FILTER BANKS AND WAVELETS EEN698 1

A Wavelet-based Pitch-shifting Method

Alexander G. Sklar

Abstract— In audio engineering, Pitch shifting is a term to signal). On the other hand, stationary signals do not occur
describe the process of changing the pitch of an audio signal in practice; consider the signal in the figure 1, containing
(which is related to the logarithm of the frequency), but opposite two frequencies ocurring at different times and its Fourier
to resampling, pitch shifting retains the duration of the original
signal. This process is very useful for example in correcting Transform magnitude spectrum in figure 2:
slightly out-of-tune singers or instruments. The dual problem
is changing the signal duration without affecting its pitch; this
is called time stretching. One of the most well-known techniques
for performing pitch shifting is called Phase Vocoding, but this
method presents some artifacts. This project aims to overcome
some of the deficiencies of the Phase Vocoder by using a wavelet-
based approach.
Index Terms— Pitch shift, vocoder, phase vocoder, wavelet,
Morlet

I. I NTRODUCTION
ITCH shifting describes the process of changing the pitch
P of an audio signal (which is related to the logarithm
of the frequency), but, opposite to resampling, pitch shifting
retains the duration of the original signal. This process is very
useful for example in correcting slightly out-of-tune singers or
instruments. The dual problem is changing the signal duration
without affecting its pitch; this is called time stretching. One of
the most well-known techniques for performing pitch shifting Fig. 1. A non-stationary signal
is called Phase Vocoding. This is a frequency-based technique,
whose main idea is to chop the input signal into very short
segments, perform a frequency domain analysis, change the
frequency content of that portion, and convert back to the time
domain. Basically, this is summarized as follows:

Algorithm 1 Typical frequency-domain algorithm

function freq_algorithm(in: x, out: y)
yb = [];
xb = buffer(x, N, M);
foreach segment in xb do
yb=fft(segment);
yb=modify(yb);
yb=[yb; ifft(yb)];
end
y=unbuffer(yb, N, M);

We split the input signal into (possibly overlapping by M

samples) blocks of length N , transform, modify and trans-
form back. This is a Short-Time Fourier Transform (STFT)
approach.
A. FT, STFT and the Wavelet Transform
For stationary signals, the Fourier transform contains all
the necessary frequency information, since all the present
frequencies are present all the time (this is after all, a stationary
asklar@gmail.com
Fig. 2. A non-stationary signal spectrum according to the FT
We note that the FT does not possess any time-localization
information. Frequencies occur at all instants in time. To over-
come this lack of time information, the STFT was developed.
In the next figure we see a spectrogram of the same signal).
We see that the frequencies are well localized in time as well
as in frequencies.
One of the difficulties the Phase vocoder encounters is the
“smearing” effect that affects transient components (such as a
2 FILTER BANKS AND WAVELETS EEN698

Fig. 3. A non-stationary signal’s spectrogram Fig. 5. A non-stationary signal with unresolved frequency components

bass drum or snare drum). This is due to the fact that despite
the STFT being “localized” both in time and frequency, it is
not localized enough. Consider the effect of computing the
spectrogram with a longer and a shorter window width in
figures 4 and 5.
quency1 and time). The main idea behind the use of wavelets
is to decompose a signal into a (possibly infinite) sum of
functions which are both time- and frequency-localized. This
is very different to the FT or STFT where the basis functions
were sinusoids (which are frequency-localized but not time-
localized since they extend from −∞ to +∞).

B. The Morlet Wavelet

The Morlet wavelet is named after Jean Morlet2 , although
it was originally formulated by Goupillaud, Grossmann and
Morlet in 1984. It is a three-parameter function of time:
1 1 2
Ψσ (t) = cσ π − 4 e− 2 t (eiσt − κσ ) (1)
− 12 σ2
where κσ = e is defined by the admissibility criterion3
and the normalization constant cσ is:
 2
 1
3 2 −2
cσ = 1 + e−σ − 2e− 4 σ
The Fourier transform of the Morlet wavelet is:

