MITEL MT9094AP Datasheet

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ISO

-CMOS ST-BUS FAMILY

MT9094



Digital Telephone (DPhone-II)

Features

• Programmable µ-Law/A-Law codec and filt ers

• Program mable CCITT (G .711)/sign-magni tude coding

• Program mab le trans mit , receiv e and si de-t one gains

• DSP-based:

i) Speakerphone switching algorithm ii) DTMF and single tone generator iii) Tone Ringer

• Differential interf ace to telepho ny tra nsdu cers

• Differential audio paths

• Singl e 5 volt pow er su ppl y

Applications

• Fully f eatu red dig ital t eleph one set s

• Cellula r phone sets

• Local area com m unications stations

ISSUE 2 May 1995

Ordering Information

MT9094AP 44 Pin PLCC

-40°C to +85°C

Description

The MT9094 DPhone-II is a fully featured integrated digital telephone circuit. Voice band signals are converted to digital PCM and vice versa by a switched capacitor Filter/Codec. The Filter/Codec uses an ingenious differential architecture to achieve low noise operation over a wide dynamic range with a single 5V supply. A Digital Signal Processor provides handsfree speaker-phone operation. The DSP is also used to generate tones (DTMF, Ringer and Call Progress) and control audio gains. Internal registers are accessed through a serial microport conforming to INTEL MCS-51™ specifications. The device is fabricated in Mitel's low power ISO technology.

-CMOS

DSTo

DSTi

F0i

C4i

VSSD

VDD

VSSA

VSS

SPKR

VBias

VRef

Digital Signal Processor Filter/Codec Gain

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22.5/-72dB

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C-Channel Registers

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∆1.5dB

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Tx & Rx

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ENCODER

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DECODER

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STATUS

Control

Registers

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-7dB

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Timing

Circuits

LCD Driver

S1 S12

BP WD PWRST

Figure 1 Functional Block Diagram

7dB

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Transd ucer

Interface

New Call

Tone

Generator

S/P &

P/S

Converter

Serial

Port

MCS-51

(

Compatible)

MICMIC+

MM+

HSPKR+ HSPKRSPKR+

SPKR-

DATA 2 DATA 1 SCLK CS

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MT9094

PWRSTICVBias

VRef

NCM-VSSA

M+

MIC+

MIC-

VSS SPKR

VSSD

4443424140

2425262728

S3S4S5S6S7

39 38 37 36 35 34 33 32 31 30 29

SPKR+ SPKRHSPKR+ HSPKRVDD BP S12 S11 S10 S9 S8

DSTi

DSTo

C4i

F0i

VSSD

SCLK DATA 2 DATA 1

65432

7 8 9 10 11 12 13 14 15 16 17

1819202122

44 PIN PLCC

Figure 2 - Pin Connections

Pin Description

Pin # Name Description

1M+Non-Inverting M icr oph on e (Inp ut). Non-inverting input to microphone amplifier from the

handset microphone. 2NCNo Connect. No internal connection to this pin. 3V

5ICInternal Connection . Tie externally to V 6 PWRST

Bias Voltage (Output). (VDD/2) volts is available at this pin for biasing external amplifiers.

Bias

Connect 0.1 µF capacitor to V

Reference voltage for codec (Outp ut). Nominally [(VDD/2)-1.5] volts. Used internally.

Ref

Connect 0.1 µF capacitor to V

SSA

for normal operation.

Power-up Reset (Input). CMOS compatible input wit h Schmit t Trigger (active low). 7DSTiST-BUS Serial Stream (Input). 2048 kbit/s input stream composed of 32 eight bit channels;

the first four of which are used by the MT9094. Input level is TTL compati ble. 8DSToST-BUS Serial Stream (Output). 2048 kbit/s output stream compose d of 32 eight bit

channels. The MT9094 sources digital signals during the appropriate channel, time coincident

with the channels used for DSTi. 9C4i

10 F0i

4096 kHz Clock (Input). CMOS level compatible.

Frame Pulse (Input). CMOS level compatible. This input is the frame s ynch ronizati on pulse

for the 2048 kbit/s ST-BUS stream.

