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Lecture 7

Real-time Digital Signal Processing

with the TMS320C6x

Dr. Konstantinos Tatas
Outline
Digital Signal Processing Basics
Aliasing
Discrete-time signals
Digital Filtering
FFT
Effects of finite fixed-point wordlength
Efficient DSP application implementation on
TMS320C6x processors
FIR filter implementation
IIR filter implementation
Digital Signal Processing Basics
A basic DSP system is composed of:
A Digital Processing system (P/ASIC/FPGA)
A DAC converting processed samples to analog output
Real-time signal processing: All processing operation
must be complete between two consecutive samples

Sampling
Quantization
Binary Coding
Sampling rate must be at least twice as
much as the highest frequency component
Aliasing
When sampling at a rate of fs samples/s, if k is any
positive or negative integer, its impossible to distinguish
between the sampled values of a sinewave of f0 Hz and
a sinewave of (f0+kfs) Hz.
Discrete-time signals
A continuous signal input is denoted x(t)
A discrete-time signal is denoted x(n),
where n = 0, 1, 2,
Therefore a discrete time signal is just a
collection of samples obtained at regular
intervals (sampling frequency)

Common Digital Sequences
Unit-step sequence:

Unit-impulse sequence:

<
>
=
0 , 0
0 , 1
) (
n
n
n u

=
=
=
0 , 0
0 , 1
) (
n
n
n o
1
0 1 2 3 4
. . .
n
u(n)
-1 -2
1
0 1 2 3 4
. . .
n
(n)
-1 -2
The z transform
Discrete equivalent of the Laplace
transform
z-transform properties
Linear

Shift theorem

Note:
)) ( ( ) ( ( )) ( ) ( ( n y bZ n x aZ n by n ax Z + = +
m
z n x Z m n x Z

= )) ( ( )) ( (
) ( )) ( ( z X n x Z =
Transfer Function
z-transform of the output/z transfer
of the input

Pole-zero form

) (
) (
) (
z X
z Y
z H =
Pole-zero plot

System Stability
Position of the poles affects system stability
The position of zeroes does not
Example 1
A system is described by the following
equation:
y(n)=0.5x(n) + 0.2x(n-1) + 0.1y(n-1)
Plot the systems transfer function on the z
plane
Is the system stable?
Plot the systems unit step response
Plot the systems unit impulse response

The Discrete Fourier Transform
(DFT)
Discrete equivalent of the continuous
Fourier Transform
A mathematical procedure used to
determine the harmonic, or frequency,
content of a discrete signal sequence

The Fast Fourier Transform (FFT)
FFT is not an approximation of the DFT, it
gives precisely the same result
Digital Filtering
In signal processing, the function of a filter is to remove
unwanted parts of the signal, such as random noise, or
to extract useful parts of the signal, such as the
components lying within a certain frequency range
Analog Filter:
Input: electrical voltage or current which is the direct analogue of
a physical quantity (sensor output)
Components: resistors, capacitors and op amps
Output: Filtered electrical voltage or current
Applications: noise reduction, video signal enhancement, graphic
equalisers
Digital Filter:
Components: Digital processor (PC/DSP/ASIC/FPGA)
Output: Filtered samples (requires DAC)
Applications: noise reduction, video signal enhancement, graphic
equalisers
Averaging Filter

Ideal Filter Frequency Response
Realistic vs. Ideal Filter Response
FIR filtering
Finite Impulse Response (FIR) filters use
past input samples only
Example:
y(n)=0.1x(n)+0.25x(n-1)+0.2x(n-2)
Z-transform: Y(z)=0.1X(z)+0.25X(z)z^(-
1)+0.2X(z)(z^-2)
Transfer function:
H(z)=Y(z)/X(z)=0.1+0.25z^(-1)+0.2(z^-2)
No poles, just zeroes. FIR is stable

FIR filter design

Inverse DFT of H(m)
FIR Filter Implementation
y(n)=h(0)x(n)+h(1)x(n-1)+h(2)x(n-2)+h(3)x(n-3)
Example 2
A filter is described by the following equation:
y(n)=0.5x(n) + 1x(n-1) + 0.5x(n-2), with initial
condition y(-1) = 0
What kind of filter is it?
Plot the filters transfer function on the z plane
Is the filter stable?
Plot the filters unit step response
Plot the filters unit impulse response

IIR Filtering
Infinite Impulse Response (IIR) filters use
past outputs together with past inputs
IIR Filter Implementation
y(n)=b(0)x(n)+b(1)x(n-1)+b(2)x(n-2)+b(3)x(n-3) +
a(0)y(n)+a(1)y(n-1)+a(2)y(n-2)+a(3)y(n-3)
FIR - IIR filter comparison
FIR
Simpler to design
Inherently stable
Can be designed to have linear phase
Require lower bit precision
IIR
Need less taps (memory, multiplications)
Can simulate analog filters

Example 3
A filter is described by the following equation:
y(n)=0.5x(n) + 0.2x(n-1) + 0.5y(n-1) + 0.2y(n-2),
with initial condition y(-1)=y(-2) = 0
What kind of filter is it?
Plot the filters transfer function on the z plane
Is the filter stable?
Plot the filters unit step response
Plot the filters unit impulse response

Software and Hardware
Implementation of FIR filters
Fixed-Point Binary Representation
Representation of a number with integer and
fractional part:

This is denoted as Qnm representation
The binary point is implied
It will affect the accuracy (dynamic range and
precision) of the number
Purely a programmers convention and has no
relationship with the hardware.

