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7-45 features ? programmable - law / a - law codec and filters ? programmable ccitt (g.711)/sign-magnitude coding ? programmable transmit, receive and side-tone gains ? dsp-based: i) speakerphone switching algorithm ii) dtmf and single tone generator iii) tone ringer ? differential interface to telephony transducers ? differential audio paths ? single 5 volt power supply applications ? fully featured digital telephone sets ? cellular phone sets ? local area communications stations 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 2 -cmos technology. figure 1 functional block diagram aaaa a aaa a aaa a aaa a aaa a aaa a aaa a aaa a aaa a aaa a aaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaa aa a aa a aa a aa a aa a aa a aa a aa a aa a aa a aaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa aaaa digital signal processor filter/codec gain dsto dsti f0i c4i vssd vdd vssa vss spkr vbias vref s1 s12 bp wd pwrst ic cs sclk data 1 data 2 spkr- spkr+ hspkr- hspkr+ m+ m- mic+ mic- transducer interface 22.5/-72db d 1.5db tx & rx encoder 7db decoder -7db c-channel registers control registers timing circuits lcd driver new call tone generator s/p & p/s converter serial port ( mcs-51 compatible) status issue 2 may 1995 mt9094 digital telephone (dphone-ii) iso 2 -cmos st-bus ? family ordering information MT9094AP 44 pin plcc -40c to +85c
mt9094 7-46 figure 2 - pin connections pin description pin # name description 1m+ non-inverting microphone (input). non-inverting input to microphone amplifier from the handset microphone. 2nc no connect. no internal connection to this pin. 3v bias bias voltage (output). (v dd /2) volts is available at this pin for biasing external amplifiers. connect 0.1 f capacitor to v ssa . 4v ref reference voltage for codec (output). nominally [(v dd /2)-1.5] volts. used internally. connect 0.1 f capacitor to v ssa . 5ic internal connection. tie externally to v ss for normal operation. 6 pwrst power-up reset (input). cmos compatible input with schmitt trigger (active low). 7dsti st-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 compatible. 8dsto st-bus serial stream (output). 2048 kbit/s output stream composed of 32 eight bit channels. the mt9094 sources digital signals during the appropriate channel, time coincident with the channels used for dsti. 9c4i 4096 khz clock (input). cmos level compatible. 10 f0i frame pulse (input). cmos level compatible. this input is the frame synchronization pulse for the 2048 kbit/s st-bus stream. 11 v ssd digital ground. nominally 0 volts. 12 nc no connect. no internal connection to this pin. 13 sclk serial port synchronous clock (input). data clock for mcs-51 compatible microport. ttl level compatible. 1 6 5 4 3 2 44 43 42 41 40 7 8 9 10 11 12 13 14 15 16 39 38 37 36 35 34 33 32 31 30 23 18 19 20 21 22 24 25 26 27 28 17 29 pwrst ic vbias vref nc m- vssa mic+ mic- m+ dsti dsto c4i f0i vssd nc sclk data 2 data 1 cs wd spkr+ spkr- hspkr+ hspkr- vdd bp s12 s11 s10 s9 s8 ic nc nc vssd s1 s3 s4 s5 s6 s7 s2 44 pin plcc vss spkr
mt9094 7-47 notes: intel and mcs-51 are registered trademarks of intel corporation, santa clara, ca, usa. 14 data 2 serial data transmit. 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 compatible 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 output. active high. 18 ic internal connection. tie externally to v ss for normal operation. 19, 20 nc no connection. no internal connection to these pins. 21 v ssd digital ground. nominally 0 volts. 22-3 3 s1-s12 segment drivers (output). 12 independently controlled, 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 respect to the bp pin, energizes its respective lcd segment. 34 bp backplane drive (output). a two-level output voltage for biasing an lcd backplane. 35 v dd positive power supply (input). nominally 5 volts. 36 hspkr- inverting handset speaker (output). output to the handset speaker (balanced). 37 hspkr+ non-inverting handset speaker (output). output to the handset speaker (balanced). 38 spkr- inverting speaker (output). output to the speakerphone speaker (balanced). 39 spkr+ non-inverting speaker (output). output to the speakerphone speaker (bala nced). 40 v ss spkr power supply rail for analog output drivers. nominally 0 volts. 41 mic- inverting handsfree microphone (input). handsfree microphone amplifier inverting input pin. 42 mic+ non-inverting handsfree microphone (input). handsfree microphone amplifier non-inverting input pin. 43 v ssa analog ground. nominally 0 v. 44 m- inverting microphone (input). inverting input to microphone amplifier from the handset microphone. pin description (continued) pin # name description
mt9094 7-48 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, being received by the telephone 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. 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 (transducer 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 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 description 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. the reader is referred 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 the text of the functional descriptions. 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 -law, with true-sign/ alternate digit inversion or true-sign/inverted magnitude coding, respectively. optionally, sign- magnitude 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 dsp 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 .
