wireless and cellular communications-unit four
TRANSCRIPT
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UNIT IV
Modulations and Signal Processing
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Outline
4.1 Digital and Analog Modulations Techniques4.2 Amplitude Modulation
4.3 Angle Modulation
4.4 Frequency vs. Amplitude Modulations4.5 Advantages of Digital Modulation
4.6 Performance of a modulation scheme
4.7 Equalization, Diversity, and Channel Coding4.8 Speech Coding
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4.1 Digital and Analog Modulations
Techniques
Modulation is the process of encoding
information from a message source in a
manner suitable for transmission
It generally involves translating a baseband
message signal (called the source) to a
bandpass signal at frequencies that are very
high when compared to the baseband
frequency
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The bandpass signal is called the modulated
signal and the baseband message signal iscalled the modulating signal
Modulation may be done by varying the
amplitude, phase, or frequency of a high
frequency carrier in accordance with the
amplitude of the message signal
Demodulation is the process of extracting the
baseband message from the carrier so that itmay be processed and interpreted by the
intended receiver (also called the sink)
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The ultimate goal of a modulation technique isto transport the message signal through aradio channel with the best possible qualitywhile occupying the least amount of radiospectrum
The three basic modulation schemes areAmplitude Modulation(AM), FrequencyModulation(FM) , and Phase Modulation(PM)
FM and PM belong to Angle Modulation
For digital modulation digital data (0 and 1) istranslated into an analog signal (basebandsignal)
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Digital modulation is required if digital data
has to be transmitted over a medium that onlyallows for analog transmission
Best example for wired networks is the old
analog telephone system: to connect a
computer to this system a modem is needed
The modem then performs the translation of
digital data into analog signals and vice versa
Digital transmission is used, for example, in
wired local area networks or within a
computer
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In wireless networks digital transmission cannot
be used directly
Here, the binary bit-stream has to be translated
into an analog signal rst
The three basic methods for this translation are
Amplitude Shift Keying (ASK), Frequency ShiftKeying (FSK), and Phase Shift Keying (PSK)
Apart from the translation of digital data into
analog signals, wireless transmission requires
an additional modulation, an analog
modulation that shifts the center frequency of
the baseband signal generated by the digital
modulation up to the radio carrier
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For example, digital modulation translates a
1Mbit/s bit-stream into a baseband signal with
a bandwidth of 1 MHz
There are several reasons why this baseband
signal cannot be directly transmitted in a
wireless system
1. Antennas: For the 1 MHz signal we
dont need an antenna some hundred
meters high practically, With 1 GHz,antennas a few centimetres in length
can be used
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2. Frequency division multiplexing: Using
only baseband transmission, FDM couldnot be applied and Analog modulationshifts the baseband signals to differentcarrier frequencies
3. Medium characteristics: Path-loss,penetration of obstacles, reflection,scattering, and diffraction depend heavilyon the wavelength of the signal and
Depending on the application, the rightcarrier frequency with the desiredcharacteristics has to be chose
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Modulation in a transmitter
Demodulation and data reconstruction in a receiver
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Amplitude shift keying
Amplitude Shift Keying (ASK): is the most
simple digital modulation scheme
The two binary values, 1 and 0, are
represented by two different amplitudes
ASK illustrated by a figure in the next slide,
this simple scheme only requires low
bandwidth, but is very susceptible tointerference
Effects like multipath propagation, noise, or
path loss heavily influence the amplitude
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In a wireless environment, a constant
amplitude cannot be guaranteed, so ASK istypically not used for wireless radio
transmission
ASK can also be applied to wireless infrared
transmission, using a directed beam or diffuse
light
Amplitude shift keying (ASK)
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Frequency shift keying
A modulation scheme often used for wireless
transmission is frequency shift keying (FSK)
The simplest form of FSK, also called binary
FSK (BFSK), assigns one frequency f1 to thebinary 1 and another frequency f2 to the
binary 0
A very simple way to implement FSK is toswitch between two oscillators, one with the
frequency f1 and the other with f2, depending
on the input
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To avoid sudden changes in phase, special
frequency modulators with continuous phase
modulation (CPM) can be used
Sudden changes in phase cause high
frequencies, which is an undesired side-effect
Frequency shift keying (FSK)
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Phase shift keying Phase shift keying (PSK) uses shifts in the
phase of a signal to represent data
The figure in the next slide shows a phase shift
of 180or as the 0 follows the 1 (vice versa)
This simple scheme, shifting the phase by
180each time the value of data changes, is
also called binary PSK (BPSK)
A simple implementation of a BPSK modulatorcould multiply a frequency f with +1 if the
binary data is 1 and with 1 if the binary data
