Content

1. Introduction
2. Characteristics of signals
3. Analog signals in frequency domain
4. Analog LTI systems
5. Sampling theorem and reconstruction
6. Discrete time signals in frequency domain
7. Discrete time LTI systems
8. Digital filters
9. Correlation
10. Advanced topics

173
6 Discrete time signals within the frequency domain
6.1 Discrete Fourier transform (DFT)
§ Introduction
§ Characteristics
6.2 The DFT as approximation
§ the discrete time Fourier transform
§ the Fourier transform
§ the Fourier sequence
6.3 Fast Fourier transform (FFT)
6.4 Windowing and spectral analysis
6.5 Spectral analysis of stationary and non-stationary signals
§ Motivation
§ Spectrogram

174
6.1 Discrete Fourier Transform (DFT) - Introduction
Motivation
§ Numerical calculation of the spectrum, which means of the frequency content of a (analog or discrete time) signal,
e.g.,
- Calculation of the spectrum of a communication signal
- Vibration analysis of mechanical objects
- Definition of the harmonic overtone at the output of a non-linear amplifier
- Search of sine-signals within the noise

§ Calculation of the convolution or of the correlation of two signals

- Is particularly then applied, if very long sequences have to be convolved (correlated) in real-time.

§ The DFT can be very efficiently calculated by means of the FFT (Fast Fourier Transform).

175
6.1 Discrete Fourier Transform (DFT) - Introduction
§ Fourier transform (general spectrum)
+ +

>(:) = T <(!)e*$%&() d! <(!) = T >(:)e$%&() d:

*+ *+

§ Fourier series (line spectrum of a periodic signal using the Fourier coefficients)

,4/% +
1
q: = T <6 (!)e*$%&:(4) d! <6 (!) = V q: e$%&:(4)
)# :-*+
*,4/%

§ Discrete time Fourier transform (periodic spectrum of the discrete time signal)
+ ($ /%
1
>! (:) = V <[']e*$%&",$( <['] = T >4 (:)e$%&",$( d:
"-*+
:4
*($ /%

176
6.1 Discrete Fourier Transform (DFT) - Introduction

Derivation of the discrete Fourier transform

The spectrum >['] of the periodic sequence <['] is given by the discrete Fourier transform:
qrs
' ,
Fourier series: !(#) = ∑.
41&. K" e t with the time period ) = e · )4 and sampling points e per period
.67 .67 .67
9 3@, *$ 5 ) 9 3@, *$ ) 9 3@, + *$ )
q: = ∫ <(!) e d! = ∫ <(!) e 8⋅5$
d! ≅ ∫ ∑"-*+ < ' Q ! − ' ⋅ )4 e 8⋅5$
d!
, 3 7⋅,$ 3 7⋅,$ 3
.67:5
$ .6
9 *$ 9 7*9
= ∑ 7*9
<['] e 8⋅5$ ⋅ )4 = ∑"-# <['] e*$ 8 " :
7⋅,$ "-# 7

Discrete Fourier transform (DFT) Inverse DFT

9
> l = eq: = ∑7*9 "-# <[']e
*$%& : "/7
<['] = 7 ∑7*9
:-# >[l]e
$%& : "/7

§ <['] and >[l] form a transform couple.

§ If <['] is given, the frequency domain >[l] can be calculated using the DFT.
§ If >[l] is given, <['] can be calculated using the IDFT.

177
6.1 Discrete Fourier Transform (DFT) - Introduction

Annotations
§ The values >[l], which are in general complex, are often called DFT-coefficients.
The parameter ' in <['] is the discrete time variable or the time index or the sample index.
Analog to this, the parameter l in >[l] is called discrete frequency variable.
§ The sequence of the DFT-coefficients >[l] is periodic with period e.
§ Usually, >[l] is calculated only for one period, which is l = 0,1, … e − 1.
§ Usually, however not exclusively, the DFT is applied to time signals with finite length.
§ After the inverse transform, the sequence can be interpreted as a periodic signal with the period e.
The basic period for ' = 0,1, … e − 1 matches (i.e., is equal) with the original signal <['].
§ The step size (sampling distance) in time domain is the inverse of the period in frequency domain and vice versa.
! time (continuous)
< < < < "* = 1/& distance of sampling points in time domain
!; = = = >⋅= ; %; = ? = >⋅? " = ' ( "* signal length in time domain
o o &* = 1/" distance of sampling points in frequency domain
& = ' ( &* signal length in frequency domain
= sampling frequency )*
" #
' =" =# ="(& number of samples (sampling points)
' '
178
6.1 Discrete Fourier Transform (DFT) - Introduction

Matrix notation -./

0123+
; D = E 6 * e. -
§ With the introduction of the rotation phasor (=complex rotation vector) +,%
-./
1 0123+
]7 = e *$%&/7
, 6* = E; D e -
'
3,%

we can write the DFT, respectively IDFT as follows:

7*9

> l = V < ' ⋅ ]7 :⋅" l = 0,1, . . . , e − 1

"-#

7*9
1
<' = V > l ⋅ ]7 *:⋅" ' = 0,1, … , e − 1
e
:-#

§ As already mentioned, DFT, respectively IDFT are mostly evaluated only for one period:
l = 0,1, … , e − 1, respectively ' = 0,1, … , e − 1

179
6.1 Discrete Fourier Transform (DFT) - Introduction

Matrix notation -./

0123+
; D = E 6 * e. -
§ Using the vectors +,%
<[0] >[0] 1
-./
0123+
<[1] >[1] 6* = E; D e -
©7 = ; ´7 = '
⋮ ⋮ 3,%

<[e − 1] >[e − 1]

and the so-called DFT-matrix

1 1 ⋯ 1
1 ]7 9 ⋯ ]7 7*9
¨7 = ⋮ ⋮ ⋱ ⋮
1 ]7 7*9 ⋯ ]7 7*9 (7*9)

the N-points–DFT can be written using a Matrix multiplication:

s E = tE u E

180
6.1 Discrete Fourier Transform (DFT) - Introduction

Matrix notation -./

0123+
; D = E 6 * e. -
§ The IDFT, thus, simply results in +,%
&$
u E = tE sE 1
-./
0123+
6* = E; D e -
'
3,%
*9
§ Thereby, ¨7 is the inverse of the DFT-matrix.

§ One can show that following holds:

&$ 1
tE = tE ∗
r
§ * thereby means that all elements of the matrix have to be taken as conjugated complex.

181
6.1 Discrete Fourier Transform (DFT) - Introduction

Rotation phasor
vE = e&'!(/E

§ Important relation: ]7 "⋅7 = e*$%& " = 1

i.e., ]7 is periodic in ' and l with period e.

