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A Quadrature Signals Tutorial: Complex, But Not Complicated
posted by Rick Lyons
Comments (3) |

Polyphase Filters and Filterbanks
posted by Kyle Kastner
Comments (8) |

DSPRelated Finally on Twitter!
posted by Stephane Boucher
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Noise shaping
posted by Markus Nentwig
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Frequency-Domain Periodicity and the Discrete Fourier Transform
posted by Eric Jacobsen
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Python scipy.signal IIR Filter Design Cont.
posted by Christopher Felton
Comments (7) |

Engineering the Statistics
posted by Sami Al Dalahmah
Comments (2) |

Implementing a full-duplex UART using the TMS320VC33 serial port
posted by Manuel Herrera
Comments (2) |

How to build a test environment for audio DSP code in C using GNU/Linux
posted by Gabriel Rivas
Comments (2) |

Discrete Wavelet Transform Filter Bank Implementation (part 2)
posted by David Valencia
Comments (7) |

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Most Popular DSP Code Snippets

Discrete Haar Wavelet Transform
by Duraisamy Sundararajan |
See article at http://www.dsprelated.com/showabstract/3639.php for the theory....

Saving a Matlab figure to file using command-line
by |
Once again, this is another example that Matlab does weird things with its figure window. If y...

Computing FFT Twiddle Factors
by Rick Lyons |
This text, figures, and code snippet are also available in downloadable PDF format Typical applic...

Echo Filter
by Miguel De Jesus |
An Audio Echo Filter is an FIR filter that repeats a sound after a given Delay attenuating the repet...

Flanger Audio Effect
by Gabriel Rivas |
This is an implementation of the flanger effect using my previous entry that is a fractional delay...

Generate ideal QPSK..64QAM symbol error rates (reference result)
by Markus Nentwig |
Calculates the symbol error rate of QPSK/16QAM/64QAM/256QAM/1024QAM modulation for additive white Ga...

Zero Crossing Counter
by |
In some DSP applications, it can be very helpful to know how many times your signal has crossed the ...

GSM (GMSK) power spectrum equation
by Markus Nentwig |
A simple, empirical model for the power spectrum shape of a GMSK-modulated GSM signal, generated acc...

Basic Kalman filter
by Sylvain B |
This is the most basic implementation of a Kalman filter. For measured position, velocity and accel...

Equal Loudness Curves (ISO226)
by |
What is loudness? No, it is not the setting of your volume knob. In audio processing, lo...

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DSP Discussion Groups

The Delay of Minimum Phase FIR filter
by wyonghao in Comp.DSP on May 18 2013
Dear all, I would like to ask is there any way to have analytical estimation value of delay of mini...

A purely theoretical question
by Michael P. in Comp.DSP on May 18 2013
Hi group I have a purely theoretical question. When doing a FFT of an unwindowed pure sine sig...

Convolving input samples with a filter
by Michael P. in Comp.DSP on May 17 2013
Hi group I am coding a program to do some filtering on an audio file. So using the method descri...

Adding AWGN to signal
by bradsdr in Comp.DSP on May 17 2013
Hi all, I always get confused while adding AWGN in my simualtions via AWGN block. What is the r...

File System Library Problem
by B S in TMS320C6x on May 16 2013
Hi, I have been trying to use Fat16/32 File System Library by third party given at the following ...

Recursive Least Squares
by in Comp.DSP on May 16 2013
We all know that Guass probably invented block least-squares, but who invented Recursive Least Squar...

Help on FIR decimation
by joaoneves in Comp.DSP on May 16 2013
Hi all I have a FIR decimation code, which is meant to decimate audio from 44100 Hz to 11025. My p...

fft in verilog
by fren...@yahoo.com in DSP & FPGA on May 15 2013
Hello I have some questions related to verilog. How can I create a N by N matrix in verilog? In M...

regarding ADC
by anja...@gmail.com in TMS320C28x on May 15 2013
Hi,I am working on F2812 processor for BLDC motor control.As part of thios ,I need to use the ADC in...

PC hard disk error correction question
by Rick Lyons in Comp.DSP on May 14 2013
Hi Guys, Can you give me your opinion on what would be the effect on personal computers (PC...

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