fisica

© All Rights Reserved

Als PDF, TXT **herunterladen** oder online auf Scribd lesen

3 Aufrufe

fisica

© All Rights Reserved

Als PDF, TXT **herunterladen** oder online auf Scribd lesen

- Computerised Speech Processing in Hearing Aids Using FPGA Architecture
- Sampling Theorem
- Acoustic Emission Sensing in Wireless Sensor Networks
- Fundamentals of Multimedia Slide
- Lecture Fourier Transform
- All
- Digital Communications_Viva Voce Questions
- Dsp Labwork
- TechRef Trigger
- Eps10 201 Digital Prps-m3
- Digital Image Processing_Lecture Notes
- Toolkit Manual
- TechRef_Fourier Source PS15
- Pid Algorithm
- Analog vs Digital - Difference and Comparison _ Diffen
- Santhosh Wave Let Ppt
- MUESTREO Y RECONSTRUCCIÓN DE SEÑALES.pdf
- Sampling
- subidiealF
- Adaptive Wiener Filter

Sie sind auf Seite 1von 16

Anna Maeser

April 21, 2008

Abstract

Sampling Theory is the process of converting a function of continuous

space or time into a discrete sequence of numbers. This paper summa-

rizes the history behind sampling theory and shows how it can be used

along with the Discrete Fourier Transform and the Short-Time Fourier

Transform in applications. These transforms are used in various elds

such as physics, engineering, and mathematics. Applications to music are

discussed in terms of ltering, compression, and denoising. MATLAB is

used for the analysis.

1 Introduction

This paper considers the sampling theorem and applications to ltering,

compression, and denoising. E.T. Whittaker, Harold Nyquist, A. Ko-

telnikov, and Claude E. Shannon are the most prominent contributors to

the development of sampling. As a result, the sampling theorem is often

referred to as the Shannon sampling theorem, the Whittaker-Shannon

sampling theorem, the Whittaker-Shannon-Kotelnikov sampling theo-

rem, and the Nyquist-Shannon sampling theorem [5]. Whittaker rst

proved the sampling theorem from the point of view of interpolation the-

ory in 1915 through his study of Cardinal Functions. His second son, J.M.

Whittaker, further developed this idea and was the mathematician to coin

the term Cardinal Series in 1929 [6]. In 1928, Harry Nyquist published

a paper titled Certain Topics in Telegraph Transmission Theory reveal-

ing an important discovery known as the Nyquist rate that species the

lowest rate of discrete samples required to reproduce an original continu-

ous signal [4].

All of these advances laid the foundation for Claude Shannons dis-

covery of the sampling theorem in 1948 and its correlation to Fourier

transforms (5). Shannon is often referred to as the father of information

theory. An important application of Shannon sampling is that a band-

limited signal can be replaced by a discrete sequence of samples from the

signal without losing any information [4]. It is remarkable that a certain

function can be completely reconstructed from a discrete set of samples

1

taken at equal intervals. This is important in practice because it is often

necessary to convert analog models into digital to make use of modern

technology. In real-world applications, functions are usually given as se-

quences of values because they are measured over discrete sets of values.

Discrete Fourier transforms (7) and short-time Fourier transforms (8) will

be used to analyze these discrete functions throughout this paper [1]. This

includes the derivation of the sampling theorem with the use of Fourier

transforms and inverse Fourier transforms.

Joseph Fourier was a French mathematician who claimed that any

function dened on a nite interval has a Fourier series expansion (1). He

expanded on the works of Swiss mathematicians Daniel Bernoulli and

Leonhard Euler in the mid 1700s that was met with extreme doubt.

Fourier presented his theory to the French Academy in 1807 and was met

with similar doubts. He presented The Propogation of Heat in Solids

and the judges, mainly Lagrange, criticized his use of the Fourier series

as an expression of the initial temperature distribution. After his work

was rejected at this time, Fourier resubmitted his revised theory to the

Academy four years later and was met with great praise and acceptance.

