Analog Devices AD1843 Datasheet

Serial-Port 16-Bit

a

FEATURES
Single Chip Integrated Speech, Audio, Fax and Modem

Codec

Highly Configurable Stereo ∑∆ ADCs and Quad ∑∆ DACs
Supports V.34, V.32bis, and Fallback Modem Standards

As Well As Voice Over Data

Dual Digital Resamplers with Programmable Input and

Output Phase and Frequency

Three On-Chip Phase Lock Loops for Synchronization to

External Signals, Including Video
Thirteen Analog Inputs and Seven Analog Outputs
Advanced Analog and Digital Signal Mixing and Digital-

to-Digital Sample Rate Conversion
Programmable Gain, Attenuation and Mute
On-Chip Signal Filters

Digital Interpolation and Decimation

Analog Output Low Pass
1 Hz Resolution Programmable Sample Rates from 4 kHz

to 54 kHz Derived from a Single Clock Input
80-Lead PQFP and 100-Lead TQFP Packages
Operation from +5 V or Mixed +5 V/+3 V Supplies
FIFO-Buffered Serial Digital Interface Compatible with

ADSP-21xx Fixed-Point DSPs
Advanced Power Management
VHDL Model of Serial Port Available; Evaluation Board

and MAFE Board Available

SIMPLIFIED FUNCTIONAL BLOCK DIAGRAM

4

LIN

2

MIC

2

AUX1

2

AUX2

2

AUX3

1

MIN

MOUT

LOUT1

HPOUTL
HPOUTC
HPOUTR

LOUT2

SUM

REV. 0

MUTE

MUTE

2

DRIVER

MUTE

4

2

GAM

∑ ∑ ∑ ∑ ∑

MUTE

LEFT AND

RIGHT CHANNELS

VOLTAGE REFERENCE

V

REF

Information furnished by Analog Devices is believed to be accurate and
reliable. However, no responsibility is assumed by Analog Devices for its
use, nor for any infringements of patents or other rights of third parties
which may result from its use. No license is granted by implication or
otherwise under any patent or patent rights of Analog Devices.

20 dB

GAM GAM GAM GAM

LEFT AND

RIGHT CHANNELS

GAM = GAIN
ATTENUATION
MUTE

GAM

AAFILTL AAFILTR

GAM

S
E
L
E
C
T
O
R

GAM

PGA

DAC

DAC

SoundComm Codec

AD1843

GENERAL PRODUCT DESCRIPTION

The AD1843 SoundComm™ Codec is a complete analog front
end for high performance DSP-based telephony and audio ap­plications. The device integrates the real-world analog I/O re­quirements for many popular functions thereby reducing size,
power consumption, and system complexity. The AD1843
SoundComm is the world’s first codec which can support four
different sample rates simultaneously, without any beat fre­quency noise issues. This is essential for highly integrated audio/
modem/fax products since the sample rates associated with au­dio are very much distinct from the sample rates associated with
telephony-oriented data communication. It is also the first codec
to offer on-chip digital phase lock loops for sample rate synchro­nization to external clock signals. This sample rate flexibility is
enabled through Analog Devices’ Continuous Time Oversampling
(CTO) technology.

The main elements of the AD1843 are its extensive input and mix­ing section, its two channels of sigma-delta (∑∆) analog-to-digital
conversion, its four channels of ∑∆ digital-to-analog conversion, its
digital filters, and the clock and control circuitry for implementing
the device’s different modes. The AD1843 permits flexible sample­rate selection through programming and external synchronization,
many input and output options, and many mixing options.

SoundComm is a trademark of Analog Devices, Inc.

SYNC XTAL CONV BIT

3

2

3

3

S
E
L
E
C
T
O
R

ATTN

ATTN

GNDA

CLOCK GENERATION

ATTN

MUTE

∑

4

V

CC

ATTN

MUTE

∑

3

FIFO

FIFO

GNDD

LAW

89

∑∆

∑∆

FILTLCMOUT

AD1843

∑∆

ADC

∑

MUTE

MUTE

∑

FILTR

S
E
L
E
C
T
O
R

M
U
T
E

M
U
T
E

One Technology Way, P.O. Box 9106, Norwood, MA 02062-9106, U.S.A.
Tel: [phone] Fax: [phone]

(continued on page 11)

CONTROL

REGISTERS

µ/A

ADC

D

I

G

I
T
A

µ/A

LAW

µ/A

LAW

V

L

DAC1

I
N
T
E

2
R
F
A
C
E

DAC2

DD

© Analog Devices, Inc., 1996

CLKOUT

SCLK
SDFS

SDI
SDO
BM

CS

TSO
TSI

XCTL [1:0]

PDMNFT

RESET

PWRDWN

AD1843–SPECIFICATIONS

STANDARD TEST CONDITIONS UNLESS OTHERWISE NOTED

Temperature 25 °C
Digital Supply (V
Analog Supply (V
Sample Rate (F

) 5.0 V

DD

) 5.0 V

CC

) 48 kHz

S

Input Signal 1008 Hz
Analog Output Passband 20 Hz to 20 kHz
ADC FFT Size 2048
DAC FFT Size 8192
V

IH

V

IL

V

OH

V

OL

I

OH

I

OL

ANALOG INPUT

2.0 V

0.8 V

2.4 V

0.4 V
–2 mA
2mA

DAC Conditions

Autocalibrated
0 dB Attenuation
0 dB Relative to Full Scale
16-Bit Linear Mode
No Output Load
Mute Off
DAC1 Single-Ended
DAC2 Differential

Min Typ Max Units

Full-Scale Input Voltage (RMS Values Assume Sine Wave Input)

All Inputs with ADRFLT & ADLFLT = 0 and LINLSD & LINRSD = 0 1 V rms
(LINLP, LINRP, AUX1L, AUX1R, AUX2L, 2.55 2.828 3.1 V p-p
AUX2R, AUX3L, AUX3R, MIN)
All Inputs with ADRFLT & ADLFLT = 0 and LINLSD & LINRSD = 1 2 V rms
(LINLP & LINLN, LINRP & LINRN) 5.1 5.656 6.2 V p-p
All Inputs with ADRFLT & ADLFLT = 1 and LINLSD & LINRSD = 0 1.127 V rms
(LINLP, LINRP, AUX1L, AUX1R, AUX2L, AUX2R, AUX3L, 2.8 3.156 3.5 V p-p
AUX3R, MIN)
All Inputs with ADRFLT & ADLFLT = 1 and LINLSD & LINRSD = 1 2.254 V rms
(LINLP & LINLN, LINRP & LINRN) 5.6 6.312 7.0 V p-p
MIC with +20 dB Gain (LMGE & RMGE = 1

and ADRFLT & ADLFLT = 0) 0.1 V rms
(MICL, MICR) 0.25 0.2828 0.31 V p-p
MIC with 0 dB Gain (LMGE & RMGE = 0

and ADRFLT & ADLFLT = 0) 1 V rms
(MICL, MICR) 2.55 2.828 3.1 V p-p

AUX, SUM and MIN Input Impedance* 10K Ω

(AUX1L, AUX1R, AUX2L, AUX2R, AUX3L, AUX3R, SUML,
SUMR, MIN)

LIN Input Impedance* (LINLP, LINLN, LINRP, LINRN) 40K Ω
MIC Input Impedance* (MICL, MICR) 20K Ω
Input Capacitance* (All Inputs) 15 pF

ADC Input Conditions

Mic 20 dB Gain Disabled
LIN Single-Ended

(LINLSD & LINRSD = 0)
Autocalibrated
0 dB PGA Gain
–1.0 dB Relative to Full Scale
Line Input
16-Bit Linear Mode

PROGRAMMABLE GAIN AMPLIFIER–ADC

Min Typ Max Units

Step Size (0 dB to 22.5 dB) (All Steps Tested) 1.3 1.5 1.7 dB
PGA Gain Range Span* 21.5 22.5 23.5 dB

INPUT (AUX1, AUX2, AUX3, MIN, MIC)
ANALOG AMPLIFIERS/ATTENUATORS

Min Typ Max Units

Step Size (+12.0 dB to –30 dB) (All Steps Tested) 1.25 1.5 1.75 dB
Step Size (–31.5 dB to –34.5 dB) (All Steps Tested) 1.1 1.5 1.9 dB
Input Gain/Attenuation Range* 45.5 46.5 47.5 dB
Mute Attenuation* –80.0 dB

REV. 0–2–

DIGITAL DECIMATION AND INTERPOLATION FILTERS–AUDIO MODE*

AD1843

Min Max Units

Passband 0 0.40 × F

Hz

S

Passband Ripple 0 –0.016 dB
Transition Band 0.4 × F
Stopband

1

0.6 × F

0.6 × F

S
S

Hz

S

Hz
Stopband Rejection 91.8 dB
Group Delay 15/F

S

s
Group Delay Variation Over Passband 0.0 µs

DIGITAL DECIMATION AND INTERPOLATION FILTERS–MODEM MODE*

Min Max Units

Passband 0 0.442 × F

Hz

S

Passband Ripple 0 –0.220 dB
Transition Band 0.442 × F
Stopband

2

0.542 × F

0.542 × F

S
S

Hz

S

Hz
Stopband Rejection 75.7 dB
Group Delay 19/F

S

s
Group Delay Variation Over Passband 0.0 µs
Sample Rate 24 kHz

DIGITAL DECIMATION AND INTERPOLATION FILTERS–RESAMPLER MODE*

Min Max Units

Passband 0 0.4 × F

Hz

S

Passband Ripple 0 –0.035 dB
Transition Band 0.4 × F
Stopband

3

0.5 × F

0.5 × F

S
S

Hz

S

Hz
Stopband Rejection 92.2 dB
Group Delay 25/F

S

s
Group Delay Variation Over Passband 0.0 µs

ANALOG-TO-DIGITAL CONVERTERS

Min Typ Max Units

Audio Dynamic Range (–60 dB Input, THD+N Referenced to Full Scale,

A-Weighted, ADRFLT & ADLFLT = 0) 80 85 dB

Modem Dynamic Range (–60 dB Input, THD+N Referenced to Full Scale,

300 Hz to 4 kHz Analog Output Passband, LINRSD & LINLSD = 1,
ADRFLT & ADLFLT = 1, F

= 12.8 kHz) 87 90 dB

S

Audio THD+N (Referenced to Full Scale) 0.03 %

–74 –70 dB

Modem THD+N (–3.0 dB Referenced to Full Scale,

300 Hz to 4 kHz Analog Output Passband, LINRSD & LINLSD = 1,
ADRFLT & ADLFLT = 1, F

= 12.8 kHz) 0.02 %

S

–78.5 –74 dB
Audio Signal-to-Intermodulation Distortion* (CCIF Method) –94 –80 dB
ADC Crosstalk*