§ The rotation phasor ]7 can be shown in the complex Gaussian plane

in comprehensible manner. imaginary
axis
W86
Example: ]V W85 j W87

W84 W88= W80

1 real
axis

W83
W81
W8 2

182
6.1 Discrete Fourier Transform (DFT) - Introduction
imaginary
axis
Rotation phasor and Kronecker symbol W86
§ The discrete Q-function is now W85 j W87
7*9
:*: ; W 1 for l = l J
V ]" = e ⋅ Q:,: ; with the Kronecker−Symbol Q:,: ; =E W88= W80
0 otherwise W84
W-# 1 real
axis
§ e ⋅ Q:,: ; fulfills the same task in the discrete Fourier transform as the Dirac pulse
W83 W81
for the continuous Fourier transform.
W82

§ The above equation shows that if we sum up all phasors, we will get zero,
except all phasors point to ]"# (i.e., 3 o’clock), which is enforced by l = l J .
In this case, we get:
W80·0 W80·1 W80·2 W80·3 W80·4 W80·5 W80·6 W80·7

For e → ∞, we see the analogy to the Q-function.

183
6.1 Discrete Fourier Transform (DFT) - Characteristics
§ Linearity
Ø<9 ['] + ∞<% ['] ∘−• Ø>9 [l] + ∞>% [l]

§ Periodicity
> l => l+e
<['] = <[' + e]
§ Parseval-Theorem
7*9 7*9
%
1 %
V <['] = V >[l]
e
"-# :-#

The energy of a sampled signal can be calculated in time as well as in frequency domain.
§ Symmetry for a real-valued time signal
e e 0, 1, 2, … e even
> + ± = >∗ −± , ±=E
2 2 0.5, 1.5, … e odd

184
6.1 Discrete Fourier Transform (DFT) – Example 1
! Y = 1 for Y = 0, … , 3, r = 4; constant function
-./
0123+
input ![%] Expectation: for the continuous Fourier transform → Q(0) ; D = E 6 * e. -

1 discrete: only >[0] ≠ 0 +,%

-./
u
1 0123+
&; : 6* = E; D e -
/ v ⋅.⋅! '
<[0] = B1⋅e =4 3,%

% .,!
u u
&; : &; : .
x[0] x[1] x[2] x[3] <1 = B 1 ⋅ e/ v ⋅.⋅- = B e/ v = 1 + −j + −1 + j = 0
.,! .,!
As ![%] is an even function,
output <[U] u
&; :
u
v; : .
<[U] does not contain
/ v ⋅.⋅& / any imaginary part.
<2 = B1⋅e = Be v = 1 + −1 + 1 + −1 = 0
4
.,! .,!
u u
&; : w; : .
<[3] = B 1 ⋅ e/ v ⋅.⋅u = B e/ v = 1 + j + (−1) + (−j) = 0
U .,! .,!

X[0]X[1]X[2]X[3]

.6< .6<
9 X ⋅"⋅: 9 ⋅"⋅#
Control by inverse transform: < ' = ∑ > l e = = ⋅4⋅e = = 1 for ' = 0, … , 3
K :-# K

185
6.1 Discrete Fourier Transform (DFT) – Example 2
<[0] = 1, <[1] = 0, <[2] = −1, <[3] = 0, N = 4; "cosine"
-./
input ![%] ; D = E 6 * G-34+
j
W43 Expectation: for the continuous
1 +,%
Fourier transform → Q : + Q(−:) -./
% 1
0 W4 2 W44= W40 6* = E ; D G-.34+
x[0] x[1] x[2] x[3] '
1 > 0 = 0 (corresponds to the sum) 3,%
-1

9⋅# 9⋅%
W41 > 1 =1⋅]
K + −1 ⋅ ]
K = 1 ⋅ 1 + −1 ⋅ −1 = 2
periodic "3o ' clock " "9 o ' clock "
output <[U] continuation 2 %⋅# %⋅%
2 >[2] = 1 ⋅ ]
K + −1 ⋅ ]K = 1 ⋅ 1 + −1 ⋅ 1 = 0

"3o ' clock " "3o ' clock "
U U X⋅# X⋅%
>[3] = 1 ⋅ ]
K + −1 ⋅ ]
 = 1 ⋅ 1 + −1 ⋅ −1 = 2
K
X[0] X[1] X[2] X[3] X[-2]X[-1] X[0] X[1]
"3o ' clock " "9 o ' clock "
interval interval

§ Since we have a real and even input signal, exactly the same values are in the right half as in the left: >[e − l] = >[l]
§ If we would have a real and odd input signal, the conjugated complex values would be in the right half as in the left:
>[e − l] = > ∗ [l]
186
6.1 Discrete Fourier Transform (DFT) – Example 3
<[0] = 0, <[1] = 1, <[2] = 0, <[3] = −1, N = 4; "sine" -./

input ![%] ; D = E 6 * G-34+

W43 +,%
j -./
1 1
% 6* = E ; D G-.34+
0 W44= W40 '
W42 3,%

1 > 0 = 0 (corresponds to the sum)

-1
9⋅9 9⋅X
output: Real part=0 W41 >[1] = 1 ⋅ ]
K + −1 ⋅ ]K = 1 ⋅ −j + −1 ⋅ j = −2j

"6o ' clock "
imaginary part: "12 o ' clock "

<[U] periodic > 2 = 1 ⋅ ]K%⋅9 + −1 ⋅ ]K%⋅X = 1 ⋅ −1 + −1 ⋅ −1 = 0

2j 2j  
continuation "9 o `clock " "9 o `clock "

U >[3] = 1 ⋅ ]KX⋅9 + −1 ⋅ 
]KX⋅X = 1 ⋅ j + −1 ⋅ −j = 2j

U "12 o ' clock " "6o ' clock "

-2j -2j

§ Since we have a real and odd input signal, we get in the right half the conjugated complex as from the left half:
>[e − l] = > ∗ [l]
187
 -./

; D = E 6 * G-34+
6.1 Discrete Fourier Transform (DFT) 1
+,%
-./

6* = E ; D G-.34+
'
3,%

§ Per frequency point l, the algorithm turns out

one full turn and then adds up all complex phasors.

§ The DFT algorithm turns the complex phasors

of a signal backwards.

e
tim

Model of a complex sine-shaped signal with three periods.