He was then awarded the Academys Grand Prize in Mathematics [1]. His

outstanding accomplishments continue to accelerate advances in applied

mathematics. Several preliminary denitions having to do with the works

of Joseph Fourier must be mentioned before sampling theory can be fully

understood.

Fourier series deal with expanding a periodic function in terms of the

specic set of functions 1, cos x, cos 2x, cos 3x, ..., sin x, sin2x, sin3x, ...

Denition 1. A Fourier Series expansion of a function f can be ex-

pressed

f(x) = a0 +

n=1

(an cos nx + bn sin nx), (1)

where the coecients a0, an, and bn are called Fourier coecients of f

and are given by the following formulae:

a0 =

1

2

f(x) dx,

an =

1

bn =

1

The functions cos x and sinx can be written using complex exponen-

tials as

cos x =

e

ix

+ e

ix

2

, sin x =

e

ix

e

ix

2i

,

which allows us to dene the Fourier series in the complex form.

2

Denition 2. Let f be a 2p-periodic piecewise smooth function. The

complex form of the Fourier series is

f(x) =

n=

cne

i

n

p

t

, (3)

where the Fourier coecients cn are given by

cn =

1

2p

p

p

f(t)e

i

n

p

t

dt, (n = 0, 1, 2, ...). (4)

The continuous Fourier Transform is denoted

f() and is a general-

ization of the complex Fourier series. It denes a relationship between a

signal in the time domain and its representation in the frequency domain.

Denition 3. The Fourier Transform of a function f is given by

f() =

1

f(x)e

ix

dx, (< < ). (5)

Denition 4. The Inverse Fourier Transform is dened

f(x) =

1

f()e

ix

d, (< x < ). (6)

As described previously, it is often necessary to sample a continuous

function to obtain a discrete representation. The next important de-

nition, the Discrete Fourier Transform, denoted X(k), can be viewed as

sampling from the continuous Fourier transform.

Denition 5. Given an N-sequence x=((x(0), x(1),...,x(N-1)), the N-

point Discrete Fourier Transform of x is dened by

X(k) =

1

N

N1

j=0

x(j)e

2ijk/N

, k = 0, 1, ..., N 1. (7)

The Fourier transform provides the intensity of a frequency; however,

in order to discover how long a specic frequency will occur, the Short-

Time Fourier Transform, also known as the Gabor transform, is necessary.

The short-time Fourier transform is a technique in which Fourier trans-

forms are taken of short consecutive segments of a signal. This transform

provides a clear representation of the signal and a frequency image of the

signal locally in time [2].

Denition 6. The Short-time Fourier Transform is dened as

Vgf(x, ) =

1

f(t)g(t x)e

it

dt, (8)

where t represents time and represents the bandwidth of the signal. g is

the lter window which is commonly represented by a Gaussian, f(x) =

e

x

2

, or an indicator function, f(x) = 1 if |x| < a and 0 otherwise. The

bar represents the complex conjugate.

3

Grasping these preliminary denitions and their purpose is important

for the understanding of sampling theory. This paper will introduce the

theory behind sampling in terms of band-limited functions, time-limited

functions, and the Fourier transform as well as provide an important ap-

plication to the theorem. Sampling plays a crucial role in communications,

signal processing, transmission, and information theory. This paper will

discuss sampling analysis on continuous sound les by transforming a con-

tinuous signal into a discrete sequence in which MATLAB can be utilized.

The purpose behind ltering will be discussed in terms of current appli-

cations with visual representations.

2 Theory Behind the Sampling Theorem

2.1 Band-Limited Functions and Band Width

Sampling analog signals is a prerequisite to digital signal processing. In

analog transmission, such as a telephone system, it is required that the

analog speech signal be sampled before becoming a part of the digital

telephone system. A digital signal processing device that converts a con-

tinuous signal to a discrete signal, known as a sampler, creates a sequence

of impulses by keeping track of the level of the signal over time at equal

intervals. These impulses are then coded as numbers and can be processed

digitally with the use of an analog-to-digital converter (ADC). This might

make it appear that information from the original signal is lost due to the

sampling. Although this is valid to some extent, it can be seen that the

lost information is unimportant as long as the sampling rate is high enough

[3].