11 V

Digital Ground. Nominally 0 volt s .

SSD

12 NC No Connect. No internal connection to this pin. 13 SCLK Serial Port Synchronous Clock (Input). Data clock for MCS-51 com pati ble mi c roport. TTL

level compatible.

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MT9094

Pin Description (continued)

Pin # Name Description

14 DATA 2 Serial Data T ransmit. In an alternate mode of operation, this pin is used for data transmit from

MT9094. In the default mode, serial data transmit and receive are performed on the DATA 1 pin and DATA 2 is tri-stated.

15 DATA 1 Bidirectional Serial Data. Port for microprocessor serial data transfer compat ible with

MCS-51 standard (default mode). In an alternate mode of operation , this pin becomes the data receive pin only and data transmit is performed on the DATA 2 pin. Input level TTL compatible.

16 CS

Chip Select (Input). This input signal is used to select the device for microport data transfers.

Active low. (TTL level compatible. ) 17 WD Watchdog (Output). Watchdog timer outp ut. Active high. 18 IC Internal Connecti on. Tie externally to V

19,

NC No Connection. No internal connection to these pins.

for normal operation.

20 21 V

Digital Ground. Nominally 0 volt s .

SSD

22-33S1-S12 Segm en t Drivers (Output). 12 independently contro lled, two level , LCD segment drivers. An

in-phase signal, with respect to the BP pin, produces a non-energized LCD segment. An

out-of-phase signal, with resp ect to the BP pin, energizes its re spective LCD segm ent . 34 BP Backp lan e Drive (Outpu t ). A two-level output voltage for biasing an LCD backplane. 35 V

Positive Pow er Su pp ly (Inpu t ). Nominally 5 volts.

36 HSPKR- Inverting Han dset Speak er (Outp ut). Output to the handset speaker (balanced). 37 HSPKR+ Non-Inverting Handset Speaker (Output). Output to the handset speaker (balanced). 38 SPKR- Inverting Spea ker (Output). Out put to the speakerphone speaker (balanced). 39 SPKR+ Non-Inv ertin g Speaker (O utp ut). Output to the speakerphone speaker (balanced). 40 V

Power Supply Rail for Analog Outp ut Drive rs. Nominally 0 Volts.

SPKR

41 MIC- Inverting Handsfree Microp hone (In pu t ). Handsfree microphone amplifier inverting input

pin. 42 MIC+ Non-inverting Han dsfree Microp ho ne (Input). Handsfree microphone amplif ier

non-inverting input pin. 43 V

Analog Ground. Nominally 0 V.

SSA

44 M- Inverting Micr op hon e (In put). Inverting input to microphone amplifie r from the handset

microphone.

NOTES: Intel and MCS-51 are registered trademarks of Intel Corporation, Santa Clara, CA, USA.

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MT9094

Overview

The Functional Block Diagram of Figure 1 depicts the main operations performed within the DPhone-II. Each of these functional blocks will be described in the sections to follow. This overview will describe some of the end-user features which may be implemented as a direct result of the level of integration found within the DPhone-II.

The main feature required of a digital telephone is to convert the digital Pulse Code Modulated (PCM) information, be ing rece ived by the telephon e set, into an analog electrical signal. This signal is then applied to an appropriate audio transducer such that the information is finally converted into intelligible acoustic energy. The same is true of the reverse direction where acoustic energy is converted first into an electrical analog and then digitized (into PCM) before being transmitted from the set. Along the way if the signals can be manipulated, either in the analog or the digital domains, other features such as gain control, signal generation and filtering may be added. More complex processing of the digital signal is also possible and is limited only be the processing power available. One example of this processing power may be the inclusion of a complex handsfree switching algorithm. Finally, most electro-acoustic transducers (loudspeakers) require a large amount of power to develop an effective acoustic signal. The inclusion of audio amplifiers to provide this power is required.

essential that the older methods be available for backward compatibility. As an example; once a call has been established, say from your office to your home, using the D-Channel signalling protocol it may be necessary to use in-band DTMF signalling to manipulate your personal answering machine in order to retrieve messages. Thus the locally generated tones must be of network quality and not just a reasonable facsimile. The DPhone-II DSP can generate the required tone pairs as well as single tones to accommodate any in-band signalling requirement.