2
1 1
+ω 2 )
Ψ̂σ (ω) = cσ π − 4 ((σ − ω) eσω + ω) e− 2 (σ (2)
The "central frequency" ωΨ is the position of the global
Fig. 4. A non-stationary signal with unresolved time components maximum of Ψ̂σ (ω) which, in this case, is given by the
solution of the equation:
We see that in the figure 4, the precise location of the 2 2
− 1 e−σωΨ


(ωΨ − σ) − 1 = ωΨ
beginnings of the tones is uncertain. Likewise, the precise
frequency in figure 5 isn’t obvious. This is because the The parameter σ in the Morlet wavelet allows trade between
frequency resolution is inversely proportional to the analysis time and frequency resolutions. Conventionally, the restriction
window length. This is known as the Heisenberg uncertainty σ > 5 is used to avoid problems with the Morlet wavelet at
principle low σ (high temporal resolution).
In summation, the STFT provides constant bandwidth anal- 1 Actually, 1
ysis, i.e. high resolution in the high frequencies and low frequency .
2 Jean Morlet is a French geophysicist who did pioneering work in the field
resolution in the low frequencies. of wavelet analysis in collaboration with Alex Grossman. Morlet invented the
To overcome these problems, the wavelet transform was term “wavelet” (ondelette) to describe equations similar to those that had been
devised. Just as the STFT, the Wavelet transform is a two- around since the 1930s.
3 This criterion states that
R
R Ψσ (t) dt = 0. It is fundamental for the
dimensional representation of a one-dimensional signal. The existence of an inverse wavelet transform as well as for the applicability of
parameters are the scale and translation (analogous to fre- Parseval’s identity.
A WAVELET-BASED PITCH-SHIFTING METHOD
For signals containing only slowly varying frequency and
amplitude modulations (audio, for example) it is not necessary
to use small values of σ. In this case, κσ becomes very small
(e.g. σ > 5 ⇒ κσ < 10−5 ) and is, therefore, often neglected.
Under the restriction σ > 5, the frequency of the Morlet
wavelet is conventionally taken to be ωΨ ' σ.
C. Discrete vs. Continuous Wavelet transform
To be orthonormal, a set of wavelet functions must be
complete (i.e. there has to be enough of them), and each must
be orthogonal to the others (i.e. no overlap or projection). In
addition, each wavelet function is normalized to have unit total
energy.
For the Morlet wavelet, there are two possibilities:
1) the continuous wavelet transform: here the wavelet is
slid along the time function. Therefore, each wavelet
overlaps the ones next to it, which gives lots of redun-
dant information. This is not orthogonal.
2) the discrete wavelet transform: we skip along the time
function, only doing the wavelet transform where the
wavelet no longer overlaps the previous ones.
• For the Morlet wavelet, this is approximately orthogonal,
but not exactly. Due to the extended tails of the Gaussian,
it is not possible to construct a truly orthogonal set for
the Morlet.
• For other wavelets such as the Daubechies, it is possible
to construct an exactly orthogonal set.
So the moral is: if we want to do the continuous wavelet
transform, then we aren’t worried about orthogonality, and we
can use the Morlet. If you want to do the discrete wavelet
transform, don’t use the Morlet.
D. A wavelet approach to pitch shifting
De Gersem et al. suggest an algorithmic framework for pitch
shifting using the Complex Morlet wavelet. Firstly, we take
the CWT of the original signal to obtain a time-frequency
representation of it. The second step is to scale the frequency
axis of this “grid” by a constant factor c. Since we are keeping
the same time scale, when we convert this grid back to the time
domain, the result is a pitch-shifted version of the original. The
algorithm is as follows:
Algorithm 2 Pitch-shifting algorithm where
coefs=cwt(f,scales); ψa,b (t) = p ψ
absc=abs(coefs);
phac=angle(coefs); and ψ (t) is the “mother” wavelet.
phac_unwrap=unwrap(phac); The inverse operation is defined as:
coefs_shifted=absc.*exp(i*phac_unwrap*c);
scales_shifted=scales/c; Z ∞
1 b−t
f_shifted=icwt(coefs_shifted,scales_shifted); f (t) =
Cψ 0 a2 −∞
II. P REVIOUS W ORK
A good pitch shifting algorithm should be able to transpose
an audio signal a tone up or down without any changes in
the duration of the signal. It should also preserve the transient
3
components and formants should be transposed accordingly.
We analyze the effects of formant shifting in a later section.
As we have already stated, there are many algorithms nowa-
days, both of public domain and commercial ones, which carry
out pitch shifting with varying degrees of success. To mention
a few, the Phase vocoder is a very well-known technique which
is also very well suited for real time processing since it can
be implemented quite efficiently.
A. The Phase Vocoder
A Phase Vocoder is often implemented using Fast Fourier
Transform (FFT) analysis. The FFT of a signal will produce
its frequency spectrum, with the limitation being that the
transform does not preserve any temporal information of the
signal, i.e. we can gain information of a signal’s frequency
content, but not when in time those frequencies occurred.
To overcome this limitation, the technique of the Short Time
Fourier Transform (STFT) was developed. The STFT is a se-
ries of windowed Fourier Transforms which slightly overlap in
time. By segmenting the signal and then performing individual
FFTs on each block, some of the time information can be
preserved (with a resolution equal to the window’s length)
while still achieving the spectrum of the signal. Windowing
is needed to avoid frequency leakage because of the signal
truncation ([4], [5]).
As we have already stated, the output can be viewed as a
2D discrete grid with time and frequency as the axis. Once
the processing has been done to create the 2D analysis grid,
the grid values can be changed to produce any desired effect.
The original time domain signal can be reconstructed by a
series of IFFTs followed by an overlap and add of each
individual block. Pitch shifting in particular, involves scaling
the phase information of the individual vocoder components
while maintaining the magnitude components ([1]).
B. The Continuous Wavelet Transform (CWT)
The CWT offers just another way to generate a time-
frequency representation of a signal. The CWT of a function
f (t) is defined as
Z ∞
∗
F (a, b) = f (t) ψa,b (t) dt (3)
−∞
 