mt9094 7-49 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 -law companding is selected. further, the filter/codec is powered down due to the pufc bit (transducer control register, 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. a reference voltage (v ref ), for the conversion requirements of the coder section, and a bias voltage (v bias ), for biasing the internal analog sections, are both generated on-chip. v bias is also brought to an external pin so that it may be used for biasing any external gain plan setting amplifiers. a 0.1f capacitor must be connected from v bias to analog ground at all times. likewise, although v ref may only be used internally, a 0.1f capacitor from the v ref pin to ground is required at all times. it is suggested that the analog ground reference point for these two capacitors be physically the same point. to facilitate this the v ref and v bias pins are situated 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 serial port dsp gain* filter/codec transducer interface pcm pcm C72 to +22.5 db (1.5db steps) dtmf, tone ringer & handsfree C72 to +22.5 db (1.5db steps) receive filter gain 0 to C7 db (1 db steps) side-tone C9.96 to +9.96db (3.32 db steps) side-tone nominal gain transmit filter gain 0 to +7db (1 db steps) m -law C11 db a -law C18.8 db -6 db speaker gain 0 to C24 db (8 db steps) receiver driver speaker phone driver 0.2db* tone ringer (input from dsp) transmit gain m u x m -law C6.3 db a -law C3.7 db m -law 6.1db a -law 15.4db mic+ micC m+ mC -6 db hspkr+ hspkrC spkr+ spkrC speakerphone speaker (40 w nominal) (32 w min) handsfree mic transmitter microphone handset receiver (150 w ) transmit internal to device external to device digital domain analog domain 75 75 note: *gain the same for a-law and m - law receive figure 3 - audio gain partitioning
mt9094 7-50 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 8khz sampling rate. the rx filter function can be altered 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 as 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 0 -stg 2 (address 0bh). transmit and receive filter gains are controlled by the txfg 0 -txfg 2 and rxfg 0 -rxfg 2 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. side-tone filter gain is controlled by the stg 0 -stg 2 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/ 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 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. ps2 ps1 ps0 micro-program 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) 0 1 0 dtmf generation plus transmit 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
mt9094 7-51 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 2 en and ch 3 en). power up reset program 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 the 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 (125sec) for execution. gain control program 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 defined 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 autonulling (see auto in the receive gain control register, address 1dh) offsets are reduced to within one bit 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 gain control program 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. table 1 dtmf signal to distortion: the sum of harmonic and noise power in the frequency band from 50hz to 3500hz is typically more than 30db below the power in the tone pair. all individual harmonics are typically more than 40db below the level of the low group tone. frequency (hz) coef actual frequency % deviation 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%
mt9094 7-52 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 ringer and gain control program 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 coefficient 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. 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 filter/codec receive gain control 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 into 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. 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 algorithm (i.e., build-up time, hang-over time and switching time). 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 txmute 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.
mt9094 7-53 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/hnstmic 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 -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. ? 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 -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 access to the internal control and status registers of mitel basic rate, layer 1, transceivers is through 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 1 en (see st-bus/timing control). figure 4 - handset speaker driver lcd 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 n ) 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 logic high (segment on). refer to the lcd driver characteristics for pin loading information. microport 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 (cs ) and a synchronous data clock pin (sclk). on power-up reset (pwrst ) or with a software reset (rst), the data1 pin becomes a bidirectional (transmit/receive) serial port while the data2 pin is internally disconnected and tri-stated. hspkr+ hspkr- 75 w 75 w 1000 pf 150 ohm load (speaker) 1000 pf ground mt9094
mt9094 7-54 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 for the duration of this two-byte transfer. 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 session. the rising edge of cs will tri-state the output driver of data1 which will remain tri-stated as long as cs is high. receive data is sampled and transmit data is made available on data1 concurrent with the falling edge of sclk. 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 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 , and a framing pulse f0i . 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 125second period translating into an 8 khz frame rate. valid frame pulse occurs when f0i is figure 5 - serial port relative timing d 0 d 1 d 2 d 3 d 4 d 5 d 6 d 7 0a 5 a 4 a 3 a 2 a 1 a 0 r/w d 7 d 0 command/address data input/output command/address data 1 receive data 1 or data 2 transmit sclk cs a ? ? a ? a ? ? delays due to mcs-51 internal timing which are transparent. the dphone-ii: -latches received data on the falling edge of sclk -outputs transmit data on the falling edge of sclk the falling edge of cs indicates that a command/address byte will be transmitted from the micropro cessor. the su bsequent byte is always data followed by cs returning high. a new command/address byte may be loaded only by cs cycling high then low again. the command/address byte con tains: 1 bit - read/write 6 bits - addressing data 1 bit - not used, write logic "0" d 0 d 1 d 2 d 3 d 4 d 5 d 6 d 7 d 0 d 1 d 2 d 3 d 4 d 5 d 6 d 7 d 0 d 1 d 2 d 3 d 4 d 5 d 6 d 7 d 0 d 1 d 2 d 3 d 4 d 5 d 6 d 7
