Datasheet AS2521T, AS2521P, AS2521BT, AS2521BP, AS2520T Datasheet (Austria Mikro Systeme International)

Preliminary AS2520/21/20B/21B

Austria Mikro Systeme International AG

with Loudhearing and Handsfree

Telephone Speech Circuit

Key Features

❑ Line/speech circuit, loudhearing, handsfree and

dc/dc converter on one 28 pin CMOS chip

❑ Operating range from 13 to 100 mA (down to 5

mA with reduced performance)

❑ Soft clipping control eliminating harsh distortion
❑ Volume control of receive signal with squelch and

automatic loop gain compensation

❑ Line loss compensation pin selectable
❑ Low noise (max. - 72 dBmp)
❑ Real or complex impedance adjustable
❑ NET 4 compatible
❑ Dynamically controlled voice switching
❑ Same monitor amplifier for loudhearing, handsfree

and tone ringer

❑ Very few external components
❑ Power derived from ring signal by switching

converter during ringing

Typical Application

La

General Description

The AS2520/21/20B/21B are CMOS integrated
circuits that contain all the audio functions needed to
form a high comfort, line-powered telephone.

The devices incorporate line adaptation, speech
circuit, loudhearing and handsfree - all supervised by
the novel voice and power control circuit. A switching
converter is also provided for converting the ring
signal. The interface to a dialler/controller is made
very simple to allow easy adaptation to a telecom
microcontroller.

The AS2520 series incorporate volume control for the
earpiece and the loudspeaker (AS2520 digital with +/­keys and AS2521 analogue with potentiometer). The
volume control circuit automatically compensates the
loop gain to ensure acoustic stability.

Package

Available in 28 pin SOP and DIP.

Lb

3V

TELEPHONE

1 2 3
4 5 6
7 8 9

0 #

*

DIALLER

µCONTROLLER

LCD DRIVER

SPEECH CIRCUIT

WITH

LOUDHEARING,

HANDSFREE,

DC/DC CONVERTER

HSM

HFM

AS2520

Figure 1: Typical Handsfree Telephone Application

Rev. 5.1 Page 1 May 1999

Preliminary AS2520/21/20B/21B
Pin Description

Pin # Name Type Description

1 LS AI
2 CI AI

3 RO AO

4 V
5 A

DD

GND

Supply

Supply
6 STB AI
7 LLC DI

8 LSI AO

9 TI AI

DI

10 RTH AI

11 CM AO

12 V

PP

Supply

13 LO AO

14 V

SSP

Supply

15 MT DI

16 PD DI

17 LE DI
18 HS DI

Line Current Sense Input

This input is used for sensing the line current.

Complex Impedance Input

Input pin for the capacitor in the complex impedance.

Receive Output

This is the output for driving a dynamic earpiece with an impedance of 140
to 300 ohm.

Positive Voltage Supply

This is the supply pin for the circuit.

Analogue Ground

This pin is the analogue ground for the amplifiers.

Side Tone Balance Input

This is the input for the side tone cancellation network.

Line Loss Compensation Selection Pin

LLC = V
LLC = A
LLC = V

: High range -6 dB from 45 mA to 75 mA;

DD

: Low range -6 dB from 20 mA to 50 mA;

GND

: No regulation gain independent of line current;

SS

Loudspeaker Amplifier Input

This is the input for applying the receive signal to the loudspeaker
amplifier.

Tone Input

This switchable input is intended for transmitting DTMF or other signals
like messages on TAMs (Telephone Answering Machines) onto the line in
off-hook conditions and when in ringing mode to apply a PDM signal to the
loudspeaker (see also table 1).

Receive Threshold Input

The sensibility of the receive peak detector can be adjusted by applying
the signal from RO to the RTH input through a voltage divider.

Converter Make Output

This is an output for controlling the external switching converter. It
converts the ring signal into a 4V supply voltage and is activated when PD
= high and HS, LE, MT = low.

Loudspeaker Power Supply

High power supply for the output driver stage.

Output for Loudspeaker

Output pin for an ac coupled 32

Ω (25 to 50 Ω)loudspeaker.

Negative High Power Supply

This pin is the negative high power supply for the loudspeaker amplifier.