188
6.2 The DFT as approximation of the Fourier Transform

§ If the sampled time signal <['] is of

finite length (' = 0, … , e − 1),
the following will hold:
ä
>! l => l
e
§ In words: The output of the discrete time
Fourier transform and of the DFT are identical
at the sampling points in frequency

: = lä/e

§ Example on the right: (sampling frequency

:4 = 1 kHz)

189
6.2 The DFT as approximation of the Fourier Transform
Proof
§ For the Fourier Transform of a sampled signal, the following holds
+

>4 (:) = V <[']e*$%& ",$(

"-*+
§ If the sampled signal <['] is of finite length (' = 0, … , e − 1), then it will apply
7*9

>4 (:) = V <[']e*$%& ",$(

"-#
§ The DFT of the signal <['] is given by
7*9

>[l] = V <[']e*$%& :"/7

"-#
§ The comparison of both equations delivers
1
> l = >4 l = >4 lä4
)4 e
190
6.2 The DFT as approximation of the Fourier Transform

§ If a time-continuous signal <(!) is limited to an interval of duration )# , the Fourier transform >(:) can be
approximated with the DFT at the discrete frequency points : = lä4 /e:

ä4 e e
> l ≈ )4 > l for l = − ,…, − 1
e 2 2

(holds for even e, similar for odd e)

§ The length of the sampled signal e)4 has to be chosen with the same length
or longer than the signal duration )# , i.e., e)4 ≥ )0.
§ If the length of the sampled signal is longer than the signal duration, this is called Zero Padding,
which means that all sampling points behind the signal duration are filled with zeros.
§ Using zero padding, one obtains a better graphical resolution of the spectrum
(: = lä4 /e, e gets greater with zero padding).

191
6.2 The DFT as approximation of the Fourier Transform
Example:

192
6.2 The DFT as approximation of the Fourier Transform
§ The distance
ä 1
ä4 = =
e e)4

of two frequency points is called frequency resolution of the DFT,

the duration e)4 is called signal duration.

§ The frequency resolution is equal to the inverse of the signal duration.

§ The frequency resolution can be improved through zero-padding.

§ The term resolution refers here to the distance of two sample points in frequency domain of
the DFT spectrum. It should not be mixed up with the minimal frequency distance, which two
sine oscillations must have to separate them in the DFT spectrum.

193
6.2 The DFT as approximation of the Fourier Transform

§ The DFT would be perfect, if our signal is limited in time and frequency domain.
§ However, this is impossible, since a signal limited in time has an infinite spectrum and
a signal limited in frequency has an unlimited time series.
§ Thus, using DFT, we always violate (hopefully only slightly) the sampling theorem.
§ We will see later (“Windowing”) how we can minimize this violation of the sampling theorem.

194
6.2 The DFT as approximation of the Fourier Transform
§ Under the condition that exactly one or several full periods of a periodic signal are sampled,
the Fourier coefficients can be calculated according to the following formula:
1
q: ≈ >[l] for l = −e/2, . . . , e/2 − 1
e
§ The same equation will hold if the periodic signal is band-limited and the sampling theorem is maintained.

§ For correct sampling, one has to keep the following points in mind:
- The length of the measuring windows has to be equal to one or more full periods of the signal.
- The value at the left edge is sampled.
- The value at the right edge is not sampled anymore.

195
6.2 The DFT as approximation – Choice of parameters
§ For the analysis of a signal by means of the DFT (FFT), it is necessary to carefully choose
the following parameters:
§ P2 sampling frequency (corresponds to period in frequency domain)
§ &2 sampling time
§ T number of samples
§ & measuring duration
§ r2 distance between two frequency points after the transform (frequency resolution)

§ The following relations hold:

%5 = 1/35

% = r%5
35 1 1
p5 = Δ3 = = =
r r%5 %
§ Hence, only two parameters are free to be chosen, all other parameters result immediately from this choice.
196
6.2 The DFT as approximation – Choice of parameters
Choice of the sampling frequency :4
§ The right choice is mainly given by the sampling theorem. The following has to apply:

35 > 23IJK

§ :YZ[ is thereby the highest frequency appearing in the signal.

§ If the signal, which has to be analyzed, is not band-limited, an anti-aliasing filter has to be connected
before sampling, which limits the band.
§ Usually, the signals to be analyzed are disturbed with noise (superimposed with additive noise).
Noise signals in general have a very wide spectrum.
§ Even if the wanted signal is band-limited, it is recommended to use an anti-aliasing filter in order to
band-limit the noise.

197
6.2 The DFT as approximation – Choice of parameters
Support for the choice of the amount e of the samples
§ The measuring duration ) = e)4 has to be as long or longer as the signal duration.
§ e should be, if possible, a power of two to calculate the DFT using the FFT
§ For the DFT, the frequency distance ä4 between two samples in frequency equals :4 /e.
If you want to choose ä4 , e must be
:4 N=12
e≥
ä4
§ To increase the spectral resolution,
e has to be increased (e.g., using Zero Padding).

N=16
§ If a periodic signal is analyzed and
the period duration )# of the signals is known,
then e has to be chosen e) = )# .

198
6.2 The DFT as approximation – Applications
Example
§ Given a sine oscillation, switched on at ! = 0 and damped in time
)
*>
< ! = r# sin 25 :# ! ⋅ e D !
with the time constant 7 = 1 s, r# = 1 V and the frequency :# = 1 Hz.
§ The Fourier transform of this function results in
25:# r#
>(:) =
(j25: + 1/7)% + (25:# )%

§ Now, by means of a DFT-analysis, an approximation of the Fourier transform shall be calculated.

§ Choice of the parameter:
§ The frequency of the signals is P! = 1 Hz. We have to choose for the sampling rate a much higher value:
P! = 10 Hz (&2 = 0.1 s).
§ The signal is approximately faded away after 5X. Thus, we choose a measuring duration of & = 6.4 s
(thus T = 64, a power of two)
Ø T = 64, & = 6.4 s, r2 = 0.156 Hz

199
6.2 The DFT as approximation – Applications
§ Input: damped sine oscillation
and sampling Dirac comb

units of <: V

§ Result of the DFT analysis

units: Vs

200
6.2 The DFT as approximation – Applications
Mean value, root mean square value and total harmonic distortion of a signal
§ In many applications, the following values are important:
§ Mean value (DC-value, linear mean value)
§ Root mean square value (RMS-value)
§ Root mean square value of the AC component (AC-part)
§ Total harmonic distortion (harmonic content)

§ These variables can simply be calculated using a DFT-analysis

§ Sampling of a period of the signal (e…number of samples)
§ Calculation of the DFT → >[l]
$
§ Mean value: 2L- ≈ 2 0
E
!
§ Root mean square value of the µth harmonic: 2" ≈ 2[[] for l = 1, … , e/2 − 1 (if e even)
E

201
6.2 The DFT as approximation – Applications

Mean value, root mean square value and total harmonic distortion of a signal

§ Root mean square value of the AC component: 2M- ≈ ∑E/!&$ !