1 t−b
|a| a
da ∞
 
Z
F (a, b) ψ db (4)
a
Where
∞ 2
|ψ (ω)|
Z
Cψ = dω
−∞ |ω|
4 FILTER BANKS AND WAVELETS EEN698

One can wonder, does any function constitute a wavelet?

The answer is, unfortunately no. The selection of the wavelet
function ψ (t) is not arbitrary and must meet two main
restrictions.
1) The wavelet function must be absolutely integrable and
square integrable (ψ ∈ L1 ∩ L2 ).
Z ∞
|ψ (t)| dt < ∞
−∞

and Z ∞
2
|ψ (t)| dt < ∞
−∞

2) The wavelet must have zero mean value (Admissibility

Condition).
Z ∞
Fig. 6. The Complex Morlet wavelet
ψ (t) dt = 0
−∞

Besides these restrictions the wavelet function must be ap-

propriate for the type of application. The CWT projects the
signal onto a series of time-shifted and dilated versions of
the wavelet function. To capture a desired aspect of a signal,
the wavelet should possess similar properties. For the case of
musical signals, the energy distribution and phase behavior
are very important characteristics. Thus the Complex Morlet
wavelet has been selected by [2] as the most appropriate for
audio representations.

C. The Complex Morlet Wavelet

The Complex Morlet wavelet is defined in a similar way to
the Morlet wavelet:
1 j2πfc t − ft2 Fig. 7. The Complex Morlet wavelet frequency response (note that since
ψ (t) = √ e e b this is a complex signal, the magnitude is not symmetric and the phase is not
πfb
antisymmetric).
where fb is the bandwidth parameter and fc is the wavelet
center frequency.
For our purposes, we will take D. The Phase Vocoder and the CWT
As we have already stated, both the CWT and the STFT
fb = 2
represent a signal on a time-frequency (or translation-scale)
5
fc = grid. However, the STFT is an equispaced time-frequency
2π lattice whose time resolution is the window size and frequency
so that we are left with resolution inversely proportional to it. Since we are concerned
with music applications, we are interested not only in the
t2 steady-state evolution, but also in the transients. Some exper-
ψ (t) = c ej5t e− 2
iments have shown the fundamental importance of transients
with in audio perception. For example, we know that if we take
1 two instruments producing the same note and exchange both
c= √ notes’ onsets, there is a great deal of cross-confusion, i.e.
2π
given two instruments I1 and I2 producing notes n1 ∈ I1
In our calculations we will use an unnormalized version of and n2 ∈ I2 , people will classify ni → Ij for i = j as well as
the wavelet function, so that c = 1: for i 6= j with equal measure. Besides instruments, transients
t2
are crucial in speech for example for fricative discernment.
ψ (t) = ej5t e− 2 (5) In music applications, transients also appear as a product of
percussive instruments such as drum hits. The STFT phase
A WAVELET-BASED PITCH-SHIFTING METHOD
vocoder uses too fine a resolution in the low frequencies and
too coarse a resolution on the high frequencies. This results
in lost timing as well as the addition of beating or warbling
effects. The term smearing is used to describe the undesirable
side effects of Phase Vocoder processing. This is one of the
motivations for using the CWT.
The CWT, by definition, provides a grid that is not a discrete
one as in the STFT, but a continuous one, as the name implies.
The grid is therefore, a variable scale one. The scale is given
by the dilation of the wavelet function. The time-frequency