mt9094 7-55 logic low coincident with a falling edge of c4i . c4i has a frequency (4096mhz) which is twice the data rate. this clock is used to sample the data at the _ bit-cell position on dsti and to make data available on dsto at the start of the bit-cell. c4i is also used to clock the dphone -ii internal functions (i.e., dsp, filter/codec, hdlc) and to provide the channel timing requirements. the dphone -ii uses channels 1, 2 & 3 of the 32 channel frame. these channels are always defined, beginning with the first channel after frame pulse, as shown in figure 6 (dsti and dsto channel assignments). channels are enabled independently by the three control bits ch 1 en -ch 3 en residing in the timing control register (address15h). ch 1 en - c-channel channel 1 conveys the control/status information for mitels layer 1 transceiver. the full 64kb/s bandwidth is available and is assigned according to which transceiver is being used. consult the data sheet for the selected transceiver for its bit definitions and order of bit transfer. when this bit is high register data is transmitted on dsto. when low, this timeslot is tri-stated on dsto. receive c-channel data (dsti) is always routed to the register regardless of this control bit's logic state. c-channel data is transferred on the st-bus msb first by the dphone -ii . ch 2 en and ch 3 en - b1-channel and b2-channel channels 2 and 3 are the b1 and b2 channels, respectively. these bits (ch 2 en and ch 3 en) are used to enable the pcm channels from/to the dphone -ii as required. transmit pcm on dsto when high, pcm from the filter/codec and dsp is transmitted on dsto in the selected st-bus channel. when low, dsto is forced to logic 0 for the corresponding timeslot. if both ch 2 en and ch 3 en are enabled, default is to channel 2. receive pcm from dsti when high, pcm from dsti is routed to the dsp and filter/codec in the associated channel. if both ch 2 en and ch 3 en are enabled the default is to channel 2. new call tone the new call tone generator produces a frequency shifted square-wave used to toggle the speaker driver outputs. this is intended for use where a ringing signal is required concurrently with an already established voice conversation in the handset. programming of the dsp for new call generator is exactly as is done for the tone ringer micro-program except that the opt bit (dsp control register, address 1eh) is set high. in this mode the dsp does not produce a frequency shifted squarewave output to the filter codec section. instead the dsp uses the contents of the tone coefficient registers, along with the tone warble rate register, to produce a gated squarewave control signal output which toggles between the programmed frequencies. this control signal is routed to the new call tone block when the nct en control bit is set (general control register, address 0fh). nct en also enables a separate gain control block, for controlling the loudness of the generated ringing signal. with the gain control block set to 0db the output is at maximum or 6 volts p-p. attenuation of the applied signal, in three steps of 8 db, provide the four settings for new call tone (0, -8, -16, -24 db). the nct gain bits (nctg 0 -nctg 1 ) reside in the fcodec gain control register 2 (address 0bh). figure 6 - st-bus channel assignment f0i dsti, dsto 125 m s channel 1 c-channel channel 3 b2-channel channel 2 b1-channel channels 4 - 31 not used msb first for c, b1 - & b2- channels not used
mt9094 7-56 watchdog to maintain program integrity an on-chip watchdog timer is provided for connection to the microcontroller reset pin. the watchdog output wd (pin 17) goes high while the dphone -ii is held in reset via the pwrst (pin 6). release of pwrst will cause wd to return low immediately and will also start the watchdog timer. the watchdog timer is clocked on the falling edge of f0i and requires only this input, along with v dd , for operation. if the watchdog reset word is written to the watchdog register (address 11h) after pwrst is released, but before the timeout period (t=512msec) expires, a reset of the timer results and wd will remain low. thereafter, if the reset word is loaded correctly at intervals less than 't' then wd will continue low. the first break from this routine, in which the watchdog register is not written to within the correct interval or it is written to with incorrect data, will result in a high going wd output after the current interval 't' expires. wd will then toggle at this rate until the watchdog register is again written to correctly. x=dont care test loops detail lbio and lboi loopback register (address 16h) lbio setting this bit causes data on dsti to be looped back to dsto directly at the pins. the appropriate channel enables ch 1 en -ch 3 en must also be set. lboi setting this bit causes data on dsto to be looped back to dsti directly at the pins. 5-bit watchdog reset word w4 w3 w2 w1 w0 xxx01010 dphone-ii register map address (hex) write read 00-09 reserved reserved 0a fcodec gain control register 1 verify 0b fcodec gain control register 2 verify 0c reserved reserved 0d reserved reserved 0e transducer control register verify 0f general control register verify 10 reserved reserved 11 watchdog register not used 12 lcd segment enable register 1 verify 13 lcd segment enable register 2 verify 14 c-channel register (to dsto) c-channel register (from dsti) 15 timing control register verify 16 loop-back register verify 17-1c reserved reserved 1d receive gain control register verify 1e dsp control register verify 1f reserved reserved 20 transmit audio gain register verify 21 transmit dtmf gain register verify 22 reserved reserved 23 tone coefficient register 1 verify 24 tone coefficient register 2 verify 25 reserved reserved 26 tone ringer warble rate register verify 27-3f reserved reserved
mt9094 7-57 register summary this section contains a complete listing of the dphone -ii register addresses, the control/status bit mapping associated with each register and a definition of the function of each control/status bit. the register summary may be used for future reference to review each of the control/status bit definitions without the need to locate them in the text of the functional block descriptions. addresses 00h and 09h are reserved addresses 0ch and 0dh are reserved note: bits marked "-" are reserved bits and should be written with logic "0". receive gain setting (db) rxfg 2 rxfg 1 rxfg 0 transmit gain setting (db) txfg 2 txfg 1 txfg 0 (default) 0 -1 -2 -3 -4 -5 -6 -7 0 0 0 0 1 1 1 1 0 0 1 1 0 0 1 1 0 1 0 1 0 1 0 1 (default) 0 1 2 3 4 5 6 7 0 0 0 0 1 1 1 1 0 0 1 1 0 0 1 1 0 1 0 1 0 1 0 1 fcodec gain control register 1 address = 0ah write/read verify power reset value x000 x000 76543210 - rxfg 2 rxfg 1 rxfg 0 txfg 2 txfg 1 txfg 0 - rxfg n = receive filter gain n txfg n = transmit filter gain n side-tone gain setting (db) stg 2 stg 1 stg 0 (default) off -9.96 -6.64 -3.32 0 3.32 6.64 9.96 0 0 0 0 1 1 1 1 0 0 1 1 0 0 1 1 0 1 0 1 0 1 0 1 fcodec gain control register 2 address = 0bh write/read verify power reset value 0x00 x000 76543210 -- nctg 1 nctg 0 stg 2 stg 1 stg 0 - stg n = side-tone gain n gain (db) nctg 1 nctg 0 0 (default) -8 -16 -24 0 0 1 1 0 1 0 1 nctgn = new call tone gain n
