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MX93032
APPLICATION NOTE [M1 VERSION]
1.0 GENERAL DESCRIPTIONS
* MX93032-M1 is an engine chip on which digital answering machine (DAM) with full duplex speaker phone is implemented. Besides, CAS tone detection capability and FSK decoder for Caller-ID service has also been implemented as well. * It provides speech processing / management modules, including speech compression/decompression, silence management, telephone line signal processing, Macronix MX29F16A FLASH management, line echo cancellation and acoustical echo cancellation, etc. * The MX93032-M1 has built-in DSP mode and MCU mode. In DSP mode, users do not need external microprocessors and can effectively reduce the overall system cost in DAM with full-duplex speakerphone system. In MCU mode, the MX93032-M1 is controlled by an MCU (micro controller) via a 8 bit parallel or serial interface.
2.0 FEATURES
* Switchable 4.8kbps/12.8kbps speech compression/ decompression algorithm for message recording/playback. * Minimum 15 or 40 minutes of recording time for one 16M bits MX29F16A FLASH based on 12.8K or 4.8K compression rate, respectively. * Low (4.8kbps) or High (12.8kbps) compression rate with silence management technique for voice prompt making. Voice prompt can be stored in Flash memory. * Support total 127 messages, maximum 7 OGMs and maximum 8 personal mailboxes. * DTMF generation and detection with near-end echo cancellation. * Provide single low crystal solution (FLL). * Built-in DSP mode and MCU mode. In DSP mode, user can mask the system code into the MX93032M1. In MCU mode, user can control the MX93032-M1 via MCU (micro controller). Provide standard sample program written by 8051 assemble for customer development. Full duplex speakerphone with both acoustical echo canceller (AEC) and line echo canceller (LEC) Built-in CAS signal detector for monitoring the presence CPE alerting signal in Calling Identity Delivery on Call Waiting. FSK decoder for Calling Identity Delivery on CID Type I and CID Type II Caller ID Type II can be turned on under speaker phone mode Auto detecting MCU mode and DSP mode
* * *
* * *
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3.0 BLOCK DIAGRAM DSP MODE
Display
Speaker Microphone MX93002
Keypad
MX93032-M1 Commands
DAA Telephone Line
FLASH
MCU MODE
Display
Speaker Microphone
MCU Commands
MX93032-M1 Commands
MX93002 DAA Telephone Line
Keypad
FLASH
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4.1 MCU MODE
4.0 PIN CONFIGURATIONS
ED11 ED12 ED13 VDD GND ED14 ED15 X1\VDD X2\GND A19 A18 A17 IPT8 FCE2\ FCE1\ RST\ EROM SDEN\ SDATA A16 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100
MX93032-M1
3
A15 OPT9 OPT8 OPT7 OPT6 OPT5 OPT4 OPT3 OPT2 OPT1 OPT0 X32I X32O PACK\ CDR1 PDN IPT5 IPT4 IPT3 P/S IPT1 IPT0 VDD GND SI/PHDB7 SO/PHDB6 PHDB5 PHDB4 PHDB3 PHDB2 80 79 78 77 76 75 74 73 72 71 70 69 68 67 66 65 64 63 62 61 60 59 58 57 56 55 54 53 52 51
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30
ED10 ED9 ED8 ED7 ED6 ED5 ED4 ED3 ED2 ED1 GND VDD ED0 HOLD\ CDX1 EDCE\ EPCE\ ERD\ EWR\ EAD0 EAD1 EAD2 EAD3 EAD4 EAD5 EAD6 FLLEN\ GND EAD7 EAD8
50 49 48 47 46 45 44 43 42 41 40 39 38 37 36 35 34 33 32 31 EAD9 EAD10 EAD11 EAD12 EAD13 EAD14 GND VDD EAD15 NMI\ INT1\/SCKL PHILO CDR0 PHRD\ CMCK CFS CDX0 PHWR\ PHDB0 PHDB1
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4.2 DSP MODE
ED11 ED12 ED13 VDD GND ED14 ED15 X1\VDD X2\GND A19 A18 A17 IPT8 FCE2\ FCE1\ RST\ EROM SDEN\ SDATA OPT11 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100
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OPT10 OPT9 OPT8 OPT7 OPT6 OPT5 OPT4 OPT3 OPT2 OPT1 OPT0 X32I X32O XF\ CDR1 IPT6 IPT5 IPT4 IPT3 IPT2 IPT1 IPT0 VDD GND BIO7 BIO6 BIO5 BIO4 BIO3 BIO2 80 79 78 77 76 75 74 73 72 71 70 69 68 67 66 65 64 63 62 61 60 59 58 57 56 55 54 53 52 51
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30
ED10 ED9 ED8 ED7 ED6 ED5 ED4 ED3 ED2 ED1 GND VDD ED0 HOLD\ CDX1 EDCE\ EPCE\ ERD\ EWR\ EAD0 EAD1 EAD2 EAD3 EAD4 EAD5 EAD6 FLLEN\ GND EAD7 EAD8
