telecommunications switching systems 13
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DPCM, DM
L-13
DPCM
Designed to take advantage of sample-to-sample redundancies.
Since the range of differences is less than therange of individual samples, fewer bits arerequired to encode difference samples.
Sampling rate, band limiting filter and smoothingfilter are identical to those used in conventionalPCM systems.
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Concept
Store the previous sample using S/Hcircuit and use a subtractor to measure thechange.
The change in the signal is then quantizedand encoded for transmission.
DPCM
The previous input value is reconstructed by afeedback loop that integrates the encodedsample differences.
In essence the feedback signal is an estimate ofthe input signal as obtained by integrating theencoded sample differences.
Thus, the feedback signal is obtained in thesame manner used to reconstruct the waveformin the decoder.
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Functional Block Diagram - DPCM
Advantage of feedback loop
Feedback loop at transmitter side does notallow quantization errors to getaccumulated.
If the feedback signal drifts from the inputsignal, as a result of an accumulation of
quantization errors, the next encoding ofthe difference signal automaticallycompensates for the drift.
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DPCM (Analog Integration)
31001 (1)1.123.1
52010 (2)1.734.7
20000 (0)0.322.3
22010 (2)1.701.7
X+PXBDPN
DPCM (Digital Integration)
001 (1)1.123.1
010 (2)1.734.7
000 (0)0.322.3010 (2)1.701.7
BDPN
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DPCM (Digital Differencing)
2 (010)
1 (001)0 (000)
2 (010)
D
3233.1
5354.7
2222.3
2021.7
REG = D+PPBN
DPCM Decoders The decoders are exactly like the feedback
implementations in the encoders. This reinforces the fact that the feedback loop
generates an approximation of the input signal(delayed by one sample).
If no channel errors occur, the decoder output(before filtering) is identical to the feedbacksignal.
Thus the closer the feedback signal matches theinput, the closer the decoder output matches theencoder input.
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DPCM
A/D process can be uniform or companded
Some DPCM systems also use Higher OrderPrediction.
The feedback signal of DPCM systemrepresents first order prediction of the next
sample value, an the sample difference is aprediction error.
Higher order prediction
DPCM concept can be extended to incorporatemore than one past sample value into theprediction circuitry.
The additional redundancy available from allprevious samples can be weighted and summedto produce a better estimate of the next inputsample.
With a better estimate, the range of theprediction error decreases to allow encodingwith fewer bits.
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Third Order Prediction
For systems with constant predictorcoefficients, results have shown that mostrealizable improvement occurs when usinglast three sample values.
DPCM (3rd Order Prediction)
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DPCM
DPCM systems with first-order predicationtypically provides a 1-bit-per-sample reduction incode length relative to PCM systems withequivalent performance.
Extended DPCM systems utilizing third orderprediction can provide reductions of 2 bits persample.
Thus a standard DPCM systems can provide 64-kbps PCM quality at 56-kbps and third-orderprediction can provide comparable quality at 48-kbps.
ADPCM
ITU-T established a 32 kbps ADPCMstandard (Recommendation G.721)
The 32-kbps rate imples a 2:1 savings inchannel BW w.r.t standard PCM.
The G.721 standard is conceptually similar
to DPCM (3rd
order prediction) but uses 8th
order predictor, adaptive quantization andadaptive prediction.
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MOS
The mean opinion score(MOS) is a method ofevaluating speech quality.
The MOS method usestrained listeners toevaluate the speechquality on a scale of 1:5.
Subjective evaluation ofG.721 algorithm usingMOS method of
evaluating speech quality(with carbon microphone)is shown:
Delta Modulation
DM can be considered as a special case ofDPCM using only 1-bit per sample of thedifference signal.
The single bit specifies merely the polarity of thedifference sample and thereby indicates whetherthe signal has increased or decreased since thelast sample.
In this way the input is encoded as a sequenceof ups and downs in a manner resembling astaircase.