mt9094 7-58 address 10h is reserved note: bits marked "-" are reserved bits and should be written with logic "0". pufc when high, the filter/codec is powered up. when low, the filter/codec is powered down. if pufc, spkr en and hsspkr en are all low then the vref/vbias circuit is also powered down. tfhp when high, an additional high pass function (passband beginning at 400hz) is inserted into the transmit path. when low, this highpass filter is disabled. dial en when high, a first order lowpass filter is inserted into the receive path (3db = 1khz). when low, this lowpass filter is disabled. side en when high, the sidetone path is enabled (assuming stg 2-0 are not all low). when low, the sidetone path is disabled. mic en when high, the selected transmit microphone is enabled to the transmit filter section. when low, the microphone path is muted. mic/hnstmic when high, the handsfree microphone (pins mic) is muxed into the transmit path. when low, the handset microphone (pins m) is muxed into the transmit path. both are contingent on "mic en". spkr en when high, the handsfree loudspeaker driver is powered up. when low, this driver is powered down. hsspkr en when high, the handset speaker driver is powered up. when low, this driver is powered down. transducer control register address = 0eh write/read verify power reset value 0000 0000 76 543 2 1 0 pufc tfhp dial side mic/ spkr hsspkr mic en en en hnstmic en en rst active high reset. performs the same function as pwrst but does not affect the microport or the watchdog circuits. to remove this reset a pwrst must occur or this bit must be written low. datasel when high, the microport transmit and receive are performed on separate pins. data1 is receive while data2 is transmit. when low, the microport conforms to intel mcs-51 mode 0 specifications; data1 is a bi-directional (transmit/receive) serial data pin while data2 is internally disconnected and tri-stated. a/u when high, a-law (de)coding is selected. when low, -law (de)coding is selected. sign-mag/ccitt when high, sign-magnitude bit coding is selected, when low, true ccitt pcm coding is selected. rxa/u when high, the receiver driver nominal gain is set at -9.7 db. when low this driver nominal gain is set at -12.3 db. mica/u when high, the transmit amplifier nominal gain is set at 15.4 db. when low this amplifier nominal gain is set at 6.1 db. sidea/u when high, the side-tone nominal gain is set at -18.8 db. when low this nominal gain is set at -11 db. nct en when high, the new call tone generator output from the dsp is selected as the source for the loudspeaker path. when low, the codec output is selected for the loudspeaker path. note that spkr en must also be set high for new call tone to function. general control register address = 0fh write/read verify power reset value 0000 0000 76 543 2 1 0 rst data sign-mag/ mic side nct rx a/ m a/ m a/ m a/ m en sel ccitt
mt9094 7-59 note: bits marked "-" are reserved bits and should be written with logic "0". watchdog reset word - xxx01010 watchdog register address = 11h write power reset value xxx0 1010 76543210 ---w 4 w 3 w 1 w 2 w 0 twelve segment control bits used for the lcd outputs. when high the respective segment is on. when low the respective segment is off. lcd segment enable register 1 address = 12h write/read verify power reset value 0000 0000 76543210 sc 8 sc 7 sc 6 sc 5 sc 4 sc 2 sc 3 sc 1 twelve segment control bits used for the lcd outputs. when high the respective segment is on. when low the respective segment is off. lcd segment enable register 2 address = 13h write/read verify power reset value xxxx 0000 76543210 - - --sc 12 sc 10 sc 11 sc 9 c-channel register address = 14h write/read power reset value write = 1111 1111 76543210 d 7 d 6 d 5 d 4 d 2 d 1 d 0 d 3 micro-port access to the st-bus c-channel information. read = not applicable
mt9094 7-60 addresses 17h - 1ch are reserved note: bits marked "-" are reserved bits and should be written with logic "0". all bits active high: ch 2 en and ch 3 en channels 2 and 3 are the b1 and b2 channels, respectively. pcm associated with the dsp, filter/codec and trans- ducer audio paths is conveyed in one of these channels as selected in the timing control register. transmit b1 and b2 data on dsto when high pcm from the filte r/codec and dsp is transmitted on dsto in the associated channel. when low dsto is forced to logic 0 for the corresponding timeslot. if both ch 2 en and ch 3 en are enabled, data defaults to channel 2. receive b1 and b2 data on dsti when enabled pcm from dsti is routed to the dsp and filter/codec in the associated channel. if both ch 2 en and ch 3 en are enabled, data input defaults to channel 2. ch 1 en channel 1 conveys the control/status information for the layer 1 transceiver. the full 64kb/s bandwidth is available and is assigned according to which transceiver is being used. consult the data sheets for the transceiver selected. when high register data is transmitted on dsto. when low this timeslot is tri-stated on dsto. receive c-channel data (dsti) is always routed to the register regardless of this control bit's logic state. timing control register address = 15h write/read verify power reset value xx0x 0000 76543210 -- --ch 3 en ch 1 en ch 2 en - lb io active high enables data from the st-bus input to be looped back to the st-bus output directly at the pins. the dsto tri-state driver must also be enabled using one of the channel enable signals. lb oi active high enables data from st-bus output to be looped back to the st-bus input directly at the pins. loop-back register address = 16h write/read verify power reset value x00x xxxx 76543210 - lbio lboi - - - --
