Telecommunication voice transmission

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    Voice Transmission Voice Transmission----Basic ConceptsBasic Concepts----

    Voice---is analog in character and moves in the form ofwaves.3-important wave-characteristics :

    Am litude

    Prof. Murat Torlak EE4367 Telecom. Switching & Transmission

    FrequencyPhase

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    Voice Digitization Voice Digitization How!How!Digitization is just a discrete electrical voltage.Electrical pulses can be varied to represent characteristics of ananalog voice signal.

    5-different VD-techniques:PAM = pulse amplitude modulationPDM = pulse duration modulationPPM = Pulse position modulation

    Prof. Murat Torlak EE4367 Telecom. Switching & Transmission

    PCM = Pulse code modulationADPCM = Adaptive differential PCM

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    Pulse Amplitude ModulationPulse Amplitude ModulationFirst step in digitizing an analog waveformEstablishes a set of discrete times at which the input signal

    waveform is sampled.The sampling process is equivalent to amplitude modulationof a constant amplitude pulse train, thus, PAM.N uist Sam lin Rate : The minimum sam lin fre uenc

    Prof. Murat Torlak EE4367 Telecom. Switching & Transmission

    required to extract all information in a continuous, time-varying waveform.

    Nyquist Criterion: f s>2 BWf s: sampling rate, BW: bandwidth of the input signal

    PAM samples

    AmplitudemodulationPulse train

    Input signal

    Low-passfilter

    Output signal

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    Spectrum of PAM SignalSpectrum of PAM SignalThe PAM spectrum can be derived by observing that acontinuous train of impulses has a frequency spectrumconsisting of discrete terms at multiples of the samplingfrequency.

    Prof. Murat Torlak EE4367 Telecom. Switching & Transmission

    Source: Digital Telephony by Bellamy

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    Foldover DistortionFoldover DistortionIf the input is undersampled (f s

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    Aliasing AliasingThe example below demonstrates an aliasing processoccurring in speech if a 5.5 kHz signal is sampled at an 8-KHz rate. Notice that the sample values are identical tothose obtained from a 2.5 KHz signal.Thus, after the sampled signal passes through the 4 KHzoutput filter, a 2.5 KHz signal arises that did not come from

    Prof. Murat Torlak EE4367 Telecom. Switching & Transmission

    e source.

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    PAM SystemPAM SystemComplete PAM system includes a bandlimiting filter beforesampling to ensure that no source related signals get foldedback into the desired signal bandwidth.End-to-End PAM system

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    Pulse Coded Modulation (PCM)Pulse Coded Modulation (PCM)PCM is an extension of PAM wherein each analog sample isquantized into a discrete value for representation as adigital codewordPAM system can be converted to PCM if we add ADC at thesource and DAC at the destination.

    Prof. Murat Torlak EE4367 Telecom. Switching & Transmission

    Source: Digital Telephony by Bellamyc

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    QuantizationQuantizationQuantization process has a set of quantization intervalsassociated in a one-to-one fashion with a binary codeword.

    Binary codeword corresponds to a discrete amplitude

    Quantization process introduces a certain amount of erroror distortion into the signal samples.

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    Quantization NoiseQuantization NoiseQuantization errors of a PCM encoder are generallyassumed to be distributed randomly and uncorrelated toeach other

    Treated as an additive noiseSignal-to-quantizing-noise ratio (SQR) to define amount ofquantization distortion.

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    Quantization NoiseQuantization NoiseThe error e(t)=y(t)-x(t) is limited in amplitude to q/2,where q is the height of the quantization interval.

    A sample value is equally likely to fall anywhere within aquantization interval implying a uniform probability densityof amplitude 1/q.

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    A sinewave amplitude

    v rms amplitude

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    Uniformly Encoded PCMUniformly Encoded PCMNumerical value of each codeword is proportional to thequantized amplitude that it represents

    Minimum digitized voice quality requires 26 dB SQR toprovide adequate quality for small signalsqmax =0.123A, and encoding from A to A, we need 4 bits

    Dynamic Range (DR): Capability of transmitting a large

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    range of signal amplitudesExpressed in dB as the ratio of the maximum amplitude signalto the minimum amplitude signal

    PCM performance equation for uniform coding

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    CompressionCompressionVarious compression-expansion characteristics can be chosen toimplement a compandor