This brings us to the idea of a band-limited functions and band width.

Denition 7. A function f(x) is band-limited if its Fourier transform

(5) vanishes outside a nite interval. Then there exists a positive number

W such that

f() = 0 for all || > W. The number W is called the band

width of f.

The majority of functions are not considered band-limited; however, most

functions can be approximated by band-limited functions [1].

The denition of band-limited functions leads to an important result.

Basically, every band-limited function with band width W can be written

as an innite sum. The coecients in this sum specically represent

function values at a discrete set of equally spaced sample points. The

sampling theorem for band-limited functions results.

Theorem 1. The Sampling Theorem for Band-Limited Functions

Suppose that f is band-limited with band width W. Then for all x,

f(x) =

n=

f(

n

W

)

sin(Wx n)

(Wx n)

, n = 0, 1, 2, ... (9)

We can see from this theorem that f is completely determined by its

sample values f(

n

W

). Harold Nyquist recognized that the larger the band

width, the more sample points per unit length are required for (9) to

4

hold. As a result, the smallest number of sample points per unit length

are required for (9) to be true is known as the Nyquist Sampling Rate.

This rate corresponds to the smallest W of f and is twice the bandwidth

of the signal or 2W[1].

2.2 Time-Limited Functions

A time-limited function is one in which the function itself vanishes outside

a nite time interval. This diers from the band-limited function where

the Fourier transform of the function vanishes outside a nite frequency

interval. The function and the Fourier transform in (9) can be exchanged

to obtain the sampling theorem for time-limited functions.

Theorem 2. The Sampling Theorem for Time-Limited Functions

Suppose t represents time and f(t) = 0 for all |t| > T. Then for all ,

f() =

n=

f(

n

T

)

sin(T n)

(T n)

, n = 0, 1, 2, ... (10)

The result is similar to the sampling theorem for band-limited func-

tions. In this case, the Fourier transform,

f of the function is completely

constructed by the coecients. In other words, sampling at the points

n

T

formulates

f [1].

2.3 Proof of the Sampling Theorem

In this section, the goal is to prove (9). That is, we will prove

f(x) =

n=

f(

n

W

)

sin(Wx n)

Wx n

for a bandlimited functionf(x).

Proof. By denition, we know that

f vanishes outside [W, W] [1]. By

the Inverse Fourier Transform (6), we have

f(x) =

1

f()e

ix

d

=

1

W

W

f()e

ix

d. (11)

On the -axis,

f can be extended periodically with period 2W due to

the fact that

f vanishes outside [W, W]. Using the complex form of the

Fourier series (3),

f can be represented

f() =

n=

cne

i

n

W

(|| < W) (12)

=

n=

cne

i

n

W

.

Using the complex Fourier coecients (4), we have

5

cn =

1

2W

W

W

f()e

i

n

W

d. (13)

Noticing the relationship between cn and f(x) in (11) and (13), we

can simplify and get

cn =

2

2W

f(

n

W

). (14)

One more integral must be solved before continuing the proof. The

solution (15) uses the identity e

ix

= cos x + i sinx.

1

2W

W

W

e

i

n

W

e

ix

d =

1

2W

W

W

e

i(Wxn)/W

d

=

1

2W

W

W

cos(

Wx n

W

) d

=

sin(Wx n)

Wx n

. (15)

We can now complete the proof using f(x) in Equation (11),

f() in

Equation (12), and cn in Equation (14).

f(x) =

1

W

W

f()

n=

cne

i

n

W

e

ix

d

=

1

W

W

f()

n=

c

n

2

2W

f(

n

W

) e

i

n

W

e

ix

d.

Interchanging the order of the integral and sum, we obtain

f(x) =

n=

f(

n

W

)

1

2W

W

W

e

i

n

W

e

ix

d

=

n=

f(

n

W

)

sin(Wx n)

Wx n

using (15).