mt9094 7-61 note: bits marked "-" are reserved bits and should be written with logic "0". auto when high autonulling of the transmit pcm is enabled. when low, autonulling is disabled. this bit is used in conjunction with the ps2 - ps0 bits of the dsp control register at address 1eh. b5-b0 these 6 bits (indicated below in hexadecimal) are decoded to control rx pcm gain: note: b0-b5 of addresses 20h and 21h are encoded in the same manner b5-b0 gain setting (db) b5-b0 gain setting (db) 3f +22.5 1f -25.5 3e +21.0 1e -27.0 3d +19.5 1d -28.5 3c +18.0 1c -30.0 3b +16.5 1b -31.5 3a +15.0 1a -33.0 39 +13.5 19 -34.5 38 +12.0 18 -36.0 37 +10.5 17 -37.5 36 +9.0 16 -39.0 35 +7.5 15 -40.5 34 +6.0 14 -42.0 33 +4.5 13 -43.5 32 +3.0 12 -45.0 31 +1.5 11 -46.5 30 +0.0 10 -48.0 2f -1.5 0f -49.5 2e -3.0 0e -51.0 2d -4.5 0d -52.5 2c -6.0 0c -54.0 2b -7.5 0b -55.5 2a -9.0 0a -57.0 29-10.509-58.5 28-12.008-60.0 27-13.507-61.5 26-15.006-63.0 25-16.505-64.5 24-18.004-66.0 23-19.503-67.5 22-21.002-69.0 21-22.501-70.5 20-24.000-72.0 receive gain control register address = 1dh write/read verify power reset value 0000 0000 76543210 -autob5b4 b2b1b0 b3
mt9094 7-62 address 1fh is reserved address 22h is reserved note: bits marked "-" are reserved bits and should be written with logic "0". opt: when high, the tone ringer is in new call tone mode. when low the normal tone ringer program is executed. rxmute: this bit when high turns off the receive pcm channel, substituting quiet code. txmute: this bit when high turns off the transmit pcm channel, substituting quiet code. dreset : this bit (when high) enables the dsp. if low, no programs are executed, the master clock is disabled and the program counter is reset to zero. ps2-ps0: these bits are program select bits for the dsp rom programs. ps2 ps1 ps0 micro-program 0 0 0 power up reset program 0 0 1 gain control program 0 1 0 dtmf & gain control program 0 1 1 tone ringer & gain control program 1 0 0 handsfree program 101reserved 110reserved 111reserved dsp control register address = 1eh write/read verify power reset value 0000 0000 76543210 ps2 ps1 ps0 opt rxmute - txmute dreset transmit audio gain register address = 20h write/read verify power reset value xx11 0000 76543210 - - b5 b4 b2 b1 b0 b3 this register controls the transmit speech path gain in 1.5db steps as in receive gain register (address 1dh). transmit dtmf gain register address = 21h write/read verify power reset value xx10 1110 76543210 - - b5 b4 b2 b1 b0 b3 this register controls the transmit dtmf level in 1.5db steps as in receive gain register (address 1dh).
mt9094 7-63 address 25h is reserved addresses: 27h to 2dh are transmit and receive gains and coefficients used by the filters in the handsfree decision circuit. 2eh to 3fh are scratch-pad ram locations used by the dsp algorithms as temporary storage dur- ing calculations. this register is used to program the low-group frequency of the dtmf program. the tone coefficient is calculated as follows: coef = 0.128 x frequency where: frequency is in hz (note: coef must be converted to an 8 bit binary integer) highest frequency possible: 1992.2 hz lowest frequency possible: 7.8 hz frequency resolution: 7.8 hz pre-twist: -2.1db 0.2db this register is used to program the first frequency of the squarewave ringer program. the tone coefficient is calculated as follows: coef = 8000 / frequency where: frequency is in hz (note: coef must be rounded off and converted to an 8 bit binary integer) highest frequency possible: 4000 hz lowest frequency possible: 31.4 hz frequency resolution: non-linear this tone can be disabled by writing zero to this register for single tone generation. tone coeff register 1-dtmf or tone ringer address = 23h write/read verify power reset value 0000 0000 76543210 b7 b6 b5 b4 b3 b1 b2 b0 this register is used to program the high-group frequency of the dtmf program. the tone coefficient is calculated as follows: coef = 0.128 x frequency where: frequency is in hz (note: coef must be converted to an 8 bit binary integer) highest frequency possible: 1992.2 hz lowest frequency possible: 7.8 hz frequency resolution: 7.8 hz pre-twist: 0db this register is used to program the second frequency of the squarewave program. the tone coefficient is calculated similarly to tone coefficient register 1. tone coeff register 2-dtmf or tone ringer address = 24h write/read verify power reset value 0000 0000 76543210 b7 b6 b5 b4 b3 b1 b2 b0 the tone ringer will switch between squarewave frequencies at a warble frequency defined by this register. the relationship between the duration period of each tone and the 8 bit warble coefficient is as follows: tone duration (warble frequency) = 500 / coef where: frequency is in hz, and 0 coef <256 highest frequency possible: 500 hz lowest frequency possible: 2.0 hz tone ringer warble rate-tone ringer address = 26h write/read verify power reset value 0000 0000 76543210 b7 b6 b5 b4 b3 b1 b2 b0
mt9094 7-64 applications to maintain a fully differential topology in the transmit path the suggested connection scheme for the transmit microphones is shown in figure 7. however, it is possible to use a single-ended arrangement as shown in figure 8 for the transmit interface. in this case the dynamic range of the mt9094 is reduced by half. in both figures the output drivers are connected in a fully differential manner. the mt9094 is a member of the mitel family of digital terminal equipment components. there are two transmisssion devices which connect directly with the mt9094 to complete an application; the mt8930 (snic) and the mt8971/72 (dsic/dnic). an isdn 4-wire "te" function is implemented with the mt8930/mt9094 combination. a 2-wire digital phone for pabx, key-systems and other proprietary applications is implemented with the mt8971/72/ mt9094 combination. figures 9 and 10 show the 4-wire and 2-wire applications, respectively. 7 figure 7 - application circuit - fully differential audio input st-bus port to transmission device serial microport intel mcs-51 dsti dsto c4i f0i sclk data2 data1 cs wd 6 5 4 3 2 1 44 43 42 41 40 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 lcd mt9094 +5v 0.1 m f 0.1 m f vbias 2r t av = 1+ r r + - - + t 100k vbias 100k 0.1 m f +5v 330 w 511 w + 10 m f + electret microphone 511 w + - r 2r t av = 1+ r + - t 100k vbias 100k 0.1 m f +5v 330 w 511 w + 10 m f + electret microphone 511 w 0.1 m f +40 w nom. 32 w min. 150 w 75 w +5v 75 w .1 m f 1000pf 1000pf 1000pf caps are optional lcd note: single-ended configurations reduce dynamic range by a factor of two. 0.1 m f
mt9094 7-65 figure 8 - application circuit - single-ended audio input r + C vbias t +5v 1 m f 1k 330 w 10 m f electret microphone r + C vbias t +5v 1 m f 1k 330 w 10 m f electret microphone + + + + 0.1 m f 0.1 m f vbias +5v dsti dsto c4i f0i 6543214443424140 7 8 9 10 11 12 13 14 15 16 17 39 38 37 36 35 34 33 32 31 30 29 18 19 20 21 22 23 24 25 26 27 28 lcd st-bus port to transmission device serial microport intel mcs-51 sclk data2 data1 cs wd mt9094 75 w +5v .1 m f lcd 75 w 1000pf 1000pf 40 w nom. 32 w min. 150 w 1000pf caps are optional + note: single-ended configurations reduce dynamic range by a factor of two.
