EEE543 DCS - Lecture 2 - Part III
EEE543 DCS - Lecture 2 - Part III
Pulse Code
Modulation - Part III
1
OUTLINE
Differential Pulse Code Modulation
Adaptive DPCM
Delta Modulation
2
DIFFERENTIAL PULSE CODE MODULATION (DPCM)
PCM is not a very efficient system because it generates so
many bits and requires so much bandwidth to transmit.
DPCM exploits the characteristics of the signals to improve
the efficiency of PCM
In analog messages the sample values are not independent
we can make a good guess about a sample value from
knowledge of past sample values
There is a great deal of redundancy in the Nyquist samples.
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DPCM CONT’D
Consider a simple scheme; instead of transmitting the
sample values, we transmit the difference between the
successive sample values
For the k-th sample 𝑚[𝑘] we transmit the difference
𝑑 𝑘 = 𝑚 𝑘 − 𝑚[𝑘 − 1]
At the receiver, knowing 𝑑 𝑘 and several previous sample
value 𝑚[𝑘 − 1], we can reconstruct 𝑚 𝑘 iteratively
The difference between successive samples is generally
much smaller than the sample values – the peak amplitude
𝑚𝑝 of the transmitted values is reduced considerably
𝑚𝑝
The reduction in 𝑚𝑝 reduces the quantization step, Δ𝜈 =
𝐿
Δ𝜈 2
which in turns reduces the quantization noise 𝑞𝑒 = for a
𝐿
given L
This means that for a given number of bits per sample n (or
transmission bandwidth) we can increase the SNR similarly 4
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DPCM CONT’D
For the discrete samples we can use the Taylor series to
predict the future samples
Let the kth sample of m(t) be given by: 𝑚 𝑘 = 𝑚 𝑘𝑇𝑠 then
𝑚 𝑘𝑇𝑠 ± 𝑇𝑠 = 𝑚 𝑘 ± 1 and so on
If w set 𝑡 = 𝑘𝑇𝑠 and also noting that
𝑚 𝑘𝑇𝑠 − 𝑚 𝑘𝑇𝑠 − 𝑇𝑠
𝑚ሶ 𝑘𝑇𝑠 =
𝑇𝑠
Then we can find 𝑚 𝑘 ± 1 as follows
𝑇𝑠 𝑚 𝑘 − 𝑚 𝑘 − 1
𝑚 𝑘+1 ≈𝑚 𝑘 +
𝑇𝑠
= 2𝑚 𝑘 − 𝑚 𝑘 − 1
This shows that we can find a crude prediction of the (k + 1 )th
sample from the two previous samples
The approximation improves as we add more terms in the
series on the right-hand side.
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DPCM CONT’D
To determine the higher order derivatives in the series, we
require more samples in the past
We can express the prediction formula as
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ANALYSIS OF DPCM CONT’D
The figure below shows a DPCM transmitter
Figure 2.1 a)
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ANALYSIS OF DPCM CONT’D
The receiver (figure 2.1b) is the same as the shaded region of the
transmitter
Figure 2.1 b)
The inputs and predictor output in both cases are the same also,
namely, 𝑑𝑞 𝑘 and 𝑚ෝ 𝑞 𝑘 respectively
From the receiver output is 𝑚𝑞 𝑘 = 𝑚 𝑘 + 𝑞 𝑘 ,we can obtain the
desired signal 𝑚 𝑘 plus the quantization noise 𝑞 𝑘
The quantization noise 𝑞 𝑘 is associated with the difference signal
𝑑 𝑘 which is generally much smaller than 𝑚 𝑘 12
ANALYSIS OF DPCM CONT’D
To determine the improvement in DPCM over PCM, let 𝑚𝑝 and 𝑑𝑝
be the peak amplitudes of m(t) and d(t) , respectively.
If we use the same value of L in both cases, the quantization step Δ𝜈
in DPCM is reduced by the factor 𝑑𝑝 /𝑚𝑝 .
The quantization noise power is Δ𝜈 2 Τ12 the quantization noise in
2
DPCM is reduced by the factor 𝑚𝑝 Τ𝑑𝑝 and the SNR is increased
by the same factor
The signal power is proportional to its peak value squared
(assuming other statistical properties invariant), the SNR
improvement due to prediction, 𝐺𝑝 , is at least
𝑃𝑚
𝐺𝑝 =
𝑃𝑑
Where 𝑃𝑚 and 𝑃𝑑 are the powers of m(t) and d(t) , respectively.
Notes:
For the same SNR, the bit rate for DPCM could be lower than that for PCM
by 3 to 4 bits per sample. Thus, telephone systems using DPCM can often
operate at 32 or even 24 kbit/s. 13
ADAPTIVE DPCM
Adaptive DPCM (ADPCM) can further improve the efficiency of
DPCM encoding by incorporating an adaptive quantizer at the
encoder.
Figure 2.2
For practical reasons, the number of quantization level L is fixed
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DM CONT’D
Once the quantisation operation is performed, transmission
of the signal can be achieved by sending a zero for a negative
transition, and a one for a positive transition.