3 Applications with MATLAB

In practice, it is usually necessary to work with both time- and band-

limited functions together. In reality, it is not possible to use the Fourier

transform to simultaneously obtain both exceptional time resolution and

frequency resolution. This is based on what is known as the Heisenberg

6

Uncertainty Principle. In this section, we will illustrate two ways of sam-

pling using MATLAB. The rst subsection will provide an example of

sampling a sound le using the Fourier transform. The second subsection

will provide a similar example using the short-time Fourier transform.

Both essentially consist of distorting a signal by adding a high frequency

tone to a sound clip and using transforms to lter out the high frequency

and reproduce the original signal.

3.1 Sampling with the Discrete Fourier Trans-

form

The rst application to sampling makes use of the discrete Fourier trans-

form (7). With the aid of MATLAB (code included in the Appendix), a

sound clip is read into the program and a high frequency tone is added to

half the signal in the form of a sine wave to create a constant pitch. The

formula used to create the sound is x = I sin(2n) where I is the intensity

of the tone. In this application, I is taken to be 0.2 and the length of n

is 67,337. An image of the original signal can be seen in Figure 1. The

signal with the added sine wave can be seen in Figure 2. The discrete

Fourier transform is taken of the signal and plotted in Figure 3.

Figure 1: This is what the original signal looks like.

Figure 2: This is the signal in Figure 1 with a constant tone added to half of it with

the formula x = .2 sin(2n).

7

Figure 3: This is the plot of the Fourier transform of the added noise. The spike

in the graph indicates the frequency component corresponding to the constant noise.

The modulus is taken to avoid complex numbers.

The next step is to get rid of the high frequency in the signal in order

to eliminate the constant tone. First, every frequency component from

the spike forward is set to zero as seen in Figure 4. Then, with the use

of the inverse Fourier transform, it is possible to lter the tone out of the

sound clip and reproduce the original signal (Figure 5).

Figure 4: The plot of the Fourier transform is reproduced after setting the freqency

components from the spike forward to zero.

8

Figure 5: This plot shows the original signal surrounded by the high frequency tone

that was ltered out. The rst half of the plot shows a much greater dierence between

the original le and the tone while the two are much closer in the second half. This is

a result of adding the tone to only the rst half.

3.2 Sampling with the Short-Time Fourier Trans-

form

One of the shortcomings of the discrete Fourier transform is that it does

not give information as to the time a frequency component occurs. This

is where the short-time Fourier transform comes into play. The process

starts out the same by reading a wav le into MATLAB. It is then neces-

sary to create a lter. Filter choices vary from application to application

and depend on the frequency domain required for analysis. Both the

Blackman and block lters will be used along with the short-time Fourier

transform.

3.2.1 Blackman Filter

Common lters include lowpass, highpass, bandpass, bandstop, and all-

pass lters. In this case of denoising, a lowpass lter known as the Black-

man lter is used. Lowpass lters reduce the amplitude of frequencies

above a certain cuto frequency and allows those below the cuto to

pass through. Every lowpass lter has a stopband associated with it. A

stopband is a band of frequencies in which the lter does not let signals

through. Any frequency between the upper and lower limits are not trans-

mitted. The minimum stopband associated with the Blackman lter is 74

dB.

The Blackman lter is created using the expression

w(n) = a0 a1 cos(

2n

N 1

) + a2 cos(

4n

N 1

)

9

The coecients are given by the following formulae:

a0 =

1

2

(16)

a1 =

1

2

a2 =

2

.

is typically taken to be 0.16 in the Blackman lter, which results in

w(n) = .42 .5 cos(

2n

N 1

) + .08 cos(

4n

N 1

), (17)

where N is the length of the lter and 0 n N 1. After adding

noise to a sound le, the Blackman lter is applied, resulting in a two-

dimensional graph that provides the time that the noisy frequency occurs.