mt9094 7-66 figure 9 - ccitt isdn voice/data terminal equipment - te1 figure 10 - voice/data digital telephone set circuit aaaa aaaa aaaa aaaa aaaa a a a a a ccitt isdn reference point s interface 5 volts ltx vbias lrx mt8930 snic with hdlc controller dc to dc converter c4b f0b dsto dsti c4i f0i dsti dsto handset microphone speaker hspkr+ hspkr- m+ m- mic+ spkr+ spkr- mic- (ale) (rd )(wr ) irq irq irq as (ale) e (rd ) r/w (wr ) ad0-7 ad0-7 cs sclk data1 8051 intel mcs- 51 mt9094 dphone- ? aaaa aaaa aaaa aaaa aaaa a a a a a twisted pair to central pbx dc to dc converter 5 volts lin z t lout mt8972 dnic c4i f0i c4 f0 dsto dsti dsto dsti hspkr+ hspkr- m+ m- mic+ spkr+ spkr- mic- handset microphone speaker mt9094 dphone- ? cs sclk data1 intel mcs-51 10.24 mhz
mt9094 7-67 programming examples some examples of the programming steps required to set-up various telephony functions are given. note that these steps are from the power-up reset default definition. if some other state is currently true then some programming steps may be omitted while new ones may be required. standard full-duplex handset call description address data select b-channel of operation 15h bits 2 or 3 (as required) reset dsp 1eh 00h set rx gain (ie 0db with tx autonull) 1dh 70h (or as required) set tx gain (ie 0db) 20h 30h (or as required) start rx gain program 1eh 21h select transducers and turn on sidetone and filter/codec 0eh 99h set sidetone gain 0bh 04h (for 0db or as required) optional: set codec rx and tx gain select a-law versus m -law 0ah 0fh as required (0db default) bits 1-5 (as required) half-duplex handsfree operation description address data select b-channel of operation 15h bits 2 or 3 (as required) reset dsp 1eh 00h set rx gain (ie 12 db) 1dh 38h (or as required) set tx gain (ie 0db) 20h 30 h (or as required) start handsfree program 1eh 81h select transducers and filter/codec and turn off sidetone 0eh 1eh optional: set codec rx and tx gain select a-law versus m -law 0ah 0fh as required (0db default) bits 1-5 (as required) generate tone ringer description address data select b-channel of operation 15h bits 2 or 3 (as required) reset dsp 1eh 00h set rx gain (ie 0 db with tx autonull) 1dh 70h (or as required) set tx gain (ie 0db) 20h 30h (or as required) write tone coefficient 1 23h as required write tone coefficient 2 24h as required write warble tone rate coefficient 26h as required start tone ringer program 1eh 61h select speaker and filter/codec and turn off sidetone 0eh 82h control ringer cadence by toggling rxmute 1eh 61 (on) 69 (off) 61 (on) 69 (off) etc...
mt9094 7-68 generate dtmf tones description address data select b-channel of operation 15h bits 2 or 3 (as required) reset dsp 1eh 00h set rx dtmf gain (ie -20 dbm0) 1dh 22h (or as required) set tx audio gain (ie 0db) 20h 30h (or as required) set tx dtmf gain (ie -4dbm0) 21h 2eh (or as required) write tone coefficient 1 23h as required write tone coefficient 2 24h as required start dtmf program 1eh 41h select transducers and filter/codec (pufc) and turn off sidetone 0eh as required optional: set codec rx gain 0ah as required (0db default) send tones in only rx or tx by disabling rxmute or txmute appropriately 1eh as required new call tone description address data assume that a b-channel of operation has already been selected for the concurrent handset conversation. if this is not true select one. select b-channel of operation 15h bits 2 or 3 (as required) reset dsp 1eh 00h ********************************************************************************** set rx gain (ie 0 db with tx autonull) 1dh 70h (or as required) set tx gain (ie 0db) 20h 30h (or as required) note: these two steps a required for the concurrent conversation only and do not affect new call tone generation. see standard full-duplex handset call for required programming. ********************************************************************************** write tone coefficient 1 23h as required write tone coefficient 2 24h as required write warble rate coefficient 26h as required start new call tone ringer program 1eh 71h set new call tone gain 0bh nctg2-1 (as required) select speaker 0eh 02h 9bh (assuming a concurrent handset call) enable new call tone 0fh 01h (assuming all other bits are m -law control ringer cadence by toggling between gain control and tone ringer with gain control programs 1eh 71h (on) 31h (off) 71h (on) etc...
