CSP1027 Datasheet, PDF (6/64 Pages) Agere Systems – CSP1027 Voice Band Codec for Cellular Handset and Modem Applications

CSP1027 Voice Band Codec for
Cellular Handset and Modem Applications
Data Sheet
December 1999
4 Architectural Information (continued)
4.1 Overview
The CSP1027 is a complete analog-to-digital and digi-
tal-to-analog acquisition and conversion system (see
Figure 3 on page 5) that band limits and encodes ana-
log input signals into 16-bit PCM, and takes 16-bit PCM
inputs and reconstructs and filters the resultant analog
output signal. The selectable A/D input circuits, pro-
grammable sample rates, and digital filter options allow
the user to optimize the codec configuration for either
speech coding or voice band data communications.
The on-chip digital filters meet the ITU-T G.712 voice
band frequency response and signal to distortion plus
noise specifications and are suitable for IS-54, GSM,
and JDC digital cellular applications. In addition, the
small supply current drain, when powered down,
extends battery life in mobile communication applica-
tions.
The CSP1027 is intended for both voice band voice and
data communication systems. As a result, this codec
has a variety of features not found in standard voice
band codecs:
s 3.0 V regulated power supply for a condenser micro-
phone.
s Microphone preamplifier with programmable input
ranges.
s Mute control of D/A output.
s Programmable output gain in 3 dB increments.
s Output speaker driver.
s Programmable master clock divider to set A/D and
D/A conversion rate.
s Testability loopback mode.
s High-quality dither scheme to eliminate idle channel
tones.
4.2 Description of Signal Paths
4.2.1 Sampling Frequency
The oversampling ratio of the codec is 125:1; this is the
ratio of the frequency of the oversampling clock to the
frequency of the sampling clock. Most speech applica-
tions specify a sampling frequency of 8 kHz, yielding an
oversampling frequency of 8 kHz x 125 = 1.0 MHz. The
codec will operate at sampling frequencies up to
24 kHz, with the frequency response of the digital filters
being changed proportionally. For this architectural
description, the sampling frequency, fS, is assumed to
be 8 kHz, with an oversampling frequency, fOS, of
1 MHz, unless otherwise stated.
6
4.2.2 Analog-to-Digital Path
The analog-to-digital (A/D) conversion signal path (see
Figure 3 on page 5) begins with the analog input driving
the input block. The signal from the input block is then
encoded by a second-order ∆-Σ modulator A/D. The
bulk of the antialiasing filtering is done in the digital
domain in two stages following the ∆-Σ modulator to
give a 16-bit result. The blocks will next be covered in
more detail.
4.2.3 Analog Input Block
The A/D input block operates in two modes: when the
external input gain select (EIGS) pin is low or left
unconnected, the input goes through a preamplifier and
is band limited by a second-order 30 kHz low-pass anti-
aliasing filter (see Figure 4 on page 7). When EIGS is
high, external resistors, Rin and Rfb, are used to set the
gain of an inverting amplifier (see Figure 5 on page 7).
These resistors, in combination with Cin and Cfb, cre-
ate a bandpass antialiasing filter. Note that EIGS is a
digital pin whose input levels are relative to digital
power and ground (VDD and VSS).
4.2.4 A/D Modulator and Digital Filters
A second-order ∆-Σ modulator quantizes the analog
signal to 1 bit (see Figure 3 on page 5). At the same
time, the resulting quantization noise is shaped such
that most of this noise lies outside of the baseband.
The modulator output is then digitally low-pass filtered
to remove the out-of-band quantization noise. After this
filtering, the output samples are decimated down to the
output sampling frequency. In the CSP1027, the filter-
ing and decimation are completed in two stages. The
first-stage low-pass filter shapes the modulator output
according to the sinc-cubic transfer function:
H(z) =
--1----
25
×
(_---1-----–_----z----^–_--²_---⁵_---)
(1 – z^–1)
3
The output sampling frequency of the sinc-cubic filter is
reduced by a factor of 25 from 1 MHz to 40 kHz. The
sinc-cubic filter places nulls in the frequency response
at multiples of 40 kHz, and removes most of the quanti-
zation noise above 20 kHz so that very little energy is
aliased as a result of the decimation.
The sinc-cubic filter output is then processed by a
seventh-order IIR digital low-pass filter. This filter
removes the out-of-band quantization noise between
3.4 kHz and 20 kHz, compensates for the passband
droop caused by the sinc-cubic decimator, and deci-
mates the sampling frequency by a factor of five from
40 kHz to 8 kHz.
Lucent Technologies Inc.