It is then possible to eliminate the noise associated with the sound le and

reproduce the original le. The short-time Fourier transform is often more

benecial than the discrete Fourier transform in terms of frequency and

time detail. As previously mentioned, it is unfortunately not possible

to obtain exceptional time and frequency detail at the same time. The

extent to which either is obtained is dependent upon the ltering length,

or N in (17). N can either be considered thick or thin. The rst set

of gures depicts when N is set to a length of 256 samples and is thick.

This results in more detail in terms of frequency (Figure 6). On the other

hand, Figure 7 shows when N is set to a thin length of 32, exhibiting

more extreme time detail.

10

Figure 6: The top image is a picture of the Blackman lter of length 256. The

Gaussian form mentioned in the denition of the short-time Fourier transform (8) is

reiterated here. The second image shows excellent detail in terms of frequency on

the vertical axis. Time is on the horizontal axis. The coloring in the plot represents

intensity. The brighter the color, the higher the intensity. An image such as this is

helpful to audio engineers when deciding the location of frequencies of specic noises.

Figure 7: The image shows excellent time detail on the horizontal axis. The frequency

in this image remains the same as in Figure 6; however, the frequency detail is miniscule

in comparison.

11

3.2.2 Block Filter

Another type of lter called a block lter can be used to analyze the

data. It is obvious by looking at Figure 8 that the Blackman lter is more

appropriate for analysis. The reason for this is that it is benecial for

the lter and the Fourier transform of the lter to decay quickly in both

time and frequency. The Blackman lter meets this criterion. The Fourier

transform of the block function is the sinc function, which does not decay

quickly. Therefore, the resulting plot does not provide time or frequency

information as clearly as the Blackman lter (Figure 8).

Figure 8: The top image shows the block lter of length 32. The bottom image can

be compared with Figure 7 and it is obvious that the time axis is not as clear. The

block lter is illustrating poor time-frequency localization as seen by the artifacts and

is a direct result of the slow decay of its Fourier transform.

12

4 Aliasing

When discussing sampling, it is essential to understand the problem of

aliasing. This term is simply known as undersampling. Noise is nat-

urally going to contaminate any recorded physical signal. In addition to

the actual signal f, the recorded signal will primarily look like h = f +r,

where r contains relatively high frequencies but has relatively small ampli-

tude. As a result, the spectrum of f will be narrower than the spectrum of

h. This means that when one bases a sampling rate on what they assume

the band width of f to be, the sampling rate will have a good chance

of being too low. The calculated spectrum will not be useful when this

occurs. Often times, a frequency will masquerade as another resulting in

an undesirable sound. In order to avoid this problem, the signal must be

ltered with a lowpass lter before it is sampled. This is advantageous be-

cause the relevant sampling rate will be obtained and high-frequency noise

is eliminated. Since sampling of a signal is done discretely, its spectrum

becomes periodic. The phenomenon of aliasing occurs when the periods

overlap as seen in Figure 9. The Nyquist sampling rate mentioned in sec-

tion 2.1 comes into play here as it represents the minimun sampling rate

resulting in no aliasing. It can be seen in Figure 10, by increasing the sam-

pling rate, the components of the spectrum are separated and problems

associated with undersampling are avoided [3], [5].

Figure 9: (a) shows a signal with its Fourier transform in (b). In (c), the signal is

sampled and the transform becomes periodic and overlapping (d). This overlap is due

to undersampling, resulting in aliasing.

13

Figure 10: A high sampling rate results in separate, non-overlapping components of

the spectrum.

5 Conclusion

Many aspects of modern day applied mathematics are a result of the con-

tributions of Claude Shannon and Joseph Fourier. The use of the short-

time Fourier transform is common in many scientic elds. The dierence

between the discrete Fourier transform and the short-time Fourier trans-

form lies in the amount of detail provided by each. The discrete Fourier

transform gives information about frequencies and their intensities, while

the short-time Fourier transform illustrates when a certain frequency ac-

tually occurs. Unfortunately, it is not possible to obtain extreme time

and frequency detail at the same time. It is possible to obtain one or the

other depending on the ltering length used in analysis. The longer the

lter, the more detail provided in terms of frequency. The shorter the

lter, the better the time detail. Through analysis with the Blackman

and the block lter, it is obvious that a lter with fast decay, such as

the Blackman, results in a more useful outcome. It is important to avoid

the problem of undersampling in order to discretely sample a continuous

signal and accurately obtain a desirable approximation. Signal process-

ing and applications to electrical engineering are among the many elds

making use of these ideas and the observations and discoveries of Shannon

and Fourier.