representation of the signal can have resolutions tailored to
the areas of interest. By using the CWT, one can create a
grid which has finer resolutions for increasing frequencies
which is more suited to music. Unlike the Phase Vocoder,
the analysis grid created by the CWT is not a time-frequency
grid in the strict sense. Since the CWT performs multiple
projections onto scaled wavelet functions, the output actually
represents the similarity (i.e. the correlation) between the input
signal and the wavelet function at each scale. A frequency is
often assigned to each scale value, so that one can achieve
an approximate representation of the frequency content. This
frequency is the dominant spectral component of the wavelet
function. Depending of the wavelet used, this frequency can
be a very good or very poor estimate of frequency content,
viz. if the wavelet has a very spread out response, then the
usefulness of this “central” frequency isn’t that good, just as
what happened in the spectrograms with the STFT. This is
a major reason why the Complex Morlet wavelet is used for
audio signals. This wavelet has a very narrowband frequency
spectrum, which offers a good representation of the frequency
content.
III. M ETHODOLOGY
5
The bottom line is, the RKHS condition means that there
exists some unique signal whose transform is a signal we mod-
ified. Put differently, if we begin with a signal, transformed
it to a different domain through a linear functional such as
the FT or the CWT, and modify this transformed signal in
some way, we don’t have any warranty that the modified
transformed signal corresponds to any signal in the original
domain! The condition than guaranties the existence of a time-
domain inverse is the RKHS condition, which is dependent on
the kind of modification we perform on the dual domain.
B. Mathematical Foundations of the Pitch Shifting Algorithm
As we have already stated, the classical approach to pitch
shifting is to first compute the STFT time-frequency represen-
tation of a signal. Afterwards, the grid spacing and coefficient
phases are scaled to create a synthesis grid. The inverse is
then computed to reconstruct the signal. It is summarized as
follows:
1) Compute the STFT representation of the signal.
2) Convert coefficients into polar form
3) Unwrap the phase and divide by the scaling factor c.
4) Reconstruct signal using new synthesis scale.
The crucial part here is the division of the phase by the scaling
factor c. Consider the following property (scaling property) of
the Fourier Transform:
1 ω
f (at) ⇐⇒ F
|a| a
We want to exploit this property in our approach to scale the
pitch of an audio signal. Any complex number z can be written
in polar notation, as z = |z| ej∠z . In particular,
F (ω) = |F (ω)| ej∠F (ω)
Therefore, F a = F ωa  exp j∠F ωa .
ω





A. Reproducing Kernel Hilbert Subspace
Recall the definition of group delay
A reproducing kernel Hilbert space is a function space in
d∠F (ω)
which pointwise evaluation is a continuous linear functional. τ (ω) = −
dω
Equivalently, they are spaces that can be defined by reproduc-
ing kernels. The subject was originally and simultaneously We then get
d∠F ωa


developed by Nachman Aronszajn (1907-1980) and Stephan ω
Bergman (1895-1987) in 1950. τ = −a = aτ 0 (ω)
a dω
Let X be an arbitrary set and H a Hilbert space of complex- where
d∠F ωa