Note that this means that the quantised signal must change
at each sampling point.
Figure 2.3
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DM CONT’D
The figure below illustrate the DM system and the input output
signals
Figure 2.4
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DM CONT’D
The input to the comparator is 𝑒 𝑡 = 𝑚 𝑡 − 𝑚 𝑡
where 𝑚 𝑡 is the message signal and 𝑚 𝑡 is a reference
signal
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Figure 2.6
DM CONT’D
Figure 2.7
Sol: The transmitted bit train would be 1 1 1 1 0 0 0 1 0 1 1 1 1 1 0 22
DM OUTPUT WAVEFORMS
The figure below show the original signal, the integrated signal and
the output digital data
Figure 2.8
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DM LIMITATIONS
The resulting output suffers from slope overload distortion which
can be reduced by increasing the step size
When the analog signal has a high rate of change, the DM can “fall
behind” and a distorted output occurs
This may result to granular noise as you increase the step size
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Figure 2.9
DM TRADE OFFS
Simplicity versus Quality
In order to obtain high quality DM requires very high sampling
rates, typically 20× the highest frequency of interest, as opposed to
Nyquist rate of 2×
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Figure 2.10
ADAPTIVE DELTA MODULATION (ADM)
In conventional DM the problem of keeping both quantisation noise
and slope overload noise acceptably low is solved by oversampling
i.e keeping the DM step size small and sampling at many times the
Nyquist rate
The penalty incurred is the loss of some, or all, of the saving in
bandwidth which might be expected with DM
An alternative strategy is to make the DM step size variable,
Making it larger during periods when slope overload noise would
otherwise dominate – to prevent the step overload
Making it smaller when quantisation noise might dominate – to prevent
granular noise
The use of an adaptive delta modulator required that the receiver
be adaptive also, so that the step size at the receiver changes to
match the change in ∆ at the modulator.
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ADAPTIVE DELTA MODULATION CONT’D
The figure below shows the performance of adaptive delta
modulation
Figure 2.11
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ADM CONT’D
Figure 2.12 28
COMPARISON OF PCM AND DM SYSTEMS
The principal advantage of DM over PCM is the simplicity of its
implementation
DM has worse SNR compared to PCM
PCM requires more bandwidth
In general, PCM exhibits better SNR characteristics at the same data rate.
Good voice reproduction via PCM can be achieved with 128 quantization
levels, or 7-bit coding (27 = 128).
A voice signal, conservatively, occupies a bandwidth of 4 kHz. Thus,
according to the sampling theorem, samples should be taken at a rate of
8000 samples per second. This implies a data rate of 8000 7 = 56 kbps for
the PCM-encoded digital data.
PCM is more preferred than DM for analog signals
Because repeaters are used instead of amplifiers, there is no cumulative
noise
The conversion to digital signaling allows the use of the more efficient
digital switching techniques.
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SNR CONSIDERATIONS IN DM SYSTEMS
Consider a sinusoid signal
𝑚 𝑡 = 𝐴 cos 𝜔𝑚 𝑡
In DM systems slope overload distortion will occur if
∆ ∆ 𝑓𝑠
𝐴< =
𝜔𝑚 𝑇𝑠 2𝜋 𝑓𝑚
where 𝑓𝑠 = 1/𝑇𝑠
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EXAMPLES
Example 1
For a sinusoidal modulating signal
𝑚 𝑡 = 𝐴 cos 𝜔𝑚 𝑡 , 𝜔𝑚 = 2𝜋𝑓𝑚
𝑆 3𝑓𝑠3
(𝑆𝑁𝑅)0 = = 2𝑓
𝑁𝑞 8𝜋2 𝑓𝑚 𝑀
0
1
where 𝑓𝑠 = 𝑇 is the sampling rate and 𝑓𝑀 is the cut-off frequency of a
𝑠
low-pass filter at the output of the receiver. For no slope overload
condition we have
∆ ∆ 𝑓 31
𝐴< = (𝑓𝑠 )
𝜔𝑚 𝑇𝑠 2𝜋 𝑚
EXAMPLES CONT’D
Example 2
A DM system is designed to operate at 3 times the Nyquist rate for a
signal with a 3-kHz bandwidth. The quantizing step size is 250mV.
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EXAMPLES CONT’D
Example 3
Determine the output SNR in a DM system for a 1-kHz sinusoid,
sampled at 32kHz, without slope overload, and followed by a 4-kHz
postreconstruction filter.
𝑆 3𝑓𝑠3
(𝑆𝑁𝑅)0 = = 2𝑓
𝑁𝑞 8𝜋2 𝑓𝑚 𝑀
0
Example 4
The data rate of Example 2 is 32 kb/s, which is the same bit rate
obtained by sampling at 8 kHz with 4 b per sample in a PCM
system. Find the average output SNR of a 4-b PCM quantizer for the
sampling of full scale sinusoid with fs = 8 kHz, and compare it with
the results of Example 2
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