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-1
votes
0answers
15 views

A filter in ios to eliminate certain frequency [duplicate]

Hi I am making an application where i record sound in ios using the inbuilt mic. I want to process the sound in such a way that only frequency from 16-18 KHZ remain the other frequency should be ...
0
votes
0answers
23 views

Creating block diagrams

What diagramming tools exist that are particularly intended for drawing signal processing block diagrams? I'm looking for specific answers (names of signal processing diagramming software packages), ...
-1
votes
0answers
32 views

Suggestions based on using IHS transformation on images and optimizing using Least square adjustments [closed]

I have to normalize the image using Intensity Hue Saturation transformation and further use Weighted Least Square adjustments to reduce root mean square error. I want help regarding 1) The above ...
0
votes
0answers
60 views

Can we break the Shannon capacity? [duplicate]

I have a friend working in wireless communications research. He told me that we can transmit more than one symbol in a given slot using one frequency (of course we can decode them at the receiver). ...
2
votes
1answer
50 views

performance metrics for analysing the performance of various filters

we are working with speckle noise reduction in ultrasound images by using various despeckling filtering techniques. We compare the performance of each filtering technique by using Quality metrics such ...
0
votes
0answers
31 views

Altera DSP builder Blockset [closed]

Dear friends who used Altera DSP builder Blockset before?? I installed properly but when I am designing a system using its blocks the Matlab program shows a report said that the blocks are not exist ...
0
votes
0answers
30 views

transfer function

Please help me in solving the following problem: Find the pulse transfer function and weighting sequence of the system described by the following difference equation : y(k+2)+ y(k+1)+.89y(k)=r(k), ...
1
vote
0answers
43 views

Is this a universal relation between signal power and period (frequency)?

I am Fourier analyzing at a bunch of signals of equal length and find that $$\frac{d\ln({E[P]})}{d\ln({PT'})}$$ is constant across the dataset, where $T$ is a vector of the periods (inverse of ...
-3
votes
0answers
95 views

Digital signal processing [closed]

What is the basic(physical) reason for which digital signal processing is faster than analog signal processing? I got to know that in digital processing of analog signals an analog signals are ...
1
vote
1answer
388 views

Jacobian of affine warp in Lucas-Kanade image alignment

I hate this kind of questions, but I'm really stuck. I'm trying to implement Lucas-Kanade algorithm as described in this paper (see pages 4 and 5). Unlike most explanations I've seen they don't assume ...
1
vote
2answers
58 views

Goerzel algorithm -increasing accuracy

I have an audio buffer with 1024 samples per second . The sample rate is 44100 . I am trying to id the frequency tones . Using this Gorzel algorithm, i can get the frequency but its not that accurate ...
4
votes
2answers
281 views

Separating waves of very close wavelengths

I'm new to wave problems, so excuse my simplified words. I have a signal consists of high range of frequencies . I applied a Band-pass filter to this signal and I have been able to separate it to ...
1
vote
1answer
275 views

How to calculate auto-correlation of a bpsk modulated signal or how to calculate expectation value of complex exponential function

How to calculate auto-correlation of a bpsk modulated signal or how to calculate expectation value of complex exponential function manually not by using matlab or any other software? for example, if ...
2
votes
1answer
78 views

How to distinguish between the different frequency domains?

Sometimes the terms 'Fourier domain', 'complex frequency domain', 'Frequency domain' and 's domain' are used interchangeably. Take those answers here for example: ...
3
votes
4answers
111 views

Verb for subtracting the mean?

Is there a verb for removing the mean or DC component of a signal? I'm trying to name a flag for doing just that. "Normalise" can mean many things, so I don't think that would be a good choice.
0
votes
1answer
32 views

How can I graph a simulation of a FM 16 bits codification?

I have to develop a software where the user inputs a 16 bits binary sequence, which is supposed to be the codification of a FM (or even AM) signal, then I should graph it according to Frequence / Time ...
3
votes
1answer
120 views

Locking on to a square wave signal with minimum oversampling

I'm designing a device that will have an IR photodiode connected to a low power microcontroller's ADC pin. At times, another device will be transmitting a 48KHz square wave, and I'd like to be able to ...
3
votes
2answers
65 views

Linear model of the modified error function

We can write the error function ($E(w)=1/2\sum_{n=1}^{N}\{y(x_n,w)-t_n\}^2$) as a linear model using its partial derivatives. Is it possible to do the same thing about the modified error function? ...
4
votes
2answers
1k views

How to find poles of transfer function by looking at the step response?

How to find poles of transfer function by looking at the step response? Given a step response graph like such: How would I find the sketch for its poles on the complex plane? The only thing I can ...
2
votes
1answer
68 views

Clarifying some notation in Bishop book

In Bishop book, page 4, section 1.1, there's a notation I don't seem to understand what's meant by it. The whole paragraph, with which the section begins, is: We begin by introducing a simple ...
0
votes
1answer
176 views

How do I implement IFFT to filter low frequencies from a group,after Freq detection by FFT?

I use a combination of Visual Basic Programming interfaced with FAMOS which is a DSP software like MATLAB but much easier.I am trying to separate signals from a group,which requires filtering each ...
0
votes
1answer
172 views

Is possible to get overshoot on bessel filter?

I tried to use bessel filter from http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html, but in case large dataset i have overshoot as i know bessel filter shouldnt have overshoot, so is code ...
2
votes
2answers
83 views

Resampling necessary?

I have a file that is composed of n data records with a sample rate s. Each of these records has a duration d. Now due to postprocessing I have to change the duration to d=d/3 thus afterwards I'll ...
4
votes
1answer
142 views

basic bandpass filter

I am attempting to implement a basic bandpass filter but the center frequency of my filter seems to be irrelevant as I can change it all I want but it has no effect on the filter. where am I going ...
2
votes
4answers
2k views

Optimized Ansi C libraries for DSP

I am new to DSP? Where is a good resource for open source DSP algorithms? MATLAB is great at making protos but once we move to C coding, it takes time and we end up making too many mistakes. I would ...
3
votes
1answer
799 views

How to avoid denormalized numbers?

The same floating-point AMD X86-64 digital signal processing system mentioned in my previous question has a problem where it sometimes slows down substantially when signals attain values very near ...
6
votes
2answers
220 views

What are less computationally demanding alternatives to the Viterbi Decoder?

What are less computationally demanding alternatives to the Viterbi Decoder? Ideally what I would like is a list of the most commonly used approximate methods, along with brief pros and cons.
15
votes
1answer
422 views

What approximation techniques exist for the square super-root function?

I need to implement an approximation to the inverse of $x^x$, i.e. the square super-root (ssrt) function. For example, $\mathrm{ssrt}(2) \approx 1.56$ means that $1.56^{1.56} \approx 2$. I'm not as ...