LIN Inputs (Input L, Ground R, Read R; Input R, Ground L, Read L) –80 dB
Line to MIC (Input LIN, Ground and Select MIC, Read Both Channels) –80 dB
Line to AUX1, AUX2, AUX3, MIN –80 dB

Interchannel Gain Mismatch (Difference of Gain Errors) ±0.5 dB
ADC Offset Error 10 50 mV

REV. 0 –3–

AD1843

DAC1 DIGITAL-TO-ANALOG CONVERTERS

Min Typ Max Units

Audio Dynamic Range (–60 dB Input, THD+N Referenced to Full Scale,

A-Weighted, DA1FLT = 0) 77 80 dB

Audio THD+N (Referenced to Full Scale, DA1FLT = 0) 0.03 %

–74 –70 dB

Audio Signal-to-Intermodulation Distortion* (CCIF Method) –92 –80 dB
Interchannel Gain Mismatch (Difference of Gain Errors) ±0.5 dB
DAC Crosstalk* (Input L, Zero R, Measure LOUT1R; Input R, Zero L,

Measure LOUT1L) –77 dB

Total Out-of-Band Energy*

(Measured from 0.6 × F

Audible Out-of-Band Energy*

(Measured from 0.6 × F
Tested at FS = 8.0 kHz) –72 dB

DAC2 DIGITAL-TO-ANALOG CONVERTERS

Audio Dynamic Range (–60 dB Input, THD+N Referenced to Full Scale,

A-Weighted, DA2FLT = 0) 78 80 dB

Modem Dynamic Range (–60 dB Input, THD+N Referenced to Full Scale,

300 Hz to 4 kHz Analog Output Passband, DA2FLT = 1, RDA2G5:0
& LDA2G5:0 = 000101 [4.5 dB], F

Audio THD+N (Referenced to Full Scale, DA2FLT = 0) 0.03 %

Modem THD+N (–3.0 dB Referenced to Full Scale,

300 Hz to 4 kHz Analog Output Passband, DA2FLT = 1, RDA2G5:0
& LDA2G5:0 = 000101 [4.5 dB], F

Audio Signal-to-Intermodulation Distortion* (CCIF Method) –86 –80 dB
Interchannel Gain Mismatch (Difference of Gain Errors) ±0.5 dB
DAC Crosstalk* (Input L, Zero R, Measure LOUT2R; Input R, Zero L,

Measure LOUT2L) –80 dB

Total Out-of-Band Energy*

(Measured from 0.6 × F

Audible Out-of-Band Energy*

(Measured from 0.6 × F
Tested at F

= 8.0 kHz) –72 dB

S

DC Offset 525mV

to 100 kHz in Audio Mode) –60 dB

S

to 22 kHz in Audio Mode,

S

Min Typ Max Units

= 12.8 kHz) 87 90 dB

S

–77 –70 dB

= 12.8 kHz) 0.016 %

S

–81 –76 dB

to 100 kHz in Audio Mode) –60 dB

S

to 22 kHz in Audio Mode,

S

DAC1 AND DAC2 ANALOG AMPLIFIERS/ATTENUATORS

Min Typ Max Units

Step Size (+12.0 dB to –30.0 dB) (All Steps Tested) 1.25 1.5 1.75 dB
Step Size (–31.5 dB to –34.5 dB) (All Steps Tested) 1.1 1.5 1.9 dB
Step Size (–36.0 dB to –82.5 dB)* 1.3 1.5 1.7 dB
Output Attenuation Span* 81.5 82.5 83.5 dB
Mute Attenuation* –80 dB

DIGITAL MIX ATTENUATORS

Min Typ Max Units

Step Size (0 dB to –94.5 dB)* (All Steps Tested) 1.3 1.5 1.7 dB
Output Attenuation Span* 93.5 94.5 95.5 dB
Mute Attenuation* –90 dB

REV. 0–4–

AD1843

ANALOG OUTPUT

Min Typ Max Units

LOUT1 Full-Scale Output Voltage 0.707 V rms

(RMS Values Assume Sine Wave Input) 1.8 2.0 2.2 V p-p

LOUT2 Full-Scale Single-Ended Output Voltage 0.707 V rms

(RMS Values Assume Sine Wave Input) 1.8 2.0 2.2 V p-p

LOUT2 Full-Scale Differential Output Voltage 1.414 V rms

(RMS Values Assume Sine Wave Input) 3.6 4.0 4.4 V p-p

LOUT1 Output Impedance* 600 Ω
LOUT2 Output Impedance* 1 Ω
LOUT1 External Load Impedance* 10 kΩ
LOUT2 External Load Impedance* 2 kΩ
MOUT External Load Impedance* 10 kΩ
HPOUT External Load Impedance* 16 32 Ω
HPOUT THD+N (Referenced to Full Scale, 32 Ω External Load Impedance) 0.10 %

Output Capacitance* 15 pF
External Load Capacitance* 100 pF
CMOUT 2.10 2.25 2.40 V
External CMOUT Load Current* 10 µA
CMOUT Output Impedance* 4 kΩ
Mute Click* (Muted Output Minus Unmuted Midscale DAC1 and DAC2 Outputs) ±5mV

SYSTEM SPECIFICATIONS

System Frequency Response Ripple* (Line-In to Line-Out) 1.0 dB
Differential Nonlinearity* ±1 Bit
Phase Linearity Deviation* 5 Degrees

–60 dB

Max Units

STATIC DIGITAL SPECIFICATIONS

Min Max Units

High-Level Input Voltage (V

Digital Inputs, Except SCLK 2.0 V

XTALI and SCLK 2.4 V
Low-Level Input Voltage (V
High-Level Output Voltage (V
Low-Level Output Voltage (V

)

IH

) –0.3 0.8 V

IL

) 2.4 V

OH

) 0.4 V

OL

+ 0.3 V

DD

+ 0.3 V

DD

Input Leakage Current (GO/NOGO Tested) –10 10 µA
Output Leakage Current (GO/NOGO Tested) –10 10 µA

TIMING PARAMETERS (GUARANTEED OVER OPERATING TEMPERATURE AND DIGITAL SUPPLY RANGE)

Min Typ Max Units

Serial Data Frame Sync [SDFS] Period (t

)

1

(Master Mode, FRS = 1 [16 Slots per Frame], SCF = 0 [SCLK = 12.288 MHz]) 20.833 µs
Frame Sync [SDFS] HI Pulse Width (t
Clock [SCLK] to Frame Sync [SDFS] Propagation Delay (t
Data [SDI] Input Setup Time to SCLK (t
Data [SDI] Input Hold Time from SCLK (t
Clock [SCLK] to Output Data [SDO] Valid (t
Clock [SCLK] to Output Data [SDO] Three-State [High-Z] (t
Clock [SCLK] to Time Slot Output [TSO] Propagation Delay (t
RESET and PWRDWN LO Pulse Width (t

SCLK

SDFS

SDI

SDO

t

2

t

tSt

PD1

H

BIT 15 BIT 14 BIT 0

t

DV

)80ns

2

)10ns

S

)10 ns

H

)15ns

DV

) 100 ns

RPWL

BIT 0BIT 14BIT 15

)15ns

PD1

)15ns

HZ

)15ns

PD2

SCLK

t

SDFS

t

PD1

t

HZ

SDI OR SDO

TSO

RESET

PWRDWN

151413

t

RPWL

1

3 2 1 0 151413

LAST
VALID

TIME SLOT

t

PD2

151413

Figure 1. Timing Diagrams

REV. 0 –5–

AD1843

WARNING!

ESD SENSITIVE DEVICE

POWER SUPPLY (33 Ω HPOUT LOAD)

Min Typ Max Units

Power Supply Range—Analog V
Power Supply Range—Digital V
Total Power Supply Current—5.0 V

(5.0 V

and V

CC

Supplies) 210 250 mA

DD

Total Power Supply Current—5.0 V

(5.0 V

Analog/3.0 VDD Digital Supplies) 150 175 mA

CC

Analog Supply Current—5.0 V
Digital Supply Current—5.0 V
Digital Supply Current—3.0 V
Digital Power Supply Current—V
Analog Power Supply Current—V
Power Dissipation—5.0 V
Power Dissipation—5.0 V
Power Dissipation—5.0 V

CC

DD

and VDD Operating

CC

/3.0 VDD Operating*

CC

Operating 60 75 mA

CC

Operating 150 175 mA

DD

Operating* 90 100 mA

DD

Power Down (PWRDWN LO) 1 mA

DD

Power Down (PWRDWN LO) 0.5 mA

CC

and V

CC

/3.0 V

CC

and VDD Power Down (PWRDWN LO)

CC

Operating (Current × Nominal Supply) 1250 mW

DD

Operating* (Current × Nominal Supply) 875 mW

DD

(Current × Nominal Supply) 7.5 mW

Power Dissipation—5.0 V

/3.0 VDD Power Down* (PWRDWN LO)

CC

(Current × Nominal Supply) 5mW

Power Supply Rejection (100 mV p-p Signal @ 1 kHz)* 40 dB

(At Both Analog and Digital Supply Pins, for ADC, DAC1 and DAC2)

CLOCK SPECIFICATIONS*

Input Crystal/Clock Frequency 24.576 MHz
Input Clock Duty Cycle (When an External Clock Is Used Instead of a Crystal) 25/75 75/25 %
Initialization Sample Rate Change Time 0 ms

PACKAGE CHARACTERISTICS

PQFP θ
PQFP θ
TQFP θ

(Thermal Resistance [Junction-to-Ambient]) 96 °C/W

JA

(Thermal Resistance [Junction-to-Case]) 8.75 °C/W

JC

(Thermal Resistance [Junction-to-Ambient]) 30.6 °C/W

JA

TQFP θJC (Thermal Resistance [Junction-to-Case]) 4.6 °C/W

NOTES

1

The stopband repeats itself at multiples of 64 × FS, where FS is the sampling frequency. Thus the audio mode digital filter will attenuate to –91.8 dB or better across

the frequency spectrum except for a range of ±0.6 × F

2

The stopband repeats itself at multiples of 64 × FS, where FS is the sampling frequency. Thus the modem mode digital filter will attenuate to –75.7 dB or better across

the frequency spectrum except for a range of ±0.542 × F

3

The stopband repeats itself at multiples of 64 × FS, where FS is the sampling frequency. Thus the resampler mode digital filter will attenuate to –92.2 dB or better

across the frequency spectrum except for a range of ±0.5 × F

*Guaranteed, not tested.
Specifications subject to change without notice.