2 2 1 0 2 1
) ... . ... (
k n n
d d d d d d

=
=
1
2
n
m i
i
r d N
Examples
x = 0100 1000 0001 1000b
Q0.15 => x= 2^(-1) + 2^(-4) + 2^(-11)+2^(-12)
Q1.14 => x= 2^0 + 2^(3) + 2^(10) + 2^(11)
Q2.13 => x = 2^1 + 2^(2) + 2^(9) + 2^(10)
Q7.8 => x = ?
Q12.3 => x = ?

EFFECTS OF FINITE FIXED-
POINT BINARY WORD LENGTH
Quantization Errors
Coefficients
Truncation
Rounding
Data Overflow

ADC converts an analog signal x(t) into a digital signal
x(n), through sampling, quantization and encoding
Assuming x(n) is interpreted as the Q15 fractional
number such that 1 x(n) < 1
dynamic range of fractional numbers is 2. Since the
quantizer employs B bits, the number of quantization
levels available is 2B
The spacing between two successive quantization levels
is = 2/2^B = 2^(1-B)
Therefore the quantization error is |e(n)|/2

Coefficient Quantization Error
Effects on FIR filters
Location of zeroes changes
Therefore, frequency response changes
Effects on IIR filters
Location of poles and zeroes change
Could move poles outside of unit circle,

Roundoff error

Overflow error
signals and coefficients normalized in the range
of 1 to 1 for fixed-point arithmetic, the sum of
two B-bit numbers may fall outside the range of
1 to 1.
Severely distorts the signal
Overflow handling
Saturation arithmetic
Clips the signal, although better than overflow
Should only be used to guarantee no overflow, but should not
be the only solution
Scaling of signals and coefficients
Coefficient representation
Fractional 2s complement (Q) representation is
used
To avoid overflow, often scaling down by a
power of two factor (S) (right shift) is used.
The scaling factor is given by the equation:
S=Imax(|h(0)|+|h(1)|+|h(2)|+)
Furthermore, filter coefficient larger than 1,
cause overflow and are scaled down further by a
factor B, in order to be less than 1

Example 1
Given the FIR filter
y(n)=0.1x(n)+0.25x(n-1)+0.2x(n-2)
Assuming the input range occupies of the full range
Develop the DSP implementation equations in Q-15
format. What is the coefficient quantization error?
Solution:
S=1/4((|h(0)|+|h(1)|+|h(2)|) = (0.1+0.25+0.2)=3.25/4
Overflow cannot occur, no input (S) scaling required
No coefficents > 1, no coefficient (B) scaling required

Example 2
Given the FIR filter
y(n)=0.8x(n)+3x(n-1)+0.6x(n-2)
Assuming the input range occupies of the full range
Develop the DSP implementation equations in Q-15
format. What is the coefficient quantization error?
Solution:
S=1/4((|h(0)|+|h(1)|+|h(2)|) = (0.8+3+0.6)=4.4/4 =
1.1
Therefore: S=2
Largest coefficient: h(1) = 3, therefore B=4
ys(n)=0.2xs(n)+0.75xs(n-1)+0.15xs(n-2)

Simple FIR Hardware
implementation
VHDL 4-tap filter entity
ENTITY my_fir is
Port (clk, rst: in std_logic;
sample_in: in std_logic_vector(length-1
downto 0);
sample_out: out std_logic_vector(length-1
downto 0)
);
END ENTITY my_fir;
Example (continued)
VHDL 4-tap filter architecture
ARCHITECTURE rtl of my_fir is
type taps is array 0 to 3 of std_logic_vector(length-1 downto 0);
Signal h: taps_type;
Signal x_p1, x_p2, x_p3: std_logic_vector(length-1 downto 0); --past samples
Signal y: std_logic_vector(2*length-1 downto 0);
Begin
Process (clk, rst)
Begin
If rst=1 then
x_p1 <= (others => 0);
x_p2 <= (others => 0);
x_p3 <= (others => 0);
Elsif rising_edge(clk) then
x_p1 <= sample_in; --delay registers
x_p2 <= x_p1;
x_p3 <= x_p2;
End if;
End process;
y <= sample_in*h(0)+x_p1*h(1)+x_p2*h(2)+x_p3*h(3);
Sample_out <= y(2*length-1 downto length);
End;
FIR Software Implementation
int yn=0; //filter output initialization
short xdly[N+1]; //input delay samples array

interrupt void c_int11() //ISR
{
short i;
yn=0;
short h[N] = { //coefficients };
xdly=input_sample();
for (i=0; i<N; i++)
yn += (h[i]*xdly[i]);
for (i=N-1; i>0; i--)
xdly[i] = xdly[i-1];
output_sample(yn >> 15); //filter output
return; //return from ISR
}

IIR C implementation
int yn=0; //filter output initialization
short xdly[N+1]; //input delay samples array
Short ydly[M]; //output delay array

interrupt void c_int11() //ISR
{
short i;
yn=0;
short a[N] = { //coefficients };
short b[M] = { //coefficients };
xdly=input_sample();
for (i=0; i<N; i++)
yn += (a[i]*xdly[i]);
for (i=0; i<M; i++)
yn += (b[i]*ydly[i]);
for (i=N-1; i>0; i--)
xdly[i] = xdly[i-1];
ydly = yn >> 15;
for (i=M-1; i>0; i--)
ydly[i] = ydly[i-1];
output_sample(yn >> 15); //filter output
return; //return from ISR
}