Each of the programmable parameters within the functional blocks is accessed through a serial microcontroller port compatible with Intel MCS-51 specifications.

Functional Descripti on

In this section, each functional block within the DPhone-II is described along with all of the associated control/status bits. Each time a control/ status bit(s) is described it is followed by the address register where it will be found. T he reader is r eferred to the section titled ‘Register Summary' for a complete listing of all address map registers, the control/status bits associated with each register and a definition of the function of each control/status bit. The Register Summary is useful for future reference of control/status bits without the need to locate them within th e tex t o f th e f unctional des crip ti o ns.

The DPhone-II features Digital Signal Processing (DSP) of the voice encoded PCM, complete Analog/ Digital and Digital/Analog conversion of audio signals (Filter/CODEC) and an analog interface to the external world of electro-acoustic devices (T ransducer Interface). These three functional blocks combine to provide a standard full-duplex telephone conversation utilizing a common handset. Selecting transducers for handsfree operation, as well as allowing the DSP to perform its handsfree switching algorithm, is all that is required to convert the full-duplex handset conversation into a half-duplex speakerphone conversation. In each of these modes, full programmability of the receive path and side-tone gains is available to set comfortable listening levels for the user as well as transmit path gain control for setting nominal transmit levels into the network.

The ability to generate tones locally provides the designer with a familiar method of feedback to the telephone user as they proceed to set-up, and ultimately, dismantle a telephone conversation. Also, as the network slowly evolves from the dial pulse/ DTMF methods to the D-Channel protocols it is

Filter-CODEC

The Filter/CODEC block implements conversion of the analog 3.3kHz speech signals to/from the digital domain compatible with 64kb/s PCM B-Channels. Selection of companding curves and digital code assignment are register programmable. These are CCITT G.711 A-law or µ-Law, with true-sign/ Alternate Digit Inversion or true-sign/Inverted Magnitude coding, respectively. Optionally, signmagnitude coding may also be selected for proprietary applications.

The Filter/CODEC block also implements transmit and receive audio path gains in the analog domain. These gains are in addition to the digital gain pad provided in the DS P section and provide an overall path gain resolution of 0.5dB. A programmable gain, voice side-tone path is also included to provide proportional transmit speech feedback to the handset receiver so that a dead sounding handset is not encountered. Figure 3 depicts the nominal half-channel and side-tone gains for the DPhone-II.

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MT9094

On PWRST (pin 6) the Filter/CODEC defaults such that the side-tone path, dial tone filter and 400Hz transmit filter are off, all programmable gains are set to 0dB and µ-Law companding is selected. Further, the Filter/CODEC is powered down due to the PuFC bit (Tra n s d ucer Control Regi ster, address 0Eh) being reset. This bit must be set high to enable the Filter/ CODEC.

The internal architecture is fully differential to provide the best possible noise rejection as well as to allow a wide dynamic range from a single 5 volt supply design. This fully differential architecture is continued into the Transducer Interface section to provide full chip realization of these capabilites.

SERIAL

PORT

DSP GAIN*

FILTER/CODEC

A reference voltage (V

), for the conversion

Ref

requirements of the CODER section, and a bias voltage (V sections, are both generated on-chip. V

), for biasing the internal analog

Bias

is also brought to an external pin so that it may be used for biasing any external gain plan setting amplifiers. A

0.1µF capacitor must be connected from V analog ground at all times. Likewise, although V

Bias

Ref

may only be used internally, a 0.1µF capacitor from the V

pin to ground is required at all times. It is

Ref

suggested that the analog ground reference point for these two capacitors be physically the same point. To facilitate this the V

Ref

and V

pins are situated

Bias

on adjacent pins.