ABSOLUTE MAXIMUM RATINGS*

Min Max Units

Power Supplies

Digital (V
Analog (V

) –0.3 6.0 V

DD

) –0.3 6.0 V

CC

wide at multiples of 64 × FS.

S

wide at multiples of 64 × FS.

S

wide at multiples of 64 × FS.

S

*Stresses greater than those listed under “Absolute Maximum Ratings” may cause

permanent damage to the device. This is a stress rating only and functional
operation of the device at these or any other conditions above those indicated in the
operational section of this specification is not implied. Exposure to absolute
maximum rating conditions for extended periods may affect device reliability.

Input Current

(Except Supply Pins) ±10.0 mA

Analog Input Voltage (Signal Pins) –0.3 V
Digital Input Voltage (Signal Pins) –0.3 V
Ambient Temperature (Operating) 0 +70 °C
Storage Temperature –65 +150 °C
ESD Tolerance (Human Body 1000 V

+ 0.3 V

CC

+ 0.3 V

DD

Model Range Description Option

AD1843JS 0°C to +70°C 80-Lead PQFP S-80
AD1843JST 0°C to +70°C 100-Lead TQFP ST-100

Model per Method 3015.2
of MIL-STD-883B)

CAUTION

ESD (electrostatic discharge) sensitive device. Electrostatic charges as high as 4000 V readily
accumulate on the human body and test equipment and can discharge without detection.
Although the AD1843 features proprietary ESD protection circuitry, permanent damage may
occur on devices subjected to high energy electrostatic discharges. Therefore, proper ESD
precautions are recommended to avoid performance degradation or loss of functionality.

4.75 5.25 V

2.85 5.25 V

Min Typ Max Units

Typ Units

ORDERING INFORMATION

Temperature Package Package

REV. 0–6–

PIN CONFIGURATIONS

80-Lead PQFP

AD1843

SDO

SDFS

GNDD

V

DD

TSI

TSO

GNDD

V

DD

CS

BM

AUX3R

AUX3L

AUX2R

AUX2L

AUX1R

AUX1L

MICR

MICL

MIN

V

DD

CLKOUT

V

FILTL

AAFILTL

CONV3

LINRP

GNDD

SDI

SCLK

79 78 77 76 75 74 73 72 71 70 69 68 67 66 65 64 63 6280 61

1

2

3

4

5

6

7

8

9

10

11

12

13

14

15

16

17

18

19

20

CC

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

FILTR

GNDA

AAFILTR

BIT3

LINRN

DD

GNDD

V

AD1843

TOP VIEW

(Not to Scale)

LINLP

LINLN

CONV2

BIT2

LOUT2RP

LOUT2RN

DD

GNDD

V

LOUT2LP

LOUT2LN

CONV1

BIT1

MOUT

LOUT1R

VDDV

GNDD

GNDA

LOUT1L

DD

XTALO

REF

V

CMOUT

XTALI

GNDA

60

59

58

57

56

55

54

53

52

51

50

49

48

47

46

45

44

43

42

41

GNDD
XCTL1
XCTL0
SYNC3
SYNC2
SYNC1
GNDD
V

DD

RESET

PWRDWN

V

DD

PDMNFT
GNDA
HPOUTL
HPOUTC
HPOUTR
V

CC

SUML
SUMR
V

CC

PIN DESCRIPTION
Serial Interface

Pin Name PQFP TQFP I/O Description

SCLK 79 99 I/O Serial Clock. SCLK is a bidirectional signal that supplies the clock as an output

to the serial bus when the Bus Master (BM) pin is driven HI and accepts the clock
as an input when the BM pin is driven LO. When the AD1843 is configured in
master mode, the SCLK frequency may be set to either 12.288 MHz or 16.384 MHz
with the SCF bit in Control Register Address 26.

SDFS 2 2 I/O Serial Data Frame Sync. SDFS is a bidirectional signal that supplies the frame

synchronization signal as an output to the serial bus when the Bus Master (BM)
pin is driven HI and accepts the frame synchronization signal as an input when
the BM pin is driven LO.

SDI 80 100 I Serial Data Input. SDI is used by peripheral devices such as the host CPU or a

DSP to supply control and playback data information to the AD1843. All control
and playback transfers are 16 bits long, MSB first.

SDO 1 1 O Serial Data Output. SDO is used to supply status/control register readback and

capture data information to peripheral devices such as the host CPU or a DSP.
All status/control register readback and capture data transfers are 16 bits long,
MSB first. A three-state output driver is used on this pin.

BM 10 12 I Bus Master. When BM is tied HI the AD1843 is the serial bus master. The

AD1843 will then supply the serial clock (SCLK) and the frame sync (SDFS)
signals for the serial bus. No more than one device (AD1843/CPU/DSP) should
be configured as the serial bus master. When BM is tied LO, the AD1843 is con­figured as a bus slave, and will accept the SCLK and SDFS signals as inputs. The
logic level on this pin must not be changed once RESET is deasserted (driven HI).

REV. 0 –7–

AD1843

PIN CONFIGURATIONS

100-Lead TQFP

SDO

SDFS

GNDD

V

TSO

GNDD

V

BM

AUX3R

AUX3L

AUX2R

AUX2L

AUX1R

AUX1L

MICL

MICR

MIN

V

TSI

CS

NC
NC

NC

NC

DD

NC

DD

CC

DD

CLKOUT

SDI

1

2

3

4

5

6

7

8

9

10

11

12

13

14

15

16

17

18

19

20

21

22

23

24

25

GNDD

SCLK

NC

99 98 97 96 95 94 93 92 91 90 89 88 87 86 85 84 83 82 81 80 79 78 77100 76

CONV3

V

NC

BIT3

GNDD

DD

V

CONV2

NC

AD1843

TOP VIEW

(Not to Scale)

BIT2

GNDD

V

DD

CONV1

NC

BIT1

GNDD

VDDV

DD

NC

XTALO

XTALI

75

74

73

72

71

70

69

68

67

66

65

64

63

62

61

60

59

58

57

56

55

54

53

52

51

GNDD
XCTL1
NC
XCTL0
SYNC3
SYNC2
SYNC1
NC
GNDD
V

DD

RESET

PWRDWN

NC
V

DD

PDMNFT
NC
GNDA
HPOUTL
HPOUTC
HPOUTR
V

CC

SUML
SUMR
NC
V

CC

26 272829 30 313233 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50

NC = NO CONNECT

NC

GNDA

FILTR

AAFILTR

FILTL

AAFILTL

NC

LINRP

LINRN

LINLP

LINLN

NC

LOUT2RP

LOUT2RN

LOUT2LP

LOUT2LN

NC

MOUT

LOUT1R

GNDA

LOUT1L

CMOUT

V

REF

NC

GNDA

Serial Interface (Continued)

Pin Name PQFP TQFP I/O Description

CS 9 11 I Chip Select. When CS is set HI, the serial interface I/O pins will be in their normal

active states. When CS is reset LO, SCLK, SDFS, and SDO are three­stated; SCLK, SDFS and SDI inputs are ignored; and TSO drives out the logic
level received on TSI.

TSO 6 7 O Time Slot Output. TSO is asserted HI by the AD1843 simultaneously with the LSB

of the last time slot used by the AD1843. It is used to daisy-chain multiple AD1843s
on a common TDM serial bus. If the power-down (

PWRDWN) pin is asserted or if
the chip select pin (CS) is deasserted, TSO is set to the logic level on the TSI pin,
allowing powered-down or unselected AD1843s on a daisy-chain to be skipped.

TSI 5 6 I Time Slot Input. Asserting TSI HI indicates to the AD1843 that it should use

the next six time slots beginning on the next SCLK period. It also enables TSO
to be asserted at the end of these six time slots. TSI is ignored (but should be tied
LO) when the AD1843 is the bus master since the bus master uses the first time
slots in a TDM frame.

XCTL[1:0] 59, 58 72, 74 I/O External Control. These signals reflect the status of bits (Data 8 and 9) in Control

Register Address 28 of the AD1843. They may be used for signaling or controlling
external logic.

REV. 0–8–

AD1843

Analog Signals

Pin Name PQFP TQFP I/O Description

LINLP 28 35 I Line Input Left Channel Positive Differential Signal.
LINLN 29 36 I Line Input Left Channel Negative Differential Signal.
LINRP 26 33 I Line Input Right Channel Positive Differential Signal.
LINRN 27 34 I Line Input Right Channel Negative Differential Signal.
MICL 18 21 I Microphone Input Left Channel. Microphone input for the left channel. This

signal can be either line level or –20 dB from line level.

MICR 17 22 I Microphone Input Right Channel. Microphone input for the right channel.

This signal can be either line level or –20 dB from line level.
AUX1L 16 20 I Auxiliary #1 Left Channel Line Input.
AUX1R 15 19 I Auxiliary #1 Right Channel Line Input.
AUX2L 14 18 I Auxiliary #2 Left Channel Line Input.
AUX2R 13 17 I Auxiliary #2 Right Channel Line Input.