Mute Input

Dialling mute input (see also table 1).
MT = V

MT = V

: Tx and Rx channels muted;

DD

: Tx and Rx channels not muted.

SS

Power Down Input

Input for powering down the speech circuit and loudhearing/handsfree
(see table 1).

Loudhearing Enable Input

Input for enabling loudhearing/handsfree, active high (see table 1).

Handset Switch Input

This is an input that is pulled high by the hook switch (handset) or µC
when off-hook (see table 1).

Rev. 5.1 Page 2 May 1999

Preliminary AS2520/21/20B/21B

19
22
20
21

M1
M2
M4
M3

AI
AI

23 VOL D/AI

Microphone Inputs

Differential inputs for handset microphone (electret).

Handsfree Microphone Inputs

These are the input pins for the handsfree microphone (electret).

Volume Control Input

Volume control for the receive signal.
AS2520: Digital control with +/– keys or from µC;
AS2521: Analogue dc control with potentiometer.

24 SS AO

Supply Source Control Output

This N-channel open drain output controls the external high power source
transistor for supplying (V

) the loudspeaker amplifier in off-hook

PP

loudhearing/handsfree mode.

25 CS AO

Current Shunt Control Output

This N-channel open drain output controls the external high power shunt
transistor for the modulation of the line voltage and for shorting the line
during make period of pulse dialling.

26 V

SS

Supply

27 LI AI/O

Negative Power Supply
Line Input

This input is used for power extraction and line current sensing.

28 RI AI

Receive Input

This is the input for the receive signal.

DI: Digital Input AI: Analogue Input
DO: Digital Output AO: Analogue Output
DI/O: Digital Input/Output AI/O: Analogue Input/output

Operating Modes

I/O Pins Digital Inputs Tone Input Outputs
MODE HS LE PD MT TI CM LI RO LO
Idle (on-hook) 0 0 0 0 Not connected Low - PD PD
Ringing 0 0 1 0 PDM signal to LO (DI) SW - - ‘TI’
POT 1 0 0 0 Not connected Low ‘M1/M2’ ‘RI/STB’ ­POT/pulse dialling 1 0 1 1 Not connected Low V

BE

POT/DTMF dialling 1 0 0 1 DTMF to LI and RO (AI) Low ‘TI’ ‘TI’ ­Handsfree 0 1 0 0 Not connected Low ‘M3/M4’ ‘RI/STB’ ‘LSI’
Handsfree/pulse dial 0 1 1 1 Not connected Low V

BE

Handsfree/DTMF dial 0 1 0 1 DTMF to LI and RO (AI) Low ‘TI’ ‘LSI’
Loudhearing 1 1 0 0 Not connected Low ‘M1/M2’ ‘RI/STB’ ‘LSI’
Loudhearing/pulse dial 1 1 1 1 Not connected Low V

BE

Loudhearing/DTMF dial 1 1 0 1 DTMF to LI and RO (AI) Low ‘TI’ ‘LSI’
TAM without LSP 1 0 1 0 Signal to LI (AI) Low ‘RI/STB’
TAM with LSP 1 1 1 0 Signal to LI (AI) Low ‘RI/STB’ ‘LSI’

- -

-

Melody feedback 0 1 1 0 PDM signal to LO (DI) Low ‘RI/STB’
Test mode 1 0 0 0 1 Reserved for testing
Test mode 2 0 0 1 1 Reserved for testing

Table 1: Operating Modes

Rev. 5.1 Page 3 May 1999

Preliminary AS2520/21/20B/21B
Functional Description

The AS252x contains all the voice circuits needed in
a high feature telephone instrument, i .e.:

• line adaptation (ac impedance, dc characteristics,

2/4-wire conversion, power extraction)

• handset speech circuit

• loudhearing with enhanced anti-Larsen

• handsfree with dynamic loop gain control

• switching converter

The line adaptation includes line driver, ac
impedance (return loss), 2 to 4 wire converter, dc
mask and power extraction circuit for extracting the
maximum dc power from the line to supply the whole
device and peripheral circuits.

The handset speech circuit consists of a transmit and
a receive path with mute, dual soft clipping and line
regulation (pin option). A volume control is provided
with squelch and loop gain compensation to improve
signal-to-noise ratio and to assure acoustic stability.