is 0
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To receive the signal correctly, the receiver
must synchronize in frequency and phase withthe transmitter. This can be done using a
Phase Lock Loop (PLL)
Compared to FSK, PSK is more resistant tointerference, but receiver and transmitter are
also more complex
Phase shift keying (PSK)
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4.2 Amplitude Modulation
In amplitude modulation, the amplitude of a highfrequency carrier signal is varied in accordance to
the instantaneous amplitude of the modulating
message signal
If Accos(2fct)is the carrier signal and m (t) is the
modulating message signal, the AM signal can be
represented as
SAM(t)=Ac[1+m(t) cos(2fct)] The modulation index k of an AM signal is defined
as the ratio of the peak message signal amplitude
to the peak carrier amplitude
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For a sinusoidal modulating signal m(t)=(Am/Ac)cos (2fmt) , the modulation index is given by
Where Am is the modulated and Ac carrier signals
The modulation index is often expressed as a
percentage, and is called percentage modulation For instance , if Am = 0.5 Ac and the signal is said
to be 50% modulated the correspondingsinusoidal modulating and AM signalrepresented by figures in the next slide
A percentage of modulation greater than 100%will distort the message signal if detected by anenvelope detector
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A sinusoidal modulating signal
AM signal with modulation index 0.5
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AM signal may be equivalently expressed as
SAM (t) = Re { g(t) exp (j2fct) }
where g (t) is the complex envelope of the AM
signal given by
g(t)=Ac{1+m(t)}
The spectrum of an AM signal(function of f)
can be shown to be
where () is the unit impulse function, and
M(f) is the message signal spectrum
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The AM spectrum consists of an impulse atthe carrier frequency, and two sidebands
which replicate the message spectrum The sidebands above and below the carrier
frequency are called the upper and lowersidebands, respectively
The bandwidth of an AM signal is equal to
where fm is the maximum frequency containedin the modulating message signal
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The total power in an AM signal can be shownto be
where * represents the average value
If the modulating signal is the
above equation may be simplified as
where is the power in the carriersignal , is the power in themodulating signal m(t), and k is themodulation index
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Example:
A zero mean sinusoidal message is applied to atransmitter that radiates an AM signal with 10
kW power. Compute the carrier power if the
modulation index is 0.6. What percentage of
the total power is in the carrier?
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4.3 Angle Modulation
The two most important classes of angle
modulation being frequency modulation and
phasemodulation
Angle modulation varies a sinusoidal carriersignal in such a way that the angle of the
carrier is varied according to the amplitude of
the modulating baseband signal
In this method, the amplitude of the carrier
wave is kept constant (this is why FM is called
constant envelope)
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There are a number of ways in which the
phase (t) of a carrier signal may be varied in
accordance with the baseband signal
Frequency modulation (FM) is a form of angle
modulation in which the instantaneous
frequency of the carrier signal is varied
linearly with the baseband message signal
m(t), as shown in equation
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where Ac is the amplitude of the carrier, fc is
the carrier frequency, and kf is the frequency
deviation constant (measured in units of
Hz/volt)
If the modulating signal is a sinusoid of
amplitude Am , and frequency fm , then the FMsignal may be expressed as
f=kfAm
Where f is the peak frequency deviation of
the transmitter
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For sinusoidal modulating signal with
modulating frequency fm
The frequency modulation index f defines
the relationship between the message
amplitude and the bandwidth of the
transmitted signal, and is given by
Where Am is the peak value of the modulatingsignal, f is the peak frequency deviation of
the transmitter and W is the maximum
bandwidth of the modulating signal
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If the modulating signal is a low pass signal, as
is usually the case, then W is equal to thehighest frequency component fmax =fm present
in the modulating signal, hence
Example
A sinusoidal modulating signal,
m(t) = 4cos24 x 103t, is applied to an FM
modulator that has a frequency deviation
constant gain of 10 kHz/V. Compute
(a) The peak frequency deviation, and
(b) The modulation index
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Phase modulation (PM) is a form of angle
modulation in which the angle (t) of the
carrier signal m(t) is varied linearly with the
baseband message signal as shown in
equation
Where k
is the phase deviation constant
(measured in units of radians/volt)
From the above equations, it is clear that anFM signal can be regarded as a PM signal in
which the modulating wave is integrated
before modulation
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Note that an FM signal can be generated by
first integrating m (t) and then using the result
as an input to a phase modulator, conversely a
PM wave can be generated by first
differentiating m(t) and then using the result
as the input to a frequency modulator
The phase modulation index p , is given by
Where is the peak phase deviation of the
transmitter
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4.4 Frequency Modulation vs.