mt9094 7-69 * excluding pwrst which is a schmitt trigger input. note 1: power delivered to the load is in addition to the bias current requirements. note 2: i ddft is not additive to i ddc1 . absolute maximum ratings parameter symbol min max units 1 supply voltage v dd -v ss -0.3 7 v 2 voltage on any i/o pin v i /v o v ss -0.3 v dd +0.3 v 3 current on any i/o pin (transducers excluded) i i /i o 20 ma 4 storage temperature t s -65 +150 c 5 power dissipation (package) plastic p d 750 mw 6 static discharge esd 2.0 kv 7 latch-up current i lu 100 ma recommended operating conditions - voltages are with respect to v ss unless otherwise stated. characteristics sym min typ max units test conditions 1 supply voltage v dd 4.7555.25v 2 input voltage (high) * v ih 2.4 v dd v noise margin = 400mv 3 input voltage (low) * v il v ss 0.4 v noise margin = 400mv 4 operating temperature t a -40 +85 c 5 clock frequenecy (c4i )f clk 4092 4096 4100 khz power characteristics characteristics sym min typ max units test conditions 1 supply current (clock enabled, all functions off i ddc1 6ma 2 supply current by function filter/codec dsp handset driver (bias only, no signal) speaker driver (bias only, no signal) timing control, c-channel, st-bus, etc. total all functions enabled i ddf1 i ddf3 i ddf4 i ddf5 i ddf6 i ddft 1.5 1.5 1.5 1.5 1.0 7.0 14 ma ma ma ma ma ma see note 1. see note 1. see note 2.
mt9094 7-70 ? dc electrical characteristics are over recommended temperature and range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. 1 ttl compatible pins only. dc electrical characteristics ? (except lcd drive pins) - voltages are with respect to ground (v ss ) unless otherwise stated. characteristics sym min typ ? max units test conditions 1 input high voltage ttl inputs v in 2.0 v 2 input low voltage ttl inputs v il 0.8 v 3 vbias voltage output v bias v dd /2 v max. load = 100 m a 4 input leakage current 1 i iz 0.1 10 m av in = v dd to v ss 5 positive going threshold voltage (pwrst only) negative going threshold voltage (pwrst only) v t+ v t- 3.3 1.5 6 output high current ttl o/p i oh -10 -16 ma v oh = 2.4v dsto, wd , data1, data2 7 output low current ttl o/p i ol 510 mav ol = 0.4v dsto, wd , data1, data2 8 output voltage v ref (v dd /2)-1.5 vno load 9 output leakage current 1 i oz 0.01 10 m av out = v dd and v ss 10 output capacitance c o 15 pf 11 input capacitance c i 10 pf dc electrical characteristics ? - lcd drive pins - voltages are with respect to ground (v ss ) unless otherwise stated. characteristics sym min typ max units test conditions 1 output high voltage both segment and backplane v oh 4.8 --- --- volts io = 1ma, v dd = 5v 2 output low voltage both segment and backplane v ol --- --- 0.2 volts io = 1ma, v dd = 5v 3 segment output load --- --- 1200 pf 4 backplane output load --- --- 7200 pf 5 frequency 62 62.5 63 hz
mt9094 7-71 ? ac electrical characteristics are over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. * note: mica/u , refer to general control register, address 0fh. ac characteristics ? for a/d (transmit) path - 0dbm0 = 1.421v rms for m -law and 1.477v rms for a-law, at the codec. (v ref = 0.5 volts and v bias = 2.5 volts). all parameters pertain exclusively to the filter/codec except absolute half-channel gain and transmit idle channel noise. characteristics sym min typ ? max units test conditions 1 analog input equivalent to overload decision a li3.17 a li3.14 5.79 6.0 vp-p vp-p m -law a-law both at codec 2 absolute half-channel gain. transmit filter gain = 0db setting g ax1 g ax2 5.4 14.7 6.1 15.4 6.8 16.1 db db mica/u =0* mica/u =1* mic or m to pcm 1020hz all other transmit filter settings (1 to 7db) are in addition to 0db setting g ax1 g ax2 -0.15 -0.15 +0.15 +0.15 db db mica/u =0* mica/u =1* from nominal mic or m to pcm 1020hz 3 gain tracking vs. input level ccitt g.714 method 2 g tx -0.3 -0.6 -1.6 0.3 0.6 1.6 db db db 3 to -40 dbm0 -40 to -50 dbm0 -50 to -55 dbm0 4 signal to total distortion vs. input level ccitt g.714 method 2 d qx 35 29 24 db db db 0 to -30dbm0 -40 dbm0 -45 dbm0 5 transmit idle channel noise n cx n px 15 -72 17.5 -66 dbrnc0 dbrn0p m -law a-law 6 gain relative to gain at 1020hz <50 hz 60 hz 200 hz 300-3000 hz 3000-3400 hz 4000 hz >4600 hz g rx -0.25 -0.9 -25 -30 0.0 0.25 0.25 -12.5 -25 db db db db db db db 7 absolute delay d ax 360 m s at frequency of minimum delay 8 group delay relative to d ax d dx 750 380 130 750 m s m s m s m s 500-600 hz 600-1000 hz 1000-2600 hz 2600 - 2800 hz 9 power supply rejection f=1020 hz f=0.3 to 3 khz f=3 to 4 khz f=4 to 50 khz pssr pssr1 pssr2 pssr3 37 40 35 40 db db db db 100mv rms signal m -law pssr1-3 not production tested