Loudhearing and handsfree functions are also
provided. The loudhearing function includes an anti­Larsen circuit to prevent acoustic howling.

The handsfree circuit has a novel voice control
system which is virtually independent of any
background noise and works in a dynamic half
duplex mode as close to full duplex as the acoustic
loop gain allows.

The switching converter is used to extract the
available power from the ring signal and provides a
4V supply voltage. This allows the same loudspeaker
to be used for loudhearing/handsfree and tone
ringing.

L +

30

300

CI

Ω

Ω

B

Z

CS

LLC

LS

V

Vss

STB

LI

DD

RI

LINE DRIVER

IMPEDANCE

SYNTESIZER

EXTRACTION

VDD

SWITCHING

CONVERTER

SS

POWER

VDD

DC

CONTROL

PD

AS2520/21/20B/21B

ST-AMP

PP

V

MT or PD

LEVEL

DETECTOR

LOGIC

INTERFACE

TX-AGC

VOICE

&

POWER

CONTROL

RX-AGC

RING

A

MT

HS

LEVEL

DETECTOR

GND

A

PDM

INPUT

MI-AMP

MI-AMP

RO-AMP

V

LO-AMP

M1

M2

M3

M4

RTH

GND

A

RO

LSI

PP

LO

CM

PD

LE HS

MT

VOL

TI

Vss

P

Figure 2: Block Diagramme

Rev. 5.1 Page 4 May 1999

Preliminary AS2520/21/20B/21B

DC Conditions

The normal operating range (off-hook) is from 13 mA
to 100 mA. Operating range with reduced
performance is from 5 mA to 13 mA (parallel
operation). In the normal operating range all
functions are operational.

In the line hold range from 0 to 5 mA the device is in
a power down mode and the voltage at LI is reduced
to maximum 3.5V.

The dc characteristic (excluding diode bridge) is
determined by the voltage at LI and the resistor R1 at
line currents above 13 mA as follows:

VLS = VLI + I

LINE

ž R1
The voltage at LI is 4.5V.
Below 13 mA the AS252x provides an additional

slope in order to allow parallel operation (see figure

3).

8

(V)

7

6

5

4

3

2

1

0

Typically
No ac signals
Tamb: 25°C

Line Current

VLS

VLI

1009080706050403020100

(mA)

(see application notes). The dc resistance of R1
should be kept at 30 ohm to ensure correct dc
condition.

Return loss and sidetone cancellation can be
determined independent of each other (see figure 4).

Speech Circuit

The speech circuit consists of a transmit and a
receive path with soft clipping, mute, line loss
compensation and sidetone cancellation.

Transmit
The gain of the transmit path is 36.5 dB in handset

mode (from M1/M2 to LS) and 46.5 dB in handsfree
mode (from M3/M4 to LS). The microphone inputs
have an input impedance of 15 kohm.

The unique dual soft clipping control circuit limits the
output voltage at LI to 2V

. Dual means that the soft

PEAK

clipping incorporates both a very fast control circuit to
eliminate harsh sidetone distortion and a slower
regulation circuit to limit the output voltage at 2V

PEAK

independent of the line impedance. The attack time
is 30 µs/6 dB. The overdrive range is 30 dB. When
mute is active, pin MT high, the gain is reduced by >
60 dB.

Receive
The gain of the receive path is 3 dB (test circuit

figure 8) from RI to RO. The receive input is the
differential signal of RI and STB. Also the receive
channel provides soft clipping to avoid acoustic
shock and harsh distortion.

Figure 3: DC Mask

When mute is active during dialling the gain is
reduced by > 60 dB. During DTMF dialling a MF

When the PD pin is high (during pulse dialling) the
speech circuit and other part of the device not
operating are in a power down mode to save current.
The CS pin is pulled to V

in order to turn the

SS

external shunt transistor on to keep a low voltage
drop at the LS pin during make periods.

AC Impedance

The synthesised ac impedance of the circuit is set on
chip and by an external resistor and an external
capacitor (for complex impedance).

When R1 is set to 30 ohm, the ac impedance is 1000
ohm real, and the complex part can be set by a
capacitor connected to pin 2 (CI).

For 600 ohm telephones it is recommended to
connect a resistor and a capacitor from pin LS to V

SS

comfort tone is applied to the receiver. The comfort
tone is the DTMF signal with a level that is -30 dB
relative to the line signal.

