pack (TQFP) available or 44-pin EIAJ quad flat pack
(QFP).
■ Operates in systems with a 3.0 V to 5.0 V digital
power supply and a 5.0 V analog supply.
■ Low-power 0.9 µm CMOS technology, fully static
design, typical power of 68 mW when active and
0.05 mW in standby with a 3.3 V digital supply and a
5.0 V analog supply.
■ A low-power inactive (standby) state without stopping
clock or removing power supply.
■ Sampling rates up to 24 kHz.
■ On-chip programmable sampling clock generator
allows input clock to be an integer multiple of
125 times the sampling rate or an integer multiple of
the sampling rate.
■ Programmable phase adjust of both codec sampling
clock and baseband codec clock.
■ Two on-chip clock dividers for generating the output
clock for the baseband codec and the output clock
for other processors.
■ Regulated microphone power supply.
■ Microphone preamplifier, with programmable input
ranges of 0.16 Vp and 0.5 Vp.
■ Output amplifier, with programmable gain settings,
0 dB to –45 dB in –3 dB steps.
■ High-pass filters selectable via control registers.
■ Power-on reset pulse generator.
■ Standard 16-bit serial I/O interface.
■ Serial I/O multiprocessor mode compatible with the
Lucent Technologies Microelectronics Group’s
DSP16A and DSP1610/1616/1617/1618 Digital Signal Processors.
2 Description
The Lucent CSP1027 is a high-precision linear voiceband ∆-Σ (delta-sigma) codec designed for cellular
handset and modem applications. The device is fabricated in low-power CMOS technology and designed for
low-voltage (3.0 V to 5.0 V) digital systems. The
CSP1027 is packaged in a 44-pin EIAJ quad flat pack
(QFP) or a 48-pin EIAJ thin quad flat pack (TQFP). In
the 48-pin TQFP, the CSP1027 occupies a total volume
of 0.0784 cm
The CSP1027 has a variety of significant programmable features not found in standard voice band codecs.
The analog interface includes a microphone preamplifier with programmable gain settings, an output amplifier with gain programmable in 3 dB steps over a 45 dB
range, and a regulated microphone power supply. An
inactive mode allows a low-power standby state, and a
mute function provides suppression of the analog output. On-chip antialiasing and anti-imaging filtering
includes a selectable high-pass filter. The CSP1027
meets ITU-T G.712 voice band specifications.
The programmable features of the CSP1027 are set by
writing four on-chip control registers through the serial
I/O interface. The codec’s digital input/output uses a
linear 16-bit two’s complement data format that is also
transferred through the serial I/O interface. The
CSP1027 interfaces easily to the 16-bit serial ports of
digital signal processors and other devices. The serial
interface supports the Lucent fixed-point DSP family
serial multiprocessor mode. This allows up to eight
compatible devices, including two CSP1027s, to interface to each other on a common 4-wire bus using a
time-division-multiplexing scheme.
3
.
CSP1027 Voice Band Codec forData Sheet
Cellular Handset and Modem ApplicationsDecember 1999
3Pin Information ........................................................................................................................................... 3
4Architectural Information ............................................................................................................................ 5
6.7Analog Power and Ground ............................................................................................................. 32
7Application Information ............................................................................................................................. 33
External Input Gain Select.
—37RESNC* Reserved.
3438V
3539V
3640V
SSA
REG
DDA
PA nal og Ground.
ARegulated Output Voltage for Electrect Condenser Microphone.
PA nal og 5.0 V Power Supply.
3741AOUTNAInverting Analog Output of Output Amplifier.
3842RESNC* Reserved.
3943AOUTPANoninverting Analog Output of Output Amplifier.
4044V
SSA
PA nal og Ground.
4145MICINAAnalog Input for Microphone.
4246REFCAExternal Capacitor Connection for Internal Voltage Regulator.
4347AUXINAAnalog Input from Auxiliary.
4448V
* Indicates no connection.
† Indica tes 3-state output.
‡ Indica tes pull-up device on input.