AUX3L 12 16 I Auxiliary #3 Left Channel Line Input.
AUX3R 11 15 I Auxiliary #3 Right Channel Line Input.
MIN 19 23 I Monaural (Mono) Line Input.
MOUT 35 44 O Monaural (Mono) Line Output.
LOUT1L 36 45 O Line Output #1 Left Channel.
LOUT1R 34 43 O Line Output #1 Right Channel.
HPOUTL 47 58 O Headphone Output Left Channel.
HPOUTC 46 57 Headphone Common Return.
HPOUTR 45 56 O Headphone Output Right Channel.
LOUT2LP 32 40 O Line Output #2 Left Channel Positive Differential Signal.
LOUT2LN 33 41 O Line Output #2 Left Channel Negative Differential Signal.
LOUT2RP 30 38 O Line Output #2 Right Channel Positive Differential Signal.
LOUT2RN 31 39 O Line Output #2 Right Channel Negative Differential Signal.
SUML 43 54 I Mixer Line Input Left Channel.
SUMR 42 53 I Mixer Line Input Right Channel.

Clocks

Pin Name PQFP TQFP I/O Description

CLKOUT 76 95 O Clock Output. This signal is a buffered version of XTALO (with a duty cycle

restored to at least 60%/40%), the crystal clock output. This pin is enabled by
default but can be three-stated by programming a bit in Control Register
Address 28. The CLKOUT frequency is 24.576 MHz.

SYNC[3:1] 57, 56, 55 71, 70, 69 I Sync Inputs. These SYNC signals are used as the clock source inputs to three

receptive PLLs in the AD1843. These pins accept a clock at, or at a multiple of,
the desired sample rate for A-to-D and D-to-A conversions. These inputs are
ignored if a sample rate is programmed directly, but should never be left floating.

CONV[3:1] 75, 71, 67 94, 89, 84 O Conversion Clock Outputs. These output clocks have an average period equal to (or 128

times) the internal sample rates of the AD1843. These clock outputs are three-stated
by default but can be enabled by programming bits in Control Register Address 28.

BIT[3:1] 74, 70, 66 92, 87, 82 O Bit Clock Outputs. These output clocks can be individually programmed to

multiples of the sample rates. Support for V.34 or V.32 bit rates is available.
These clock outputs are three-stated by default but can be enabled by
programming bits in Control Register Address 28.

REV. 0 –9–

AD1843

Miscellaneous

Pin Name PQFP TQFP I/O Description

XTALI 61 76 I 24.576 MHz Crystal Input. When using a crystal as the clock source, the crystal

should be connected between the XTALI and XTALO pins. This crystal should
be 24.576 MHz for the normal sampling rate range, i.e., 4 kHz to 54 kHz. A
clock input (perhaps the CLKOUT of another AD1843) may be driven into
XTALI in place of a crystal. The external clock input must be greater than or equal
to 512 times the maximum desired AD1843 sampling frequency.

XTALO 62 77 O 24.576 MHz Crystal Output. When using a crystal as the clock source, the crystal

should be connected between the XTALI and XTALO pins. If a clock is driven
directly into XTALI, then XTALO should be left unconnected.

PWRDWN 51 64 I Power Down. PWRDWN is active LO. The assertion of this signal will initialize

the on-chip Control Registers to their default values, and will completely and
quietly power down the AD1843. If a crystal is not connected between XTALI
and XTALO, there must be a 24.576 MHz clock input on XTALI for at least
5 ms after this signal is asserted LO for proper operation. The AD1843 will not
be completely powered down until after this 5 ms period elapses. The AD1843
always finishes an in-progress power-up sequence before initiating a power-down
sequence, and vice versa. If the
is in progress, the 24.576 MHz clock signal on XTALI must persist for a worst
case maximum of 479 ms (power up = 470 ms, autocalibration = 4 ms, power
down = 5 ms) after
Address 0, Bit 15) is set to a “1,” the power-down sequence is complete. See
the “Power Management” section for important additional details.

RESET 52 65 I Reset. RESET is active LO. The assertion of this signal will initialize the on-chip

registers to their default values, and will completely power down the AD1843.
RESET is similar to PWRDWN, except that when PWRDWN is asserted, power
down is “quiet” and performed synchronously to the internal clocks. When
is asserted, power down is “noisy” and performed asynchronously to the internal
clocks.

PDMNFT 49 61 I Power-Down Mono Feedthrough. When the AD1843 mixer is powered down,

and PDMNFT is asserted HI, the Mono Input (MIN, PQFP Pin 19) is routed to
the Mono Output (MOUT, PQFP Pin 35), and the signal applied to MIN will
feedthrough to MOUT. When the AD1843 mixer is powered down and
PDMNFT is deasserted LO, the feedthrough of MIN to MOUT will be muted.
When the AD1843 mixer is not powered down, and MIN to MOUT feedthrough
is desired, the Mono Input Mix Mute (Control Register Address 8, Bit 15) and the
Mono Output Mute (Control Register Address 8, Bit 6) must be unmuted. During
power-down feedthrough, the signal applied to the MIN input appears only at
the MOUT output. During normal operation, the signal applied to the MIN
input appears at both the MOUT and the LOUT1 outputs. The state of the
PDMNFT pin should be changed when the AD1843 mixer is powered up. If the
state of PDMNFT is changed when the AD1843 is in total power-down, audible
pops and clicks will likely result.

CMOUT 38 47 O Common-Mode Voltage Output. Nominal 2.25 volt reference available externally

for dc-coupling and level-shifting. CMOUT should not be used where it will sink
or source current.

V

REF

FILTL 25 31 I Left Channel Filter. This pin requires a 1.0 µF capacitor to analog ground for

FILTR 23 29 I Right Channel Filter. This pin requires a 1.0 µF capacitor to analog ground for

AAFILTL 24 30 I Left Channel Antialias Filter. This pin requires a 1000 pF capacitor to analog

AAFILTR 22 28 I Right Channel Antialias Filter. This pin requires a 1000 pF capacitor to analog

39 48 I Voltage Reference Filter. Voltage reference filter point for external bypassing only.

proper operation.

proper operation.

ground for proper operation.

ground for proper operation.

PWRDWN is asserted. When INIT (Control Register

PWRDWN pin is asserted while a power-up sequence

RESET

REV. 0–10–

POWER SUPPLIES

Pin Name PQFP TQFP I/O Description

AD1843

V

CC

20, 41, 44 25, 51, 55 I Analog Supply Voltage (+5 V).
GNDA 21, 37, 40, 48 27, 46, 49, 59 O Analog Ground.
V

DD

4, 8, 50, 53, 5, 10, 62, 66, I Digital Supply Voltage (+5/3 V).

63, 64, 68, 72, 79, 80, 85, 90,

77 96
GNDD 3, 7, 54, 60, 4, 9, 67, 75, I Digital Ground.

65, 69, 73, 78 81, 86, 91, 97
NC 3, 8, 13, 14, No Connect. May be left floating.

24, 26, 32, 37,
42, 50, 52, 60,
63, 68, 73, 78,
83, 88, 93, 98

(continued from page 1)

The versatility of the device is shown by the following examples
of functions it can perform:

• Stereo audio input and/or quad output, simultaneously at dif-

ferent sample rates

• Stereo audio output with simultaneous full duplex modem or

fax operation with frequency and phase resampling

• Mono audio input and stereo audio output with simultaneous

modem receive and transmit for simultaneous voice and data
communications

• Dual independent audio inputs with audio output for echo-

cancelling speakerphones

Audio Functional Description

The AD1843 SoundComm codec provides a complete audio so­lution with very few external components required. Dynamic
range of the device exceeds 80 dB over the 20 kHz audio band
and sample rates from 4 kHz to 49 kHz are supported (up to
54 kHz for a single channel if other channels are powered
down). The audio functionality of this device is a superset of
that found in the Analog Devices AD1848 SoundPort

®

device
which has set the business audio standard throughout the com­puter industry.

Inputs to the device include a stereo microphone pair, a stereo
line pair, a stereo CD input pair (AUX1), a stereo synthesized
music input pair (AUX2), a dual phone line input (AUX3), a
mono input, and a stereo input from an FM synthesizer (SUM).
All of these inputs (except SUM) are multiplexed to the two ∑∆
A/D converters and are mixable directly as analog signals with
the outputs of the D/A converters. All analog input signals (ex­cept SUM) can be amplified, attenuated or muted before mix­ing with the outputs of the D/A converters.

The device has two pairs of ∑∆ DACs which accept 8- or 16-bit
digital data from the serial port. Each DAC pair’s independent
sampling rate can either be programmed by Control Register
(with 1 Hz resolution) or synchronized to an external input.
The second pair of DACs can be used to replace the music syn­thesis DAC pair found on many audio products for PCs. Out­puts from the AD1843 include a line output, a mono output, a
stereo headphone output with its own current return path, and a

differential stereo output for connection to a DAA. The line and
differential outputs are looped back to the ADC input selector.

The AD1843’s mixing and routing capabilities are extensive.
The digital data from both DAC channels after interpolation
can be routed back to the ADC decimators, to support digital­to-digital sample rate conversion (digital resampling). Digital
data from the ADC can also be routed to the two stereo DAC
pairs, for a digital loopback mode which is helpful for device­level and board-level test. Digital data from either stereo DAC
can be mixed with the digital data feeding the other DAC, and
the analog signal from DAC2 can be mixed with the analog out­put from DAC1.

Sample rates are independently programmable in the range of
4 kHz to 54 kHz to a 1 Hz resolution or sample rates can be
synchronized to an external source. Up to three different signals
can be applied to the device’s three digital phase lock loop
SYNC inputs for external synchronization.

These SYNC inputs can also be used in a special mode for au­dio/video synchronization. In this mode, an NTSC or PAL de­rived clock signal (approximately 15 kHz) is applied to the
SYNC inputs and the device produces one of a variety of stan­dard audio sample rates (32 kHz, 44.056 kHz, 44.1 kHz and
48 kHz, and most of these divided by the integers 1 through 8).
In this manner, video and audio sample rates which are math­ematically unrelated can be locked together.

Data Communications/Telephony Functional Description

The AD1843 includes all data conversion, filtering, and clock
generation circuitry needed to implement an echo-cancelling
modem with a companion digital signal processor. Software­programmable sample rates and clocking modes support all
established modem standards including those for the V.34
standard.

The AD1843 utilizes advanced ∑∆ technology to move the
entire echo-cancelling modem implementation into the digital
domain. The device maintains 90 dB typical dynamic range
throughout all filtering and data conversion across a 9.6 kHz
passband. Purely DSP-based echo cancellation algorithms can
maintain robust bit error rates under worst-case signal attenua­tion and echo amplitude conditions. The AD1843’s on-chip
interpolation filter resamples (both frequency and phase) the re­ceived signal after echo cancellation in the DSP, freeing the pro-

SoundPort is a registered trademark of Analog Devices, Inc.

cessor for other voice or data communications tasks.