valued functions on X. H is a reproducing kernel Hilbert
τ 0 (ω) = −
space iff the linear map f 7→ f (x) dω
from H to the complex numbers is continuous for any x in
X. By the Riesz representation theorem, this implies that for
a given x there exists an element Kx of H with the property ω    ω   Z   
ω
F =  exp j∠ − τ dω
 
that: f (x) = hKx , f i ∀f ∈ H (∗) a
F
a a
def
The function K(x, y) = Kx (y) is called a reproducing   ω   Z 
0
=  exp j∠ − a τ (ω) dω
 
kernel for the Hilbert space. In fact, K is uniquely determined F
a
by condition (*).  
1   ω 
 Z
For example, when X is finite and H consists of all f (at) ⇐⇒  exp j∠ − a τ 0 (ω) dω
|a| a
F
complex-valued functions on X, then an element of H can
be represented as an array of complex numbers. If the usual 1
f (at) ⇐⇒ |F (Ω)| exp (ja∠F (Ω))
inner product is used, then Kx is the function whose value is |a|
1 at x and 0 everywhere else. The magnitude is scaled by a constant factor, so we won’t
R In other contexts, (*) amounts to saying f (x) = take this into account since this corresponds only to a dif-
X
K(x, y)f (y) dy. ference in volume. The phase portion of the FT is related to
6 FILTER BANKS AND WAVELETS EEN698

the group delay of the signal’s frequency content. To maintain
the same time duration when changing the frequencies, the
slope of the phase needs to be scaled to either slow down or
speed up the group delay to compensate for the time scaling
side effect. The hardest part in terms of computation is the
transformation of the input signal into the 2D time-frequency
representation. The quality of this transform will directly affect
the output pitch shift. It was stated earlier how the CWT results
in a similar 2D signal representation. More care has to be
taken when modifying coefficients of the CWT however, as
the reconstruction conditions are much more complex than
that of the STFT Phase Vocoder. Any modifications in the dual
domain must be made while satisfying the Reproducing Kernel
Hilbert Subspace (RKHS) property. Changing the phase com-
ponents of the Complex Morlet CWT maintains this property.
Thus the algorithm used in the Phase Vocoder implementation
can be used with the Complex Morlet CWT.
C. The Pitch Shifting Algorithm
The pseudo code for the entire CWT based pitch shift
algorithm is as follows:
function pitch_shift(in c: real, in x:
real[], out y: real[])
coefficients=cwt(x,scales);
magnitude=abs(coefficients);
phase=unwrap(angle(coefficients));
coefs_shifted=magnitude*exp(j*phase*c);
scales_shifted=scales/c;
y = icwt(coefs_shifted,scales_shifted);
where ⊗ denotes convolution.
The final result of Equation 6 can be quickly found in the
frequency domain by taking the inverse Fourier transform of:
F̃ (an , b) = |a|f˜ (ω) ψ̃ (aω)
p
E. Computation of the Inverse CWT
The inverse CWT is defined by equation 4. We see that it in-
volves a double integral, which can be performed numerically
through some integration technique such as the trapezoidal
rule, quadrature integration, etc. The trapezoidal rule approx-
imates the area under a curve by fitting a trapezoid between
every adjacent sample point. There is also an approximate
reconstruction formula given by [2] which simplifies to a
single integration over the dilation scales.
Z ∞
1
f (t) ≈ Kψ F (a, t) 2/3 da (7)
−∞ a
IV. R ESULTS
A. Algorithm Testing
The algorithm was implemented in MATLAB, and several
tests were carried out, which we describe below.
Example 1: We pitch-shift a middle-C tone (256 Hz) by
a factor of 2 and 12 (an octave up and down respectively).
The spectrogram of the original and the two shifted signals is
shown in figures 8, 9 and 10 respectively, and the FT of the
relevant frequency region is shown in figure 11.

The main component of the algorithm is the development

of the cwt() and icwt() functions for the Complex Morlet
Wavelet. To implement these continuous time operations on
a computer, some level of discretization and approximation
must be made. The CWT can be well implemented using a
series of convolutions carried out in the frequency domain.