"1$ 2" (for even r)

§ Root mean square value >`x[ of the periodic signal <(!): 29N5 ≈ 2L- ! + 2M- !

∑z/q{|
syq Ps q
§ Total harmonic distortion: Q ≈ (for even r)
∑z/q{|
sy| Ps q

The total harmonic distortion is defined as the relation of the root mean square values of all harmonics to the
root mean square value of the complete AC component (including the fundamental signal). Thus, it is a measure
for the deviation from a pure sinusoidal signal.

202
6.3 Fast Fourier Transform (FFT)
James W. Cooley,
*1926, † June 29 John Wilder Tukey
2016, (* 16. June 1915,
† 26. July 2000)

§ The FFT (Fast Fourier Transform), published in 1965 in a famous article of Cooley and Tukey,
is the most important algorithm in digital signal processing.

§ The FFT is not a novel transform – but an efficient method to calculate the DFT.

§ Starting point of the FFT is the question:

What is the Fourier transform of a signal with only one sample point?

! 0 , r = 1 ⟹ 2 0 = ![0]
Ø The number itself

203
6.3 Fast Fourier Transform (FFT)
We start with the definition of the DFT:

E&$
2 [ = J ! Y vE 4" [ = 0,1, . . . , r − 1
41/
with
§ l discrete frequency variable
§ ' discrete time variable
§ e number of samples, number of frequency samples
§ >[l] discrete Fourier transform at the frequency point : = l:4 /e
or the lth DFT-coefficient
§ <['] sample of <(!) at the point ! = ')4 = '/:4
§ ]7 rotation phasor ]7 = e*$%&/7

204
 3/-
<[U] = B ![%]x3 .~ U = 0,1, . . . , T − 1
6.3 Fast Fourier Transform (FFT) .,!

Calculation cost
§ In order to calculate the DFT at one frequency point, e multiplications and e − 1 summations have to be processed.

§ If the DFT should be calculated at all e frequency points, e2 multiplications and e(e − 1) summations
have to be processed.

§ The calculation cost, measured in the number of multiplications, is thus ∂}.

§ The FFT-algorithm provides another method, with which the DFT can be calculated with
significantly less calculation cost.

§ Thereby, it is presumed that e is a power of two, which means it can be written in the form e = 2\ with ∑ ∈ ℕ.

205
6.3 Fast Fourier Transform (FFT)
FFT-algorithm
§ At first, we split the sequence <['] into two subsequences

<9 ' = < 2'

<% ' = < 2' + 1

for ' = 0, … , e/2 − 1. The first subsequence consists of all samples with even index 2'
and the second subsequence of all samples with odd index 2' + 1.
§ Now, we can write for the DFT-sum (l = 0, … , e − 1):
7*9 7/%*9 7/%*9

> l = V < ' ]7 ": = V <[2']]7 %": + V <[2' + 1]]7 (%"@9):

"-# "-# "-#
7/%*9 7/%*9

= V <9 [']]7 %": + ]7 : V <% [']]7 %":

"-# "-#

206
 3 3
& /- & /-
< U = B !- % x3&.~ + x3~ B !& % x3&.~
6.3 Fast Fourier Transform (FFT) .,! .,!

§ With
%& %&
%": *$ 7 %": *$ ":
]7 =e =e 7/% = ]7/% ":

we can write
7 7
% *9 % *9
> l = V <9 ' ]7⁄% ": + ]7 : V <% ' ]7⁄% ":
"-# "-#

= >9 l + ]7 : >% l

for l = 0, … , e − 1.

§ >1[l] and >2[l] are the DFTs of <1['] and <2['].

§ <1['] and <2['], however, have only e/2-points.
§ In other words: We have split the e-point-DFT into two DFTs with e/2-points and
saved nearly half of the calculation cost.

207
 >[l] = >9 [l] + ]7 : >% [l]
6.3 Fast Fourier transform (FFT)

§ Graphic representation of the

segmentation of an 8-point DFT
into 2 DFTs with 4-points

208
6.3 Fast Fourier Transform (FFT)
Calculation cost
§ Each of the two e/2-DFTs requires e/2 % multiplications.
§ The multiplication with the factor ]7 : needs additional e multiplications.
§ Total cost: 2 · (e⁄2)% +e
§ Compared to the original e2 operations, the calculation cost is significantly reduced for large e.

If N is a power of two, the calculation can further be simplified:

§ Both of e/2-DFTs are split into e/4-DFTs each.
§ These will be further split until only 1-point DFTs remain.
§ The complete segmentation can only be done, if e = 2\ is a power of two.
§ The number of segmentation steps is then ∑ = log2(e).

209
6.3 Fast Fourier Transform (FFT)
§ Segmentation of a 8-point DFT
in 4 times 2-point DFTs

210
6.3 Fast Fourier Transform (FFT)
§ Complete segmentation
of an 8-point DFT

§ If we study this schematic, we can

see that each segmentation level
consists of e/2 so-called Butterfly-
Operations (butterfly-graphs).

211
6.3 Fast Fourier Transform (FFT)

Butterfly Operation
§ The processing of a butterfly operation requires
two complex multiplications, which is equivalent to
2 • 4 real multiplications.
§ Thus, a calculation cost of @ multiplications
per segmentation level is needed.
§ Total calculation cost for a @-point FFT: @ log2(@)
§ Example 1: 1024-points DFT
§ Direct calculation of the DFT after the defining equation leads to approximately 106 multiplications.
§ Calculation by means of the FFT only leads to approximately 104 operations.
§ Example 2: A song of 1 minute length is sampled with -@ = 44.1 kHz, resulting in @ = 221 samples. Using a
computer with D = 0.2 µs for one multiplication, the following times result for calculating the spectrum:
§ Direct calculation of the DFT: 0.2 µs · 2H(HI = 244.3 h ≈ 10 days
§ Calculation by means of the FFT: 0.2 µs · 221 · log H 2HI = 8,8 s
212
6.3 Fast Fourier Transform (FFT)
§ With
vE Q#E/! = vE E/! vE Q = −vE Q

the Butterfly operation can be reduced

further to only one multiplication:

E
§ The total calculation cost results in log ! ( r).
!

213
6.3 Fast Fourier Transform (FFT)
§ Complete graph of an
8-point FFT

214
6.3 Fast Fourier Transform (FFT)
Annotations
§ e should be chosen as a power of two (2, 4, 8, 16, 32, 64, 128, 256,…).