References

[1] N.H. Asmar. Partial Dierential Equations with Fourier Series and

Boundary Value Problems. (Pearson Education, Inc.), 398:546-571,

2005.

[2] J.S. Byrnes and J.L. Byrnes. Recent Advances in Fourier Analysis and

Its Applications. (Kluwer Academic Publishers), 436:441-442, 1990.

[3] C. Gasquet and P. Witomski. Fourier Analysis and Applications: Fil-

tering, Numerical Computation, Wavelets. (Springer-Verlag New York

Inc.), 343-360, 1999.

[4] A.J. Jerri. The Shannon Sampling Theorem-Its Various Extensions

and Applications: A tutorial Review. (The Institute of Electrical and

Electronics Engineers, Inc.), 1565-1567, 1977.

14

[5] R.J. Marks II. Introduction to Shannon Sampling and Interpolation

Theory. (Springer-Verlag New York Inc.), 1-5:37, 1991.

[6] A.I. Zayed. Advances in Shannons Sampling Theory. (CRC Press,

Inc.), 1-5:19, 1993.

Appendices

MATLAB code from Section 3.1

y = wavread(boxachoc.wav); length(y);

L=2^16;

j=zeros(1,(L/2+1000));

n=[0:(L/2+1000),j];

x=.2*sin(2*n)

x=x+y(1:2000+L+1,1);

L=length(x)-1;

wavplay(x,11000)

zp=zeros(L,1);

xp=[x(1:L/2); zp; x(L/2+1:L)];

N=length(xp);

X=fft(xp)/sqrt(N);

colormap(hot)

figure(1);

plot(x);

figure(2);

plot(abs(X(1:L)));

X(30000:N)=0;

X(N-31000:N)=0;

figure(3);

plot(abs(X(1:L)));

XX=sqrt(N)*(ifft(X));

xlabel(Frequency)

ylabel(Intensity)

title(Fourier Transform without High Frequency Tone)

filtered=[XX(1:L/2);XX(3*L/2:2*L)];

filtered=real(filtered);

wavplay(filtered,11000)

figure(4);

plot(1:L+1,x,1:L+1,filtered,g)

MATLAB code from 3.2

N =256;

M=256;

w = .42 - .5*cos(2*pi*(0:M-1)/(M-1)) ...

+ .08*cos(4*pi*(0:M-1)/(M-1)); %blackman window i.e. tight frame

%w=ones(1,N); %To use block filter

w=w;

15

y = wavread(boxachoc.wav);

L=2^16;

j=zeros(1,(L/2)+1000); %adding a tone to half the signal

n=[0:(L/2)+1000,j];

x=.2*sin(2*n);

x=x+y(1:2000+L+1,1);

wavplay(x,11000)

nframes=67000;

Xtwz = zeros(N,nframes); % pre-allocate STFT output array

M = length(w); % M = window length, N = FFT length

zp =zeros(N-M,1) ; % zero padding (to be inserted)

R=1; % padding

M1=M/2;

xoff = 0;

% current offset in input signal x

for m=1:nframes

xt = x(xoff+1:xoff+M);

xtw = w .* xt; % apply window to current frame

xtwz = [xtw(M1+1:M); zp; xtw(1:M1)]; % windowed, z-padded

Xtwz(:,m) = fft(xtwz); % STFT for frame m

xoff = xoff +R; % advance in-pointer by hop-size R

end;

colormap(hot)

imagesc(abs(Xtwz));

plot(w);

figure(2)

colormap(hot)

subplot(211);plot(w)

subplot(212);

xlabel(Time)

ylabel(Frequency)

imagesc(abs(Xtwz(N/2:N,:)));

