Preliminary AS2520/21/20B/21B Telephone Speech Circuit with Loudhearing and Handsfree Austria Mikro Systeme International AG General Description 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 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. La Lb 3V 1 2 3 4 5 6 7 8 9 * 0 # TELEPHONE SPEECH CIRCUIT WITH LOUDHEARING, HANDSFREE, DC/DC CONVERTER DIALLER µCONTROLLER LCD DRIVER 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 1 LS AI 2 CI AI Description 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. 3 RO AO 4 VDD Supply 5 AGND Supply 6 STB AI 7 LLC DI 8 LSI AO 9 TI AI DI 10 RTH AI 11 CM AO 12 VPP Supply 13 LO AO 14 VSSP Supply 15 MT DI 16 PD DI 17 LE DI 18 HS DI Rev. 5.1 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 -6 dB from 45 mA to 75 mA; LLC = VDD: High range -6 dB from 20 mA to 50 mA; LLC = AGND: Low range gain independent of line current; LLC = VSS: No regulation 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 = VDD: Tx and Rx channels muted; MT = VSS: Tx and Rx channels not muted. 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). Page 2 May 1999 Preliminary AS2520/21/20B/21B 19 22 20 21 23 M1 M2 M4 M3 VOL D/AI 24 SS AO 25 CS AO 26 27 VSS LI Supply AI/O 28 RI AI DI: DO: DI/O: AI 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. Supply Source Control Output This N-channel open drain output controls the external high power source transistor for supplying (VPP) the loudspeaker amplifier in off-hook loudhearing/handsfree mode. 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. Negative Power Supply Line Input This input is used for power extraction and line current sensing. Receive Input This is the input for the receive signal. Digital Input Digital Output Digital Input/Output AI: AO: AI/O: Analogue Input Analogue Output Analogue Input/output Operating Modes I/O Pins MODE Digital Inputs Tone Input HS LE PD MT Idle (on-hook) 0 0 0 0 Ringing 0 0 1 POT 1 0 POT/pulse dialling 1 POT/DTMF dialling Outputs TI CM LI RO LO Not connected Low - PD PD 0 PDM signal to LO (DI) SW - - ‘TI’ 0 0 Not connected Low ‘M1/M2’ ‘RI/STB’ - 0 1 1 Not connected Low VBE - - 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 VBE - 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’ Loudhearing/pulse dial 1 1 1 1 Not connected Low VBE Loudhearing/DTMF dial 1 1 0 1 DTMF to LI and RO (AI) Low ‘TI’ 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’ 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 ‘RI/STB’ ‘LSI’ ‘LSI’ ‘LSI’ Table 1: Operating Modes Rev. 5.1 Page 3 May 1999 Preliminary AS2520/21/20B/21B 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. 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) Loudhearing and handsfree functions are also provided. The loudhearing function includes an antiLarsen circuit to prevent acoustic howling. • handset speech circuit • loudhearing with enhanced anti-Larsen • switching converter 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 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 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. • handsfree with dynamic loop gain control CI 30 Ω L+ SS VPP LI MI-AMP LINE DRIVER IMPEDANCE SYNTESIZER CS LS M3 MI-AMP LEVEL DETECTOR M4 VDD RTH LEVEL DETECTOR VOICE & POWER CONTROL DC CONTROL AGND AGND VDD 300 Ω Vss RO-AMP PD ZB RO VDD AS2520/21/20B/21B LSI MT VPP RI STB M2 TX-AGC POWER EXTRACTION LLC M1 HS MT or PD RX-AGC ST-AMP PD LE HS PDM INPUT RING LOGIC INTERFACE SWITCHING CONVERTER CM A MT VOL TI LO-AMP LO VssP Figure 2: Block Diagramme Rev. 5.1 Page 4 May 1999 Preliminary AS2520/21/20B/21B (see application notes). The dc resistance of R1 should be kept at 30 ohm to ensure correct dc condition. 