§ Indicates pull-up resistor on input.
** Indicates pull-down device on input.
DDA
PA nal og 5.0 V Power Supply.
4
Lucent Technologies Inc.
Data SheetCSP1027 Voice Band Codec for
December 1999Cellular Handset and Modem Applications
Cellular Handset and Modem ApplicationsDecember 1999
4 Architectural Information
(continued)
4.1 Overview
The CSP1027 is a complete analog-to-digital and digital-to-analog acquisition and conversion system (see
Figure 3 on page 5) that band limits and encodes analog 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, programmable 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 applications.
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:
■ 3.0 V regulated power supply for a condenser micro-
phone.
■ Microphone preamplifier with programmable input
ranges.
■ Mute control of D/A output.
■ Programmable output gain in 3 dB increments.
■ Output speaker driver.
■ Programmable master clock divider to set A/D and
D/A conversion rate.
■ Testability loopba ck mode.
■ High-quality dither scheme to eliminate idle channel
tones.
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 antialiasing filter (see Figure 4 on page 7). When EIGS is
high, e xternal r esistor s, Rin and Rfb , are use d to set t he
gain of an inverting amplifier (see Figure 5 on page 7).
These resistors, in combination with Cin and Cfb, create a bandpass antialiasing filter. Note that EIGS is a
digital pin whose input levels are relative to digital
power and ground (V
and VSS).
DD
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 filtering and decimation are completed in two stages. The
first-stage low-pass filter shapes the modulator output
according to the sinc-cubic transfer function:
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 applications 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, f
be 8 kHz, with an oversampling frequency, f
1 MHz, unless otherwise stated.
6
, is assumed to
S
, of
OS
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 quantization 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 decimates the sampling frequency by a factor of five from
40 kHz to 8 kHz.
Lucent Technologies Inc.
Data SheetCSP1027 Voice Band Codec for
December 1999Cellular Handset and Modem Applications
4 Architectural Information
(continued)
Following the low-pass filtering and decimation to 8 kHz, the 16-bit two's complement PCM can go directly to the
output register,
cdx(A/D)
, or go to a third-order IIR digital high-pass filter and then to the output register. The –3 dB
corner frequency of the high-pass filter is approximately 270 Hz. This filter exceeds the VSELP preprocessing
requirements of IS-54 for attenuation of 60 Hz and 120 Hz signals. The high-pass filter is selected by writing the
HPFE field in the
cioc3
register (see Table 10 on page 29). The default value upon reset is the high-pass filter
enabled (HPFE = 0).
Vin1
Vin2
EXTERNAL
COMPONENTS
Cin
MICIN
Cin
AUXIN
Rin
Rin
PREAMPLIFIER
AND AA-FILTER
+
AND
FILTERS
–
A/D
5-7592 (F)
Figure 4. CSP1027 A/D Path When in the Preamplifier Mode (EIGS = 0)
AUXIN
INTERNAL
SIGNAL GND
RfbCfb
RinCin
MICIN
EXTERNAL
COMPONENTS
–
+
INTERNAL
SIGNAL GND
+
FILTERS
–
A/D
AND
5-7593 (F)
Figure 5. CSP1027 A/D Path in the External Gain Select Mode (EIGS = 1)
4.2.5 A/D Path Frequency Response
The composite digital filters (decimator, LPF, and HPF)
meet the ITU-T G.712 voice band frequency response
specifications and are suitable for IS-54, JDC, and
GSM digital cellular applications. Figures 6 through 9
show the A/D and D/A frequency response without the
optional high-pass filter (HPF). Figures 10 and 11 show
the group delay characteristics of the A/D and D/A without the high-pass filter. Figures 12 and 13 show the frequency response of the high-pass filter. Figures 14 and
15 show the group delay characteristics of the highpass filter. In all figures, the frequency is normalized to
the sampling frequency f
(i.e., frequency/fS). T o get the
S
actual frequency, multiply the normalized frequency by
f
. The absolute delay and delay distortion have been
S
normalized to the sampling period 1/f
(i.e., delay x fS).