§ If the number of samples is not a power of two, one can add zeros (zero-padding) until e is power of two.

§ Since the FFT-algorithm is the most known DSP-algorithm, many software algorithms already exist
so that nowadays only very rarely somebody has to program it her-/himself.

215
6.3 Fast Fourier Transform (FFT)
x[0]
§ Example: “saw tooth”
x[2]
<[0] = 0, <[1] = 1, <[2] = 2 <[3] = 3, N = 4;
x[1]
input
WN1
x[3]
1
0 2
0 6

j
W43 1 -2
2 -2+2j
W44= W40
W42 4 1
1 1 -2

1 -2 -j
W4 1
3 -2-2j
216
6.4 Windowing and spectral analysis
§ For any application of the DFT (FFT), a sample sequence limited in time domain is required.
§ Often, however, one would like to perform a spectral analysis of a continuous sampling signal.
§ Consequently, it is necessary to temporally limit the continuous sequence.
§ One talks then about the Windowing of the sampling signal.
§ We look at the following applications, which can be performed by means of an appropriate Windowing
and subsequent FFT:
§ Detection of sine signals
§ Short-time spectral analysis of audio signals

217
6.4 Windowing and spectral analysis
The algorithm of the DFT assumes a periodic continuation of the signal. However, it is very unlikely that the
signal to be analyzed shows an integer number of periods within the analyzed time window.
Ø Only then, the periodic continuation of the analyzed time window corresponds
to the original function.
Ø Only in this case, the DFT represents a good approximation of the Fourier transform.
periodic
continuation

§ The period of the function, which has to be analyzed, and the

sampled time window match.
→ DFT is a good approximation of the Fourier transform.
Time Time
window window
periodic
continuation
§ The period of the function, which has to be analyzed, and the
sampled time window do not match.
→ DFT is not a good approximation of the Fourier transform.
Time Time
window window

218
 § The period of the signal, which has to be analyzed,

6.4
Windowing and and the sampled time window do not match.
→ DFT is not a good approximation of
spectral analysis the Fourier transform.

§ The period of the signal, which has to be analyzed, and

the sampled time window match.
→ DFT is a good approximation of the Fourier
transform.

periodic
continuation

bath tub
due signal jump 219
6.4 Windowing and spectral analysis
§ Example 1: given the following signal:

!(#) = Å/ cos( 2=3/ #)

The frequency :# has to be measured, thus it is unknown. The analysis is done with FFT.
For that, the signal has to be sampled and the FFT-parameter have to be chosen carefully.
§ For a successful application of the FFT, it is important to have a rough idea of the signal,
which has to be analyzed, such as
§ In which frequency domain is the signal?
§ What is the amplitude range?
§ Is it superimposed by noise?
§ A rough overview of these parameters is necessary so that the parameter of the FFT (such as e and :4 )
can be chosen adequately.

220
6.4 Windowing and spectral analysis
§ Presumption: The frequency :# to be measured is in the kHz domain.

§ Possible choice of the FFT-Parameter

§ Sampling frequency :4 = 8 kHz (sampling theorem)
9
§ Sampling time )4 = ( = 125 µs
$
§ Signal-recording length: 4 ms → 32 samples are available

§ Target frequency resolution: ä4 ≤ 100 Hz

9
Ø Measuring duration ) ≥ U = 10 ms
$
Ø Number of samples e ≥ 10 ms⁄0.125 ms = 80
Ø Choice: e = 128

221
6.4 Windowing and spectral analysis

§ Result in the frequency domain:

§ One can observe a peak at

Ω = 0.785 (corresponds to
: = 1 kHz), but no single line,
although a periodic signal shows a
line spectrum.

§ Unwanted secondary peaks,

so-called Side lobes show up.

222
6.4 Windowing and spectral analysis

§ Example 2: We will now analyze what happens if we cut a signal <(!) and then perform the Fourier transform.
We cut <(!) anywhere – if possible, there, where the signal is no longer large.
*_)
< ! = Ee for t > 0
0 for ! < 0
+ + 9 + 9
>(:) = ∫*+ <(!) ⋅ e*$%&() d! = ∫# e*_) e*$%&() d! = e*_) e*$%&() #
= (Lorenz-curve)
*_*$%&( _@$%&(
*_)
§ Now, we cut: < ! = Ee für 0 ≤ ! < )
0 otherwise
, 9 , 9*D>?5><.6@5
>(:) = ∫# e*_) e*$%&() d! = e*_) e*$%&() #
=
*_*$%&( _@$%&(

*D>?5><.6@5
§ Compared to the uncut signal, we obtained an additional term .
_@$%&(
This term is not really large for large ), but it has the nasty characteristic to oscillate.

§ On top of the smooth Lorenz-curve, we get small oscillations (ripples) due to the truncation.
223
6.4 Windowing and spectral analysis
§ Conclusion: Never cut without any need and especially not so rough.
§ Instead: Gently weight the signal to zero.
§ The hard cutting of the signal (in example 1 at 4 ms) means that the signal <['] was multiplied
with a window function Ω['] (here rectangle-shaped):

!R [Y] = ![Y] ⋅ Ç[Y]

§ In frequency domain, this multiplication corresponds to the convolution operation:

>`! (Ω) = >! (Ω) ∗ ]! (Ω)

§ Leakage: Through this convolution, parts of the spectral energy of the signal
gets spread over a wide frequency domain.

224
6.4 Windowing and spectral analysis

§ This rectangular window is inappropriate for most applications because of its very high side lobes.
§ For example, if we have in the signal a second frequency with a significantly smaller amplitude near by the
strong peak, it could be covered completely by the side lobes of the strong peak and, thus, it can not be
recognized.

§ Common recommendations for the DFT

§ Many zeros do well
§ Choose your sampling interval )4 fine enough so that you always have a significantly higher Nyquist frequency
as you expect from the spectrum, which means that you need >[l] only for l ≪ e.
Thus, you will approximately get rid of the consequences of the periodic continuation of the spectrum.
§ Use the window functions, e.g., Hamming window.

225
6.4
Windowing and spectral analysis –
Rectangular window
Main lobe

Amplitude in dB

Frequency

227
 6.4 Windowing and spectral analysis – Rectangular
Rectangular window, time domain Frequency domain in dB, N=41

Side Lobe Level

1 0≤Y ≤r−1
Ç9 (Y) = É -13 dB
0 else

Scaled amplitude in dB
Amplitude

Samples Frequency

228
6.4 Windowing and spectral analysis

§ Already Fourier knew about the problem of the leakage (side lobes).
§ At first, the leakage seemed very confusing: Where do all these frequencies come from?
Finally, it turned out that the additional frequencies stem from the jump from zero to the first point
and from the last point back to zero.
§ People assumed that this leakage cannot be avoided because one has to start and stop with
the data anywhere and exactly there has to be a jump.
§ Maurice S. Bartlett suggested to get rid of the jump discontinuity by multiplication of the data
with a so-called weighting function.

