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Polytechnic University of the Philippines

College of Engineering
Electronics and Communications Engineering Department
3rd floor, CEA bldg. Anonas Street, corner Pureza NDC compound,
Sta. Mesa Manila

Differential PCM Modulation

Delta PCM Modulation
Adaptive Delta PCM Modulation

In Partial Fulfillment of the
Requirements for Digital Communications

Billones, Liza May E.


Engr. Christian Lazana


“Differential Pulse Code Modulation”, is a modulation technique invented by

the British Alec Reeves in 1937. It is a digital representation of an analog
signal where the magnitude of the signal is sampled regularly at uniform
intervals. Every sample is quantized to a series of symbols in a digital code,
which is usually a binary code.

PCM is used in digital telephone systems. It is also the standard form for
digital audio in computers and various compact disc formats. Several PCM
streams may be multiplexed into a larger aggregate data stream. This
technique is called Time-Division Multiplexing, or (TDM). TDM was invented
by the telephone industry, but today the technique is an integral part of
many digital audio workstations such as Pro Tools. In conventional PCM, the
analog signal may be processed (example by amplitude compression) before
being digitized. Once the signal is digitized, the PCM signal is not subjected
to further processing such as digital data compression.

Some forms of PCM combine signal processing with coding. Older versions of
these systems applied the processing in the analog domain as part of the
A/D process; newer implementations do so in the digital domain. These
simple techniques have been largely rendered obsolete by modern
transform-based signal compression techniques.

Differential (or Delta) pulse-code modulation encodes the PCM values as

differences between the current and the previous value. For audio this type
of encoding reduces the number of bits required per sample by about 25%
compared to PCM. Adaptive DPCM is a variant of DPCM that varies the size of
the quantization step, to allow further reduction of the required bandwidth
for a given signal-to-noise ratio.

A natural refinement of this general approach is to predict the current

sample based on the previous M samples utilizing linear prediction (LP),
where LP parameters are dynamically estimated. Block diagram of a DPCM
encoder and decoder is shown below. Part (a) shows DPCM encoder and part
(b) shows DPCM decoder at the receiver.


Delta modulation (DM or Δ-modulation) is an analog-to-digital and digital-to-

analog signal conversion technique used for transmission of voice
information where quality is not of primary importance. DM is the simplest
form of differential pulse-code modulation (DPCM) where the difference
between successive samples is encoded into n-bit data streams. In delta
modulation, the transmitted data is reduced to a 1-bit data stream.

Its main features are:

1. The analog signal is approximated with a series of segments.

2. Each segment of the approximated signal is compared to the original

analog wave to determine the increase or decrease in relative

3. The decision process for establishing the state of successive bits is

determined by this comparison.
4. Only the change of information is sent, that is, only an increase or
decrease of the signal amplitude from the previous sample is sent
whereas a no-change condition causes the modulated signal to remain
at the same 0 or 1 state of the previous sample.

To achieve high signal-to-noise ratio, delta modulation must use

oversampling techniques, that is, the analog signal is sampled at a rate
several times higher than the Nyquist rate.

Derived forms of delta modulation are continuously variable slope delta

modulation, delta-sigma modulation, and differential modulation. The
Differential Pulse Code Modulation is the super set of DM.

The DM codifies each signal sample using a single bit, which is determined
from the previous sample's value and the current one. The bit specifies
whether the new sample is higher than the previous one, hence only
describing the variation in the information (and not its contents). The
resulting set of bits can be drawn as a staircase that approximates to the
original signal, being 1 a rise of the stair and a 0 a descent of it. Remember
that the number of samples per second is determined by the Nyquist

For example, consider a standard 4 KHz voice channel. You need to take
8000 samples per second to preserve the original signal. Given this and
using the DM, you can codify the signal with 8000 bps or 8 Kbps. OTOH, if
you were using PCM with 256 levels (8 bits each), the same signal could
occupy 8000 * 8 bps or 64 Kbps. It is clear that the DM needs to transmit a
lot less data than PCM. But, as you can imagine, DM doesn't provide any
quality: it does not adapt well to signals with sudden changes (such as a
shouting voice), although it might work for monotonous voices. On the
opposite side, PCM with 256 levels provides very good quality.