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. Return loss and sidetone cancellation can be determined independent of each other (see figure 4). Speech Circuit 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 speech circuit consists of a transmit and a receive path with soft clipping, mute, line loss compensation and sidetone cancellation. Transmit 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: 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. VLS = VLI + ILINE ž R1 The voltage at LI is 4.5V. The unique dual soft clipping control circuit limits the output voltage at LI to 2VPEAK. Dual means that the soft clipping incorporates both a very fast control circuit to eliminate harsh sidetone distortion and a slower regulation circuit to limit the output voltage at 2VPEAK 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. Below 13 mA the AS252x provides an additional slope in order to allow parallel operation (see figure 3). 8 (V) 7 VLS 6 5 VLI 4 3 Receive Typically No ac signals Tamb: 25°C 2 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. 1 0 0 10 20 30 40 50 Line Current 60 70 80 90 100 (mA) Figure 3: DC Mask When mute is active during dialling the gain is reduced by > 60 dB. During DTMF dialling a MF 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. 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 VSS in order to turn the external shunt transistor on to keep a low voltage drop at the LS pin during make periods. Volume Control The synthesised ac impedance of the circuit is set on chip and by an external resistor and an external capacitor (for complex impedance). 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. 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). 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. For 600 ohm telephones it is recommended to connect a resistor and a capacitor from pin LS to VSS The volume control is common for both the earpiece and the loudspeaker. Any increase will be compensated to ensure acoustic stability. AC Impedance Rev. 5.1 Page 5 May 1999 Preliminary AS2520/21/20B/21B 20 to 50 mA or 45 to 75 mA depending on selected range. 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. 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. 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. 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. Sidetone Anti-Clipping (not AS2520B/21B) A good sidetone cancellation is achieved by using the following equation: The anti-clipping circuit is activated in loudhearing and handsfree mode. The circuit prevents harsh distortion at very high signal levels. ZBAL/ZLINE = R5/R1 The sidetone cancellation signal is applied to the STB input. 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. 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. 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. 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 R1 30 ohm ZLINE ZBAL 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. R5 300 ohm Figure 4: Sidetone Bridge 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. 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 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. 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 Rev. 5.1 Page 6 May 1999 Preliminary AS2520/21/20B/21B 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). 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. 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. 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. 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. SIDE TONE VTX PEAK DETECTOR AGC ZAC VLINE 2/4 VTH 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. VRX ± 10 dB VOL 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 VRX > VTX (see figure 5). 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. 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. 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. 