2Y r−1
Maurice S. Bartlett 0≤Y≤
(* 1910 in Scrooby, England, ÇS6Q,T>,, Y = r−1 2
† 2002 in Exmouth, England) 2Y r−1
2− ≤Y ≤r−1
r−1 2

229
 6.4 Windowing and spectral analysis – Bartlett window
Bartlett (“Dreieck“) window, time domain Frequency domain in dB, N=41

2% T−1
0≤%≤
y\Ä= % = T−1 2
2% T−1
2− ≤%≤T−1
T−1 2

Scaled amplitude in dB
Samples Frequency

230
6.4 Windowing and spectral analysis
Modified rectangular window according to Bartlett
Modified Bartlett window in time domain, N=41

§ A window, which is
adjustable between
rectangle and Bartlett.
Scaled absolute amplitude in dB, N=41

Scaled amplitude in dB
§ The side lobe suppression
is also situated between
Samples
rectangle and Bartlett.

Frequency
231
6.4 Windowing and spectral analysis

§ Bartlett’s idea was partially correct – the high-frequency side lobes have been reduced but
they did not disappear completely. His idea was good, but not perfect.
§ Bartlett was a little bit disappointed, because he expected that all “wrong” frequencies will disappear
as soon as the curve is continuous.
§ According to Bartlett, the following effects generate high-frequency side lobes:
§ Discontinuities (steps or jumps)
§ Discontinuities in the derivative (corners)
§ Discontinuities in the second derivative (change of the gradient)
§ Discontinuities in every derivative
§ The high-frequency decline of the side lobes in the spectrum 1/: (R@9) results from
the lowest non-continuous derivative æ.

Examples
– Rectangular: spectrum sin(<)/<, the side lobes decline with 1/: #@9 , i.e., 1/:
– Triangular: spectrum (sin(<)/<)% , the side lobes decline with 1/: 9@9 , i.e., 1/: %
– Sine: continuous in all derivatives, side lobes decline with 1/: +

232
6.4 Windowing and spectral analysis – Hann window

§ A window, which weighs its coefficients much more effective, but is

nevertheless relatively simple, is the Hann-window after J. von Hann.

§ It is well known as Hann-window or "Raised-Cosine-window" and is defined by

1 2=Y
ÇU644 Y = 1 − cos 0≤Y ≤r−1
2 r−1
§ The width of the main lobe becomes j = 7.8 5 /e, which is twice
as big compared with the rectangular window. The attenuation of the first
side lobe is already 31.5 dB.
Julius Ferdinand von Hann
(* 1839 Castle Haus, Wartberg over Aist,
† 1921 in Vienna)

233
 6.4 Windowing and spectral analysis – Hann window
Hann window, time domain Frequency domain in dB, N=41

1 25'
ΩOA"" ' = 1 − cos
2 e−1

Scaled amplitude in dB
0≤' ≤e−1

Samples Frequency

234
6.4
Windowing and spectral analysis
– Hamming window
§ With the Hamming-window, the side lobe suppression is improved further.
Through the insertion of a parameter ∞, the Hann-window is modified and, ´
thus, a reduction of the unwanted spectral parts is achieved:
25'
ΩOARR5"P ' = 1 −∞ − ∞ ⋅ cos 0≤' ≤e−1
e−1

§ Through a proper choice of ∞, a complete cancellation of the main maxima Richard Wesley Hamming
within the stop band can be achieved. ( *1915 in Chicago, Illinois;
† 1998 in Monterey, Kalifornien)
§ The maximum side lobe level of the Hann window is given by the first side lobe by ; = 55/e.
<F6
aABCCD:E D 8
If we force = 0 , then it follows that ∞ = 0.46.
aABCCD:E (9)
§ Hence, we define a new window function called Hamming-window:
25'
ΩbARR5"P ' = 0.54 − 0.46 ⋅ cos 0≤' ≤e−1
e−1
§ From above equation, we calculate for ΩbARR5"P (0) = ΩbARR5"P (e − 1) = 0.08.
Therefore, the Hamming-window is also called the "Raised-Cosine-window" with Platform.

235
 6.4
Windowing and spectral analysis
– Hamming window
Hamming window, time domain Frequency domain in dB, N=41

%&"
ΩOARR5"P ' = 0.54 − 0.46 ⋅ cos
7*9

0≤' ≤e−1

Scaled amplitude in dB
Samples Frequency

236
6.4
Windowing and spectral analysis
– Blackmann window
§ The Blackmann window is also an often used window function:

2= Y 4= Y
ÇST62"<64 (Y) = 0.42 − 0.5 cos + 0.08 cos
r−1 r−1

§ In comparison to the Hann and Hamming window, the Blackmann window

shows the largest side lobe rejection.

§ The frequency response has a relatively wide main lobe, the side lobe level
is very good: j = 12 5/e, the first side lobe is already damped to 57 dB.

R. B. Blackman and J. W. Tukey, “The measurement of power

spectra from the point of view of communications
engineering,“ Bell Sys. Tech. J., vol. 37,pp. 185-282 and 485-
569, 1958; reprinted by Dover, New York, 1958

237
 6.4
Windowing and spectral analysis
– Blackmann window
Blackman window, time domain Frequency domain in dB, N=41

ΩcWAd:RA" ' =
25' 45'
0.42 − 0.5 cos + 0.08 cos
e−1 e−1

Scaled amplitude in dB
Samples Frequency

238
6.4 Windowing and spectral analysis
§ Rectangular window and corresponding e` = 32
amplitude spectrum

1 for 0 ≤ Y ≤ rR − 1
Ç Y =É
0 otherwise
'?EÅ sin(ΩrR /2)
v5 Ω = e !
Ω⁄2

Ø Width of the main peak: 45/eÇ

Ø Side lobe suppression: 13 dB

241
6.4 Windowing and spectral analysis
§ Hann window e` = 64

ÇU644 Y
2=Y
= 0.5 − 0.5cos
r−1

Ø Width of the main peak:

7.85/eÇ
Ø Side lobe suppression:
31.5 dB

242
6.4 Windowing and spectral analysis - Overview
window function spectrum

Example
§ Measurement signal with 128 points
§ Calculated and spectrum with rectangular rectangular
different window functions
§ Rectangle
§ Hann
§ Hamming
Hann