S
Lucent T echnologies Inc.
To obtain the actual delay, divide the normalized delay
by f
. The templates shown in Figures 7 through 9,
S
11 through 13, and 15 correspond to the limits in the
ITU-T G.712 specification where f
= 8.0 kHz.
S
4.2.6 PCM Saturation Versus Analog Input Levels
16-bit two's complement saturation is employed to prevent wraparound during input overload conditions. The
saturation is hard-limiting:
0x7fff = maximum positive level
0x8000 = minimum negative level
The analog levels that correspond to the saturation lev-
els for the three input modes are outlined in T ab le 14 on
page 51.
7
CSP1027 Voice Band Codec forData Sheet
Cellular Handset and Modem ApplicationsDecember 1999
4 Architectural Information
–20
–40
–60
–80
LOG MAGNITUDE (dB)
–100
–120
Figure 6. A/D or D/A Path Frequency Response Over 5.0 f
0.00.050.100.150.200.250.300.350.400.450.50
FREQUENCY (fs = 1)
Figure 15. A/D or D/A Path Group Delay Distortion (HPF Enabled)
5-7603 (F)
12
Lucent Technologies Inc.
Data SheetCSP1027 Voice Band Codec for
December 1999Cellular Handset and Modem Applications
4 Architectural Information
(continued)
4.2.7 Digital-to-Analog Path
Starting at the bottom right of Figure 3 on page 5, the
∆-Σ D/A conversion process begins with a 16-bit two's
complement PCM signal read from the DI serial input.
The PCM is interpolated up to 1 MHz in two stages and
low-pass filtered at each stage to attenuate 8 kHz
images.
The PCM input is latched into the
cdx(D/A)
register at
a nominal word rate of 8 kHz. The signal is then optionally high-pass filtered. This filter has the same transfer
function as the A/D high-pass filter.
A digital sample-and-hold increases the word rate by a
factor of 5 from 8 kHz to 40 kHz. The seventh-order IIR
digital low-pass filter then removes the spectral images
between 4 kHz and 20 kHz and predistorts the passband to compensate for the filtering done during the
interpolation up to the 1 MHz word rate. The transfer
function of this low-pass filter is the same as the one
employed in the A/D converter.
The output of the low-pass filter feeds a programmable
gain adjustment block that serves as a volume control.
The gain can be changed in 3 dB increments from
0 dB to –45 dB. The attenuation level is set by writing
the OGSEL field in the
cioc0
register (see Table 7 on
page 26).
The digital modulator block further increases the word
rate by a factor of 25 from 40 kHz to 1 MHz. Through
quantization and noise shaping, the digital ∆-Σ modulator creates 1-bit output words at 1 MHz.
The modulator 1-bit output drives a structure combining
a 1-bit D/A converter and a second-order switchedcapacitor filter having a cutoff frequency of 8 kHz
(based on a 1 MHz clock). This is all shown as the D/A
block in Figure 3 on page 5.
4.3 Programmable Features
4.3.1 Active/Inactive Modes
The CSP1027 has active and inactive modes of operation which are selected by the ACTIVE field in the
cioc0
register (see Table 7 on page 26). The default
value upon reset and powerup is ACTIVE = 0 (i.e.,
inactive). In the inactive mode, the codec clocks are
disabled, data transfers by the codec are disabled, and
analog bias currents are shut off. This state is useful in
battery-powered applications when prolonged periods
of inactivity are expected. It takes approximately
600 ms for the codec to reach full steady-state performance in going from inactive to active. This is primarily
due to the charging of the large external capacitors,
C
and C
REF
useful after 100 ms.
4.3.2 Input Select
When the A/D preamplifier is selected (EIGS = 0), the
INSEL field of
the preamp input between the MICIN and AUXIN
inputs. When external gain select is used (EIGS = 1),
the INSEL field has no effect.
