Sign in

username:

password:



Not a member?

Search Online Books



Search tips

Free Online Books

Ads

Chapters

See Also

Embedded SystemsFPGAElectronics
Chapter Contents:

Search Physical Audio Signal Processing

  

Book Index | Global Index


Would you like to be notified by email when Julius Orion Smith III publishes a new entry into his blog?

  

Feedforward Comb Filters

The feedforward comb filter is normally implemented as shown in Fig.1.17, in which the direct signal ``feeds forward'' around the delay line and sums (scaled) with the delay-line output.

Figure 1.17: The feedforward comb filter.
\begin{figure}\input fig/ffcf.pstex_t
\end{figure}

The ``difference equation'' for the feedforward comb filter is

$\displaystyle y(n) = b_0 x(n) + b_M x(n-M). \protect$ (2.2)

We see that the feedforward comb filter is a particular type of FIR filter. It is also a type of TDL in which the output is formed as a linear combination of the delay line's input and output.

Note that the feedforward comb filter can implement the echo simulator of Fig.1.8 by setting $ b_0=1$ and $ b_M=g$. Thus, the feedforward comb filter is a computational physical model of a single discrete echo. This is one of the simplest examples of acoustic modeling using signal processing elements. The feedforward comb filter models the superposition of a ``direct signal'' $ b_0 x(n)$ plus an attenuated, delayed signal $ b_M x(n-M)$, where the attenuation is due to ``air absorption'' and/or spherical spreading losses, and the delay can be ascribed to acoustic propagation over the distance $ cMT$ meters, where $ T$ is the sampling period in seconds, and $ c$ is sound speed in meters per second. In cases where the simulated propagation delay needs to be more accurate than the nearest integer number of samples $ M$, some kind of delay-line interpolation needs to be used (which we address in §3.2). Similarly, when air absorption needs to be simulated more accurately, the constant attenuation factor $ b_M$ can be replaced by a linear, time-invariant filter $ G(z)$ giving a different attenuation at every frequency. Due to the physics of air absorption, $ G(z)$ is generally lowpass in character [358, p. 560], [47,326].


Order a Hardcopy of Physical Audio Signal Processing

Previous: Comb Filters
Next: Feedback Comb Filters

written by Julius Orion Smith III
Julius Smith's background is in electrical engineering (BS Rice 1975, PhD Stanford 1983). He is presently Professor of Music and Associate Professor (by courtesy) of Electrical Engineering at Stanford's Center for Computer Research in Music and Acoustics (CCRMA), teaching courses and pursuing research related to signal processing applied to music and audio systems. See http://ccrma.stanford.edu/~jos/ for details.


Comments


No comments yet for this page


Add a Comment
You need to login before you can post a comment (best way to prevent spam). ( Not a member? )