Xtwz(N/8:7*N/8,:)=0;

xlabel(Time)

ylabel(Frequency)

figure(3)

imagesc(abs(Xtwz(N/2:N,:)));

Xtwz(N/8:7*N/8,:)=0; % filter out tone

figure(4)

imagesc(abs(Xtwz(N/2:N,:)));

X=real(sum(Xtwz))/M;

figure(5)

plot(X);

wavplay(X,11000)

16

- Computerised Speech Processing in Hearing Aids Using FPGA ArchitectureHochgeladen vonEditor IJACSA
- Sampling TheoremHochgeladen vonSyamantak Kshirsagar
- Acoustic Emission Sensing in Wireless Sensor NetworksHochgeladen vonakozy
- Fundamentals of Multimedia SlideHochgeladen vonBradford Gomez
- Lecture Fourier TransformHochgeladen vonashmitashrivas
- AllHochgeladen vonpalavicino
- Digital Communications_Viva Voce QuestionsHochgeladen vonAllanki Sanyasi Rao
- Dsp LabworkHochgeladen vonsachinshetty001
- TechRef TriggerHochgeladen vonVladimirCoello
- Eps10 201 Digital Prps-m3Hochgeladen vonChandra Bose Kn
- Digital Image Processing_Lecture NotesHochgeladen vonJerrin Thomas Panachakel
- Toolkit ManualHochgeladen vonAura Auro
- TechRef_Fourier Source PS15Hochgeladen vonMartin Chris Svendsen
- Pid AlgorithmHochgeladen vonArjun Raj
- Analog vs Digital - Difference and Comparison _ DiffenHochgeladen vonCharles Downs
- Santhosh Wave Let PptHochgeladen vonSanthosh Guduru
- MUESTREO Y RECONSTRUCCIÓN DE SEÑALES.pdfHochgeladen vonDaniel Servin de la Mora
- SamplingHochgeladen vonarsh
- subidiealFHochgeladen vonndokuch
- Adaptive Wiener FilterHochgeladen vonbinukiruba
- Explanation About the Principal of Working, Construction & the Circuitry of Cathode Ray OscilloscopeHochgeladen vonRahul Gupta
- mobisys09_soundsense.pdfHochgeladen vonMayank Pawar
- PixelatorHochgeladen vonDavid Esteves Ruiz
- Signal Representation & Analysis IntroductionHochgeladen vongaurav_juneja_4
- Types of Data AcquisitionHochgeladen vonFranch Maverick Arellano Lorilla
- MRSP_Part-1.pdfHochgeladen vonanon_326727214
- mce441_8nHochgeladen vonmasinac91
- gsm based temperature monitering and controlHochgeladen vonHirankur Khillare
- 1984-09_HP Journal PapersHochgeladen vonElizabeth Williams
- cosine_filter_theory.pdfHochgeladen vonDung Le

