signal. ...
-2
votes
1answer
35 views

Why 1536 FFT size for 15 MHz in LTE

Why we use 1536 point FFT for 15 MHz bandwidth, when we can use 1024 point FFT, as in bandwidth of 15 MHz we have only 900 subcarriers.
1
vote
0answers
59 views

Hilbert transform of a bandpass signal

I have a bandpass signal x. I am taking hilbert transform of it. I want to reconstruct amplitude and phase of the signal.Therefore, i am using Hilbert transform. ...
1
vote
0answers
68 views

Hilbert Transform, filters - two different phase graphs

Relationship between magnitude response and phase response for minimum phase is: Phase plots are obtained from Hilbert transform of real magnitude values. Input to Hilbert transform was a vector ...
3
votes
2answers
73 views

derivative filter transfer function

In many of the papers it is said that the derivative filter transfer function is given by: $$H(z) = \dfrac{1}{8T}\left(-z^{-2} - 2z^{-1} + 2z + z^{2}\right)$$ But no one gave the detailed information ...
0
votes
1answer
60 views

Filters - Hilbert Transform and Absolute Value of Magnitude Response

Following relationship between magnitude response and phase response for minimum phase: I have implemented in C++ code using the Hilbert Transform C code from file found online ht.c. However, I am ...
0
votes
0answers
121 views

Hilbert Transform - C or C++ code implementation

For a minimum phase system relationship of magnitude response to phase response is: In the search for C code implementation I have found a ht.c file on www.physionet.org. However, this code ...
0
votes
2answers
72 views

Raised-Cosine Filter - phase response graphs for three kinds of filter

In the previous post it has been mentioned that the phase response of Raised-Cosine filter is linear and is given with ϕ(ω)=−ωT, where T is the delay. But, I have also read in literature that it is ...
0
votes
2answers
40 views

Raised-Cosine Filter - calculating magnitude response

Here is the formula to compute magnitude of a raised-cosine filter: I know that 0 <= Beta <= 1. I am not sure what are the values for f (frequency) and Ts. I assume that some normalization ...
0
votes
1answer
87 views

Raised-Cosine Filter - calculating phase response

I want to compute phase response of an raised-cosine FIR filter. I have trouble finding the complex transfer function of the filter in literature (there is a real valued transfer function of ...
1
vote
3answers
155 views

IIR Filter - calculating the phase response

Using formulae from Audio EQ Cookbook from here it was easy to implement biquad IIR filters in C++, i.e. to calculate coefficients b0, b1, b2, a0, a1, a2. Now, I want to calculate the phase response ...
0
votes
1answer
31 views

Shift a signal by a given number of samples

Is shifting a signal by a given number of samples, say $n$ samples, the same as the following in matlab: signal(n:end) instead of signal(1:end). That is, if the size of $signal$ is 200x1, does it ...
1
vote
1answer
57 views

How to find phase shift and process I/Q channel

I am currently working of 24GHz FMCW radar signal processing. The Module I'm using has 2 receiving antennas and each has 2 outputs (I and Q channel). So overall I've 4 channels. I'm using combination ...
1
vote
1answer
23 views

Path loss effect on BER

Does including log distance path loss to a communication system cause a drastic difference to the BER of the system? In other words, if I have a simulation for a communication system without path ...
0
votes
0answers
15 views

Parameter selection for log distance path loss model

I am currently referring to the following book MIMO-OFDM Wireless Communications with Matlab in which the log distance path loss model is given by the equation: $P_{LD}(d)[dB] = PL_F(d_0) + ...
0
votes
1answer
11 views

Comparison of Baseband simulation results and experimental results at Passband

Consider that I have simulated a communication system at baseband and would implement the same on hardware but using a carrier at 2.4GHz(Passband conversion). Would it make sense to compare the ...
0
votes
2answers
50 views

Log Distance Path Loss Model

While simulating a MISO system, I am trying to consider the effect of varying the distance between the transmitter and receiver on the BER. My search for the effect of varying distance has pointed me ...
0
votes
1answer
43 views

Simulation at Baseband or Passband

A naive question I guess. Would it make a difference in the results when a system is simulated at Baseband or at Passband. If I have a plot of a BER vs SNR curve that has been plotted at Baseband, ...
0
votes
0answers
44 views

Raised-Cosine filter - design

I have a requirement to design bandpass Raised-Cosine filter with the magnitude and phase response as outlined in the picture below. I have read some literature online (still learning DSP) and I have ...
5
votes
2answers
260 views

Shift a signal by fraction of a sample

I have a signal with a sample time of 0.5 micro seconds and I would like to shift this signal by a fraction of the sample time, say by 3 nano seconds. I have read a few online resources about ...
0
votes
0answers
52 views

Biquad high-pass filter - calculating filter slope

I have to design high-pass IIR Biquad filter with cutoff frequency of 80Hz, sampling frequency 24kHz, and filter slope of 24dB/oct @ 80Hz. I have used formula for HPF biquad coefficients from here. ...
0
votes
0answers
67 views

FIR Filters: determining optimal number of taps and phase slope

As a design input for an low-pass FIR filter I have 6dB@1.6kHz, 24 dB/oct, sample frequency 48kHz. I need to specify phase response, and for that I need a phase slope. I have found, from Julius O. ...
0
votes
0answers
62 views

Half Sine Pulse Shaping design

The standard of 802.15.4 specfies the following 'The chip sequences representing each data symbol are modulated onto the carrier using O-QPSK with halfsine pulse shaping.' For pulse shaping the ...
0
votes
2answers
105 views