Volume Control
On the AS2520 the receive gain can be changed by

pressing the volume keys. The + key increases the
gain by 10 dB in 5 steps and the – key decreases the
gain by 10 dB in 5 steps. The gain is reset by next
off-hook. The volume can also be controlled via a
microcontroller.

The AS2521 uses a potentiometer to control the
receive gain. The volume is an indirect dc control to
avoid that noise is introduced from the potentiometer.

The volume control is common for both the earpiece
and the loudspeaker. Any increase will be
compensated to ensure acoustic stability.

Rev. 5.1 Page 5 May 1999

Preliminary AS2520/21/20B/21B

The acoustic stability is provided as follows:
When the volume is increased, e.g. by 10 dB, the

receive gain maintains the same as long as no
receive signal is applied. Applying a receive signal
will cause a 10 dB increase of the receive gain and a
corresponding decrease of the transmit gain. This
squelch function improves the signal-to-noise ratio.

In other words, a certain increase of the volume
introduces a similar amount of dynamic voice
switching, controlled by the receive signal, also in the
handset mode.

Sidetone
A good sidetone cancellation is achieved by using

the following equation:

Z

BAL/ZLINE

= R5/R1

The sidetone cancellation signal is applied to the
STB input.

By using two separate Wheatstone Bridges for return
loss and sidetone cancellation it is very easy to
calculate the sidetone balance network (see figure 4).
This unique configuration provides a sidetone
cancellation less sensitive to tolerances on the
external balance network and totally independent of
the ac impedance and its tolerances.

20 to 50 mA or 45 to 75 mA depending on selected
range.

Loudhearing
The loudhearing mode is enabled when HS and LE

are high. In order to prevent acoustic coupling
between the handset microphone and the
loudspeaker, the AS252x incorporate an anti-Larsen
circuit.

The anti-Larsen circuit decreases the gain of the
loudspeaker amplifier when a microphone signal is
applied. If no signal is applied from the microphone,
the loudspeaker amplifier is at its full gain.

Anti-Clipping (not

AS2520B/21B)

The anti-clipping circuit is activated in loudhearing
and handsfree mode. The circuit prevents harsh
distortion at very high signal levels.

Furthermore, the circuit assures that the integrity of
the whole telephone circuit is maintained under
extreme load conditions, since it prevents that the
supply voltage drops below a certain minimum level.

The attack time is fast (120 µs/6 dB) for preventing
harsh distortion when the amplitude rapidly
increases. For avoiding chopper effects and to
assure low distortion, the decay time is longer,
approx. 128 ms/6 dB.

A good and stable sidetone cancellation improves the
handsfree function considerably and ensures a safe
margin against acoustic instability under all
circumstances.

R1

Z

LINE

Z

BAL

30 ohm

R5
300 ohm

Figure 4: Sidetone Bridge

Furthermore, the dual Wheatstone bridge makes it
very simple to adapt the circuit to different PTT
requirements as these two parameters (return loss
and sidetone balance) are independent of each
other.

Line Loss

Compensation

The line loss compensation (Rx and Tx AGC
controlled by the line current) is a pin option. When it
is activated, the transmit and receive gains are
changed by -6 dB in 1 dB steps at line currents from

When the anti-clipping circuit has been activated by a
large receive signal, the channel control will increase
the Tx gain corresponding to the reduction in Rx gain
caused by the anti-clipping.

Handsfree
The handsfree function allows voice communication

without using the handset (full 2-way speaker phone).
Two voice controlled attenuators prevent acoustic
coupling between the loudspeaker and the
microphone.

A conventional voice switching circuit has a channel
control with three states, namely idle, transmit or
receive. In idle state, when no signal is applied, both
the transmit and the receive channels are attenuated
by approx. 20 dB to keep the total loop gain below 0
dB.

When a signal is applied to the microphone, the
circuit switches to transmit state, i.e. the gain in the
transmit channel is increased and the gain in the
receive channel is decreased accordingly. And vice
versa when a receive signal is applied.

