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College of Engineering
Electronics and Communications Engineering Department
3rd floor, CEA bldg. Anonas Street, corner Pureza NDC compound,
Sta. Mesa Manila
Presented
In Partial Fulfillment of the
Requirements for Digital Communications
By:
BS ECE IV-5
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.
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
theorem.
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.
ADPCM was developed in the early 1970s at Bell Labs for voice coding, by P.
Cummiskey, N. S. Jayant, and James L. Flanagan.
1. http://everything2.com/title/Adaptive+Differential+Pulse+Code+Modulation
2. http://www.encyclopedia.com/doc/1O12-adaptivedigitalplscdmdltn.html
3. Jerry D. Gibson, Toby Berger, and Tom Lookabaugh (1998). Digital Compression for
Multimedia. Morgan Kaufmann. ISBN 9781558603691.
http://books.google.com/books?
id=aqQ2Ry6spu0C&pg=PA265&dq=G.722+adpcm+subband+split&lr=&as_brr=3&
as_pt=ALLTYPES&ei=VqFNSfbxB5CIkATU4LjJCQ.
4. http://blog.julipedia.org/2005/10/delta-modulation.html
5. http://en.wikipedia.org/wiki/Delta_modulation
6. http://www.ece.drexel.edu/commweb/dpcm/dpcm.html
7. http://www.birds-eye.net/definition/d/dpcm-
differential_pulse_code_modulation.shtml