D. Computation of the CWT

Recall the definition of the CWT from Equation 3 and the
Complex Morlet definition from equation 5, by substitution of
the Complex Morlet wavelet into the CWT formula:
Z t − b  t − b 2 !∗
p1 1
∞
F (a, b) = f (t) exp j5 − dt
|a| a 2 a
Z t − b  t − b 2 !
−∞

p1 1
∞
= f (t) exp −j5 − dt
|a| a 2 a Fig. 8. Spectrogram of a middle-C tone
Z  b − t  1  b − t 2 !
−∞

p1 ∞
= f (t) exp j5 − dt We see that the algorithm performs well: the frequencies
|a| a 2 a
Z
−∞
b − t are shifted correctly up and down, and the duration of the clip
p1
∞
= f (t) ψ dt is preserved.
|a| −∞ a

We rewrite this as Example 2: In the next test case, we wanted to see how the
  algorithm performed as to preserving the temporal character-
1 b istics of a signal. To do this, a middle C tone burst was used
F (a, b) = p f (b) ⊗ ψ (6)
|a| a as the test signal, and shifted an octave higher.
A WAVELET-BASED PITCH-SHIFTING METHOD
Fig. 9. Spectrogram of a higher C tone
Fig. 10. Spectrogram of a lower C tone
The signal is shown in figures 12, 13 and 14 as well as the
spectrogram of the original and shifted signals.
We notice there is some smearing of the frequency content,
but considering the abrupt frequency changes, the smearing is
not too large and there is negligible distortion upon playback.
Example 3: Harmonic relations. In this test we verify
that a shifted C major scale preserves the harmonic relations
between successive notes. Note that if we were to shift
frequencies (as opposed to pitch), we would be shifting the
frequency content by an additive (or subtractive, for that mat-
ter) value. Instead, we are changing the frequency content by
a multiplicative factor, thus changing the pitch by a constant,
additive term.
The spectrogram of the original and shifted signals are
shown in figures 15 and 16 respectively.
By listening to the resultant audio files and from inspection of
the spectrograms, we see that the spacing (exponential spacing,
that is) between tones is definitely maintained. However, some
attenuation can be noticed near the end of the signal.
7
Fig. 11. Fourier Transform of the three C tones
Fig. 12. C tone Burst
These results show that our algorithm certainly works. A
final test was performed involving speech and music signals.
For both of these classes of signals, the algorithm performs
fairly well. In music signals, there is a very small distortion
artifact in between notes, which corresponds to discontinuity
in the time-frequency grid. Voice signals are very susceptible
to distortion when pitch shifting. Smearing can result in
a loss of speech coherency, which is almost unnoticeable
for instrumentals but is very noticeable for voice. Despite
this, even in the extreme cases of the 12 semitone shifts,
much of what the speaker is saying is still clear. The major
distortion that can be noticed is the timbre changes as pitch
is altered. The voice of the singer transforms from beast-like
to chipmunk-like as the pitch is changed. Unfortunately this
is one thing the algorithm does not account for.
1) Formants: The algorithm shifts each and every fre-
quency. This can cause the unique characteristics of a signal
to change, namely the formants. These are created by a
particular person’s vocal tract and are what makes a certain
voice distinguishable from someone else’s. They correspond
8 FILTER BANKS AND WAVELETS EEN698
Fig. 13. C tone Burst spectrogram
Fig. 14. Shifted C tone Burst spectrogram
to the resonant frequencies associated with the normal modes
inside a person’s vocal tract, and are of course different for
every possible vowel/consonant.
To maintain these identifiable characteristics while pitch
shifting, certain frequencies should not be emphasized as much
as others. This is a difficult task to do as the CWT method
treats all frequencies equally. The term formant preservation
is used to describe pitch shifting while maintain the character
or timbre of a signal but it is a complex topic involving
psychoacoustics.
V. C ONCLUSIONS
There are several possible sources of distortion for the
pitch-shifting algorithm. First, we are computing the CWT
only in a discrete lattice (since that’s all we can represent
in a computer). A finer sampling lattice consisting of more
projections per octave may provide a more accurate time-
frequency representations of the signal. Another major source
of errors may be in the ICWT calculations. The approximate
Fig. 15. C major scale spectrogram
Fig. 16. Shifted C major scale spectrogram
formula giving in Equation 7 was used instead of the tra-
ditional, double-integral inverse formula. This approximation
introduces error; in addition, the integral was computed using
a very crude trapezoidal approximation, further contributing
to errors in the result.
The CWT has been shown to be an alternative to the Phase
Vocoder in representing a signal in a time-frequency format.
Our method still has some areas that need improvement.
Mainly, the ICWT algorithm has to be refined, and the underly-
ing numerical approximation technique should be investigated.
Further work on this pitch shifter may also include research
into the use of alternate wavelets. Work can also be done on
finding a method of preserving vocal formants, thus avoiding
the chipmunk effect.
A WAVELET-BASED PITCH-SHIFTING METHOD 9