Rev. 5.1 Page 6 May 1999

Preliminary AS2520/21/20B/21B

SIDE TONE

Line Output Signal

This approach has some disadvantages. It requires a
high degree of discipline, since the three state
channel control gives a very distinct half duplex with
a relative high switching time constant to avoid
chopper effects. Furthermore, the system is very
sensitive to the environment,- noise, line conditions
and acoustics (echo).

Apart from keeping a distinct discipline, the user can
not do anything to minimise the effect of these
constraints, since the parameters of the voice
switching (thresholds, time constants, noise
discrimination, etc.) can not be changed or adapted
to the actual conditions by the user.

The dynamic voice control system of the AS252x
have been designed to overcome the above
constraints. The basic philosophy behind the AS252x
is that telephone circuits should not have any
automatic regulations preventing the user from
having all information about the actual conditions
which should enable her/him to act accordingly, i.e.
to comply with the given constraints.

Now, assuming subscriber A has a handsfree
telephone and is calling subscriber B, who has a
normal telephone. The B subscriber does not
necessarily know that A is using a handsfree
telephone and will therefore not automatically comply
to the discipline of a half duplex conversation. Hence,
the disadvantages by using half duplex should apply
to the A subscriber only.

Secondly, if A is in a noisy environment, the B
subscriber should hear it, so that he speaks up to
increase the signal-to-noise ratio at the A subscriber.

The traditional 3-state switching system has two
major drawbacks: first of all, when no one is talking,
the circuit is in idle state and the B subscriber gets
the feeling that the line is dead, since the background
noise does not activate the voice switching.
Secondly, the B subscriber does not speak up, since
she/he does not hear the background noise.

The concept of the AS252x, however, does not
exclude the human factor, but provides the
information about the actual conditions to the user
and allows her/him to act accordingly, i.e. to speak
up, to change the volume, etc.

In more technical terms, the AS252x works in the
following manner:

When no signal is applied neither from the line nor
from the microphone, the circuit is in the only static
state, which is transmit channel full open and receive
channel attenuated by up to 30 dB.

The advantages of using the transmit state as the
static (idle) state are that the B subscriber hears an
open line (the line is not dead), does not miss the
initial word of a sentence when the A subscriber
starts talking, and hears the level of the background
noise at A´s end which will actuate her/him to speak
up accordingly.

When the A subscriber starts talking, the circuit
remains in the static state.

The dynamic state of the voice switching can only be
activated by the receive signal. Applying a receive
signal above a certain level will cause the circuit to
enter the dynamic state.

V

TX

AGC

PEAK

DETECTOR

Z

AC

V

TH

± 10 dB

VOL

2/4

V

LINE

V

RX

Figure 5: Channel Control System

The signal for controlling the channel attenuation is
taken after the sidetone amplifier. With the volume at
0 dB (neutral) the threshold for entering the dynamic
state (VTH) is 15 mV assuming that V

> VTX (see

RX

figure 5).
In the dynamic state the channel attenuation is

controlled by a voltage controlled amplifier. The
attack time is 4 ms/6 dB and the hold time is 200 ms.

A speech compression is activated when a transmit
signal with a high amplitude reaches a level
corresponding to approximately 460 mV on the line.

300
(mV)
250

200

150

100

50

0

Sidetone Cancellation: 11 dB
Volume Control: 0 dB (neutral)

Microphone Input Signal

(mV)

1.501.251.000.750.500.250.00

Figure 6: Speech Compression

Rev. 5.1 Page 7 May 1999

Preliminary AS2520/21/20B/21B

AS252x

The speech compression allows a higher gain in the
transmit channel, i.e. the microphone gets more
sensitive at low sound pressure levels on the
microphone, which enables the user to move further
away from the telephone. This means that a constant
signal is provided on the line practical independent of
the microphone signal level. Any reduction of gain by
the compressor in the transmit channel will
automatically be given to the receive channel.

Switching Converter
The ac ringing signal is utilised to extract the power

necessary to the tone ringer circuit. A switch mode
power supply is used to obtain a high efficiency dc
conversion.

This approach allows the use of the same
loudspeaker and amplifiers for both loudhearing and
tone ringing. It also allows an acoustic feedback of
the melodies during programming with the same
sound pressure level as during ringing.

When a ringing signal is applied, PD is pulled high
and the oscillator is enabled. The switching converter
is controlled by the output CM, which is turned high
and low with a duty cycle controlled by the voltage at

.