The transmit filter is designed to meet CCITT G.714 specifications. The nominal gain for this filter path is 0dB (gain control = 0dB). An anti-aliasing filter is

TRANSDUCER INTERFACE

µ-Law –6.3 dB Α-Law –3.7 dB

-6 dB

HSPKR+

Handset Receiver (150Ω)

PCM

Receive

–72 to

+22.5 dB

(1.5dB

steps)

DTMF,

Tone

Ringer &

Handsfree

–72 to

+22.5 dB

(1.5dB

steps)

Transmit

Receive Filter Gain 0 to –7 dB

(1 dB steps)

Side-tone

–9.96 to +9.96d B

(3.32 dB steps)

Side-tone

Nominal

Gain

µ-Law –11 dB Α-Law –18.8 dB

Transmit

Filter Gain

0 to +7dB

(1 dB steps)

-6 dB

Speaker Gain

0 to –24 dB

(8 dB steps)

µ-Law 6.1dB Α-Law 15.4dB

Transmit

Gain

Receiver

Driver

Speaker

Phone

Driver

0.2dB*

Tone

Ringer

(input

from DSP)

M U X

HSPKR–

SPKR+

SPKR–

MIC+ MIC–

M+ M–

Speakerphone

Speaker

(40Ω nominal)

(32Ω min)

Handsfree mic

Transmitter microphone

DIGITAL DOMAIN

Internal to Device External to Device

Note: *gain the same for A-Law and m

−

ANALOG DOMAIN

Law

Figure 3 - Audio Gain Partitioning

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MT9094

included. This is a second order lowpass implementation with a corner frequency at 25kHz. Attenuation is better than 32dB at 256 kHz and less than 0.01dB within the passband.

An optional 400Hz high-pass function may be included into the transmit path by enabling the Tfhp bit in the Transducer Control Register (address 0Eh). This option allows the reduction of transmitted background noise such as motor and fan noise.

The receive filter is designed to meet CCITT G.714 specifications. The nominal gain for this filter path is 0 dB (gain control = 0dB). Filter response is peaked to compensate for the sinx/x attenuation caused by the 8 kH z sam p li ng rat e.

The Rx filter function can be alt ered by enabling the DIAL EN control bit in the Transducer Control Register (address 0Eh). This causes another lowpass function to be added, with a 3dB point at 1000Hz. This function is intended to improve the sound quality of digitally generated dial tone received a s PCM.

Transmit sidetone is derived from the Tx filter and is subject to the gain control of the Tx filter section. Sidetone is summed into the receive path after the Rx filter gain control section so that Rx gain adjustment will not affect sidetone levels. The side-tone path may be enabled/disabled with the SIDE EN bit located in the Transducer Control Register (address 0Eh). See also STG

-STG

(address 0Bh).

Transmit and receive filter gains are controlled by the TxFG0-TxFG2 and RxFG0-RxFG2 control bits respectively. These are located in the FCODEC Gain Control Register 1 (address 0Ah). Transmit filter gain is adjustable from 0dB to +7dB and receive filter gain from 0dB to -7dB, both in 1dB increments.

main purpose is to provide both a digital gain control and a half-duplex handsfree switching function. The DSP will also generate the digital patterns required to produce standard DTMF signalling tones as well as single tones and a tone ringer output. A programmable (ON/OFF) offset null routine may also be performed on the transmit PCM data stream. The DSP can generate a ringer tone to be applied to the speakerphone speaker during normal handset operation so that the existing call is not interrupted.

The main functional control of the DSP is through two hardware registers which are accessible at any time via the microport. These are the Receive Gain Control Register at address 1Dh and the DSP Control Register at address 1Eh. In addition, other functional control is accomplished via multiple RAM-based registers which are accessible only while the DSP is held in a reset state. This is accomplished with the DRESET

bit of the DSP Control Register. Ram-based registers are used to store transmit gain levels (20h for transmit PCM and 21h for transmit DTMF levels), the coefficients for tone and ringer generation (addresses 23h and 24h), and tone ringer warble rates (address 26h). All undefined addresses below 20h are reserved for the temporary storage of interim variables calculated during the execution of the DSP algorithms. These undefined addresses should not be written to via the microprocessor port. The DSP can be programmed to execute the following micro-programs which are stored in instruction ROM, (see PS0 to PS2, DSP Control Register, address 1Eh). All program execution begins at the frame pulse boundary.