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DM
Since each encoded sample contains arelatively small amount of information (1-bit), DMsystems require a higher sampling rate thanPCM or multi-bit DPCM systems.
In fact, the sampling rate is necessarily muchhigher than the minimum (Nyquist) samplingrate.
The relatively high sampling rate produces awider separation between replicas of samplespectrum, thus preventing fold-over distortion.
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DM The main attraction of DM is its simplicity. A/D is provided by a simple comparator. A positive difference produces a 1 and a negative
difference produces a 0. D/A function in feedback path and decoder, is a two-
polarity pulse generator. In its simplest for, a capacitor can work as integrator.
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Slope Overload
sqfdt
tdx
)(becauseoccursoverloadSlope
max
)(overloadSlopepreventTo
dt
tdxqfs
Granular Noise
Granular noise is predominant consideration forslowly changing signals, whereas slope overloadis dominant during rapidly changing signals.
Granular noise is small if step sizes are small,but small step sizes increase the likelihood ofslope overload.
The design of DM necessarily involves a trade-off between two types of distortion.
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DM
Slope overload noise has strong componentsidentical in frequency and approximately inphase with a major component of input.
Distortion that is correlated in this manner to thespeech energy is effectively masked by thespeech energy and therefore is less noticeablethen uncorrelated distortion.
In fact, overload noise in much lessobjectionable to a listener than random orgranular noise at an equivalent power level.
Adaptive Delta Modulation
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ADM
In ADM, if DAC o/p amplitude is still belowthe sample amplitude, the next step size isincreased till DAC catches up with analogsignal.
When an alternative 1s and 0s occur, stepsize is reduced.
Dithering Dithering involves the deliberate addition of
noise to our input signal. It helps by smearing out the little differences in
amplitude resolution. The key is to add random noise in a way that
makes the signal bounce back and forthbetween successive levels.
Of course, this in itself just makes the signalnoisier. But, the signal smoothes out byaveraging this noise digitally once the signal isacquired
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Dithering
G.7xx, including G.711, G.721, G.722, G.726, G.727,G.728, G.729, is a suite of ITU-T standards for audiocompression and de-compression.
It is primarily used in telephony.
In telephony, there are 2 main algorithms defined in thestandard, -law algorithm and A-law algorithm.
Both are logarithmic, but the later a-law was specificallydesigned to be simpler for a computer to process.
G.7xx: Audio (Voice) Compression Protocols (G.711, G.721, G.722,
G.726, G.727, G.728, G.729)
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824/40
Extensions ofRecommendation
G.721 adaptivedifferential pulse codemodulation to 24 and40 kbit/s for digitalcircuit multiplicationequipment application
ITU-TG.723
1624/32
Coding at 24 and 32 kbit/sfor hands-freeoperation in systemswith low frame loss
ITU-TG.722.1
16647 kHz audio-coding within
64 kbit/sITU-TG.722
832Adaptive differential pulse
code modulation(ADPCM)
ITU-TG.721
864Pulse code modulation
(PCM)ITU-TG.711
832ADPCMIntel, IMA(ADPCM) DVI
sampling rate
(kHz)
bit rate
(kb/s)description
standardized
byName
Comparison of G.7xx Protocol
IMA: Interactive Multimedia AssociationDVI: Digital Visual/ Video Interface/Interactive
88Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction(CS-ACELP)
ITU-TG.729
816Coding of speech at 16 kbit/s using low-delaycode excited linear prediction
ITU-TG.728
8var5-, 4-, 3- and 2-bit/sample embedded adaptivedifferential pulse code modulation (ADPCM)
ITU-TG.727
816/24/32/4040, 32, 24, 16 kbit/s adaptive differential pulsecode modulation (ADPCM)
ITU-TG.726
85.6/6.3Dual rate speech coder for multimediacommunications transmitting at 5.3 and 6.3kbit/s
ITU-TG.723.1
samplingrate (kHz)
bit rate(kb/s)
descriptionstandardized
byName
Comparison of G.7xx Protocol