4.3.3 A/D Input Ranges
When the preamplifier is used (EIGS = 0), the IRSEL
field of the
selects the 500 mVp range when IRSEL = 0 and the
160 mVp range when IRSEL = 1. IRSEL has no effect
when the external gain select mode is used (EIGS = 1).
When EIGS = 1, the inverting amplifier of Figure 5 on
page 7 replaces the preamplifier. The input range in
this mode is the following:
. However, the codec is functionally
REG
cioc0
(see Table 7 on page 26) switches
cioc0
register (see T able 7 on page 26)
Rin
--------- -
FULL-SCALE
V
=
Rfb
1.578 V p
×
This is followed by a second-order active Chebychev filter having a cutoff frequency of 35 kHz.
The passband ripple of the analog filters is small
enough such that they have virtually no effect on the
passband response.
The output amplifier buffers the analog filter output.
The frequency responses of the A/D and D/A paths are
essentially the same. See Figures 6 through 15 for the
magnitude and delay responses versus frequency.
Lucent T echnologies Inc.
4.3.4 Output Mute Function
The D/A converter output can be selectively muted with
the MUTE field in the
cioc0
register (see Table 7 on
page 26). The default value upon reset is muted
(MUTE = 0). The mute function is implemented (Figure
3 on page 5) internally by a MUX following the D/A
input. Placing the mute function here causes the signal
at the analog output to gradually decay/rise over
approximately 1 ms upon muting/unmuting. This effect
is due to the impulse response and group delay of the
digital filters. This implementation will reduce any
potentially undesirable transient effects such as pops,
when the D/A is muted.
13
CSP1027 Voice Band Codec forData Sheet
Cellular Handset and Modem ApplicationsDecember 1999
4 Architectural Information
(continued)
4.3.5 Output Gains
The D/A converter output can be programmed in 3 dB
increments with the OGSEL field in the
cioc0
register
(see Table 7 on page 26) to serve as a volume control.
4.3.6 Loopback Mode
The codec has a programmable loopback mode, represented by the TEST field in the
cioc0
register, (see
Table 7 on page 26). As shown in Figure 3 on page 5,
when TEST = 0, the codec is in its normal mode of
operation. When TEST = 1, the loopback mode is activated. In loopback mode, the 1-bit PDM output signal
from the analog modulator is received by the analog
demodulator. At the same time, the 1-bit signal output
from the digital modulator is received by the sinc-cubic
filter in the A/D. This results in the analog input being
looped back to the analog output through the A/D and
D/A, and the digital input being looped back to the digital output through the digital filters. The loopback mode
can be useful for evaluating analog performance of the
codec in the target system without going through the
digital filters. This mode is also useful for evaluating the
response of the digital filters or in evaluating the read/
write functions of the
codec
and
cdx
registers without
having to provide an analog input to the A/D.
4.3.7 High-Pass Filter Select
The high-pass filter in the A/D and D/A can be enabled
or disabled with the HPFE field in the
cioc3
register
(see Table 10 on page 29).
4.3.8 Dither
A dithering scheme is employed in the CSP1027 which
decorrelates the periodic quantization noise of the D/A
modulator to make it white noise.
The DITHER field in the
cioc3
register (see Table 10
on page 29) disables this feature. The default value
upon reset is DITHER = 0 (i.e., enabled). When the
DITHER is disabled, the signal-to-noise ratio will generally be about 2 dB higher. The DITHER should be
enabled if the CSP1027 is used in an audio application,
i.e., where this device interfaces to an audio transducer. If the CSP1027 is used in an application other
than audio, such as data communications, the DITHER
can be disabled if so desired.
4.4 Power-On Reset
4.4.1 Internal
The CSP1027 has a power-on reset circuit that is
ORed internally with the inversion of the reset pin,
RSTB, to form the internal reset (see Figure 16 on
page 15). The power-on reset circuit’s inverted output
is also an output pin, PORB. The PORB can be used to
provide power-on reset to the system.