PS1 PS0 Micro-program

PS2

0 0 0 Power up reset program 0 0 1 Transmit and receive gain control

program; with autonulling of the transmit PCM, if the AUTO bit is set (see address 1Dh)

Side-tone filt er gain is controlled by the STG

-ST G

control bits located in the FCODEC Gain Control Register 2 (address 0Bh). Side-tone gain is adjustable from -9.96dB to +9.96dB in 3.32dB increments.

Law selection for the Filter/CODEC is provided by the A/µ companding control bit while the coding scheme is controlled by the sign-mag/CCITT

bit. Both of these reside in the General Control Register (address 0Fh).

Digital Signal Processor

The DSP block is located, functionally, between the serial ST-BUS port and the Filter/CODEC block. Its

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and receive gain control program (autonull available via the AUTO control bit)

0 1 1 Tone ringer plus transmit and

receive gain control program (autonull available via the AUTO

control bit) 1 0 0 handsfree switching program 101

1 1 0 Last three selections reserved 111

0 1 0 DTMF generation plus transmit

MT9094

Note: For the DSP to function it must be selected to

operate, in conjunction with the Filter/Codec, in one of the B-Channels. Therefore, one of the B-Channel enable bits must be set (see Timing Control, address 15h : bits CH

EN and CH3EN).

Power Up reset Pro gram

A hardware power-up reset (pin 6, PWRST) will initialize the DSP hardware registers to the default values (all zeros) and will reset the DSP program counter. The DSP will then be disabled and the PCM streams will pass transparently through t he DSP. The RAM-based registers are not reset by the PWRST pin but may be initialized to their default settings by programming the DSP to execute the power up reset program. None of the micro-programs actually require the execution of the power up reset program but it is useful for pre-setting the variables to a known condition. Note that the reset program requires one full frame (125µSec) for execution.

Gain Contr ol Progra m

Gain control is performed on converted linear code for both the receive and the transmit PCM. Receive gain control is set via the hardware register at address 1Dh (see bits B0 - B5) and may be changed at any time. Gain in 1.5dB increments is available within a range of +22.5dB to -72dB. Normal operation usually requires no more than a +20 to -20 dB range of control. However, the handsfree switching algorithm requires a large attenuation depth to maintain stability in worst case environments, hence the large (-72 dB) negative limit. Transmit gain control is divided into two RAM registers, one for setting the network level of transmit speech (address 20h) and the other for setting the transmit level of DTMF tones into the network (address 21h). Both registers provide gain control in

1.5dB increments and are encoded in the same manner as the receive gain control register (see address 1Dh, bits B0 - B5). The power up reset program sets the default values such that the receive gain is set to -72.0 dB, the transmit audio gain is set to 0.0dB and the transmit DTMF gain is set to -3.0 dB (equivalent to a DTMF output level of -4dBm0 into the network).

Optional Offset Nulling

Transmit PCM may contain residual offset in the form of a DC component. An offset of up to ±fifteen linear bits is acceptable with no degradation of the parameters def ined in CCITT G.714. The DPhone-II filter/CODEC guarantees no more than ±ten linear bits of offset in the transmit PCM when the autonull

routine is not enabled. By enabling aut onulling (see AUTO in the Receive Gain Control Register, address 1Dh) offsets a re reduced to within ±one bi t of zero. Autonulling circuitry was essential in the first generations of Filter/Codecs to remove the large DC offsets found in the linear technology. Newer technology has made nulling circuitry optional as offered in the DPhone-II.