REV. 0 –11–

AD1843

8 9

GNDD

V

DD

FILTL

CMOUT

V

REF

FILTR

∑

S

ELE

C

T

O

R

CLOCK GENERATION

HPOUTR

HPOUTC

HPOUTL

LOUT2LP

LOUT2LN

LOUT2RP

LOUT2RN

CLKOUT

2

LEFT

RIGHT

LEFT

RIGHT

GN/AT

MUTE

SELECTO

R

20 dB

LEFT

RIGHT

43

GNDA

V

CC

ADC

DAC1

DAC2

D

IGI

TAL

I

NTE

RFACE

LOUT1L

LOUT1R

AUX3L

AUX2R

AUX1L

AUX2L

AUX1R

LINRP

LINRN

MICR

MICL

GN/AT = GAIN/

ATTENUATION

DRIVER

CONTROL

REGISTERS

AD1843

AAFILTL

AAFILTR

SUML

SUMR

LINLP

LINLN

MOUT

RESET

PWRDWN

∑

GN/AT

MUTE

∑

GN/AT

MUTE

∑

∑

GN/AT

MUTE

∑

∑

GN/AT

MUTE

∑

GN/AT

MUTE

∑

∑

MUTE

MUTE

GN/AT

MUTE

MUTE

∑∆

DAC

∑

∑

MUTE

MUTE

ATTN

∑

∑

MUTE

MUTE

∑

∑

MUTE

ATTN

MUTE

MUTE

GN/AT

ATTN

∑

∑

MUTE

ATTN

FIFO

µ/A

LAW

AUX3R

MIN

FIFO

µ/A

LAW

S

ELE

C

T

O

R

∑∆

ADC

PGA

µ/A

LAW

XTALI XTALO SYNC3 SYNC2 SYNC1 CONV3 CONV2 CONV1 BIT3 BIT2 BIT1

SCLK

SDFS

SDI

SDOBMCS

TSO

TSI

XCTL [1:0]

PDMNFT

VOLTAGE REFERENCE

∑∆

DAC

Figure 2. Detailed Functional Block Diagram

REV. 0–12–

AD1843

On-chip bit and baud clock generation circuitry allows either
synchronous or asynchronous operation of the transmit (DAC)
and receive (ADC) paths. Each path features independent
phase advance and retard adjustments via software control. The
AD1843 can also synchronize modem operation to an external
terminal band clock. Because the device has multiple input and
output channels and converters, it is well suited for telephony
applications requiring multiple channels for voice and modem.

A detailed block diagram of the AD1843 is shown in Figure 2.

DETAILED PRODUCT DESCRIPTION

The Serial-Port AD1843 SoundComm Codec integrates the key
audio and PSTN data conversion and control functions into a
single integrated circuit. The AD1843 is intended to provide a
complete, single-chip audio and fax/modem solution for PC
multimedia applications.

External circuit requirements are limited to a minimal number
of low cost support components. Dynamic range exceeds 80 dB
over the 20 kHz audio band. Sample rates from 4 kHz to
54 kHz with 1 Hz resolution are supported from a single exter­nal crystal or clock source.

The AD1843 SoundComm Codec is intended to be interfaced
through a DSP chip or an ASIC to a host bus such as ISA,
EISA or PCI. A general system architecture is shown in
Figure 3.

S
Y
S
T
E

M
B

U
S

ASIC

ADSP-21xx

AD1843

ANALOG I/O

Figure 3. AD1843 System Diagram

The SoundComm codec includes a stereo pair of ∑∆ analog-to­digital converters and two stereo pairs of ∑∆ digital-to-analog
converters. Inputs to the ADC can be selected from eight
sources of analog signals: stereo line (LIN), stereo microphone
(MIC), stereo auxiliary line #1 (AUX1), stereo auxiliary line #2
(AUX2), stereo auxiliary line #3 (AUX3), mono line (MIN),
mixer output, and DAC2 output. A mono output and a stereo
headphone driver are included on-chip. A stereo line level input
(SUM) can be mixed into the output summer. A software-con­trolled programmable gain stage allows independent gain for
each ADC channel. The ADCs’ output can be digitally mixed
with both the DAC1 and DAC2 inputs. The left and right
ADC channels can be configured for different sample rates and
digital filter function (audio, modem or resampling).

The pair of 16-bit outputs from the ADCs is available over a se­rial interface that also supports 16-bit digital input to the DACs
and control/status information. The AD1843 can accept and
generate 16-bit twos-complement PCM linear digital data, 8-bit
unsigned magnitude PCM linear data, and 8-bit µ-law or A-law
companded digital data. The data format is defined indepen­dently for each conversion resource on the AD1843.

The ∑∆ DACs are preceded by a four sample deep FIFO buffer
and a digital interpolation filter. The DAC1 and DAC2 outputs
can be mixed in the digital domain. Digital and analog attenua­tors provide independent user volume control (plus mute) over
each DAC channel. Nyquist images and shaped quantized

noise are removed from the DACs’ analog stereo output by on­chip switched-capacitor and continuous-time filters. All of the
analog inputs (except the stereo line input) can be mixed with
the DAC1 output in the analog domain. The DAC2 output can
also be mixed with the DAC1 output in the analog domain.
The DAC1 and DAC2 digital data can be fed back to the digital
half of the ADC to enable digital resampling operation. DAC1
and DAC2 can be run at different sample rates and with differ­ent digital filter functions, without any beat frequency problems.

FUNCTIONAL DESCRIPTION

This section overviews the functionality of the AD1843 and is
intended as a general introduction to the capabilities of the de­vice. As much as possible, detailed reference information has
been placed in “Control Registers” and other sections. The
user is not expected to refer repeatedly to this section.

Analog Inputs

The AD1843 SoundComm Codec accepts stereo line-level and
mic-level inputs. The mono MIN analog signal input, and LIN
(differential), MIC, AUX1, AUX2, AUX3 and post-mixed
DAC output analog stereo signals are multiplexed to the inter­nal programmable gain amplifier stage (PGA).

The PGA following the input multiplexer allows left and right
independent selectable gains for each channel from 0 dB to

22.5 dB in +1.5 dB steps. The Codec can operate either in a
global stereo mode or in a global mono mode with left-channel
inputs appearing at both channel outputs.

Analog Mixing

The MIN analog mono signal, and the MIC, AUX1, AUX2,
AUX3 and SUM analog stereo signals can be mixed in the ana­log domain with the DAC1 output. Each channel of each auxil­iary analog input can be independently gained/attenuated from
+12 dB to –34.5 dB in 1.5 dB steps or completely muted. The
mixer output is available on LOUT1 externally and as an input
to the ADCs. Even if the AD1843 is not playing back data from
its DACs, the analog mix function can still be active.

MIN allows the analog signal intended for the PC speaker to be
passed through, attenuated or mixed in the AD1843’s analog
domain. MIN can be used to accept other mono input sources.
A digital control signal pin PDMNFT (Power Down Mono
Feed Through) enables the mono input signal to be fed through
to the mono output when the AD1843 mixer is powered down.

Analog-to-Digital Datapath

The AD1843 ∑∆ ADCs incorporate a fourth-order modulator.
A single pole of passive filtering is all that is required for
antialiasing the analog input because of the ADC’s high over­sampling ratio. The ADCs include linear-phase digital decima­tion filters that low-pass filter the input. ADC input overrange
conditions will cause bits to be set that can be read.

Each channel of the mic inputs can be amplified in the analog
domain by +20 dB to compensate for the voltage swing differ­ence between line levels and typical condenser microphone levels.

Digital-to-Analog Datapath

The ∑∆ DACs are preceded by a programmable attenuator and
a low-pass digital interpolation filter. The anti-imaging interpo­lation filter oversamples and digitally filters the higher frequency
images. The attenuator allows independent control of each
DAC channel from +12.0 dB to –82.5 dB in 1.5 dB steps plus
full mute. The DACs’ ∑∆ noise shapers oversample and con­vert the signal to a single-bit stream. The DAC outputs are then

REV. 0 –13–

AD1843

filtered in the analog domain by a combination of switched-capaci­tor and continuous-time filters. They remove the very high fre­quency components of the DAC bitstream output. No external
components are required. Phase linearity at the analog output is
achieved by internally compensating for the group delay varia­tion of the analog output filters.

Changes in DAC output attenuation may be programmed to
take effect immediately, or only on zero crossings of the digital
signal, thereby eliminating “zipper” noise on playback. Each
channel has its own independent zero-crossing detector and at­tenuator change control circuitry. A timer guarantees that re­quested volume changes will occur even in the absence of an
input signal that changes sign. The time-out period is 8 milli­seconds at a 48 kHz sampling rate and 48 milliseconds at an
8 kHz sampling rate. (Time-out [ms] ≈ 384 ÷ F

[kHz]).

S

Digital Mixing

Stereo digital output from the ADCs can be mixed digitally with
the input to the DACs. Digital output from the ADCs going
out of the serial port is unaffected by this digital mix. Along the
digital mix datapath, the 16-bit linear output from the ADCs is
attenuated by an amount specified with Control Register bits.
The level of attenuation applied to the left and right channels is
independently programmable. (Note that internally the AD1843
always works with 16-bit PCM linear data, digital mixing in­cluded; format conversions take place at the input and output.)

Sixty-four steps of –1.5 dB attenuation are supported to –94.5 dB.
The digital mix datapath can also be completely muted, pre­venting any mixing of the analog input with the analog output.
Note that the level of the mixed signal is also a function of the
input PGA settings, since they affect the ADCs’ output. The
sample rate of the ADCs and the selected DAC pair must be the
same for the digital mix function to operate properly.

The attenuated digital mix data is digitally summed with the
DAC input data prior to the DACs’ datapath attenuators. The
digital sum of digital mix data and DAC input data is clipped at
plus or minus full scale and does not wrap around. Because
both stereo signals are mixed before the output attenuators,
mix data is attenuated a second time by the DACs’ datapath
attenuators. In case the AD1843 is playing back data but input
digital DAC data fails to arrive in time (“DAC underrun”), then
a midscale zero will be added to the digital mix data in place of
the unavailable DAC data.