V

PP

When off-hook the switching converter has a high
impedance (CM low) to avoid any influence on the
transmission and on pulse dialling.

The smoothing capacitor should be in the range of 10
to 68 nF. The choke coil must have an inductance of
>1mH and a dc resistance of < 15 ohm.

1µ5

La

Lb

1µ5

510

510

33 n

30V

VPP

327

BC

2.2 mH

470 µ

5k6

10 k

5V1

547

BC

CM

VPP

VssP

Figure 7: Switching Converter

Tone Input
The tone input is a digital input in ringing mode and

during melody feedback. The digital melody signal
(PDM = pulse density modulation) is directly applied
to the TI input (see also application notes for further
details).

During DTMF dialling the DTMF signal is applied
through a capacitor to the TI input and will be fed to
the line (pin LI) and to the receive output (RO) as
confidence tone.

Rev. 5.1 Page 8 May 1999

Preliminary AS2520/21/20B/21B
Electrical Characteristics

Absolute Maximum Ratings*

Supply Voltage............................................................................................................................... -0.3 ≤ V

DD

≤ 7V

Input Current..........................................................................................................................................+/- 25 mA

≤ V

≤ V

≤ V

≤ V

DD

DD

≤ 10V

IN

≤ 8V

IN

+0.3V
+ 0.3V

Input Voltage (LS)....................................................................................................................... -0.3V

Input Voltage (LI, CS, SS) ................................................................ .............................................-0.3V

Input Voltage (STB, RI).........................................................................................................-2V

Digital Input Voltage..........................................................................................................-0.3V

≤ V

≤ V

IN

IN

Electrostatic Discharge ..........................................................................................................................+/- 1000V

Storage Temperature Range ................................................................ ........................................... -65 to +125°C

Total Power Dissipation ............................................................................................................................500mW

*Exceeding these figures may cause permanent damage. Functional operation under these conditions is not permitted.

Recommended Operating Range
Symbol Parameter Conditions Min. Typ.* Max. Units

V

DD

V

PP

T

AMB

* Typical figures are at 25°C and are for design aid only; not guaranteed and not subject to production testing.

Supply Voltage (internally generated) Speech mode 3.0 4.1 5.5 V
Supply Voltage (internally regulated) Speech mode 3.0 4.1 5.5 V
Ambient Operating Temp. Range -25 +70 °C

DC Characteristics (I

= 15 mA, recommended operating conditions unless otherwise specified)

LINE

Symbol Parameter Conditions Min. Typ. Max. Units

I

DD

I

DDPD

I

DD0

V
I

OL

LI

Operating Supply Current HS = high

LE = high
HS and LE = high
PD = high, CM running

Power-Down Current
Standby Current

PD = high
All digital inputs = V

Line Voltage 13 mA< I
Output Current, Sink

VOL = 0.4V 1.5 mA

5
5
5

7
7

7
300
200

SS

< 100 mA 4.2 4.5 4.8 V

LINE

1

mA
mA
µA
µA
µA
µA

Pin CS, SS

I

OL

Output Current, Sink

VOL = 0.4V 1.5 mA

Pin CM

V

IL

V

IH

Input Low Voltage T
Input High Voltage T

= 25°C V

AMB

= 25°C 0.8 V

AMB

SS

DD

0.2 VDDV
V

DD

V

Rev. 5.1 Page 9 May 1999

Preliminary AS2520/21/20B/21B

AC Electrical Characteristics

I

= 15 mA; f = 800 Hz; recommended operating conditions unless otherwise specified.

LINE

Transmit

Symbol Parameter Conditions Min. Typ. Max. Units

A

A

∆A

A

TX

MF

TX/F

LLC

Gain (M1/M2 to LS)
Gain (M3/M4 to LS)

HS, LH modes; LLC = A

HF mode; LLC = A
Gain (TI to LS) MF mode 12 13.5 15 dB
Variation with Frequency f = 500 Hz to 3.4 kHz +/- 0.8 dB
Gain Range, LLC Speech mode; LLC = VSS or V