Amplitude Modulation
Frequency modulation (FM) is the mostpopular analog modulation technique used inmobile radio systems
In FM, the amplitude of the modulatedcarrier signal is kept constant while itsfrequency is varied by the modulatingmessage signal
Thus, FM signals have all their information inthe phase or frequency of the carrier
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This provides a nonlinear and very rapid
improvement in reception quality once acertain minimum received signal level, calledthe FM threshold, is achieved
In amplitude modulation (AM) schemes, there
is a linear relationship between the quality ofthe received signal and the power of thereceived signal since AM signals superimposethe exact relative amplitudes of the
modulating signal onto the carrier Thus, AM signals have all their information in
the amplitude of the carrier
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FM offers many advantages over amplitude
modulation (AM), which makes it a betterchoice for many mobile radio applications
Frequency modulation has better noise
immunity when compared to amplitude
modulation
Since signals are represented as frequency
variations rather than amplitude variations,
FM signals are less susceptible toatmospheric and impulse noise, which tend
to cause rapid fluctuations in the amplitude of
the received radio signal
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Also, message amplitude variations do not
carry information in FM, so burst noise does
not affect FM system performance as much as
AM systems, provided that the FM received
signal is above the FM threshold
Unlike AM, in an FM system, the modulationindex, and hence bandwidth occupancy, can
be varied to obtain greater signal-to-noise
performance Under certain conditions, the FM signal-to-
noise ratio improves 6 dB for each doubling of
bandwidth occupancy
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This ability of an FM system to trade
bandwidth for SNR is perhaps the most
important reason for its superiority over AM However, AM signals are able to occupy less
bandwidth as compared to FM signals, since
the transmission system is linear An FM is a constant envelope signal, due to
the fact that the envelope of the carrier does
not change with changes in the modulating
signal
Hence the transmitted power of an FM signal
is constant regardless of the amplitude of the
messa e si nal
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The issue of amplifier efficiency is extremely
important when designing portable subscriberterminals since the battery life of the portable
is tied to the power amplifier efficiency
The capture effect is a direct result of the
rapid nonlinear improvement in received
quality for an increase in received power
If two signals in the same frequency band are
available at an FM receiver, the one appearingat the higher received signal level is accepted
and demodulated, while the weaker one is
rejected
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This inherent ability to pick up the strongest
signal and reject the rest makes FM systems
very resistant to co-channel interference and
provides excellent subjective received quality
In AM systems, on the other hand, all of the
interferers are received at once and must bediscriminated after the demodulation process
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4.5 Advantages of Digital
Modulation Modern mobile communication systems use
digital modulation techniques
Advancements in very large-scale integration
(VLSI) and digital signal processing (DSP)technology have made digital modulation
more cost effective than analog transmission
systems
Digital modulation offers many advantages
over analog modulation
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Some advantages include greater noise
immunity and robustness to channelimpairments, easier multiplexing of various
forms of information (e.g., voice, data, and
video), and greater security
Digital transmissions accommodate digital
error-control codes which detect and/or
correct transmission errors, and support
complex signal conditioning and processingtechniques such as source coding, encryption,
and equalization to improve the performance
of the overall communication link
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In digital wireless communication systems, the
modulating signal (e.g., the message) may berepresented as a time sequence of symbols or
pulses, where each symbol has m finite states
Each symbol represents n bits of information,
where n = log2m bits/symbol
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4.6 Performance of a modulation
scheme
The performance of a modulation scheme is
often measured in terms of its bit error rate,
power efficiency and bandwidth efficiency
Power efficiency describes the ability of a
modulation technique to preserve the fidelity
of the digital message at low power levels
In a digital communication system, in order to
increase noise immunity, it is necessary to
increase the signal power
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However, the amount by which the signal
power should be increased to obtain a certainlevel of fidelity(i.e., an acceptable bit error
probability) depends on the particular type of
modulation employed
The power efficiency, p , (sometimes called
energy efficiency) of a digital modulation
scheme is a measure of how favourably this
trade-off between fidelity and signal power ismade
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It is often expressed as the ratio of the signal
energy per bit to noise power spectral density
(Eb/N0) required at the receiver input for a
certain probability of error (say 10-5)
Bandwidth efficiency describes the ability of a
modulation scheme to accommodate datawithin a limited bandwidth
In general, increasing the data rate implies
decreasing the pulse width of a digital symbol,which increases the bandwidth of the signal
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Bandwidth efficiency reflects how efficiently
