mt9092apr1 Zarlink Semiconductor, mt9092apr1 Datasheet - Page 9

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mt9092apr1

Manufacturer Part Number
mt9092apr1
Description
Fully Featured Digital Telephone Circuit With Embedded Hdlc Transceiver And Dsp For Tone Generations And Hands Free Operation
Manufacturer
Zarlink Semiconductor
Datasheet

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MT9092
Data Sheet
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 1.5 dB 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
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