THD Distortion VLI < 0.25 V
V

AGC

V

AGC

A

SCO

Z

IN

A

AD

A

MUTE

V

NO

Soft Clip Level
Soft Clip Level

HS, LH modes; VLI =

HF mode; VLI =
Soft Clip Overdrive
Input Impedance; M1/M2 and M3/M4 15 kohm
Attenuation Depth 30 dB
Mute Attenuation Mute activated 60 dB
Noise Output Voltage HS = high; T

LE = high; HS = low; T

V

IN MAX

Input Voltage Range;

Differential
M1/M2

Single ended

RMS

AMB

GND

= 25°C

GND

= 25°C

AMB

35
45

DD

36.5

46.5

38
48

-6 dB

dB
dB

2 %

2

650

30

-72

-62

+/- 1

+/- 0.5

V

PEAK

mV

PEAK

dB

dBmp
dBmp
V

PEAK

V

PEAK

Line Driver

Symbol Parameter Test Conditions Min. Typ. Max. Units

V
RL

∆Z

IN MAX

AC/TEMP

Input Voltage Range; LI +/- 2 V
Return Loss
Temperature Variation

ZRL = 1000 ohm; T

= 25°C 18

AMB

0.5

PEAK

dB

Ω/°C

Rev. 5.1 Page 10 May 1999

Preliminary AS2520/21/20B/21B

Receive

Symbol Parameter Condition Min. Typ. Max. Units

A

RX

Gain (LS to RO), Default

Volume reset 1.5

LSP Gain (LSI to LO)

∆A

A
A

TX/F

LLC

RX

Variation with Frequency f = 500 Hz to 3.4 kHz +/- 0.8 dB
Gain Range, LLC Speech mode; LLC = VSS or V

Volume Range 10 steps, each 2 dB 20 dB
THD Distortion VRI < 0.2 V
V

SC

Soft Clip Level (RO)

Soft Clip Level (LO)

VRO =
Not AS2520B/21B; VLO =

Unloaded

A
V
A
t

DECAY

t

DECAY

V

SCO

RTH

AD

NO

Soft Clip Overdrive

Threshold Voltage at RTH

Attenuation Depth

Attack Time

Decay Time

Noise Output Voltage (RO)

Channel control; VRI > 0.8 V
Channel control
HS = high; T

RMS

= 25°C

AMB

RMS

3

17.5

DD

19

-6 dB

4.5

20.5dBdB

2 %

1

1.3

30

7 15

30

25 mV

V
V

dB

dB

PEAK

PEAK

µs/6dB
µs/6dB

-72

dBmp

V

UFC

Unwanted Frequency

50 Hz.........20 kHz

-60

dBmp

Components (RO)
Z

IN

V

IN RI

A

ST

Z

IN

V

IN ST

Input Impedance, RI 8 kohm

Input Voltage Range, RI +/- 2 V

Sidetone Cancellation VRI < 0.2 V

RMS

; T

= 25°C 26 dB

AMB

PEAK

Input Impedance, STB 80 kohm

Input Voltage Range, STB +/- 2 V

PEAK

General Timings

Symbol Parameter Condition Min. Typ. Max. Units

t

VOL

t

SCA

t

SCD

t

PDA

t

PDD

Volume Key Debounce 7 ms

Soft Clip Attack Time VIN above soft clip level 0.12 ms/6dB

Soft Clip Decay Time VIN below soft clip level 128 ms/6dB

Peak Detector Attack Time VIN above V

Peak Detector Decay Time VIN below V

TH

TH

3.2 ms/V
29 ms/V

t

LPA

t

LPD

Low-Power Attack Time VPP < 3.6V 250 ms/6dB
Low-Power Release VPP > 3.6V 1 sec/6dB

