C is a high level (i.e. not assembler) programming language often used to program DSPs and other embedded microprocessors as well as being used for systems programming. Features of the language make it especially suitable for programming hardware and writing device drivers.

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21 views

Filter signal1 from evolving frequency response of signal2

I'm a composer (of music) starting out with signal processing. I'm also reasonably proficient in C at this point. I'm not even sure if what I'm trying to do is possible, because I can't yet find an ...
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1answer
28 views

How would I go about designing a difference equation for a given type of filter?

I have experience using design tools such as Xilinx's CoreGen and MatLab's DSP tool kit, but I have not really had much experience designing filters from the ground up. Reinventing the filter as it ...
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1answer
38 views

Help me understanding the super equalizer source code

Please forgive my bad English. Hello, I'm trying to understand Shibatch's super equalizer Winamp plugin source, but I'm not good at DSP and mathematics. It seems that super equalizer plugin uses ...
2
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2answers
115 views

Perform Convolution in Frequency Domain Using FFTW?

I'm trying to convolve two signals $x(n)$ and $h(n)$ in C by using the FFTW library's functions to perform a Fourier transform on each, multiply the appropriate complex components together, and take ...
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2answers
41 views

Playing back an audio file at an arbitrary rate with upsampling and downsampling

Background: My overall goal is to create an audio file player which can playback an audio file at an arbitrary rate. The rate can change continuously as the file plays. This means upsampling to ...
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1answer
21 views

How to manipulate attenuation in Parks MClellan algorithm?

I am using GNU Radio C++ source code in my app (pm_remez()) for generating FIR filters coefficients. Input parameters here are: ...
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1answer
34 views

Fast Fourier Transformation using parallel processing coding for image processing [closed]

I'm now working in parallel precessing -and i want a full code for fast Fourier transform(FFT) written parallel processing Code for any image processing application
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1answer
45 views

Kiss fft does not work after giving it more than 32 samples

I am trying to take data from an accelerometer and apply Kiss FFT to the samples. I'm using a Freescale Kinetis FRDM-K22F board. I want to use 64 samples, but when I run the program I get an error ...
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3answers
157 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 ...
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1answer
46 views

Beginner - use FFT to display spectrum/equalizer

At first, I want to say that I'm totally beginner in DSP and I don't have enough math education to understand Fast-Fourier transformation. I'm a programmer and I want to archieve something like ...
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3answers
99 views

How to implement a moving average in C without a buffer?

Is it possible to implement a moving average in C without the need for a window of samples? I've found that I can optimize a bit, by choosing a window size that's a power of two to allow for ...
3
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1answer
158 views

Denoise images with wavelets

I try to understand what following algorithm from this gimp plugin does to denoise an image: ...
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0answers
62 views

Well-regarded C/C++ General-purpose DSP libraries

I am looking to transition some of my MATLAB algorithms to C/C++ for deployment on target (PowerPC and/or ARM). Mostly 1D FIR/IIR filter in this instance. In other domains, there are well-regarded ...
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1answer
58 views

Confusion when applying Hamming window to signal

I am writing some code which takes a signal of length $n$, and processes it in 20 ms chunks with 5 ms of overlap... So basically I have a buffer of 20 ms worth of sampled data, and I am trying to ...
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0answers
43 views

Direct Form I Filter from Direct Form II

I am looking a way to make outputs match. For len 160 wouldnt the error be significant? For 40 ? I have included some sample output showing the error. Direct Form 2 Original ...
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0answers
21 views

Testing whether instantaneous phase algorithm works 2D

I need to calculate the instantaneous phase $\phi$ of a 2D wave field $U$ in C which can be derived by one of two 1D Hilbert transforms $H$: $$ q(x,z) = H_1[U(x,z)] = 1/\pi \int_R U(x-\zeta,z) ...
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2answers
599 views

C/C++ library for digital modulation [closed]

Is there a library written in C/C++ that offers functions like QPSK modulation?
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1answer
251 views

Calculating the Power spectral density

I've already asked this question on SO, but didn't get an answer, I tqake a shot here. I am trying to get the PSD of a real data set by making use of fftw3 ...
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5answers
643 views

What approximation techniques exist for computing the square root?