The power-on reset circuit is composed of two pulsegenerating elements, its output being the OR of the
two. One element is entirely internal and generates a
power-on pulse of 1.5 ms to 7.0 ms. The second element is composed of an input pin, PORCAP, a resistor
connected between PORCAP and V
ing input buffer. The user selects the capacitor value to
connect between PORCAP and ground that will generate a power-on pulse of desired width. The pin
PORCAP allows the user to lengthen the power-on
reset pulse to a width greater than the internal poweron element provides. The nominal value of the resistor
is 155 kΩ, and the threshold of the inverting input buffer
is 0.6 x V
. The formula that relates the power-on
DD
reset pulse delay to the PORCAP capacitor is as follows:
D
T
= –R x C x loge (1 – 0.6)
D
T
= 0.9163 x R x C
, and an invert-
DD
∆-Σ converters are popular due to their high tolerance
to component mismatch present in integrated circuit
fabrication processes. However, ∆-Σ converters may
suffer from periodic noise and spurious tone generation
(in-band and out-of-band) due to the coarse quantization and feedback of the ∆-Σ modulator. Although this
periodic noise may exist at very low levels (f or example,
at about –90 dBm), it may be very objectionable to the
listener while having virtually no impact on the resolution of the converter. The CSP1027 D/A uses a robust
dithering scheme which eliminates any potential problems due to this phenomenon.
14
Hence, to generate a 14.2 ms power-on reset pulse,
one would use a 0.1 µF capacitor connected between
PORCAP and V
SS
.
An internal power-on pulse can be initiated after poweron by writing a one to the TSTPOR field in the
cioc3
register (see T able 10 on page 29). This causes the
internal power-on pulse of 1.5 ms to 7.0 ms to be generated. The pulse resets the device and appears on the
PORB output pin.
Lucent Technologies Inc.
Data SheetCSP1027 Voice Band Codec for
December 1999Cellular Handset and Modem Applications
4 Architectural Information
PORCAP
C
EXTERNAL
COMPONENT
XOSCEN
XLO
XHI
CLK
ENABLE
OSCILLATOR
(continued)
DD
V
R
TSTPOR
POWER-ON
PULSE
GENERATOR
RSTB
Figure 16. Power-On Reset Diagram
CDIFS, CDIF0,
CDIF1, CDIF2,
CDIV3
ADJMOD,
ADJ
÷
F1
÷
ICLK
CDIV0
÷
CDIV2
1
M
U
X
0
ICLK0
CKO2
÷
CDIV1
CKO1
PORB
INTERNAL
RESET
CK
÷
125
OS
CK
5-7604 (F)
S
Lucent T echnologies Inc.
5-7560 (F)
Figure 17. Clock Generation
15
CSP1027 Voice Band Codec forData Sheet
Cellular Handset and Modem ApplicationsDecember 1999
4 Architectural Information
(continued)
4.5 Clock Generation
Figure 17 on page 15 shows the clock generation and
distribution for the CSP1027. The programmable dividers can customize the codec sample and master clock
rates for a variety of applications in addition to standard
8 kHz sampling, while allowing a range of values for
the crystal-controlled input clock. In Figure 17 on page
15, XOSCEN is a chip input to enable the crystal oscillator circuit. XLO and XHI are the two leads for the
crystal. CLK is the chip clock input if the crystal is not
used. CK
8 kHz. CK
typically 1 MHz. CKO1 and CKO2 are general-purpose
clocks brought out to chip pins. CDIV1 and CDIV2 are
programmable dividers with a range fr om 1 to 31.
CDIV0 is programmed to be 1 or 2, but extra clock
pulses can be added or subtracted at the output for one
period of time following a write to the control register
cioc1
ber of clocks is programmed by ADJMOD and ADJ and
causes a phase shift in the CKO1 and CK
is an integral or a fractional divider controlled by the
five programmable coefficients shown connected to it.
With the fractional divide, the period of CK
but the period of CK
The following discussion begins with the crystal oscillator and is followed by a detailed description of each
divisor block. Section 7.5 on page 45 provides some
examples of how to program the clocks.