Analog Outputs

The two mixer line-level outputs are available at external pins.
Each output channel can be independently muted. When
muted, the outputs will settle to a dc value near CMOUT, the
midscale reference voltage. The two DAC2 stereo outputs are
available at external pins differentially. The full-scale level on
these pins is established by programming bits in a Control Reg­ister. In addition, there is stereo headphone output (with a cur­rent return), and a mono output. Both the headphone output
and the mono output have a single mute control.

Digital Data Types

The AD1843 supports four data types: 16-bit twos-complement
linear PCM, 8-bit unsigned linear PCM, 8-bit companded µ-law,
and 8-bit companded A-law, as specified by control register bits.
The data type is independently assignable for each conversion
resource (i.e., ADCL, ADCR, DAC1 and DAC2). Data in all
four formats is always transferred MSB first. Eight-bit data is al­ways left-justified in 16-bit fields; said in other words, the MSBs
of all data types are always aligned; in yet other words, full-scale

representations in all four formats correspond to equivalent full­scale signals. The eight least-significant bit positions of 8-bit
data in 16-bit fields are ignored on input and zeroed on output.

The 16-bit PCM data format is capable of representing 96 dB of
dynamic range. Eight-bit PCM can represent 48 dB of dynamic
range. Companded µ-law and A-law data formats use nonlinear
coding with less precision for large-amplitude signals. The loss
of precision is compensated for by an increase in dynamic range
to 64 dB and 72 dB, respectively.

On input, 8-bit companded data is expanded to an internal lin­ear representation, according to whether µ-law or A-law was
specified in the Codec’s internal registers. Note that when µ-
law compressed data is expanded to a linear format, it requires
14 bits. A-law data expanded requires 13 bits.

COMPRESSED

INPUT DATA

EXPANSION

DAC INPUT 000/00

15

MSB LSB

15

MSB LSB

MSB LSB

8 7

3/2 2/1

3/2 2/1 015

0

0

Figure 4.µ-Law or A-Law Expansion

When 8-bit companding is specified, the ADCs’ linear output is
compressed to the format specified.

0

015

ADC OUTPUT

TRUNCATION

15

MSB LSB

3/2 2/1

MSB LSB

.

COMPRESSION

15

MSB LSB

8 7

00000000

0

Figure 5.µ-Law or A-Law Compression

Note that all format conversions take place at input or output.

Power Supplies and Voltage Reference

The AD1843 operates from either +5.0 V analog (VCC) and
digital (V

) power supplies or +5.0 V analog and +3.0 V digi-

DD

tal supplies. Independent analog and digital supplies are recom­mended for optimal performance though excellent results can be
obtained in single-supply systems. A voltage reference is included
on the Codec and its +2.25 V buffered output is available on an
external pin (CMOUT). The reference output can be used for
biasing op amps used in single supply systems. The internal ref­erence is externally bypassed to analog ground at the V

REF

pin.

Clocks and Sample Rates

The AD1843 operates from a single external clock or crystal
source. From a single clock, a wide range of sample rates can be
generated. When supplied with a single 24.576 MHz clock, the
AD1843 can be programmed to generate any sample frequency
between 4 kHz and 54 kHz with 1 Hz resolution. For modem
sample rate support, the frequency programmed can also be in­creased by 8/7 using a control bit. All sample rate changes can
be made “on the fly.”

The AD1843’s SYNC inputs can be used to synchronize the
sampling activity of the four on-chip conversion resources to ex­ternal clock signals, such as video HSYNC or an ISDN network
clock. The SYNC inputs are used by three on-chip digital phase

REV. 0–14–

AD1843

lock loops, which can be arbitrarily assigned to the conversion
resources. The lock range of these digital PLLs is 4 kHz to
54 kHz, which is the same range supported by the register­controlled clock generators.

If a SYNC input stops after its associated phase lock loop has
had a chance to initially lock, the AD1843 will continue to gen­erate a sample clock (as well as BIT clock and CONV clock)
very similar to the initial frequency, but off by at most ±1%.
The three SYNC inputs feed three on-chip Digital Phase Lock
Loops (DPLLs) which utilize a first-order loop filter with a
20 Hz corner frequency. Jitter frequencies above 20 Hz are
attenuated, and jitter frequencies below 20 Hz are interpreted as
time base drift, and are tracked. The DPLL provides 12 dB per
octave of jitter rejection. The DPLLs have been designed to tol­erate at least 2% Unit Interval (UI) of SYNC clock jitter. The
DPLLs are critically damped at all input frequencies.

Power Management

The AD1843 SoundComm codec has extensive power manage­ment capabilities. Hardware power down is performed using the
PWRDWN pin. Software power management is programmed us­ing Control Register Address 27 and 28. Several elements of the
AD1843 can be powered down on a selective basis. These blocks
include: the DAC2 to DAC1 analog mixer; the entire DAC1 con­version channel; the entire DAC2 conversion channel; the analog
half of the ADC, DAC1 and DAC2; the headphone driver; the en­tire analog mixer; the right ADC channel; the left ADC channel; all

four conversion channels; clock generator 1; clock generator 2;
clock generator 3; conversion clock outputs 1 through 3; bit clock
outputs 1 through 3; and the nominal 24.576 MHz clock output.
Refer to the descriptions of Control Register Address 27 and 28 for
further information.

For proper operation, the AD1843 must be calibrated following
power-up. This initial calibration occurs automatically without any
user intervention or programming. Subsequent to this initial
power-up autocalibration, there is no requirement to recalibrate the
SoundComm codec following software power-down sequences.
The entire AD1843 or selected portions of the device may be
powered down, allowed to idle indefinitely, then powered up
and used immediately, without the need for repeated auto­calibration. The digital information obtained during the initial
power-up calibration is retained and valid unless the

RESET or

PWRDWN pin is asserted, forcing a hardware reset. (If desired,

the user can specify that a calibration cycle occur when leaving
the software power-down state by setting ACEN (Control Reg­ister Address 28, Bit 14) to ”1.”) A hardware reset or power­down clears the calibration information, and therefore a fresh
autocalibration cycle is performed by the AD1843 following this
event. Autocalibration takes approximately 4 ms to complete.

The following table provides an indication of the power savings
associated with powering-down the various resources in the
AD1843. Note that the power savings is somewhat order-

Table I. AD1843 Power-Down Savings

+5 V Digital, +5 V Analog Supplies Total Active Operation Current: 200 mA

Software Power Down Control Register Bit(s) ICC Current Power Savings

CLKOUT Output ENCLKO Bit = 0 8 mA 4%
All Bit Clocks and ENBT3, ENBT2, ENBT1 Bits = 0

All Conversion Clocks ENCV3, ENCV2, ENCV1 Bits = 0 2 mA 1%
Clock Generator 1 C1EN Bit = 0 6 mA 3%
Clock Generator 2 C2EN Bit = 0 6 mA 3%
Clock Generator 3 C3EN Bit = 0 6 mA 3%
All Clock Generators C1EN, C2EN, C3EN Bits = 0 20 mA 10%
Headphone Driver HPEN Bit = 0 8 mA 4%
DAC2 to DAC1 Mix DDMEN Bit = 0 2 mA 1%
Analog Input to Analog Output Mix AAMEN Bit = 0 8 mA 4%
ADC Left Channel ADLEN Bit = 0 8 mA 4%
ADC Right Channel ADREN Bit = 0 8 mA 4%
ADC Left and Right Channels ADLEN, ADREN Bits = 0 38 mA 17%
DAC2 (Left and Right Channels) DA2EN Bit = 0 30 mA 15%
DAC1 (Left and Right Channels) DA1EN Bit = 0 24 mA 12%
DAC2 AND DAC1 (Left and Right Chs) DA2EN, DA1EN Bits = 0 60 mA 30%
ADC and DAC2 and DAC1 ADLEN, ADREN,

DA2EN, DA1EN Bits = 0 108 mA 54%
Analog Channel ANAEN Bit = 0 54 mA 27%
All Control Register 27 HPEN, DDMEN, AAMEN, ADLEN,

ADREN, DA2EN, DA1EN, ANAEN Bits = 0 134 mA 67%
Converter PDNI Bit = 1 140 mA 70%
All of the Above (Register 27 and ENCLKO, ENBT3, ENBT2, ENBT1, ENCV3,

Clocks and PDNI) ENCV2, ENCV1, C1EN, C2EN, C3EN, HPEN,

DDMEN, AAMEN, ADLEN, ADREN, DA2EN,

DA1EN, ANAEN Bits = 0, PDNI Bit = 1 176 mA 88%

Average, Typical Average, Typical
Absolute I

+ Normalized

DD

REV. 0 –15–

AD1843

dependent; depending upon the sequence in which the hardware
resources are powered down, the savings may be more or less
than the typical numbers given.

Mode Changing

In general, there are very few restrictions with respect to chang­ing the operating mode of the AD1843. Because of the advanced
Continuous Time Oversampling technology, the waiting period
associated with changes to the sample rate of the data converters
(“Mode Change Enable” resynchronization delay) is eliminated.
The only waiting periods associated with the AD1843 occur at
start-up, and are documented in the “START-UP SEQUENCE”
section below. Following the start-up sequence, the sample rate
of the four data conversion resources on the AD1843 may be
changed at any time, on-the-fly (presuming that they are
enabled). All gain, mute and attenuation settings of enabled
resources may also be changed at any time.

Channel Synchronization

If multiple AD1843s are used in a daisy-chained system, and it
is desired to synchronize data conversion activity among the
multiple codecs, the clock generator blocks of the AD1843s
must be enabled on the same frame (see step 5 in the “START­UP SEQUENCE” section below).

A DAC channel does not actually start processing samples until
the first rising edge of the conversion clock (CONV pin) after
the sixth rising edge of frame sync (SDFS pin) after the channel
is enabled (via a write to DA1EN or DA2EN in Control Regis­ter Address 27). The wait until the sixth rising edge of frame
sync is necessary to allow the four deep DAC FIFO to be filled
before conversion commences. The subsequent wait until the
rising edge of the conversion clock is necessary to synchronize
the serial interface based DAC channel enable command with a
conversion clock that is potentially already running (which is
particularly likely if the SYNC pin inputs and lock mode are
in use).