- 2015-Vocal Indicators of Emotional StressHochgeladen vonhord72
- art%3A10.1155%2F2009%2F898576Hochgeladen vonhord72
- Prosody Phonology and PhoneticsHochgeladen vonhord72
- 2012-11-01：【技術專題】Digital Voltage Control of Boost CRM PFC Converters.pdfHochgeladen vonhord72
- bme221labbookHochgeladen vonhord72
- 2007-02-15：【技術專題】Introduction to PID ControlHochgeladen vondaswk
- 2009 05 29：【技術專題】2nd Order SystemHochgeladen vonhord72
- WP-0005_Rev_AHochgeladen vonhord72
- 2008-Feature Extraction of Speech Signals in Emotion IdentificationHochgeladen vonhord72
- 2016-Assessment on Speech Emotion Recognition for ASDHochgeladen vonhord72
- 2016-Relations Among Detection of Syllable StressHochgeladen vonhord72
- 2003-Classification of Stress in Speech Using Linear and Nonlinear FeaturesHochgeladen vonhord72
- 2008-Speech Emotion Classification Using Machine Learning AlgorithmsHochgeladen vonhord72
- 2009-Stress and Emotion Recognition Using Log-Gabor Filter Analysis of Speech SpectrogramsHochgeladen vonhord72
- 2007-Stress and Emotion Classification Using Jitter and Shimmer FeaturesHochgeladen vonhord72
- 2009-Time-Frequency Feature Extraction From Stress and Emotio Clasification in SpeechHochgeladen vonhord72
- 1996-Feature Analysis and Neural Network-Based Classification of Speech Under StressHochgeladen vonhord72
- 2013-Very Early Detection of Autism Spectrum Disorders Based on Acoustic Analysis of Pre-Verbal Vocalizations of 18-Month Old ToddlersHochgeladen vonhord72
- 2015_A Machine Learning Based WSN System for Autism Activity RecognitionHochgeladen vonhord72
- 2000_Technical Report-Impact of Speech Under Stress on Military Speech TechnologyHochgeladen vonhord72
- 2014_Temporal Dynamics of Speech and Gesture in ASDHochgeladen vonhord72
- 2015-An Automatic Emotion Recognizer Using MFCCs and HMMHochgeladen vonhord72
- 2012_Emotion Recognition From Speech a ReviewHochgeladen vonhord72
- Invoice Email42148307Hochgeladen vonhord72
- Statistical Parametric Speech Synthesis a ReviewHochgeladen vonhord72
- EEpart2Hochgeladen vonVitor Aquino
- EEpart1Hochgeladen vonhord72
- Invent to learnHochgeladen vonsnowmanflo
- 1-s2.0-S1871519215001249-mainHochgeladen vonhord72

- RakeHochgeladen vonGagandeep Kaur
- Image enhancement using FusionHochgeladen vonpi194043
- Daa Notes FullHochgeladen vonAarif Muhammed
- A Class of Multirate Convolutional Codes by Dummy Bit InsertionHochgeladen vonJayshri Silmana
- ds question bankHochgeladen vondivskapatel04
- KPI DashboardHochgeladen vonEblebleblebl Ebl Ebl Ble
- KMP Algorithm - DSAHochgeladen vonAkhil Balaji
- Simplified Matching Pursuit as a New Method for SSVEP RecognitionHochgeladen vonAnonymous jxr3gcz46A
- Minkowski Distance based Feature Selection Algorithm for Effective Intrusion DetectionHochgeladen vonIJMER
- HandBook of biomedicalHochgeladen vonJuan Bolaños
- 08 Filtering 10Hochgeladen vonHanif Mohammad
- Ec333-Digital Signal ProcessingHochgeladen vonsaminathan123
- An_Introduction_to_Optimization_Chong_and_Zak.pdfHochgeladen vonNaveen Reddy
- ECTE301 Notes Week1Hochgeladen vonsreddybabu
- A Digital Acoustic Transceiver for Underwater Communication_2003_I3EHochgeladen vonlazzycat
- Runge_Kutta_Collocation_Methods_for_Diff.pdfHochgeladen vonelene shopova
- Digital TransmissionHochgeladen vonBaishakhi Bose
- Applied Operations Research Lt p cHochgeladen vonAyothy Senthil
- PruebaHochgeladen vonel bolas
- OBTL COE60Syllabus(revised) (3) (1)Hochgeladen vonAlvin1621
- Clustering Exercise 4-15-15Hochgeladen vonnomore891
- Donald+Gerald=RobertHochgeladen vonnutankhaire01
- Feed Forward Backpropagation Neural Network Image Compression for Better SNR, PSNR and BPPHochgeladen vonIJSTA
- Solving by IterationHochgeladen vonJose Miguel Ochoa
- Tutorial-6.pdfHochgeladen vonAnimesh Choudhary
- 2016-Assessment on Speech Emotion Recognition for ASDHochgeladen vonhord72
- pubs.pdfHochgeladen vonsd_hosseini_88
- Lot SizingHochgeladen vonMarcelo Kawakame
- SPSOHochgeladen vongondronk
- DS Practical ListHochgeladen vonDevashish Khattar