4.5.1 Crystal Oscillator
The CSP1027 has a selectable on-chip clock oscillator.
A logic 1 on the XOSCEN pin enables the crystal oscillator. A logic 0 disables the oscillator, powers it down,
and selects the input buffer connected to the CLK pin.
To use the oscillator, select a 20 MHz to 30 MHz fundamental-mode crystal with a series resistance less than
60 Ω and a mutual capacitance less than 7.0 pF. Connect the crystal between the XLO and XHI pins, and
add 10 pF capacitors between XLO and ground, and
XHI and ground. The XOSCEN pin enables and disables the crystal oscillator. See the application information on optimizing the oscillator performance.
is the internal codec sample clock, typically
S
is the internal codec oversampled clock,
OS
. This one-time increase or decrease in the num-
output. F1
S
will vary,
OS
will be constant.
S
4.5.2 Clock Divider 2
ICLKCKO2
÷
CDIV2
5-7589 (F)
Figure 18. Clock Divider 2
The CDIV2 field in
cioc0
(see Table 7 on page 26) sets
the clock divider that generates the output clock,
CKO2. The clock output is a general-purpose clock that
can be used to clock external logic or processors.
CDIV2 ranges from 1 to 31, with 0 holding the output
low. RSTB going low sets CDIV2 to ÷16. CKO2 is
active while RSTB is low and synchronized by RSTB
going high.
4.5.3 Clock Divider 0
ADJMOD,
ADJ
ICLKICLK0
÷
CDIV0
5-7588 (F)
Figure 19. Clock Divider 0
The CDIV0 field in
cioc1
(see Table 8 on page 27) sets
the clock divider that generates the internal clock 0
(ICLK0) to either divide by one or divide by two. The
ADJMOD and ADJ fields in
cioc1
are used to adjust
the phase of ICLK0 by increasing or decreasing the
rate of ICLK0 for a burst of pulses, one time only. This
event occurs each time control register
cioc1
is written
with nonzero values of ADJ. For example, let CDIV0 be
set to ÷2, ADJ to seven, and ADJMOD to one
(advance). After this word is written to the
cioc1
register, seven ICLK0 pulses will occur at the same rate as
ICLK, not divided by two. These seven clock pulses
shift the phase of CK
, CKS, and CKO1 earlier, thus
OS
advancing these clocks. If ADJMOD is set to zero
(retard), the ÷2 becomes a ÷3 for seven pulses of
ICLK0. The CDIV0 clock divider is temporarily changed
internally so that it divides by one greater, to retard the
clocks, or one less, to advance the clocks, for the specified number of ICLK0 cycles. Note that the CDIV0
clock divider must be set to divide by two in order to
advance and retard the clocks. If CDIV0 clock divider is
set to divide by 1, one can only retard the clocks.
16
Lucent Technologies Inc.
Data SheetCSP1027 Voice Band Codec for
F1MS
N
125
--------- -
×
+=
fSf
OS
125÷f
ICLK
0
MS
N
125
--------- -
×+
÷
125
÷==
f
ICLK
0
f
S
----------------
125 M
×()
SN
×()+=
1
f
S
---- -
125 N
–()
M
f
ICLK
0
----------------
×
N
MS
+()
f
ICLK
0
--------------------
×+=
December 1999Cellular Handset and Modem Applications
4 Architectural Information
(continued)
CDIV0 has values of 1 or 2, ADJMOD is 0 or 1, and
ADJMOD ranges from 1 to 127, with 0 selecting no
clock adjust. RSTB going low sets CDIV0 to ÷2. ICLK0
is active while RSTB is low and synchronized by RSTB
going high.
4.5.4 Clock Divider 1
ICLK0CKO1
÷
CDIV1
5-7587 (F)
Figure 20. Clock Divider 1
The CDIV1 field in
cioc1
(see Table 8 on page 27) sets
a clock divider that generates the CKO1 output clock.