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    Compression used in North America has the -law characteristics

    compressor

    expander

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    T1 Carrier SystemT1 Carrier SystemTelephone wires were used to transmit one audio signal ofbandwidth 4 KHzAfter introduction of PCM, the same wires are used to transmit 24TDM PCM (DS0) telephone signals wit a total BW of 1.544 MHzRepeaters every at 6000 feet

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    T1 System Signaling FormatT1 System Signaling FormatBinary code words corresponding to samples of each of the24 channels are multiplexed in a sequence

    = 100, 7 bits are used for data, 1-bit for signaling in D1signaling

    D2 channel bank needs better voice quality8 bits PCM and =255, signaling only repeats every sixth frame

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    DS-0 provides 64KbpsT-1 DS-1 provides 1.544 MbpsT-2 DS-2 provides 6.312 MbpsT-3 DS-3 provides 33.375 MbpsT-4 DS-4 provides 274.176 Mbps

    Robbed bit signaling

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    What is a Codec!What is a Codec!Codec = Coder + Dec oderModem = Modulator + Dem odulatorCodec is a voice-connector device that samples the analog POTStransmission into a stream of binary digits using PCM.For quality digital transmission over a Codec:

    8000 samples/secEach sample = 1 byte = 8 bits

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    8000 X 8 = 64,000 bits/sec----( DS-0 circuit)Uses T-1 circuit1.544 Mbps---(equivalent to 24 DS-0)

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    Codec and Modem!Codec and Modem!Codec is a voice-connector deviceModem is a data-connector device.All high speed Codecs operate at multiples of DS-0 speed:

    DS-0 provides 64 KbpsT-1 DS-1 provides 1.544 MbpsT-2 DS-2 provides 6.312 MbpsT-3 DS-3 provides 33.375 Mbps

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    T-4 DS-4 provides 274.176 Mbps

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    8

    7

    6

    5

    4

    3

    2

    1

    0

    8 possible amplitudes inexample are really 256(28) amplitudes in PCM

    analog signal

    Analog Signal to be Digitized

    Sampling rate = 8,000 times/secon1/8000 of a second

    Step 1: Sample Amplitude of Analog Signal

    Amplitude in example at sample position 1 (the gray shaded box) is 4

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    Voice Digitization:Pulse CodeModulation (PCM)

    Step 2: Represent Measured Amplitude in Binary Notation

    Step 3: Transmit Coded Digital Pulses Representing Measured Amplitude

    Power of 2

    Value

    Binary notation

    27 26 25 24 23 22 21 20

    128 64 32 16 8 4 2 1

    0 0 0 0 0 1 0 0 = 4

    8 bits = 1 byte

    0 0 0 0 0 1 0 0

    8 transmitted bits = 1 transmitted byte =1 transmitted sampled amplitude

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    Voice Compression! Voice Compression!ADPCM--adaptive differential PCM-is also known as a voicecompression technique, because of its ability to transmit 24digitized voice conversations in half the bandwidth required by

    the PCM.More advanced VCs use digital signal processors that take thedigitized PCM and further manipulate and compress it.Each voice compression technique seeks to reduce the amount of

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    - .

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    ExampleExampleSpeech digitization techniques are sometimes measured for qualityby use of an 800-Hz sine wave as a representative test signal.Assuming a uniform PCM system is available to encode the sine waveacross a given dynamic range, determine how many bits sample canbe saved by using a uniform DPCM system.A basic solution can be obtained by determining how much smallerthe dynamic range of the difference signal is in comparison to the

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    .

    amplitude of the sine wave is A, so that

    The maximum amplitude of the difference signal can be obtained bydifferentiating and multiplying by the time interval between samples:

    The savings in bits per sample can be determined as

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    ( ) sin(2 800 ) x t A t =

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    Adaptive Differential PCM Adaptive Differential PCMAdaptation logic can be added to the basic DPCM algorithmto create what is referred to as adaptive differential PCM(ADPCM).ITU has established a 32-kbps ADPCM standard (G.721) forvoice messaging and for increasing the number of voicechannels on a T1 line.

    Prof. Murat Torlak

    It uses a 8th

    order predictor, adaptive quantization, andadaptive prediction.

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    Delta ModulationDelta ModulationSpecial case of DPCM with 1-bit per sample differencesignalRequires high-sampling rate than DPCMProblem: Slope overload with sampling rate is low

    Delta Modulator

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    Delta Demodulator

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    Subband CodingSubband CodingHuman ear perceives speech by measuring the short-termenergy level of individual frequency bands

    Prof. Murat Torlak EE4367 Telecom. Switching & Transmission