300 (mV) Line Output Signal 250 In more technical terms, the AS252x works in the following manner: 200 150 Sidetone Cancellation: 11 dB Volume Control: 0 dB (neutral) 100 50 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. Rev. 5.1 0 0.00 0.25 0.50 0.75 1.00 Microphone Input Signal 1.25 1.50 (mV) Figure 6: Speech Compression Page 7 May 1999 Preliminary AS2520/21/20B/21B 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. 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. 1µ5 La 5k6 Lb 1µ5 Switching Converter 510 33 n BC 327 30V 10 k BC 547 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. CM 2.2 mH VPP VPP 470 µ 5V1 VssP 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. 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). 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 VPP. 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. When off-hook the switching converter has a high impedance (CM low) to avoid any influence on the transmission and on pulse dialling. Rev. 5.1 AS252x 510 Page 8 May 1999 Preliminary AS2520/21/20B/21B Electrical Characteristics Absolute Maximum Ratings* Supply Voltage............................................................................................................................... -0.3 ≤ VDD ≤ 7V Input Current..........................................................................................................................................+/- 25 mA Input Voltage (LS) ....................................................................................................................... -0.3V ≤ VIN ≤ 10V Input Voltage (LI, CS, SS).............................................................................................................-0.3V ≤ VIN ≤ 8V Input Voltage (STB, RI).........................................................................................................-2V ≤ VIN ≤ VDD +0.3V Digital Input Voltage .......................................................................................................... -0.3V ≤ VIN ≤ VDD + 0.3V 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 VDD Conditions Min. Typ.* Max. Supply Voltage (internally generated) Speech mode 3.0 4.1 5.5 V VPP Supply Voltage (internally regulated) 3.0 4.1 5.5 V TAMB Ambient Operating Temp. Range +70 °C Speech mode -25 Units * Typical figures are at 25°C and are for design aid only; not guaranteed and not subject to production testing. DC Characteristics (ILINE = 15 mA, recommended operating conditions unless otherwise specified) Symbol Parameter Conditions IDD Operating Supply Current Min. Typ. Max. Units HS = high 5 7 mA LE = high 5 7 mA HS and LE = high 5 7 µA PD = high, CM running 300 µA 200 µA 1 µA IDDPD Power-Down Current PD = high IDD0 Standby Current All digital inputs = VSS V Line Voltage 13 mA< ILINE < 100 mA IOL Output Current, Sink Pin CS, SS VOL = 0.4V 1.5 mA IOL Output Current, Sink Pin CM VOL = 0.4V 1.5 mA VIL Input Low Voltage TAMB = 25°C VSS VIH Input High Voltage TAMB = 25°C 0.8 VDD LI Rev. 5.1 Page 9 4.2 4.5 4.8 V 0.2 VDD V VDD V May 1999 Preliminary AS2520/21/20B/21B AC Electrical Characteristics ILINE = 15 mA; f = 800 Hz; recommended operating conditions unless otherwise specified. Transmit Symbol Parameter Conditions Min. Typ. Max. ATX Gain (M1/M2 to LS) HS, LH modes; LLC = AGND 35 36.5 38 dB Gain (M3/M4 to LS) HF mode; LLC = AGND 45 46.5 48 dB AMF Gain (TI to LS) MF mode 12 13.5 15 dB ∆ATX/F Variation with Frequency f = 500 Hz to 3.4 kHz ALLC Gain Range, LLC Speech mode; LLC = VSS or VDD THD Distortion VLI < 0.25 VRMS VAGC Soft Clip Level HS, LH modes; VLI = VAGC Soft Clip Level HF mode; VLI = ASCO Soft Clip Overdrive ZIN Input Impedance; AAD Attenuation Depth AMUTE Mute Attenuation Mute activated VNO Noise Output Voltage HS = high; TAMB = 25°C -72 dBmp LE = high; HS = low; TAMB = 25°C -62 dBmp VIN MAX