This
** preview**
has intentionally

**sections.**

*blurred***to view the full version.**

*Sign up**This preview shows
pages
1–3. Sign up
to
view the full content.*

Lowpass filter
↑
L
↓
M
C/D
↑
4
Stage 2
Stage 1
↑
2
↑
2
Sampling Rate Conversion
Overview
In discrete-time signal processing, it is often necessary to convert a signal from one
sampling rate to another. A common example is the conversion from the sampling rate of a
compact disk (CD); signal (44.1 KHz) to that of a Digital Audio Tape (DAT) signal (48 KHz).
Another example is the aud io standard in High-Definition Television (HDTV)
transmission, where at least three sampling rates are supported (32, 44.1, and 48 KHz).
Although in principle we may convert the signal back to analog form and resample at the
desired rate, it is usually preferable to perform t he entire conversion digitally. This is clue
to many considerations including the fact that conversion to analog form often introduces
noise in the signal, and that digital signal processing can be much more cost-effective and
flexible.
This MATLAB
project asks you to perform a sampling rate conversion on segments of audio
signals. The input audio signals are quantized to 8 bits and sampled with a sampling
frequency of 11,025 Hz. You are required to convert the signal to a sampling rate of 24,000
Hz in a computationally efficient manner. Although one conceptual way of realizing this
sampling rate conversion process is to upsample the signal, lowpass filter, and downsample
it, a more clever implementation can lead to an implementation that is many times more
efficient. To do this, you can exploit various aspects of class to optimize the system, such as
multistage filter implementation, filter design, and polyphase implementation. By the end
of the project, you hopefully will have a much better understanding of both the theoretical
aspect of the system as well as various issues in implementing a practical DSP system at a
software level.
Project Goal
A sampling rate converter which produces an output signal with a sampling rate which is
times the original sampling rate can be specified as shown in Figure 1.
f1997

This
** preview**
has intentionally

Figure 1: Sampling rate conversion system.
For an ideal sampling rate converter, the lowpass filter in Figure 1 is an ideal lowpass filter
with cutoff frequency The goal of this project is to design an efficient DSP algorithm that
implements the system in Figure 1 subject to the following constraints:
•
The system performs the correct sampling rate conversion from 11,025
Hz to 24,000 Hz. In particular, do not assume that 11,025 Hz

This is the end of the preview. Sign up
to
access the rest of the document.

Ask a homework question
- tutors are online