mt9094 7-72 ? ac electrical characteristics are over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. * note: rxa/u , refer to general control register, address 0fh. ? ac electrical characteristics are over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. ac characteristics ? for d/a (receive) path - 0dbm0 = 1.421v rms for m -law and 1.477v rms for a-law, at the codec. (v ref = 0.5 volts and v bias = 2.5 volts). all parameters pertain exclusively to the filter/codec except absolute gain and receive idle channel noise. characteristics sym min typ ? max units test conditions 1 analog output at the codec full scale a lo3.17 a lo3.14 5.704 5.906 vp-p vp-p m -law a-law 2 absolute half-channel gain. receive filter gain = 0db setting g ar1 g ar2 g ar3 -0.6 -12.9 -10.3 0.2 -12.3 -9.7 0.95 -11.8 -9.1 db db db pcm to spkr pcm to hspkr, rxa/u =0* pcm to hspkr, rxa/u =1* 1020hz all other receive filter settings (-1 to -7db) are in addition to 0db setting g ar1 g ar2 g ar3 -0.15 -0.15 -0.15 +0.15 +0.15 +0.15 db db db pcm to spkr pcm to hspkr, rxa/u =0* pcm to hspkr, rxa/u =1* from nominal 1020hz 3 gain tracking vs. input level ccitt g.714 method 2 g tr -0.3 -0.6 -1.6 0.3 0.6 1.6 db db db 3 to -40 dbm0 -40 to -50 dbm0 -50 to -55 dbm0 4 signal to total distortion vs. input level ccitt g.714 method 2 g qr 35 29 24 db db db 0 to -30dbm0 -40 dbm0 -45 dbm0 5 receive idle channel noise n cr n pr 15.5 -75 dbrnc0 dbrn0p m -law a-law 6 gain relative to gain at 1020hz 200 hz 300-3000 hz 3000-3400 hz 4000 hz >4600 hz g rr -0.25 -0.90 0.25 0.25 0.25 -12.5 -25 db db db db db 7 absolute delay d ar 240 m s at frequency of min. delay 8 group delay relative to d ar d dr 750 380 130 750 m s m s m s m s 500-600 hz 600-1000 hz 1000-2600 hz 2600 - 2800 hz 9 crosstalk d/a to a/d a/d to d/a ct rt ct tr -74 -80 db db g.714.16 ac electrical characteristics ? for side-tone path characteristics sym min typ ? max units test conditions 1 absolute path gain gain adjust = 0db g as1 g as2 -17.2 -13.1 -16.7 -12.6 16.2 -12.1 db db sidea/u , mica/u , rxa/u all 0 sidea/u , mica/u , rxa/u all 1 m inputs to hspkr outputs 1000hz all other settings (-9.96 to +9.96db) g as g as -0.3 -0.3 +0.3 +0.3 db db sidea/u =0 sidea/u =1 from nominal relative measurements w.r.t. g as1 & g as2
mt9094 7-73 ? ac electrical characteristics are over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. ? electrical characteristics are over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. ? electrical characteristics are over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. ac electrical characteristics ? for new call tone characteristics sym typ ? units test conditions 1 new call tone output voltage (spkr+ to spkr-) v nct1 v nct2 v nct3 v nct4 6.0 2.390 0.950 0.380 vp-p vp-p vp-p vp-p nctg0=0, nctg1=0 nctg0=1, nctg1=0 nctg0=0, nctg1=1 nctg0=1, nctg1=1 load > 34 ohms across spkr electrical characteristics ? for analog outputs characteristics sym min typ ? max units test conditions 1 earpiece load impedance e zl 260 300 ohms across hspkr 2 allowable earpiece capacitive load e cl 300 pf each pin: hspkr+ hspkr- 3 earpiece harmonic distortion e d 0.5 % 300 ohms load across hspkr (tol-15%),rxa/u =1, v o 693v rms , rx gain=0db 4 speaker load impedance s zl 34 40 ohms across spkr 5 allowable speaker capacitive load s cl 300 pf each pin spkr+ spkr- 6 speaker harmonic distortion s d 0.5 % 40 ohms load across spkr (tol-15%), v o 6.2vp-p, rx gain=0db electrical characteristics ? for analog inputs characteristics sym min typ ? max units test conditions 1 differential input voltage without overloading codec v id 2.87 1.02 vp-p vp-p mica/u =0, a/u =0 mica/u =0, a/u =1 across mic or m inputs, tx filter gain = 0db setting 2 input impedance z i 50 k w mic+, mic-, m+ or m- to v ss .
mt9094 7-74 ? timing is over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. figure 11 -st-bus timing diagram ac electrical characteristics ? - st-bus timing (see figure 11) characteristics sym min typ ? max units test conditions 1c4i clock period t c4p 243 244 245 ns 2c4i clock high period t c4h 121 122 123 ns 3c4i clock low period t c4l 121 122 123 ns 4c4i clock transition time t t 20 50 ns 5f0i frame pulse setup time t f0is 50 ns 6f0i frame pulse hold time t f0ih 50 ns 7f0i frame pulse width low t f0iw 150 ns 8 dsto delay t dstod 100 125 ns c l =50 pf 9 dsti setup time t dstis 30 ns 10 dsti hold time t dstih 50 ns 1 bit cell t t t c4p t c4h t c4l t t t dstis t dstih t t t f0is t f0ih t t t f0iw t dstod c4i dsto dsti f0i 2.4v 0.4v 2.4v 0.4v 2.4v 0.4v 2.4v 0.4v
mt9094 7-75 ? timing is over recommended temperature range & recommended power supply voltages. ? typical figures are at 25 c and are for design aid only: not guaranteed and not subject to production testing. figure 12 - serial microport timing diagram ac electrical characteristics ? - microport timing (see figure 12) characteristics sym min typ ? max units test conditions 1 receive data setup a 10 ns 2 receive data hold b 10 ns 3 transmit data delay from clock falling edge c 80 ns 50 pf 4 high z to valid data from sclk falling edge d 80 ns 50 pf 5 valid data to high z from cs rising edge e 80 ns 50 pf 6 current transmit data hold from clock falling edge f0ns 7 chip select to sclk setup and hold times g0 ns 8 sclk clock period (3 mhz) h 333 ns data 1 receive sclk cs data 1 or data 2 transmit 012 7 h a b g g d f e c 01267
mt9094 7-76 notes:

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