50 49 48 47 46 45 44 43 42 41 40 39 38 37 36 35 34 33 32 31 EAD9 EAD10 EAD11 EAD12 EAD13 EAD14 GND VDD FAD15 NMI\ INT1\ A16 CDR0 A15 CMCK CFS CDX0 OPT16PHDB0 PHDB1
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5.0 PIN DESCRIPTIONS
POWER/CLOCK/CONTROL PINS : SYMBOL PIN TYPE PIN NUMBER VDD GND X1/VDD X2/GND RST\ XF\ HOLD\ EROM NMI\ INT1\ X32O X32I FLLEN\ P/S PDN IS IS IS IS OA IS IS IS IS 23, 43, 69, 84 24, 44, 53, 70, 85 88 89 96 14 67 97 41 40 13 12 54 20 16 DESCRIPTION 5V power source Ground 32.256MHz crystal input/Connect to VDD in single low X'tal mode 32.256MHz crystal output/Connect to GND in single low X'tal mode Power-on reset External flag if UPMODX=1. This pin can be directly written by one DSP instruction. Default inactive (5V output) Hold DSP clock down and release bus Disable internal ROM, use external ROM only Non-maskable interrupt pin Interrupt pin 32.768KHz crystal output 32.768KHz crystal input 1 : Dual X'tal mode 0 : Single low X'tal mode 1:Parallel interface in MCU mode 0:Seriall interface in MCU mode 0:POWER DOWN
CODEC INTERFACE PINS : SYMBOL PIN TYPE PIN NUMBER CFS CMCK CDX0 CDR0 CDX1 CDR1 SDEN\ SDATA OA OA OA IS OA IS OB OB 35 36 34 38 66 15 98 99
DESCRIPTION Codec frame sync, 8KHz (9.6KHz). Output low in power down mode. Codec master clock, 1.536MHz. Output low in power down mode. Codec data transmit Codec data receive Codec data transmit Codec data receive Serial data enable(for MXIC codec family) Serial data transmit(for MXIC codec family)
IPT : Input port SYMBOL PIN TYPE IPT4-IPT5, IS IPT8 IPT0-IPT3 ISH
PIN NUMBER 18-17,93 22-19
DESCRIPTION Input port Input port with internal pull high resistor (R=30KW)
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OPT : Output port SYMBOL PIN TYPE PIN NUMBER OPT0-OPT9 OB OPT10-11 OB 2-11 1, 100 DESCRIPTION Output to pin, all output values are registered and may be read back when read by "IN" instruction. Output to pin, all output values are registered and may be read back when read by "IN" instruction. Only available for DSP mode MEMORY INTERFACE PINS : SYMBOL PIN TYPE PIN NUMBER EAD0-EAD15 OA/Z ED0-ED15 EDCE\ EPCE\ ERD\ EWR\ A15 A16 A17 A18 A19 61-55, 52-45, 42 IT/OA/ZR 68, 71-83, 86-87 OA/Z OA/Z OA/Z OA/Z 0A OA OA OA OA 65 64 63 62 37 (DSP MODE) 1 (MCU MODE) 39 (DSP MODE) 100 (MCU MODE) 92 91 90 95, 94 address bus for FLASH Address bus for FLASH Address bus for FLASH FLASH chip enable address bus for FLASH
DESCRIPTION External memory address bus. External memory data bus. With soft latch feed back current is 250
uA
External data chip enable External program chip enable SRAM/ROM/IO external read SRAM/ROM/IO external write address bus for FLASH
FCE1\, FCE2\ OB
uP Parallel Interface (MCU MODE) SYMBOL PIN TYPE PIN NUMBER
PHDB0-PHDB7 IS/OA/Z PHILO PHRD\ PHWR\ PACK\ IS/OA/Z IS/OA/Z IS/OA/Z OA 25-32 39 37 33 14
DESCRIPTION Parallel data bus Select High or low byte. 1: high byte , 0: low byte HOST read enable HOST write enable Response is ready
uP Serial Interface (MCU MODE) SYMBOL PIN TYPE PIN NUMBER SCLK SI SO IS OA/Z IS/Z 40 25 26
DESCRIPTION Synchronous clock HOST serial data in HOST Serial data out
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BIO: Bi-directional I/O (DSP MODE) SYMBOL PIN TYPE PIN NUMBER BIO7-BIO0 IT/OA 25-32 DESCRIPTION Input/output port when UPMODX = 1. Direction is controlled by BIO15-BIO8 (see BIOR). NOTE : IT :TTL level input IS :CMOS level Schmidt trigger input (hysteresis : 2V~3V) ISH :CMOS level Schmidt trigger input with internal pull high resistor (~30KW) OA :8mA drive level output OB :16mA drive level output Z :high impedance state ZR :high impedance state with soft latch
MULTIPLEX PINS: PIN NUMBER PIN NAME 25~32 39 37 33 14 100 1 BIO(7:0) A16 A15 OPT16 XF\ OPT11 OPT10
DSP MODE Input/output port Flash address Flash address Output port External flag Output port Output port
PIN NAME PHDB(7:0) PHILO PHRB\ PHWR\ PACK\ A16 A15
MCU MODE Host data bus High low data select Host read Host write Acknowledge to host Flash address Flash address
PIN NUMBER 88 89
PIN NAME X1 X2
FLLEN\=1 (Dual X'tal) 32.256MHz crystal input 32.256MHz crystal output