The ADC channels behave very similarly to the DAC channels.
An ADC channel does not actually start taking samples until the
first rising edge of the conversion clock (CONV pin) after the
sixth rising edge of frame sync (SDFS pin) after the channel is
enabled (via a write to ADLEN or ADREN in Control Register
Address 27). The wait until the sixth rising edge of frame sync is
present so that the ADC startup is similar to that of the DAC
startup, as well as to allow some time for stale ADC data inside
the AD1843 to be cleared. The subsequent wait until the rising
edge of the conversion clock is necessary to synchronize the
serial interface based ADC channel enable command with a
conversion clock that it potentially already running (which is par­ticularly likely if the SYNC pin inputs and lock mode are in use).

Supported Conversion Rates

With all conversion channels operating (i.e., ADC left, ADC
right, DAC1 and DAC2), the AD1843 is able to support sam­pling rates up to 49 kHz, which 2.1% higher than the nominal
maximum audio standard of 48 kHz, to accommodate timebase
drift while configured in slave mode. If either one DAC (i.e.,
either DAC1 or DAC2) or both ADC channels (i.e., ADC left
and ADC right) are shut down, then the AD1843 can support
sampling up to 54 kHz on all channels of the remaining conver-

sion resources, as long as the DFREE bit (Control Register Ad­dress 27) is asserted (i.e., set to “1”). If DFREE is not asserted,
then the maximum sampling rate for the remaining conversion
resources is 49 kHz.

Digital Filter Selection

The operative digital filter modes for the four conversion re­sources on the AD1843 SoundComm are programmed using
Control Register Address 25. ADLFLT (Bit 0) selects the digi­tal filter mode for the ADC left channel and ADRFLT (Bit 1)
selects the digital filter mode for the ADC right channel. Note
that these bits also establish the full-scale input voltage range for
these channels as well. DA1FLT (Bit 8) selects the DAC1 digi­tal filter mode, and DA2FLT (Bit 9) selects the DAC2 digital
filter mode. Note that these bits also establish the full-scale out­put voltage for these channels as well.

The three digital filter modes are audio, modem and resampler.
The specifications for these modes are given in the description
of Control Register Address 25, as well as in the “SPECIFICA­TIONS” section of this data sheet. The specifications have been
made to satisfy the demands of the applications which the
AD1843 can serve. The audio mode provides decimation and
interpolation characteristics sufficient for high quality cap­ture and playback of material from 20 Hz to 20 kHz. The mo­dem mode provides characteristics sufficient for modulation
standards up to V.34 quality. The resampling mode provides
optimal characteristics for high quality sample rate conversion.
While in the resampling mode, all images in the resampled data
stream (including those in the transition band) are attenuated to
below the quantization noise floor. Note that the maximum
sample rate for modem mode is 24 kHz.

Digital Resampling

Digital resampling is best achieved by routing the digital output
of one of the DACs back to the digital input of one of the
ADCs. This bypasses the analog portion of the DAC and ADC,
eliminating their noise and signal delay contributions. This fea­ture is enabled by bits DAADR1:0 (Digital ADC Right Channel
Source Select) and DAADL1:0 (Digital ADC Left Channel
Source Select) in Control Register Address 25.

If the “Digital Resampler Filter Mode” (DRSFLT bit = “1,”
Control Register Address 25) is enabled, the DAC2 pair is sacri­ficed, but the remaining four channels (ADC left and right,
DAC1 left and right) can still be used in any way they could
have been when not in “Digital Resampler Filter Mode.” When
in this mode, internal AD1843 hardware normally devoted to
DAC2 is reallocated to the other four channels, allowing these
channels to realize superior digital filtering. Note that the
AD1843 DOES NOT actually have to be in digital resampler
filter mode to perform digital resampling, however the superior
digital filters in this mode allow for a much higher quality digital
resampling.

Using the AD1843 in a Modem Application

The AD1843 analog performance is sufficient to support the
modem Analog Front End (AFE) function, for data modulation
standards up to and including the 28.8 kbps V.34 ITU stan­dard. The data pump function is performed in a companion
DSP, such as the ADSP-2181, for which several V.34 algo­rithms (from third party Independent Algorithm Vendors) exist.

REV. 0–16–

AUDIO FROM

EXTERNAL MPEG

DECODER

EXTERNAL

WAVEFORM

SYNTHESIZER

AUDIO FROM

CD-ROM

PSTN

AUDIO FROM DAT

OR CASSETTE

DAA

PC SPEAKER

PC ATTENTION

“BEEPER” SIGNAL

AD1843 SOUNDCOMM

LINE IN
MIC IN RIGHT
MIC IN LEFT
AUX1 IN
AUX2 IN
AUX3 IN
MONO IN
SUM IN

LINE1 OUT
HEADPHONE OUT
LINE2 OUT RIGHT
LINE2 OUT LEFT
MONO OUT

SPEAKERPHONE

CODEC

EXTERNAL POWERED

MULTIMEDIA SPEAKERS

Figure 6. Typical Configurations

SYNC2

CONV1

BIT1

SERIAL

INTERFACE

NTSC

HORIZONTAL

SYNC SIGNAL

ADSP-21xx

DSP OR ASIC

SERIAL
INTERFACE

OR PARALLEL

IDMA PORT

PORT

AD1843

HOST BUS
ISA OR PCI

SDI OR SDO

SCLK

SDFS

SDI OR SDO

SCLK

SDFS

SAMPLE PERIOD N

256 BITS 256 BITS

321015141312

MSB

SAMPLE PERIOD N

321015141312

MSB

256 BITS

FRAME M

SAMPLE PERIOD N+1

SLOT 15SLOT 0

321015141312

MSB

512 BITS

256 BITS

SLOT 31 SLOT 16

321015141312

SAMPLE PERIOD N+2

MSB

FRAME M

FRS = 0 [DEFAULT 32 SLOTS PER FRAME, 2 SAMPLES PER FRAME SYNC]

NOTE THAT AD1843 FRAME RATE IS NOT RELATED TO SAMPLE RATES

MASTER MODE

Figure 7. FRS = 0, Master Mode Timing

SAMPLE PERIOD N+1

SLOT 15SLOT 0

321015141312

MSB

256 BITS

FRAME M+1

FRS = 1 [16 SLOTS PER FRAME, 1 SAMPLE PER FRAME SYNC]

NOTE THAT SCLK CAN BE PROGRAMMED FOR EITHER 12.288 MHz

OR 16.384 MHz WHEN IN MASTER MODE

MASTER MODE

SLOT 15 SLOT 0

321015141312

SAMPLE PERIOD N+2

MSB

256 BITS

FRAME M+2

SAMPLE
PERIOD N+3

SLOT 16

SLOT 15SLOT 0

321015141312

256 BITS

MSB

512 BITS

FRAME M+1

SAMPLE
PERIOD N+3

SLOT 0

SLOT 15SLOT 0

321015141312

MSB

FRAME M+3

Figure 8. FRS = 1, Master Mode Timing

REV. 0 –17–

AD1843

SDI OR SDO

SCLK

TSI

SDI OR SDO

SCLK

TSI

SAMPLE PERIOD N

GAP

15 14 13 12

MSB

256 BITS

SAMPLE PERIOD N

321015141312

MSB

SAMPLE PERIOD N+1

SLOT 15SLOT 0

321015141312

MSB

512 BITS

SLOT 31 SLOT 16

256 BITS

GAP

3210 15141312 321015

SAMPLE PERIOD N+2

MSB

FRAME M

FRS = 0 [DEFAULT 32 SLOTS PER FRAME, 2 SAMPLES PER FRAME SYNC]

EXAMPLE SHOWING GAPS BETWEEN FRAMES

SLAVE MODE

Figure 9a. FRS = 0, Slave Mode Timing

SAMPLE PERIOD N+1

SLOT 15SLOT 0

256 BITS

321015141312

MSB

512 BITS

256 BITS

FRAME M

FRS = 0 [DEFAULT 32 SLOTS PER FRAME, 2 SAMPLES PER FRAME SYNC]

EXAMPLE SHOWING NO GAPS BETWEEN FRAMES

SLAVE MODE

SLOT 31 SLOT 16

321015141312

SAMPLE PERIOD N+2

MSB

SLOT 15SLOT 0

256 BITS

MSB

512 BITS

FRAME M+1

SLOT 15SLOT 0

256 BITS

32101514

MSB

512 BITS

FRAME M+1

SDI OR SDO

SCLK

TSI

SDI OR SDO

SCLK

TSI

Figure 9b. FRS = 0, Slave Mode Timing

SAMPLE PERIOD N SAMPLE PERIOD N+1

15 14 13 12

SLOT 15SLOT 0

3210

256 BITS

GAPGAP

15 14 13 12

MSBMSB

FRAME M

FRS = 1 [16 SLOTS PER FRAME, 1 SAMPLES PER FRAME SYNC]

EXAMPLE SHOWING GAPS BETWEEN FRAMES

SLAVE MODE

3210

FRAME M+1

Figure 10a. FRS = 1, Slave Mode Timing

SAMPLE PERIOD N

3210 15 14 13 12

15 14 13 12

SLOT 15SLOT 0

3210

256 BITS

FRAME M

FRS = 1 [16 SLOTS PER FRAME, 1 SAMPLES PER FRAME SYNC]

EXAMPLE SHOWING NO GAPS BETWEEN FRAMES

SLAVE MODE

SAMPLE PERIOD N+1

15 14 13 12

MSBMSB

FRAME M+1

SLOT 15SLOT 0

3210

MSB

256 BITS

SLOT 15SLOT 0

256 BITS

GAP

Figure 10b. FRS = 1, Slave Mode Timing

REV. 0–18–

AD1843

Modem Data Access Arrangement (DAA) devices are generally
differential on the transmit side, and single-ended on the receive
side. The DAA transmit input (generally differential) should be
connected to the DAC2 output, pins LOUT2LP and LOUT2LN,
or LOUT2RP and LOUT2RN. The DAA receive output
(generally single-ended) should be connected to one of the
ADC line inputs, LINLP or LINRP. See the “APPLICATION
CIRCUITS” section below for more detail on the electrical
connections. There are several software driver steps that are re­quired to configure the SoundComm codec for use as a modem
AFE.