DTMF and Gai n Cont rol Prog ram

The DTMF program generates a dual cosine wave pattern which may be routed into the receive path as comfort tones or into the transmit path as network signalling. In both cases, the digitally generated signal will undergo gain adjustment as programmed into the Receive Gain Control and the Transmit DTMF Gain Control registers. The composite signal output level in both directions is -4dBm0 when the gain controls are set to 2Eh (-3.0 dB). Adjustments to these levels may be made by altering the settings of the gain control registers. Pre-twist of 2.0dB is incorporated into the composite signal. The frequency of the low group tone is programmed by writing an 8-bit coefficient into Tone Coefficient Register 1 (address 23h), while the high group tone frequency uses the 8-bit coefficient programmed into Tone Coefficient Register 2 (address 24h). Both coefficients are determined by the following equation:

COEFF = 0.128 x Frequency (in Hz)

where COEFF is a rounded off 8 bit binary integer

A single frequency tone may be generated instead of a dual tone by programming the coefficient at address 23h to a value of zero. In this case thefrequency of the single output tone is governed by the coefficient stored at address 24h.

Frequency

(Hz)

697 59h 695.3 -.20% 770 63h 773.4 +.40% 852 6Dh 851.6 -.05%

941 79h 945.3 +.46% 1209 9Bh 1210.9 +.20% 1336 ABh 1335.9 .00% 1477 BDh 1476.6 -.03% 1633 D1h 1632.8 -.01%

COEF

Actual

Frequency%Deviation

Ta ble 1

DTMF Signal to distortion:

The sum of harmonic and noise power in the frequency band

from 50Hz to 350 0 Hz i s typi ca ll y mor e th an 30 dB below the power in th e t on e p air. All in div i du al ha r m on ic s are ty pically more th an 40 dB below th e le ve l of the low group to ne .

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MT9094

Table 1 gives the standard DTMF frequencies, the coefficient required to generate the closest frequency, the actual frequency generated and the percent deviation of the generated tone from the nominal.

Tone Ring er and G ain Co ntrol P rogram

A locally generated alerting (ringing) signal is used to prompt the user when an incoming call must be answered. The DSP uses the values programmed into Tone Coefficient Registers 1 and 2 (addresses 23h and 24h) to generate two different squarewave frequencies in PCM code. The amplitude of the squarewave frequencies is set to a mid level before being sent to the receive gain control block. From there the PCM passes through the decoder and receive filter, replacing the normal receive PCM data, on its way to the loudspeaker driver. Both coefficients are determined by the following equation:

COEFF = 8000/Frequency (Hz)

where COEFF is a rounded off 8 bit binary integer

The ringer program switches between these two frequencies at a rate defined by the 8-bit coeff icient programmed into the Tone Ringer Warble Rate Register (address 26h). The warble rate is defined by the equation:

Tone duration (warble frequency

in Hz) = 500/COEFF

where 0 < COEFF < 256, a warble rate of 5-20Hz is suggested.

Handsfree Program

A half-duplex speakerphone program, fully contained on chip, provides high quality gain switching of the transmit and receive speech PCM to maintain loop stability under most network and local acoustic environments. Gain switching is performed in continuous 1.5dB increments and operates in a complimentary fashion. That is, with the transmit path at maximum gain the receive path is fully attenuated and vice versa. This implies that there is a mid position where both transmit and receive paths are attenuated equally during transition. This is known as the idle state.

Of the 64 possible attenuator states, the algorithm may rest in only one of three stable states; full receive, full transmit and idle. The maximum gain values for full transmit and full receive are programmable through the microport at addresses 20h and 1Dh respectively, as is done for normal handset operation. This allows the user to set the maximum volumes to which the algorithm will adhere. The algorithm determines which path should maintain control of the loop based upon the relative levels of the transmit and receive audio signals after the detection and removal of background noise energy. If the algorithm determines that neither the transmit or the receive path has valid speech energy then the idle state will be sought. The present state of the algorithm plus the result of the Tx vs. Rx decision will determine which transition the algorithm will take toward its next stable state. The time durations required to move from one stable state to the next are parameters defined in CCITT Recommendation P.34 and are used by default by this algorith m (i.e ., b u ild- u p time , h a ng- ov e r time a nd switching time).