the allocated bandwidth is utilized and isdefined as the ratio of the throughput data
rate per Hertz in a given bandwidth
If R is the data rate in bits per second, and B is
the bandwidth occupied by the modulated
1W signal, then bandwidth efficiency B is
expressed as
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The system capacity of a digital mobile
communication system is directly related tothe bandwidth efficiency of the modulation
scheme, since a modulation with a greater
value ofB will transmit more data in a given
spectrum allocation
There is a fundamental upper bound on
achievable bandwidth efficiency
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Shannon's channel coding theorem states that
for an arbitrarily small probability of error,
the maximum possible bandwidth efficiency is
limited by the noise in the channel, and is
given by the channel capacity formula
where C is the channel capacity (in bps), B is
the RF bandwidth, and S/N is the signal-to-noise ratio
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If the modulation bandwidth exceeds the
coherence bandwidth of the radio channel, ISI
occurs and modulation pulses are spread intime
An equalizer within a receiver compensates
for the average range of expected channelamplitude and delay characteristics
Equalizers must be adaptive since the channel
is generally unknown and time varying
Diversity is another technique used to
compensate for fading channel impairments,
and is usually implemented by using two or
more receivin antennas
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As with an equalizer, the quality of a mobile
communications link is improved without
increasing the transmitted power orbandwidth
While equalization is used to counter the
effects of time dispersion (ISI), diversity
usually employed to reduce the depth and
duration of the fades experienced by a
receiver in a flat fading (narrowband) channel
One of the best techniques to mitigate the
effects of fading is diversity combining of
independently fading signal paths
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Diversity combining exploits the fact that
independent signal paths have a low
probability of experiencing deep fadessimultaneously
Thus, the idea behind diversity is to send the
same data over independent fading paths These independent paths are combined in
such a way that the fading of the resultant
signal is reduced For example, consider a system with two
antennas at either the transmitter or receiver
that experience independent fading
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If the antennas are spaced sufficiently far
apart, it is unlikely that they both experiencedeep fades at the same time
By selecting the antenna with the strongest
signal, a technique known as selection
combining, we obtain a much better signal
than if we had just one antenna
Diversity techniques can be employed at both
base station and mobile receivers
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CDMA systems often use a RAKE receiver,
which provides link improvement through
time diversity
By coding is meant the purposeful
introduction of additional bits in a digital
message stream to allow correction and/ordetection of bits in the message stream that
may have been received in error
Channel coding improves mobilecommunication link performance by adding
redundant data bits in the transmitted
message
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At the baseband portion of the transmitter, a
channel coder maps a digital message
sequence into another specific code sequence
containing a greater number of bits than
originally contained in the message
The coded message is then modulated fortransmission in the wireless channel
Channel coding is used by the receiver to
detect or correct some (or all) of the errorsintroduced by the channel in a particular
sequence of message bits
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Because decoding is performed after the
demodulation portion of the receiver, coding
can be considered to be a post detectiontechnique
The added coding bits lowers the raw data
transmission rate through the channel
(expands the occupied bandwidth for a
particular message data rate)
There are two general types of channel codes:
block codes and convolutional codes
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Channel coding is treated independently from
the type of modulation used
Although this has changed recently with theuse of trellis coded modulation schemes that
combine coding and modulation to achieve
large coding gains without any bandwidthexpansion
The three techniques of equalization, are used
to improve radio link performance (to
minimize the instantaneous bit error rate), but
the approach, cost, complexity, and
effectiveness of each technique varies widely
in practical wireless communication systems
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RAKE Receiver
In CDMA spread spectrum systems, the chip
rate is typically much greater than the flat
fading bandwidth of the channel
Whereas conventional modulation techniques
require an equalizer to undo the inter symbolinterference between adjacent symbols,
CDMA spreading codes are designed to
provide very low correlation betweensuccessive chips
Propagation delay spread in the radio channel
merely provides multiple versions of the
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If these multipath components are delayed in
time by more than a chip duration, they
appear like uncorrelated noise at a CDMAreceiver, and equalization is not required
A RAKE receiver attempts to collect the time-
shifted versions of the original signal byproviding a separate correlation receiver for
each of the multipath signals
CDMA systems often use a RAKE receiver,which provides link improvement through
time diversity
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The RAKE(M branch or M-finger) receiver, shown in the
Figure, is essentially a diversity receiver designed
specifically for CDMA, where the diversity is provided