A PPENDIX We now prove the following

The Heisenberg Uncertainty principle for L and L 1 2 Theorem 1: Heisenberg inequality for L2 signals
signals Proof: Equation 8 implies that
2 Z
a) L1 (R) functions: Assume f (t) and its Fourier trans-  df (t) 2
Z Z  
 t f (t) df (t) dt ≤ 2 2
 
form F (ω) are absolutely
integrable. Let t and ω be any t f (t) dt    dt

R dt 
R R dt 
two values such that f t 6= 0 and F (ω) 6= 0. The
corresponding effective duration ∆t and the corresponding Let
df (t)
Z
effective bandwidth ∆ω are the satisfying
A = t f (t) dt
Z ∞

 R dt
|f (t)| dt = f t  ∆t
 
Z d f (t)2
−∞ 2
= t dt
and Z ∞ R dt
∞
2 2
|F (ω)| dω = |F (ω)| ∆ω f (t)  f (t)
Z
−∞ = t − dt
2  2

R
−∞
with R∞
t |f (t)| dt 2
By hypothesis, |t| f (t) → 0 ⇒ A = − E2 .
t = R−∞
∞ d
−∞ |f (t)| dt We now recall that dt f (t) ⇐⇒ jωF (ω), we then have
using Parseval’s identity that
and R∞
ω |F (ω)| dω Z 
 df (t) 2

 dt = 1
Z
ω = R−∞
∞  ω 2 |F (ω)| dω
−∞
|F (ω)| dω R
 dt  2π R
Z ∞
Therefore,
 E 2
2
|F (ω)| ≤ |f (t)| dt
  Z
 t f (t) df (t) dt
 
−∞ −  =

  2  dt 
≤ f t  ∆t ZR
1
Z
Z ∞ 2 2

 1 ≤ t2 f (t) dt · ω 2 |F (ω)| dω
f t  ≤ |F (ω)| dω R 2π R
2π −∞
2 1 2
1 = E (∆t) · E (∆ω)
= |F (ω)| ∆ω 2π
2π E2
2 2
1 
 
 ≤ E 2 (∆t) (∆ω)
∆t∆ω |F (ω)| f t  ≥ f t  |F (ω)| 4
2π 1
1 ∆t∆ω ≥
∆t∆ω ≥ 2
2π
QED.
b) L2 (R) functions: Define
Z ∞ Z ∞
2 1 2
E= |f (t)| dt = |F (ω)| dω
−∞ 2π −∞

R∞ 2
2 −∞
t2 |f (t)| dt
(∆t) =
E
R∞ 2
2 −∞
ω 2 |F (ω)| dω
(∆ω) =
2πE
2
Claim 1: If limt→±∞ t |f (t)| = 0, then ∆t∆ω ≥ 21 .
We recall the Cauchy-Schwartz Inequality for the L2 Hilbert
space. For any L2 functions z (x) and w (x) defined in the
interval [a, b],
Z 2 Z
 b  b Z b
2 2
z (x) w (x) dx ≤ |z (x)| dx |w (x)| dx (8)
 

 a  a a

which in vector notation becomes

2
hZ, W i ≤ hZ, Zi hW, W i
or equivalently
hZ, W i ≤ kZk kW k
10 FILTER BANKS AND WAVELETS EEN698

Alexander Sklar was born in Tel-Aviv, Israel on

January 19th , 1981. He graduated from both the
Electrical Engineering and Computer Science pro-
grams at the Universidad de la República in Monte-
video, Uruguay in May 2005. His interests include
software design and engineering, music and DSP,
robotics and control.

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