An alternate method of generating ringer tones to the speakerphone speaker is available. With this method the normal receive speech path through the decoder and receive filter is uninterrupted to the handset, allowing an existing conversation to continue. The normal DSP and Filt er/CODEC receive gain c ont rol is also retained by the speech path. When the OPT bit (DSP Control Register address 1Eh) is set high the DSP will generate the new call tone according to the coefficients programmed int o registers 23h, 24h and 26h as before. In this mode the DSP output is no longer a PCM code but a toggling signal which is routed directly through the New Call Tone gain control section to the loudspeaker driver. Refer to the section titled ‘New Call Tone’.

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Quiet Code

The DSP can be made to send quiet code to the decoder and receive filter path by setting the RxMUTE bit high. Likewise, the DSP will send quiet code in the transmit (DSTo) path when the TxM UTE bit is high. Both of these control bits reside in the DSP Control Register at address 1Eh. When either of these bits are low, their respective paths function normally.

Transducer Interfaces

Four standard telephony transducer interfaces are provided by the DPhone-II. These are:

➧ The handset microphone inputs (transmitter),

pins M+/M- and the speakerphone microphone inputs, pins MIC+/MIC-. The transmit path is muted/not-muted by the MIC EN control bit. Selection of which input pair is to be routed to the transmit filter amplifier is acomplished by the MIC/HNS TM IC

control bit. Both of these reside in the Transducer Control Register (address 0Eh). The nominal transmit path gain may be adjusted to either 6.1dB (suggested for µ-Law) or 15.4dB (suggested for A-Law). Control of this gain is provided by the MICA/u control bit (General Control Register, address 0Fh). This gain adjustment is in addition to the programmable gain provided by the transmit filter and DSP.

LCD

HSPKR+

MT9094

HSPKR-

MT9094

75 Ω

1000 pF

150 ohm

load

(speaker)

75 Ω

1000 pF

ground

Figure 4 - Handset Speaker Driver

➧ The handset speaker outputs (receiver), pins

HSPKR+/HSPKR-. This internally compensated, fully differential output driver is capable of driving the load shown in Figure 4. This output is enabled/disabled by the HSSPKR EN bit residing in the Transducer Control Register (address 0Eh). The nominal handset receive path gain may be adjusted to either

-12.3dB (suggested for µ-Law) or - 9.7dB (suggested for A-Law). Control of this gain is provided by the RxA/u

control bit (General Control Register, address 0Fh). This gain adjustment is in addition to the programmable gain provided by the receive filter and DSP.

➧ The loudspeaker outputs, pins SPKR+/SPKR-.

This internally compensated, fully differential output driver is capable of directly driving 6.5vpp into a 40 ohm load. This output is enabled/ disabled by the SPKR EN bit residing in the Transducer Control Register (address 0Eh). The nominal gain for this amplifier is 0.2dB.

C-Channel

A twelve segment, non-multiplexed, LCD display controller is provided for easy implementation of various set status and call progress indicators. The twelve output pins (S

) are used in conjunction with

12 segment control bits, located in LCD Segment Enable Registers 1& 2 (addresses 12h and 13h), and the BackPlane output pin (BP) to control the on/off state of each segment individually.

The BP pin drives a continuous 62.5Hz, 50% duty cycle squarewave output signal. An individual segment is controlled via the phase relationship of its segment driver output pin with respect to the backplane, or common, driver output. Each of the twelve Segment Enable bits corresponds to a segment output pin. The waveform at each segment pin is in-phase with the BP waveform when its control bit is set to logic zero (segment off) and is out-of-phase with the BP waveform when its control bit is set to a lo gic high ( segment on). Refer to the LCD Driver Characteristics for pin loading information.

Microport

Access to the internal c ontrol and status registers of Mitel bas ic rate, layer 1, tra nsceivers is thro ugh the ST-BUS Control Channel (C-Channel), since direct microport access is not usually provided, except in the case of the SNIC (MT8930). The DPhone-II provides asynchronous microport access to the ST-BUS C-Channel information on both DSTo and DSTi via a double-buffered read/write register (address 14h). Data written to this address is transmitted on the C-Channel every frame when enabled by CH

EN (see ST-BUS/Timing Contro l ) .