Hann
Other often used windows:
§ Flattop (Turkey)
§ Kaiser-Bessel- and
§ Blackman-Harris-window

Hamming Hamming
243
6.4 Windowing and spectral analysis - Overview
Peak-width
Suppression of the
highest side lobe § Evaluation is always a compromise:
The higher the side lobe suppression,
side lobe- the wider gets the main peak.
damping
§ In signal processing mostly Hann,
Hamming and Turkey, for special
applications Kaiser-Bessel or Blackman-
window are employed.
Window Suppression of the Side lobe 6 dB-peak-width
function highest side lobe in damping (bins = 2pfs/N)
dB (dB/Oct) § Other windows are rather of
Rectangular -13 -6 1.21 „academic“ interest.
Hann -32 -18 2.00
Hamming -43 -6 1.81
Blackman -58 -18 2.35

244
6.4
Windowing and spectral analysis –
Detection of sine signals
Example
§ We want to analyze the following signal (superimposition of two sine oscillations):

< ! = r9 cos 25:9 ! + r% cos 25:% !

whereas both of the frequencies :9 and :% have to be measured, thus they are unknown.
In the example, r1 = 1 V, r2 = 0.4 V, :9 = 996 Hz, :% = 1032 Hz are chosen.

§ For the successful application of a FFT, it is important to have a „rough idea" of the signals, which have to be
analyzed, and we have to answer these questions:
§ In which frequency domain are the signal components?
§ How close are both frequencies to each other?
§ How big are their amplitudes?
§ Is noise superimposed?

245
6.4
Windowing and spectral analysis –
Detection of sine signals
§ The parameter, which have to be chosen
§ FFT-Parameter: sampling frequency :4 , sampling time )4 , number of samples e,
measuring duration ), frequency resolution ∆: = ä4
§ Window type, window width e`

§ The width of the main peak in the frequency domain depends considerably for all window types on the
window width e` in the time domain.
Ø The bigger e` , the smaller the main peak in the frequency domain.

§ With the choice of the window type, the following characteristics can be controlled:
§ Width of the main peak within the frequency domain (this is different with the same e`
for various window types)
§ Height of the side lobe

246
6.4
Windowing and spectral analysis –
Detection of sine signals
§ Assumption
§ The frequencies, which have to be measured, are approximately in the kHz domain.
§ The difference between both frequencies is at least 20 Hz.

§ Possible choice of the parameter

§ P2 = 8 kHz (sampling theorem)
§ &2 = 1⁄P2 = 125 µs
§ To separate both frequencies after the DFT, both peaks must not overlap each other and the side lobes have to be severely
suppressed → e.g., Hann window → side lobe suppression 31.5 dB
§ Width of the main peak ~É ≤ 20 Hz; ~É,.ÑÄÖ = 23~É ⁄P2 = 0.0157
§ ~É,.ÑÄÖ = 7.8 3 ⁄TÉ ≤ 0.0157 à choice: TG = 1560
§ Target: r2 ≤ 2 Hz (≈ 10 samples per main peak in the frequency domain)
-
§ & ≥ Ü = 0.5 s
5
§ T = &/&2 ≥ 0.5 s/125 µs = 4000
§ Choice: T = 4096

247
6.4 Windowing and spectral analysis –
Detection of sine signals – Results of the spectral analysis

time domain
§ One can clearly see two peaks
of different amplitudes at
Ω9 = 0.782 and
Ω9 = 0.811.

§ This corresponds to the

frequency domain frequencies
:9 = 966 Hz and
:% = 1033 Hz.

§ The amplitude relation is

frequency domain 1: 0.4.
(zoom in)
Ø Very good agreement with
original signals.

248
6.4 Windowing and spectral analysis

Discrete Fourier transform (DFT) of real signals

Time domain Frequency domain For real-valued signals, a

special real FFT can be
Real-valued measurement signal positive frequencies negative frequencies applied, with which only
the positive frequency
values of 0 to e/2 are
Re X(i) Re X(i)
calculated.
X(n) FFT
→ Further reduction of
0 N/2 N/2+1 N-1
Im X(i) Im X(i) the calculation effort.
0 N sample points N-1 IFFT
0 N/2 N/2+1 N-1
Redundant Information:
mirrored and conjugated complex
to positive frequencies

249
 6.4 Windowing and spectral analysis –
discrete Hilbert transform
Problem
Measured: real-valued sampling signal ⟹ sought-after: the phase (gradient) and/or envelope of the signal
positive frequencies negative frequencies
real-valued measurement signal Re X(i) Re X(i)
x(n)
FFT 0 N/2 N/2+1 N-1 § Attention: noise through
Im X(i) Im X(i)
0 N sample points N-1 window effects
0 N/2 N/2+1 N-1
§ <′(') is the analytical signal
Complex-valued expanded signal Put all values to zero for <('); it applies
Re x‘(n) Re X(i) Re X(i) < ' = Re <′(')

0 N sample points N-1 N/2 N/2+1 N-1

0 § Envelope of <(') results from
Im x‘(n) IFFT Im X(i) Im X(i)
abs Re <′(') + j Im <′(')
N sample points N/2 N/2+1 N-1
0 N-1 0
=0
250
6 Discrete time signals within the frequency domain
6.1 Discrete Fourier transform (DFT)
§ Introduction
§ Characteristics
6.2 The DFT as approximation
§ the discrete time Fourier Transformation
§ the Fourier transform
§ the Fourier sequence
6.3 Fast Fourier transform (FFT)
6.4 Windowing and spectral analysis
6.5 Spectral analysis of stationary and non-stationary signals
§ Motivation
§ Spectrogram

253
6.5 Time-Frequency Distribution

254
6.5 Spectral analysis of non-stationary signals

§ The Fourier transform gives the spectral composition of a signal. However, it does not give any information
about the temporal relation of this spectral component.

§ Stationary signals consist of spectral components, which do not change over time:
- All spectral components exist over the whole time.
- Temporal information is therefore not necessary.
- The Fourier transform works excellently for stationary signals.

§ Non-stationary signals consist of time-changeable spectral components.

- Many signals (for example audio signals) show typically unsteady behavior, which means the frequencies
that are contained in the signal, are at particular times stronger represented, and at particular times less
strongly represented.
- How does one define the temporal position of the spectral components?
- The Fourier transform only shows which spectral components appear, but not which components exist at which
point in time. (The temporal information is in the phase incline of the spectral function and cannot directly be
extracted from the signal).
- We need another method to define the temporal position of each spectral component and to represent them.