by the fact that the multipath components are
practically uncorrelated from one another when their
relative propagation delays exceed a chip period
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A RAKE receiver utilizes multiple correlators to
separately detect the M strongest multipath
components
The outputs of each correlator are weighted
to provide a better estimate of the
transmitted signal than is provided by a singlecomponent
Demodulation and bit decisions are then
based on the weighted out puts of the M
correlators
A weighting network is used to provide a
linear combination of the correlator output for
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The outputs of the M correlators are denoted
as Z1, Z2 ,... and ZM, They are weighted by
1,2,M respectively
The overall signal Z' is given by
The weighting coefficients, am, are normalized
to the output signal power of the correlator in
such a way that the coefficients sum to unity,as show below
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4.8 Speech Coding
Speech coders determines the quality of the
recovered speech and the capacity of the
system
Service providers are continuously met with
the challenge of accommodating more users
within a limited allocated bandwidth
Low bit-rate speech coding offers a way tomeet this challenge
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Speech coders differ widely in their
approaches to achieving signal compression
Based on the means by which they achieve
compression, speech coders are broadly
classified into two categories: Waveform
Coders and Vocoders Waveform coders essentially strive to
reproduce the time waveform of the speech
signal as closely as possible They are designed to be source independent
and can hence code equally well a variety of
signals
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Examples of waveform coders include Pulse
Code Modulation (PCM), Differential Pulse
Code Modulation (DPCM), AdaptiveDifferential Pulse Code Modulation (ADPCM),
Delta Modulation (DM), Continuously Variable
Slope Delta Modulation (CVSDM), andAdaptive Predictive Coding (APC)
Vocoders on the other hand achieve very high
economy in transmission bit rate and are in
general more complex
They are based on using a priori knowledge
about the signal to be coded(Signal specific)
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Characteristics of Speech Signals
1. The non-uniform probability distribution ofspeech amplitude
2. The nonzero autocorrelation between
successive speech samples3. The non-flat nature of the speech spectra,
the existence of voiced and Unvoiced
segments in speech4. The quasi-periodicity of voiced speech signals
are the properties that are most often
utilized in coder design
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The most basic property of speech waveforms
that is exploited by all speech coders is that
they are band limited Probability Density Function (pdf): The non-
uniform probability density function of speech
amplitudes is perhaps the next most exploitedproperty of speech
The pdf of a speech signal is characterized by a
very high probability of near-zero amplitudes,
a significant probability of very high
amplitudes, and a monotonically decreasing
function of amplitudes between these
extremes
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The exact distribution depends on the input
bandwidth and recording conditions
The two-sided exponential (Laplacian)
function given in the equation below provides
a good approximation to the long-term pdfof
telephone quality speech signals
This pdf shows a distinct peak at zero
Short-time pdfs of speech segments are also
single-peaked functions and are usually
approximated as a Gaussian distribution
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Autocorrelation Function (ACF): Another very
useful property of speech signals is that there
exists much correlation between adjacent
samples of a segment of speech
This implies that in every sample of speech,
there is a large component that is easilypredicted from the value of the previous
samples with a small random error
All differential and predictive coding schemesare based on exploiting this property
It gives a quantitative measure of the
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It gives a quantitative measure of the
closeness or similarity between samples of a
speech signal as a function of their time
separation and mathematically defined as:
where x (k) represents the kth speech sample
The autocorrelation function is often
normalized to the variance of the speech
signal and hence is constrained to have values
in the range {-1,1} with C (0) = 1 signals have
an adjacent sample correlation, C(1) , as high
as 0.85 to 0.9
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Power Spectral Density Function (PSD):The
non-flat characteristic of the power spectral
density of speech makes it possible to obtainsignificant compression by coding speech in
the frequency domain
The non-flat nature of the PSD is basically afrequency domain manifestation of the
nonzero autocorrelation property
Typical long-term averaged PSD's of speechshow that high frequency components
contribute very little to the total speech
energy
Thi i di h di h l i
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This indicates that coding speech separately in
different frequency bands can lead to
significant coding gain A qualitative measure of the theoretical
maximum coding gain that can be obtained by
exploiting the non-flat characteristics of thespeech spectra is given by the Spectral
Flatness Measure (SFM)
The SFM is defined as the ratio of the
arithmetic to geometric mean of the samples
of the PSD taken at uniform intervals in
frequency
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Mathematically,
where, Sk is the kth frequency sample of the
PSD of the speech signal
Typically, speech signals have a long-term
SFM value of 8 and a short-time SFM value
varying widely between 2 and 500
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