A serial microport, compatible with Intel MCS-51 (mode 0) specifications, provides access to all DPhone-II internal read and write registers. This microport consists of three pins; a half-duplex transmit/receive data pin (DATA1), a chip select pin

) and a synchronous data clock pin (SCLK).

(CS

On power-u p res et (PWRST

) or with a s o ftw a re reset (RST), the DATA1 pin becomes a bidirectional (transmit/receive) serial port while the DATA2 pin is internally disconnected and tri-stated.

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MT9094

All data transfers through the microport are two-byte transfers requiring the transmission of a Command/ Address byte followed by the data byte written or read from the addressed register. CS

must remain asserted fo r th e duration of this t wo - by te t ra n sfer. As shown in Figure 5, the falling edge of CS

indicates to the DPhone-II that a microport transfer is about to begin. The first 8 clock cycles of SCLK after the falling edge of CS

are always used to receive the Command/Address byte from the microcontroller. The Command/Address byte contains information detailing whether the second byte transfer will be a read or a write operation and of what address. The next 8 clock cycles are used to transfer the data byte between the DPhone-II and the microcontroller. At the end of the two-byte transfer CS

is brought high again to terminate the se ssion. The rising edge of CS will tri-state the output driver of DATA1 which will remain tr i- st ated as long as CS

is high.

Receive data is sampled and transmit data is made available on DATA1 concurrent with the falling edge of S CLK.

DATASEL; internal timing remains the same in both cases. Tri-stating on DATA2 follows CS

as it does on DATA1 when DATASEL is logic low. Use of the DATASEL bit is intended to help in adapting Motorola (SPI) and National Semiconductor (Micro-wire) microcontrollers to the DPhone-II. Note that whereas Intel processor serial ports transmit data LSB first other processor serial ports, including Motorola, transmit data MSB first. It is the responsibility of the microcontroller to provide LSB first data to the DPhone-II.

ST-BUS/Timing Control

A serial link is required for the transport of data between the DPhone-II and the external digital transmission device. The DPhone-II utilizes the ST-BUS architecture defined by Mitel Semiconductor. Refer to Mitel Application Note MSAN-126. The DPhone-II ST-BUS consists of output and input serial data streams, DSTo and DSTi respectively, a synchronous clock signal C4i framing pulse F0i

, and a

Lastly, provision is made to seperate the transmit and receive data streams onto two individual pins. This control is given by the DATASEL pin in the General Control Register (address 0Fh). Setting DATASEL logic high will cause DATA1 to become the data receive pin and DATA2 to become the data transmit pin. Only the signal paths are altered by

COMMAND/ADDRESS DATA INPUT/OUTPUT COMMAND/ADDRESS

DATA 1 Receive

DATA 1 or DATA 2 Transmit

SCLK

➀ ➁

➂

➃ ➄

D0D1D2D3D4D5D6D

➁

➂

Delays due to MCS-51 internal timing which are transparent. The DPhone-II: -latches received data on the falling edge of SCLK

The falling edge of CS byte is always data followed by CS

A new COMMAND/ADDRESS byte may be loaded only by CS The COMMAND/ADDRESS byte contains:

-outputs transmit data on the falling edge of SCLK indicates that a COMMAND/ADDRESS byte will be transmitted from the microprocessor. The subsequent

➄

➀

returning high.

0D1D2D3D4D5D6D7

D0D1D2D3D4D5D6D

1 bit - Read/Write 6 bits - Addressing Data 1 bit - Not used, write logic "0"

Figure 5 - S erial Port Rela tive Tim ing

The data streams operate at 2048kb/s and are Time Division Multiplexed into 32 identical channels of 64kb/s bandwidth. Frame Pulse (a 244nSec low going pulse) is used to parse the continuous serial data streams into the 32 channel TDM frames. Each frame has a 125µSecond period translating into an 8 kHz frame rate. Valid frame pulse occurs when F0i is

➀

➃

cycling high then low again.

0A5A4A3A2A1A0R/W

➃

D0D1D2D3D4D5D6D

➂

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