This general-purpose clock output can be used for
clocking another codec in the system, such as the
CSP1084. The ability to phase adjust the output clock
and the codec sampling clock simultaneously is an
important feature. CDIV1 ranges from 1 to 31, with 0
disabling the output. RSTB going low sets CDIV1
to ÷16. CKO1 is active while RSTB is low and synchronized by RSTB going high.
4.5.5 Sampling Clocks Generation
CDIFS, CDIF0, DSIF1,
CDIF2, CDIV3
OS
CK
The CSP1027 solves this problem in a unique way, by
providing a programmable, fractional divider, F1.
F1 is the programmable ratio between ICLK0 and
CK
. The equation for F1 is:
OS
where 3 ≤ M ≤ 64, 0 ≤ N ≤ 62, and S = {1, –1};
or M = 2, 0 ≤ N ≤ 62, and S = 1;
or M = 1, N = 0, and S = 1.
M is encoded by CDIV3 (see Table 3 on page 18), N is
encoded by CDIF0, CDIF1, and CDIF2 (see Table 5 on
page 19), and S is encoded by CDIFS (see Table 4 on
page 18).
CK
is generated by dividing CKOS by 125. The fre-
S
quency of CK
can be described by:
S
Note that when N = 0, ICLK0 is simply divided by the
integer M to create the oversampling clock, CK
OS
. This
is the preferred method for generating the sampling
clock. If N ≠ 0, the fractional division results in an oversampling clock, CK
, whose period varies with time
OS
such that the average period is the desired fraction.
This vari ation in the oversampling clock period is minimized by the clock generator but can cause distortion
in the codec. Because the denominator of the fraction
is fixed at 125, the period of the sampling clock, CK
S,
will be an integer multiple of the period of the internal
clock, ICLK0, and will not vary. This is more clearly
shown by the following equation:
ICLK0
÷
F1
÷
125
CK
S
The expanded equation below explains what is happening in the time domain:
5-7586 (F)
Figure 21. Sampling Clocks Generation
1
During each sampling period, , there are (125 – N)
The oversampling codec clock CK
is used in the front sections of the A/D and the back
, typically 1 MHz,
OS
oversampling clock cycles of period and N
sections of the D/A. The lower-frequency codec clock,
CK
, typically 8 kHz, is the sample clock at th e output
S
oversampling clock cycles of period . The N
of the A/D and the input to the D/A. The sampling clock
frequency, f
, is the oversampling clock frequency, fOS,
S
divided by 125 (the fixed oversampling ratio). The
divide by 125 must remain fixed, since it is constrained
by the architecture of the codec digital filters. Many systems, however, have fixed high-frequency clocks and
fixed sampling clocks, so it is necessary to have a great
deal of flexibility in the creation of the codec clock CK
oversampling clock cycles are evenly distributed
among the (125 – N) oversampling clock cycles to minimize the distortion due to oversampling clock cycles of
differing period. The values for CDIF[0—2] in Table 5
on page 19 have been selected to achieve the even
distribution.
.
S
Lucent T echnologies Inc.
---- -
f
S
M
----------------
f
ICLK
0
MS
()
+
------------------- -
f
ICLK
0
17
CSP1027 Voice Band Codec forData Sheet
Cellular Handset and Modem ApplicationsDecember 1999
4 Architectural Information
The procedure for selecting M, S, and N is illustrated in Section 7.5 on page 45. The ranges for the programmable
dividers are summarized in Table 2.
Table 2. Programmable Divider Summary
Clock RatioICLK/ICLK0ICLK0/CK
Variable NameCDIV0
Range of Values1, 21, 2 to 64.496125, 250 to 8062Off, 1 to 31Off, 1 to 31
Encoding0, 1See Tables 3
Table 3. CDIV3 Value for Each M
MCDIV3
100 0001
200 0010
.
.
6211 1110
6311 1111
6400 0000
.
.
(continued)
N
--------- -
M
±
125
through 5.