Configure DAC2

1. Set the DA2FLT bit (Control Register Address 25, Bit 9) to
“1,” to select the digital modem filter mode. The DAC2 out­puts can be used either as differential outputs or single-ended
outputs depending on how the pins are connected electrically;
no Control Register writes are required to configure the DAC2
outputs as either differential or single-ended.

2. Program LDA2G5:0 (Control Register Address 10, Bits 8
through 13) to “00 0101” (i.e., +4.5 dB) or RDA2G5:0
(Control Register 10, Bits 0 through 5) to “00 0101” (i.e.,
+4.5 dB), depending on whether the DAA transmit input is
connected to the left channel DAC2 output (use LDA2G5:0)
or the right channel DAC2 output (use RDA2G5:0). This
code establishes the DAC2 nominal analog output swing at

3.156 V p-p single-ended, or 6.312 V p-p differentially. The

3.156 V p-p level is equivalent to 3.17 dBm.

Configure ADC

1. Set the ADLFLT bit (Control Register Address 25, Bit 0) to
“1,” or the ADRFLT bit (Control Register Address 25, Bit 1)
to “1,” to select the digital modem filter mode. Set ADLFLT
if the DAA receive output is connected to the AD1843
LINLP input; set ADRFLT if the DAA receive output is
connected to the AD1843 LINRP input. Set the LINLSD bit
(Control Register Address 28, Bit 0) to “1” if the DAA is
connected to the AD1843 LINLP input; set the LINRSD bit
(Control Register Address 28, Bit 1) to “1” if the DAA is
connected to the AD1843 LINRP input.

2. Program LIG3:0 (Control Register Address 2, Bits 8 through

11) to “0000” (i.e., 0.0 dB) or RIG3:0 (Control Register
Address 2, Bits 0 through 3) to “0000” (i.e., 0.0 dB) de­pending on whether the left or right ADC input channel is
being used for the modem function. This code maps an ana­log input swing of 3.156 V p-p to the full dynamic range of
the 16-bit digital sample (i.e., ± 2
equivalent to 3.17 dBm.

Note that if the AD1843 is to be reconfigured dynamically, the
affected converter must be powered down before its associated
digital filter can be changed. In other words, if the digital filter
for the ADC left channel is being changed from audio mode to
modem mode, the ADC left channel must be powered down
first (using the ADLEN bit in Control Register Address 27).

15

). The 3.156 V p-p level is

Use the ADREN bit in Control Register Address 27 for the
ADC right channel, the DAC1EN bit in Control Register
Address 27 for DAC1, and the DAC2EN bit in Control Regis­ter Address 27 for DAC2.

Typical Configurations

Figure 6 below illustrates example connections between the
AD1843 SoundComm codec and other system resources. The
rich analog input and output connectivity of the AD1843 allows
a wide variety of configuration possibilities. Note that the level
of modem, speakerphone and external speaker concurrency is
application and DSP resource dependent.

SERIAL INTERFACE

The AD1843 SoundComm Codec transmits and receives both
data and control/status information through its serial port.

The AD1843 can be configured as either master or slave of the
serial interface. This is selected by using the BM pin. When
BM is tied HI, the AD1843 serves as bus master and supplies
the frame sync and the serial clock. When BM is tied LO, the
AD1843 serves as bus slave and receives the frame sync and the
serial clock. The level on BM should not be altered unless the
reset pin (

The AD1843 has six pins devoted to the serial interface: SDI,
SDO, SCLK, SDFS, TSI and TSO. The SDI pin is for serial
data input to the AD1843 and the SDO pin is for serial data
output from the AD1843. The SCLK pin is the serial interface
clock. Communication in and out of the AD1843 requires bits
of data to be transmitted after a rising edge of SCLK, and
sampled on a falling edge of SCLK. When the AD1843 is bus
master (BM pin tied HI), the SCLK frequency driven by the
AD1843 will be 12.288 MHz by default, but this can be in­creased to 16.384 MHz by setting the SCF bit in Control Regis­ter 26. When the AD1843 is bus slave (BM pin tied LO), the
SCLK frequency driven to the AD1843 may be as high as

24.576 MHz, but must not be any higher than the frequency on
the XTALI pin.

The SDFS pin is for the serial interface frame sync. When bus
master, new frames are marked by a HI pulse driven out on
SDFS one serial clock period before the frame begins. When
bus slave, new frames must be marked by a LO to HI transition
driven in on SDFS one serial clock period before the frame be­gins, but the transition back from HI to LO may occur at any
time provided the HI and LO times of SDFS are at least one
SCLK period in duration each.

When the AD1843 is bus master, frame size is controlled by the
FRS bit in Control Register 26. When FRS is set to “1,” each
frame is divided into 16 slots of 16 bits. When FRS is reset to
“0,” each frame is divided into 32 slots of 16 bits. In 32 slot
configuration, the second 16 slots of a frame must have slot as­signments that are identical to the first 16 slots of the frame; 32
slot configuration is essentially 16 slot configuration with every
other SDFS pulse missing. Although these are the frame sizes

RESET) is asserted.

REV. 0 –19–

AD1843

produced by an AD1843 serving as bus master, an AD1843
serving as bus slave does not actually require these frame sizes.
When FRS is set to “1,” a slave will operate correctly with any
number or fraction of slots, provided there are enough slots for
it to complete its necessary communication (see below). When
FRS is reset to “0,” a slave can also operate correctly with a
wide range in the number of slots per frame, however it will au­tomatically retake ownership of the serial interface bus 16 slots
after it is first given ownership of the bus in a frame.

The nominal minimum number of slots when the AD1843 is
configured in slave mode is six. The codec must be supplied
with at least 6 × 16 = 96 SCLK periods (both rising and falling
edges); SCLK may be gated (i.e., no need to be continuous)
between valid slots.

While SDFS marks the beginning of frames, AD1843 bus own­ership during a frame is controlled by the TSI (Time Slot In)
and TSO (Time Slot Out) pins. When bus slave, a level HI on
TSI grants the AD1843 bus ownership beginning with the next
SCLK period. The TSI pin is monitored only when an AD1843
does not already own the bus; once an AD1843 is given owner­ship of the bus, the level on TSI is ignored until one SCLK
period before bus ownership is relinquished. Bus ownership will
last for six slots. Coincident with the final SCLK period of the
final slot owned, the AD1843 asserts TSO HI. This allows
chaining of AD1843s onto a common serial bus by connecting
the TSO pin of one AD1843 to the TSI pin on the next later
AD1843 in a chain. In single codec systems where the
SoundComm is configured as bus slave, connect the AD1843
SDFS and TSI signals together. When an AD1843 is bus mas­ter, its function is identical to that just described for the slave,
except a bus master always owns the first six slots and its TSI
pin is ignored (but should be tied LO).

Whenever an AD1843 does not own the bus, its SDO pin will
be three-stated and its SDI pin is ignored. Figures 7 through 10
illustrate the signal, slot, sample and frame relationships for the
four basic operating modes of the AD1843 serial interface.

The AD1843 requires slots of communication each time it takes
ownership of the serial bus. The first slot is used for a Control
Word input and a Status Word output. The second slot is used

for Control Register write data input and read data output. The
remaining slots are used for playback (DAC) data input and
capture (ADC) data output, where each channel has an assigned
slot. Table II and Figure 11 illustrate these slot assignments.

Since the conversion channels of the AD1843 can be pro­grammed to run at different sample rates, a communication
mechanism indicates when playback channels request data,
when playback data is actually sent to the AD1843, and when
transmission of capture data from the AD1843 becomes neces­sary. This is facilitated by the Control and Status Words lo­cated in the first slot. The Control Word indicates which slots
in the current frame contain valid playback data. The Status
Word indicates if playback data can be sent to the AD1843 dur­ing the next frame, and which slots in the current frame contain
valid capture data. See the descriptions of the Control Word
and the Status Word below for additional detail.

Four word FIFO buffers are used on the inputs of each of the
DACs to allow data to be transferred in small bursts. This re­duces the required response time to playback data requests, and
also buffers differences between the frame sync rate and the
channel sample rate. The Status Word indicates that playback
data can be sent if there is any room in the buffers, thus tending
to keep the input buffers full. Underrun flags are available in
Control Register 1, which indicate if an input buffer ran out of
data. If an underrun occurs, a zero is used in place of the un­available data. To ensure underruns do not occur, playback data
must be sent to the AD1843 within two sample periods after the
status word indicates that the DAC FIFO is not full.

Note that the DAC Not Full status bits (DA2RQ and DA1RQ
in the Status Word Output) are updated immediately (i.e., in
the same frame as a valid write to the DAC FIFOs). If the DAC
Input Valid Flags (DA2V and DA1V in the Control Word
Input) are set (i.e., DAC data is valid) and only one location in
the DAC1 and DAC2 input FIFOs is available, then the
DA2RQ and DA1RQ status bits will reflect this valid write, and
will be reset to “0.” This is possible because the DA2V and
DA1V bits are in the most significant bits of the Control Word
and the DA2RQ and DA1RQ bits are in the least significant bits
of the Status Word, and the AD1843 uses this intervening time

Table II. AD1843 Slot Assignment

32 Slot Mode (FRS Reset to “0”)

Slot SDI Pin SDO Pin

0 & 16 Control Word Input Status Word Output
1 & 17 Control Register Data Input Control Register Data Output
2 & 18 Playback Data Input—DAC1 Left Capture Data Output—ADC Left
3 & 19 Playback Data Input—DAC1 Right Capture Data Output—ADC Right
4 & 20 Playback Data Input—DAC2 Left Reserved (Unused)
5 & 21 Playback Data Input—DAC2 Right Reserved (Unused)

16 Slot Mode (FRS Set to “1”)

Slot SDI Pin SDO Pin

0 Control Word Input Status Word Output
1 Control Register Data Input Control Register Data Output
2 Playback Data Input—DAC1 Left Capture Data Output—ADC Left
3 Playback Data Input—DAC1 Right Capture Data Output—ADC Right
4 Playback Data Input—DAC2 Left Reserved (Unused)
5 Playback Data Input—DAC2 Right Reserved (Unused)

REV. 0–20–