Phase or Frequency Shifter Using a Hilbert Transformer

Neil Robertson

In this article, we'll describe how to use a Hilbert transformer to make a phase shifter or frequency shifter. In either case, the input is a real signal and the output is a real signal. We'll use some simple Matlab code to simulate these systems. After that, we'll go into a little more detail on Hilbert transformer theory and design.


An Efficient Linear Interpolation Scheme

Rick Lyons
4 comments

This article presents a computationally-efficient linear interpolation trick that requires at most one multiply per output sample.


Simplest Calculation of Half-band Filter Coefficients

Neil Robertson

Half-band filters are lowpass FIR filters with cut-off frequency of one-quarter of sampling frequency fs and odd symmetry about fs/4 [1]*. And it so happens that almost half of the coefficients are zero. The passband and stopband bandwiths are equal, making these filters useful for decimation-by-2 and interpolation-by-2. Since the zero coefficients make them computationally efficient, these filters are ubiquitous in DSP systems. Here we will compute half-band coefficients using the window method. While the window method typically does not yield the fewest taps for a given performance, it is useful for learning about half-band filters. Efficient equiripple half-band filters can be designed using the Matlab function firhalfband [2].


How to Find a Fast Floating-Point atan2 Approximation

Nic Taylor

Context Over a short period of time, I came across nearly identical approximations of the two parameter arctangent function, atan2, developed by different companies, in different countries, and even in different decades. Fascinated...


A Beginner's Guide to OFDM

Qasim Chaudhari

In the recent past, high data rate wireless communications is often considered synonymous to an Orthogonal Frequency Division Multiplexing (OFDM) system. OFDM is a special case of multi-carrier communication as opposed to a conventional...


Sinusoidal Frequency Estimation Based on Time-Domain Samples

Rick Lyons
6 comments

The topic of estimating a noise-free real or complex sinusoid's frequency, based on fast Fourier transform (FFT) samples, has been presented in recent blogs here on dsprelated.com. For completeness, it's worth knowing that simple frequency estimation algorithms exist that do not require FFTs to be performed . Below I present three frequency estimation algorithms that use time-domain samples, and illustrate a very important principle regarding so called "exact" mathematically-derived DSP algorithms.


Launch of Youtube Channel: My First Videos - Embedded World 2017

Stephane Boucher

I went to Embedded World 2017 in Nuremberg with an ambitious plan; I would make video highlights of several exhibits (booths) to be presented to the *Related sites audience.  I would try to make the vendors focus their pitch on the essential...


A Two Bin Exact Frequency Formula for a Pure Complex Tone in a DFT

Cedron Dawg
1 comment

Introduction This is an article to hopefully give a better understanding to the Discrete Fourier Transform (DFT) by deriving an exact formula for the frequency of a complex tone in a DFT. It is basically a parallel treatment to the real case...


DFT Bin Value Formulas for Pure Complex Tones

Cedron Dawg

Introduction This is an article to hopefully give a better understanding to the Discrete Fourier Transform (DFT) by deriving an analytical formula for the DFT of pure complex tones and an alternative variation. It is basically a parallel...


Multi-Decimation Stage Filtering: Design and Optimization

AHMED SHAHEIN
1 comment

During my research on digital FIR decimation filters I have been developing various Matlab scripts and functions. In which I have decided later on to consolidate it in a form of a toolbox. I have developed this toolbox to assist and...