Rev. 5.1 Page 11 May 1999

Preliminary AS2520/21/20B/21B

10 µ

LS

RI

STB

LI

CS

Vss

PD

M1

M2

RO

M4

HS

AGND

VPP

VDD

RTH

M3

LO

LSI

TI

VOL

LE

MT

LLC

22 µ

30 ohm

300 ohm

6 k

10 µ

680 n

BC

327

10 V

1 k

200 ohm

ILINE

B

UL

600 ohm

A

100 µ

CM

CI

22 µ

25 ohm

1 k

1 k

VsSP

SS

13

21

8

9

10

5

20

3

22

19

2

4

23

7

15

17

18

16

14

12

24

26

25

27

6

11

1

28

BC

327

1 k

Test Circuit

AS252x

Figure 8: Test Circuit

Rev. 5.1 Page 12 May 1999

Preliminary AS2520/21/20B/21B

AS252x

LINE ADAPTER/TELEPHONE VOICE CIRCUIT

Application Diagramme

La

30

Ω

2k2

10 k

2N5551

33 n

10 V

BC327

470 µ

BC327

10 µ

100 µ

300

100 n

220 µ

Ω

Side tone
balance
network

10 n

100 n

100 n

AGND

15 n

15 n

100 n

100 µ

100 k

VDD

1k8

1k8

10 k

1k2

VOL+

VOL-

1k2

32

Ω

27

LI

25

CS

26

Vss

1

LS

28

RI

4

VDD

9

TI

18

HS

16

PD

15

MT

17

LE

5

A

GND

24

SS

11

CM

12

V

PP

14

VSSP

STB

RO

LSI

RTH

LLC

VOL

2

CI

19

M1

22

M2

1 µ

6

1k8

7k5

10 µ

3

8

10

21

M3

20

M4

13

LO

High

7

23

Low
Off

AS2521

AS2520

Lb

VDD

510

1µ5

INPUT FOR DTMF

TONE RINGER MELODIES

CONTROL INPUTS

1µ5

510

Ω

Ω

33 n

MPSA92

AND

(FROM µC)

5k6

2.2 mH

5V1

Figure 9: Application Diagramme

Applications Hints

Interface to Microcontroller

In off-hook condition the microcontroller can be supplied from V
MT) must be kept low until V

has reached its minimum operating voltage (>2.5V).

DD

Radio Frequency Interference

The RFI sensitivity has been minimised by the consequent use of CMOS technology and one overall ground and
by having differential inputs with a relative low input impedance.

For further application information see application notes for the AS2520 series.

Rev. 5.1 Page 13 May 1999

of AS252x. The digital inputs (HS, LE, PD, and

DD

Preliminary AS2520/21/20B/21B

AS252x

Pin Configuration

28 Pin SOP/DIP

V

GND

A

STB

LLC

RTH

V

V

LS

RO

LSI

CM

LO

SSP

CI

DD

PP

1
2
3
4
5
6
7
8
9

TI

10
11
12
13
14

28
27
26
25
24
23
22
21
20
19
18
17
16
15

RI
LI

SS

V
CS
SS
VOL
M2
M3
M4

M1
HS
LE

PD
MT

Ordering Information

Part
Number

AS2520 T 28 pin

AS2520 P 28 pin DIP Digital Yes
AS2520B T 28 pin

AS2520BP28 pin DIP Digital No

AS2521 T 28 pin

AS2521 P 28 pin DIP Analogue Yes
AS2521B T 28 pin

AS2521BP28 pin DIP Analogue No

Package
Type

SOP

SOP

SOP

SOP

Volume
Control

Soft Clip
Loudspk.

Digital Yes

Digital No

Analogue Yes

Analogue No

The devices are also available as dice on request.

Devices sold by Austria Mikro Systeme Int. AG are covered by the warranty and patent indemnification provisions appearing in
its Term of Sale. Austria Mikro Systeme Int. AG makes no warranty, express, statutory, implied, or by description regarding
the information set forth herein or regarding the freedom of the described devices from patent infringement Austria Mikro
Systeme Int. AG reserves the right to change specifications and prices at any time and without notice. Therefore, prior to
designing this product into a system, it is necessary to check with Austria Mikro Systeme Int. AG for current information. This
product is intended for use in normal commercial applications. Applications requiring extended temperature range, unusual
environmental requirements, or high reliability applications, such as military, medical life-support or life-sustaining equipment
are specifically not recommended without additional processing by Austria Mikro Systeme Int. AG for each application.

Copyright © 1999, Austria Mikro Systeme International AG, Schloss Premstätten, 8141 Unterpremstätten, Austria.
Trademarks Registered®. All rights reserved. The material herein may not be reproduced, adapted, merged, translated,
stored, or used without the prior written consent of the copyright owner.

Austria Mikro Systeme Int. AG reserves the right to change or discontinue this product without notice.

Rev. 5.1 Page 14 May 1999