Adaptive DPCM (ADPCM) is a variant of DPCM (differential pulse-code
modulation) that varies the size of the quantization step, to allow further
reduction of the required bandwidth for a given signal-to-noise ratio.
Typically, the adaptation to signal statistics in ADPCM consists simply of an
adaptive scale factor before quantizing the difference in the DPCM encoder.

ADPCM was developed in the early 1970s at Bell Labs for voice coding, by P.
Cummiskey, N. S. Jayant, and James L. Flanagan.

In telephony, a standard audio signal for a single phone call is encoded as

8000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital
signal known as DS0. The default signal compression encoding on a DS0 is
either μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe
and most of the rest of the world). These are logarithmic compression
systems where a 13 or 14 bit linear PCM sample number is mapped into an 8
bit value. This system is described by international standard G.711. Where
circuit costs are high and loss of voice quality is acceptable, it sometimes
makes sense to compress the voice signal even further. An ADPCM algorithm
is used to map a series of 8 bit µ-law (or a-law) PCM samples into a series of
4 bit ADPCM samples. In this way, the capacity of the line is doubled. The
technique is detailed in the G.726 standard.

Differential PCM encodes each discrete information symbol by transmitting

only the difference between the current signal and its predecessor. For most
real-world signals, this can be far more efficient than vanilla Pulse Code
Modulation: fewer bits may be used to represent the deltas than the total

Adaptive DPCM works similarly to DPCM, but uses a pre-defined algorithm on

each side to predict the likely value of the next information symbol delta: the
difference between this prediction and the actual value is what is
transmitted. A highly accurate algorithm, especially with the output symbols
Shannon-Fano encoded, will give a highly efficient transmission method.
A simple example algorithm would be to presume that the previous delta will
be the same as the next one. For an equally simple example input, a
triangular wave, the encoding scheme works as follows:
A ^
m |
p 7| - -
l 6| -# -#
i 5| -# # #- -# # #-
t 4| -# # # # #- -# # # #
u 3| -# # # # # # #- -# # # # #
d 2| # # # # # # # # #- # # # # # #
e 1| # # # # # # # # # # #-# # # # # #
0 ----------------------------------->
The sampled input is shown as # symbols, and the predicted next output is
shown as - characters. Only after the first and second samples are sent is it
possible to predict the next sample value; the first and subsequent
predictions is accurate, with the exception of the turning points. A table of
the source, delta, prediction, and the error is shown below:
| Source | 1 2 3 4 5 6 5 4 3 2 1 2 3 4 |
| Prev. Out | - 1 2 3 4 5 6 5 4 3 2 1 2 3 |
| Delta | - 1 1 1 1 1 -1 -1 -1 -1 -1 1 1 1 |
| Predicted | - - 3 4 5 6 7 4 3 2 1 0 3 4 |
| Error | - - 0 0 0 0 -2 0 0 0 0 2 0 0 |
It is the last line which is encoded and transmitted. For this simple case,
there are only three possible outputs: the most common case is "no
difference", meaning the algorithm was correct, and at every peak or trough
in the triangle wave, there is a ±2 bit prediction error. This signal can
therefore be coded with only three symbols, and ceil (log2(3))1 = 2 bits per
source symbol (although it's possible to use alphabet extension to increase
efficiency still further).

As with differential pulse code modulation, ADPCM may be lossy or lossless,

depending on how the predictive error is encoded. CCITT's G.726 voice-
compression standard is lossy; for encoding voice or other audio, the
difference in output quality is often unnoticeable.



3. Jerry D. Gibson, Toby Berger, and Tom Lookabaugh (1998). Digital Compression for
Multimedia. Morgan Kaufmann. ISBN 9781558603691.