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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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