I have very limited resources as I'm working with a microcontroller. Is there a taylor-series expansion, common lookup table, or recursive approach? I'd prefer to do something without using math.h's ...
2
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0answers
101 views

What is dynamic pulse-width modulation?

I'm working through exercises in a book called The Audio Programming Book and have been doing okay for nearly 300 pages, but have just got stuck. (I'm a musician, not programmer or audio specialist ...
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2answers
207 views

Confusion on using FFTW 3.3.4 (C version) for real and even data sets

I am using FFTW3 to transform the autocorrelation function of a particular function. I expect the transform to be real since the autocorrelation function is symmetric with respect to negative and ...
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2answers
308 views

Recommendations for DSP Hardware

I'm trying to figure out what a good DSP unit would be to try and learn how to do some embedded programming on? I would like something relatively inexpensive but that still has a decent amount of ...
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1answer
133 views

understanding complex fft results

i use this for complex fft. Output expected $fft[3].real= 32$ (peak at 3rd bin) $fft[61].real= 32$ ((peak at negative frequency pair of 3rd bin)) All other values negligibly small The input is ...
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2answers
253 views

problems calculating phase propagation (in C)

I'm coding a phase vocoder in C for time-stretching audio signals. The only piece that is currently not working is the phase propagation. I take overlapping frames from an input signal, apply a ...
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1answer
604 views

Equal power crossfade

I am working on an audio system where I need to implement an equal power crossfade between two signals over a specified number of samples. Initially I implemented a linear fade but this creates a dip ...
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2answers
291 views

Manipulating with frequencies through fftw

I'm using portaudio for I/O audio streaming, and I wanted to apply some equalization on audio input from microphone. I found that most people refer to fft for task like that, so I use FFTW library in ...
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1answer
1k views

Trouble Generating a Sine Wave in C/Matlab from Basics

this is a crosspost from stackoverflow as I figured there may be more who can help me here... I've been writing a basic C program to generate a sine wave and write it out to STDOUT for piping into a ...
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1answer
321 views

Implementation of FFT algorithm in C Programming [closed]

Does anyone know any FFT algorithm that can be implemented in C? I am trying to convert a built-in matlab FFT api to a function in C. Appreciate any help!
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1answer
86 views

Mat class of c++ in open cv [closed]

I am a beginner in opencv and have started learning it using visual studio 2012 and opencv 2.4.7 . But i got stuck at very beginning .I want to know about Mat used in Opencv .It is a class and an ...
0
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1answer
129 views

In voice recognition system, how to match digital input to a given word without seconds delay?

I've been looking around and found some great C/C++/Java libraries like CMU Sphinx or Windows SAPI. I did not like SAPI, and CMU Sphinx is way too complicated... would need months to understand how it ...
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2answers
3k views

Algorithm for implementing an IIR digital filter, Chebyshev type I low pass

I am trying to implement a Chebyshev type I low-pass IIR digital filter in C. I have got the SOS Matrix and scale values from Matlab. What is the direct equation or algorithm to implement such a ...
3
votes
1answer
878 views

Digital Distortion effect algorithm

I read the DAFX book by Udo Zoler about the distortion effect at pages 124-125 and it says that suitable simulation of the distortion is given by the function: ...
1
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1answer
255 views

Matlab SOS Filter in C++

I'm struggling to validate that my C++ implementation of a filter I designed in Matlab is correct. My first issue is to make sure I'm implementing the sections in the right order. This is the output ...
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1answer
52 views

Can I compute my IIR faster given limited possible input values?