Input Voltage Range; M1/M2 +/- 0.8 dB -6 dB 2 2 M1/M2 and M3/M4 Single ended % VPEAK 650 mVPEAK 30 dB 15 kohm 30 dB 60 Differential Units dB +/- 1 VPEAK +/- 0.5 VPEAK Line Driver Symbol Parameter VIN MAX Input Voltage Range; LI RL Return Loss ∆ZAC/TEMP Temperature Variation Rev. 5.1 Test Conditions Min. Typ. +/- 2 ZRL = 1000 ohm; TAMB = 25°C 18 Units VPEAK dB 0.5 Page 10 Max. Ω/°C May 1999 Preliminary AS2520/21/20B/21B Receive Symbol Parameter Condition ARX Gain (LS to RO), Default Volume reset LSP Gain (LSI to LO) ∆ATX/F Variation with Frequency f = 500 Hz to 3.4 kHz ALLC Gain Range, LLC ARX Min. Typ. Max. Units 1.5 3 4.5 dB 17.5 19 20.5 dB +/- 0.8 dB Speech mode; LLC = VSS or VDD -6 dB Volume Range 10 steps, each 2 dB 20 dB THD Distortion VRI < 0.2 VRMS VSC Soft Clip Level (RO) VRO = Soft Clip Level (LO) Not AS2520B/21B; VLO = Unloaded 2 % 1 VPEAK 1.3 VPEAK 30 dB ASCO Soft Clip Overdrive VRTH Threshold Voltage at RTH AAD Attenuation Depth tDECAY Attack Time Channel control; VRI > 0.8 VRMS µs/6dB tDECAY Decay Time Channel control µs/6dB VNO Noise Output Voltage (RO) HS = high; TAMB = 25°C -72 dBmp VUFC Unwanted Frequency Components (RO) -60 dBmp ZIN Input Impedance, RI VIN RI Input Voltage Range, RI AST Sidetone Cancellation ZIN Input Impedance, STB VIN ST Input Voltage Range, STB 7 15 25 30 50 Hz.........20 kHz VRI < 0.2 VRMS; TAMB = 25°C mV dB 8 kohm +/- 2 VPEAK 26 dB 80 kohm +/- 2 VPEAK General Timings Symbol Parameter tVOL Volume Key Debounce tSCA Soft Clip Attack Time VIN above soft clip level 0.12 ms/6dB tSCD Soft Clip Decay Time VIN below soft clip level 128 ms/6dB tPDA Peak Detector Attack Time VIN above VTH 3.2 ms/V tPDD Peak Detector Decay Time VIN below VTH 29 ms/V tLPA Low-Power Attack Time VPP < 3.6V 250 ms/6dB tLPD Low-Power Release VPP > 3.6V 1 sec/6dB Rev. 5.1 Condition Min. Typ. 7 Page 11 Max. Units ms May 1999 Rev. 5.1 I LINE B UL BC 327 10 V 600 ohm 100 µ A BC 327 22 µ 300 ohm 30 ohm 10 µ 6k 680 n Page 12 23 7 15 17 18 16 14 12 24 26 25 27 6 28 1 11 VOL LLC MT LE HS PD VsSP VPP SS Vss CS LI STB RI LS CM VDD TI LSI RO LO RTH AGND M4 M3 M2 M1 CI 4 9 8 3 13 10 5 20 21 22 19 2 22 µ 10 µ 1k 1k 1k 1k 200 ohm 25 ohm Preliminary AS2520/21/20B/21B Test Circuit AS252x Figure 8: Test Circuit May 1999 Preliminary AS2520/21/20B/21B Application Diagramme La 30 Ω 2k2 10 V Lb 27 220 µ LI CI 2 1k2 25 BC327 CS M1 1 LS 28 RI 10 µ 4 VDD VDD 9 TI INPUT FOR DTMF AND TONE RINGER MELODIES 33 n 18 16 CONTROL INPUTS (FROM µC) 1µ5 HS PD 15 MT 17 1µ5 22 15 n LE M2 1k2 300Ω 1µ 6 STB 1k8 Side tone balance network 7k5 10 n 10 µ RO 3 100 n 8 100 n LSI 10 RTH 1k8 M3 510Ω 15 n Vss LINE ADAPTER/TELEPHONE VOICE CIRCUIT 26 19 21 100 n 20 100 n 5 510Ω AGND 100 µ 24 33 n 5k6 BC327 SS 10 k MPSA92 11 AS252x M4 1k8 100 µ LO 13 7 LLC 2.2 mH Low 32 Ω VDD High CM 2N5551 AGND Off 12 5V1 470 µ 10 k VPP 23 14 VSSP VOL AS2521 AS2520 100 k VOL+ VOL- Figure 9: Application Diagramme Applications Hints Interface to Microcontroller In off-hook condition the microcontroller can be supplied from VDD of AS252x. The digital inputs (HS, LE, PD, and MT) must be kept low until VDD has reached its minimum operating voltage (>2.5V). 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 Preliminary AS2520/21/20B/21B Ordering Information 28 Pin SOP/DIP Part Number Package Type Volume Control Soft Clip Loudspk. RI LI VSS AS2520 T 28 pin SOP Digital Yes CS SS VOL M2 M3 M4 M1 HS LE AS2520 P 28 pin DIP Digital Yes AS2520B T 28 pin SOP Digital No AS2520B P 28 pin DIP Digital No AS2521 T 28 pin SOP Analogue Yes AS2521 P 28 pin DIP Analogue Yes AS2521B T 28 pin SOP Analogue No AS2521B P 28 pin DIP Analogue No LS CI RO VDD AGND STB LLC LSI TI RTH CM VPP LO VSSP 1 2 3 4 5 6 7 8 9 10 11 12 13 14 AS252x Pin Configuration 28 27 26 25 24 23 22 21 20 19 18 17 16 15 PD MT 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