Upsampling by zero padding or duplicating sequence

I need to upsample a 2 MHz signal by a factor of 3000 to convert the OQPSK modulated baseband signal to passband. When I upsample by duplicating the elements in the sequence, I seem to get less error ...
0
votes
1answer
37 views

Frequency Spectrum of a sequence

The OQPSK modulator in Matlab produces an output sequence as follows: [0.707 0.707+0.707i -0.707-0.707i ...and so on ] This is the sequence and I am trying to view the frequency spectrum of this ...
0
votes
0answers
23 views

Theoretical BER - 802.15.4

The 802.15.4 standard gives a formula for the BER calculation but they have considered interference while computing this formula according to what I understood. I am trying to find out what the ...
0
votes
0answers
32 views

Frequency response of OQPSK Modulated Signal

I am a little confused as how the frequency spectrum of an OQPSK modulated signal would look like. I have a DSSS that is OQPSK modulated with a bandwidth of 2 MHz. The modulated signal has complex ...
0
votes
0answers
73 views

BER of OQPSK Modulation

I am unable to find out a formula for the BER for the OQPSK Modulation. Currently, I am using the berawgn function from Matlab to calculate the BER for OQPSK but I am looking for an expression for ...
0
votes
0answers
31 views

FIR Filter Design - from slope and cutoff frequency [duplicate]

I am stuck with the following design requirements for FIR filter: Low-pass -6dB @ 1.6kHz Slope: 24 dB/octave. Ideally I would like to get also FIR coefficients, but any help how to design this ...
0
votes
1answer
44 views

Convert low pass continuous time filter design to bandpass, discrete time

I am trying to convert the continuous time transfer function of a second order lowpass Butterworth filter is given by: To a bandpass fourth order bandpass digital filter, I first apply the mapping ...
2
votes
0answers
52 views

Viterbi Decoder=1/2 k=4 [closed]

Woud anyone have a viterbi decoding tool or c-code or .m file for (R=1/2, K=4)? I am trying to decode a bitstream received in RFSTUDIO from a non-TI transceiver and it says that it uses this kind of ...
2
votes
1answer
144 views

Intuitive explanation of cross-correlation in frequency domain

According to the cross-correlation theorem : the cross-correlation between two signals is equal to the product of fourier transform of one signal multiplied by complex conjugate of fourier transform ...
-1
votes
1answer
52 views

Multi instance DSP C Source code

I need to design a multi instance DSP algorithm implementation. Any reference Open C source code available ? Any links on multi instance implementations. -ben
0
votes
0answers
14 views

A Suitable DSO Replacement

Currently I have a Pipeline leakage detection system the concept is that we are using a striker to strike the pipe at the generator side and receive the sound waves at the receiver side both separated ...
0
votes
1answer
67 views

Matlab FIlter coefficients to CMSIS api

I am trying to implement iir Band Pass filter design in STM32F429i discovery kit. I konw that it needs coefficients to be generated from matlab and feed them into cmsis api to achivi filtered data. ...
0
votes
0answers
49 views

MATLAB band-stop filter design for room reverberation?

I took acoustic measurements of a room using sine swept method, and I also used Pioneer sc-37 to obtain more accurate data. I used RoomEQ wizard software and measured the impulse response of the ...
1
vote
3answers
222 views

Get a N-FFT with two N/2-FFT already computed

After somme researchs on the web, I don't find the answer of my problem (or I don't understand it) and I hope this post will succeed. I'm working on a real-time FFT convolution algorithm (C++) which ...
0
votes
0answers
45 views

Developing Python code for BPSK Fiber Opticis Communication

I am a beginner both in Python and DSP but eager to take my first assignment tasks. I need to implement BPSK fiber optics communication simulation technique in Python. As per my research I found ...
0
votes
1answer
47 views

Output of a band pass filter

Signal x(t) = | cos(2πft)|, f != 0 is the input of an ideal band-pass filter with bandwidth B = 3f/2. How can i Find the ...
2
votes
1answer
92 views

Real-time block processing setup on ARM

Newbie question here. I'm working on a system that is running a 32khz sampling rate, audio, 16 bit DAC. Previously, our system had a very hard latency requirement of 1 sample. So no problem, we ...
0
votes
1answer
72 views

All pass systems- Poles and Zeros

If an All Pass system has poles at $0.8\exp\left(-j \dfrac{\pi}{4}\right)$ and $0.8\exp\left(j \dfrac{\pi}{4}\right)$, There would be $2\pi$ jumps in the phase (principal value) at angular frequency ...
0
votes
2answers
69 views

convolution of a real and complex signal

How do we define convolution of: $$x(t)= \mathrm{sinc}(W t)$$ and $$y(t)= -j\mathrm{sinc}(2W t)$$ Answer: In the frequency domain, both of them are rectangular functions and multiplication of them ...
0
votes
3answers
83 views

detect impact from digital audio

I've started a hobby project which is to detect and count the number of impacts (for example someone clapping, bouncing a ball or hitting a boxing bag). I've been reading and playing around (in java) ...
0
votes
0answers
30 views

choosing how many DWT Decomposition levels for 1D

I am building application decomposition of 1D signal sampling frequency $f_s=256$, number of samples $N=512$ - how many levels if the above signal passed through LPF of $70 Hz$ $f_s=256$, number ...
0
votes
0answers
13 views

DSP whinning noise when connected to audio source

I have a development board of a DSP. However as a beginner I dont get some things. Connecting a microphone and configuring ADC , DAC works somewhat fine instead of some white noise no matter if DAC ...