I have an IIR routine in C++ I'm trying to speed up. Currently I'm using a DF2 implementation. Since my input signal will always be either 1.0 or -1.0 is it possible for me to optimize the algorithm, ...
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2answers
622 views

Discrete Fourier Transform for beginners

I want to make image processing programs which talk about homomorphic filter. But when I read some articles, I can't get it. Maybe can anyone help it. I very need this. Sorry about my english. I ...
2
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2answers
2k views

Bandpass Filter Implementation in C

I've been googling for a while and came across the audio cookbook and Bjorn's blog explaining it further Link Here. Background info: I've modified the rtl-sdr open source code to tune to a VOR ...
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4answers
8k views

Simplest way to implement FIR filter in C/C++

What's the simplest way to implement a FIR filter in C/C++ ? (I don't want to have to use a third-party software like Matlab or Octave in order to compute the coefficiens because I have to use ...
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1answer
70 views

How to use clustering file after training

I trained the data using BOG with SVM , i trained it , now i just want to open it through xml , i don't want to train it again and again on every run , I trained ...
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2answers
2k views

Algorithm to zero pad data before FFT

I've seen some other questions on here about zero padding (like this one) but I'm still a little confused about my situation. I'm attempting to zero pad my input time series data, so I get an ...
1
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1answer
137 views

Comparing two sound files in embedded-systems (C language)

what would be the most efficient ("cheap" in terms of system resources etc.) way to compare 2 sound files? Here is the scenario. Sound A comes as an input. There is sound B stored in memory. We need ...
2
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1answer
83 views

Embedded Correlation Function

I'm trying to write a correlation function on an embedded device. Because its embedded, I need to make sure that the memory/CPU usage is minimum. Here's the deal: - I have a saved waveform sampled ...
1
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1answer
905 views

What is the most efficient type of software bandpass filter?

I'm writing a C++ application to extract four modulated carrier signals at four separate frequencies, from a single input signal. I've designed four FIR filters at t-filter.appspot.com but they run ...
2
votes
3answers
8k views

How does a low-pass filter programmatically work?

I have been working on a simple low pass filter for < 100 Hz metering in my application. But so far, I am struggling with the theory behind it all. It's cool that I got it working, but I'd really ...
0
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0answers
192 views

How to implement Hilbert-Huang transform in finding image splicing?

I have some programming experiences but am really new to these kind of algorithms. I can find a few source code about implementing HHT, and most of them are written with matlab. What I am trying to ...
0
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1answer
316 views

DSP library compatible with QT? [closed]

I have checked this and this Question in this site very much similar to my question. But I want to know are they Qt compatible? I want to start learn developing DSP application in Qt framework so I ...
0
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2answers
169 views

FFT Data Array Indexing [closed]

I've been Googling "FIR filters" and "FFT convolution" etc. and I ran across something interesting. Array Index vs. Pointers: http://www.bores.com/courses/intro/program/7_array.htm ...
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1answer
181 views

Dynamic time warping in percentage

I'd like to compare 2 speech similarity and show it in terms of 0-100%. Here's my approach: 1: Convert 2 audio waveform into audio spectrum by Fast Fourier Transform (FFT) 2: Use Dynamic Time Warping ...
9
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0answers
500 views

Real-valued ringing when zero-padding odd-length FFT

So I'm trying to write a frequency-domain interpolator that zero-pads the frequency response of a signal and inverse transforms. There's two cases I have to deal with: Even-length response - have ...
8
votes
2answers
2k views

Add odd/even harmonics to signal?

How do I add odd or even harmonics to a floating point signal? Do I have to use tanh or sin? What I'm trying to do is achieve some very simple distortion effects, but I'm having a hard time finding ...
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1answer
3k views

Using FFT to convert to frequency domain, then IFFT back to time domain C++

I've posted the same question on stackoverflow.com with little success, so I thought I would try here! I'm using C++/C to perform forwards and reverse FFT on some data which is supposed to be the ...