OS
ICLK0/CKS ICLK/CKO2ICLK0/CKO1
125M ± N
See Tables 3
through 5.
CDIV2CDIV1
0 to 310 to 31
Table 4. CDIFS Value for Each S
SCDIFS
+10
–11
18
Lucent Technologies Inc.
Data SheetCSP1027 Voice Band Codec for
December 1999Cellular Handset and Modem Applications
CSP1027 Voice Band Codec forData Sheet
Cellular Handset and Modem ApplicationsDecember 1999
4 Architectural Information
(continued)
4.6 Serial I/O Configurations
4.6.1 Codec Data Transfer
When the codec is active, ACTIVE = 1 (see Table 7 on
page 26), it loads data into the
data from the
cdx(D/A)
register (see Figure 3 on page
5) at the sampling frequency, f
on a 1 MHz oversampling frequency). The codec data
transfers occur independent of the serial input/output
data transfers described below. The data is double
buffered, allowing the codec to transfer data to or from
cdx
the
the shift registers (
while the serial I/O is shifting data into or out of
isr
and
to inactive, ACTIVE = 0, there are no codec data transfers to the
cdx(A/D)
or from the
The internal STATUS flag is set high when
loaded and
cdx(A/D)
cdx(D/A)
is emptied. Loading data from the
into the output shift register (
data from the input shift register (
due to a serial I/O transaction, clears the internal STATUS flag. The internal STATUS flag can be observed
on the data output (DO) pin in the passive mode and
causes data transfers in the active and multiprocessor
modes.
4.6.2 Codec Control Writes
The four control registers are written through the serial
port. The serial address (SADD) selects between control and data transfers. Bits 15 and 14 of the control
word being transferred select which control register,
cioc0, cioc1, cioc2
Bit[15:14] = 00,
cioc1
cioc3
, or
: Bit[15:14] = 01, etc.).
4.6.3 Serial I/O Port Overview
The CSP1027 serial I/O unit is an asynchronous, fullduplex, double-buffered channel operating at up to
20 Mbits/s that easily interfaces with other Lucent fixedpoint DSPs (i.e., DSP16A and DSP1610/1616/1617/
1618) in a single or multiple DSP environment. Commercially available codecs and time-division multiplexed (TDM) channels can be interfaced to the
CSP1027 device with little, if any, external logic.
cdx(A/D)
(which is 8 kHz based
S
osr
). When the codec is set
cdx(D/A)
and empties
.
cdx(A/D)
osr
) or loading
isr
) into the
, is written (i.e.,
cdx(D/A)
cioc0
is
:
The serial interface is a subset of the standard Lucent
DSP serial I/O and is comprised of eight pins:
■ A single passive serial input/output clock (IOCK).
■ A combined input load, output load, and synchroni-
zation (SYNC).
■ Serial data input (DI).
■ Serial data output (DO).
■ Serial address (SADD).
■ Three serial mode select pins (SMODE[2:0]).
The CSP1027's serial I/O is different from the standard
Lucent serial I/O in a number of ways:
■ The SMODE[1:0] pins configure the serial I/O port
into one of four possible ways: a passive SIO configuration, an active SIO configuration, and two multiprocessor SIO configurati ons.
■ A fixed most significant bit (MSB) first data format.
■ A fixed 16-bit data mode.
■ The serial address (SADD) is an input during the
passive and active SIO configurations to select
between data and control SIO transfers. It is
intended to be connected to the DSP’s SADD pin,
which is an output during passive and active SIO.
Note that the DSP's SADD output is inverted and is
composed of two 8-bit fields that are shifted out least
significant bit (LSB) first.
■ The multiprocessor mode time slots and serial
addresses are restricted to two sets, one of which is
selected based on the state of SMODE0.
■ The SMODE2 pin should always be tied low for the
serial I/O port to operate as described.
■ The frequency of the serial I/O interface clock input
IOCK (F
the internal oversampling clock (F
IOCK
) must be greater than the frequency of
IOCK
).
20
Lucent Technologies Inc.
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