Freescale Semiconductor, Inc.
RO +
RO -
FREQ
PI
FSR
PO -
-
+
BCLKR
Mu/A
SEQUENCE
AND
CONTROL
1.575 V
REF
1
2.4 V
REFERENCE
PDI
MCLK
BCLKT
FST
DAC
RECEIVE
SHIFT
REGISTER
DR
SHARED
DAC
PO +
-1
VDD
VSS
Freescale Semiconductor, Inc...
VAG
TG
TI -
TI +
-
+
FREQ
ADC
TRANSMIT
SHIFT
REGISTER
DT
Figure 1. MC14LC5480 PCM Codec–Filter Block Diagram
DEVICE DESCRIPTION
A PCM Codec–Filter is used for digitizing and reconstruct-
ing the human voice. These devices are used primarily for
the telephone network to facilitate voice switching and trans-
mission. Once the voice is digitized, it may be switched by
digital switching methods or transmitted long distance (T1,
microwave, satellites, etc.) without degradation. The name
codec is an acronym from ‘‘COder’’ for the analog–to–digital
converter (ADC) used to digitize voice, and ‘‘DECoder’’ for
the digital–to–analog converter (DAC) used for reconstruct-
ing voice. A codec is a single device that does both the ADC
and DAC conversions.
To digitize intelligible voice requires a signal–to–distortion
ratio of about 30 dB over a dynamic range of about 40 dB.
This may be accomplished with a linear 13–bit ADC and
DAC, but will far exceed the required signal–to–distortion
ratio at larger amplitudes than 40 dB below the peak ampli-
tude. This excess performance is at the expense of data per
sample. Two methods of data reduction are implemented by
compressing the 13–bit linear scheme to companded
pseudo–logarithmic 8–bit schemes. The two companding
schemes are: Mu–255 Law, primarily in North America and
Japan; and A–Law, primarily used in Europe. These com-
panding schemes are accepted world wide. These compand-
ing schemes follow a segmented or ‘‘piecewise–linear’’ curve
formatted as sign bit, three chord bits, and four step bits. For
a given chord, all sixteen of the steps have the same voltage
weighting. As the voltage of the analog input increases, the
four step bits increment and carry to the three chord bits
which increment. When the chord bits increment, the step
bits double their voltage weighting. This results in an effec-
tive resolution of six bits (sign + chord + four step bits) across
a 42 dB dynamic range (seven chords above 0, by 6 dB per
chord).
In a sampling environment, Nyquist theory says that to
properly sample a continuous signal, it must be sampled at a
frequency higher than twice the signal’s highest frequency
component. Voice contains spectral energy above 3 kHz, but
its absence is not detrimental to intelligibility. To reduce the
digital data rate, which is proportional to the sampling rate, a
sample rate of 8 kHz was adopted, consistent with a band-
width of 3 kHz. This sampling requires a low–pass filter to
limit the high frequency energy above 3 kHz from distorting
the in–band signal. The telephone line is also subject to
50/60 Hz power line coupling, which must be attenuated
from the signal by a high–pass filter before the analog–to–
digital converter.
The digital–to–analog conversion process reconstructs a
staircase version of the desired in–band signal, which has
spectral images of the in–band signal modulated about the
sample frequency and its harmonics. These spectral images
are called aliasing components, which need to be attenuated
to obtain the desired signal. The low–pass filter used to at-
tenuate these aliasing components is typically called a re-
construction or smoothing filter.
The MC14LC5480 PCM Codec–Filter has the codec, both
presampling and reconstruction filters, a precision voltage
reference on–chip, and requires no external components.
For More Information On This Product,
Go to: www.freescale.com
FREESCALE [ FREESCALE SEMICONDUCTOR, INC ]