255
6.5 Spectral analysis of non-stationary signals

§ Stationary signals do not change through time

!$ # = cos 2= 5 #
+ cos 2= 25 #
+ cos(2= 50 #)

§ Non-stationary signals have a time changeable spectrum

cos 2= 5 # for 0 ≤ # < 0.3 s

!! # = á cos 2= 25 # for 0.3 s ≤ # < 0.6 s
cos 2= 50 # for 0.6 s ≤ # < 1 s

256
6.5 Spectral analysis of non-stationary signals
2$ (3)

2! (3)
§ Exact information about which
frequencies occur in the signal

§ Problem: No information about at

which point in time a frequency
component occurs.

257
6.5 Spectral analysis of non-stationary signals
§ Sine signals and exponential signals
§ are infinitely stretched within time no temporal localization
Dennis Gábor
§ are infinitely small in the frequency domain exact spectral localization (* 1900 in Budapest,
† 1979 in London)

Ø A global analysis is not meaningful for a non-stationary signal.

§ We need a local analysis procedure for a time-frequency-representation (time-frequency representation, TFR)

of non-stationary signals.
§ Short-time spectral analysis or spectrograms (short-time Fourier transform):
§ We dissect the signal into short-time intervals, which are short enough so that we can assume
the signal within them to be stationary.
§ Then, we calculate the Fourier transform of each segment (Gabor 1946).
§ The short-time spectral analysis is an important component of modern language and audio coding methods
as well as a method for recognition of speech.

258
6.5 Short-time spectral analysis - Principle
§ Out of the sequence ! Y ,
blocks of the length r (r = 2e ) time domain :
are cut out by a window function
Ç[Y].

Zoom in

Frame/ overlapping
block
259
6.5 Short-time Fourier transform / spectrogram

§ For each block, the FFT is calculated and as a rule, the amount of DFT-coefficients is graphically displayed.
One obtains for each block the so-called short-time Fourier transform (STFT)
7*9
:"
*$%& 7
>4,U, ', l = V < ' − æ Ω æ e
R-#

>4,U, !, ; = T< ! − 7 Ω ∗ (7) e*$H> d7

where Ω['] represents an appropriate window function.

§ For Ω ' = 1, the STFT reduces itself to the DTFT of <['].

§ For Ω['] = Q[0] = QR,# , the STFT reduces itself to the time function.
§ The signal blocks can superimpose – as in the representation – in order to get smooth transitions.
§ >[EáE[l, '] is a function of 2 variables: the time index ' and the frequency index l.
§ >[EáE[l, '] is periodic in the frequency with half of the sampling rate.
§ The display |>[EáE[l, ']|, respectively |>[EáE[l, ']|2 is called a spectrogram.
§ To display the spectrograms, one needs 3 dimensions.

260
6.5 Short-time Fourier transform / spectrogram

time- frequency- the signal, nucleus of the Fourier

parameter parameter which has to transform (rotation phasor)
be analyzed

!âäãä (#, %) = ()(# − +), ∗ (+)eåçéè d+

signal shifted
spectrogram of the about # window function
signal !(#)

261
 Short-time Fourier transform / spectrogram -
6.5
algorithm
§ Choose a window function of finite length.
§ The window function is placed at the beginning of the signal.
§ Weight the signal with the window function.
§ Calculate the Fourier transform of the weighted signals and store the result.
§ Shift the signal function about a fixed interval to the left.
§ Go back to point 3 until the window function reaches the edge of the signal.
§ In doing so, we obtain for each window position another spectrum.
§ These different spectra contain the spectral information, which belongs to the
individual time interval. This facilitates us with a common display of the signal energy
with respect to time and frequency.

262
6.5 Short-time Fourier transform and spectrogram

§ Large weighting window (in the

time domain) results in a fine
frequency resolution due to larger
amount of considered sampling
points rR
Ø narrow-band spectrogram,
narrow-band spectrogram
Ø time information gets smudged

263
6.5 Short-time Fourier transform and spectrogram

§ Small weighting window (in the

time domain) smudge the
frequency resolution
Ø Broadband spectrogram,
wideband spectrogram
Ø Fine time information available

264
6.5 Spectrogram of the sentence “today is free”

Narrow-band spectrogram Narrow band sonagram

§ High resolution within the
frequency domain, low
resolution within the time
domain.
§ Fundamental frequency and
individual harmonic (small
parallel „bands“) are well Harmonics
recognizable and can be
defined very accurately.
§ Formants (meaning harmonics),
which reach via resonance a
maximum intensity (vocal tract?)
are hard to recognize.
§ Single glottal stops are not
visible. Voicing Voicing Voicing

265
6.5 Spectrogram of the sentence “today is free”

Broadband spectrogram
§ High resolution in the time
domain, low resolution in the
frequency domain.
§ Individual glottal stops are well
visible (the vertical curves).
§ The formants (wide horizontal
“bands”) are well to recognize.
They are characteristic for the Formants
sounds, especially for the vocals.

Wideband-sonagramm

266
6.5 Spectrogram

§ One can see the mutual dependency between time and frequency resolution.
§ Either one can achieve the high time resolution at the expense of a limited spectral information, or a high spectral
resolution is bought with time smudging.
§ For the resolution (in terms of separation of two signals) within the time domain ∆) and within
the frequency domain Ɗ, the following applies:
∆ä ( ∆) ≈ 1

§ The required window function broadens the resolution additionally.

§ An integration over the frequency variable does not result in general in the time function
7*9

V >4,U, [', l] ≠ <[']

:-#
§ An integration over the time variable does not in general result in the frequency function
7*9

V >4,U, [', l] ≠ >[l]

"-#
§ A similar representation is the so-called sub band-filtering. Here, we start from the spectrum and cut out
small frequency domains, which then are transformed in the time domain.
267
6.5 Spectrogram

§ The spectrogram is often employed in the language and music analysis.

§ If employed correctly, artefacts of signal processing do not emerge, which means everything you see corresponds to
parts of the signal. The interpretation is simple.
>4,U, (!, ;) = T<(! − 7) Ω ∗ (7)e*$H> d7
7*9

>4,U, [', l] = V < ' − æ Ω æ e*$%& :"/7

R-#
§ Resolution:
1
Δ% ⋅ Δp ≥
4=

resolution in time resolution in frequency

§ Time and frequency resolution cannot both be high. (Heisenberg‘s uncertainty principle, 1930)
We cannot estimate exactly at the same time the temporal location and the respective frequency. We can only give a
respective frequency interval in which the signal energy is concentrated for a finite time interval.
268