PIN NAME VDD GND
FLLEN\=0 (Single X'tal) Power VDD Power ground
NOTE FLLEN\: pin 54
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6.0 FUNCTIONAL DESCRIPTIONS
HOST INTERFACE (MCU MODE) The command and response of the MX93032-M1 are made up by a 16-bit code. The protocol that one command sent and one response received must strictly be obeyed by HOST ; otherwise, dead lock will occur. Meanwhile, both parallel interface and serial interface are provided by the MX93032-M1. 8-bit parallel : A 16-bit code can be accessed by HOST two times via setting PHILO pin. For writing operation, the HOST write the low byte of the command into the register of DSP, when PHILO is pulled to low, DSP does nothing but latches data. Following, PHILO is set to high and high byte of command is written into register, then DSP will perform the command issued by HOST and send out the response. DSP will pull PACK\ to be low to indicate that response is ready. HOST must read the low byte and then high byte sequentially. PACK\ will reset to high when the high byte is read by HOST. All of the description given above will be illustrated by the following timing chart.
PHILO
PHDB 0 - 7
PHWR\
PACKB\
PHRD\
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Synchronous Serial :
The protocol of synchronous serial interface will be interpreted by the timing chart given below .
16 PULSES SCLK
...
Tget SI
...
0 R15 R14 R1 R0
16 PULSES SCLK
...
Tput SO
...
START C15 0
C14
C1
C0
Note:
SCLK : synchronous clock sent by HOST, SI : HOST serial in, SO : HOST serial out, Tget and Tput should be larger than 200 uS
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Decision Rules between MCU mode and DSP Mode: 1. The contents located on 0xE000 and 0xE001 are 0x1234 and 0x5678, It will be considered as external DSP mode. This configuration is used for developing customer's control codes on MXIC's MX93032-M1 IC with standard code. 2. The contents located on 0x680E and 0x680F are 0x5678 and 0xA988, it will be considered as MCU mode. This is the only one configuration for MCU mode under developing and mass production. 3. Either the content located on 0x680E and not 0x5678 or that located on 0x680F are not 0xA988, it will be considered as DSP mode. The configuration is used for mass production for DSP mode adopted. 4. The decision priority depends on the items order. That is, the item labeled with smaller number has higher priority.
CONTROL CODE CONFIGURATION (DSP MODE) There are two configurations for customer development or production.
Configuration 1 : In this configuration, the BIOS+ALGORITHM and control codes are all running in the external high-speed EPROMs or SRAMs. The MX93032-M1 pin 97 (EROM) must be set to high.
External high-speed EPROMs or SRAMs 0000H BIOS+ALGORITHM c (address 0000H - 67F
26K words 6800H 6K words 7FFFH
control code (address 6800H - 7F
This configuration is mainly used in the development stage and the access time of high-speed EPROMs or SRAMs must be less than 30ns.
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Configuration 2 : In this configuration, the BIOS+ALGORITHM code is running in the MX93032-M1 internal ROM and the control code running in the external low-speed EPROMs. The MX93032-M1 pin 97 (EROM) must be set to low.
DSP internal ROM 0000H BIOS+ALGORITHM c (address 0000H - 67F
26K words
6800H 7FFFH
External low-speed EPROMs 0000H 1000H external voice promp data address 1000H external voice promp program address 800 control code (address E000H - FF
8000H
E000H FFFFH
max. 8K words
This configuration can be used at the development stage or for production. The control code must start with two words 1234H and 5678H.
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Configuration 3 : In this configuration, the BIOS+ALGORITHM and control codes are all running in the MX93032-M1 internal ROM. The MX93032-M1 pin 97 (EROM) must be set to low.
DSP internal ROM 0000H BIOS+ALGORITHM c (address 0000H - 67F
26K words
6800H 7FFFH 6 K words
control code (address 6800H - 7F
External low-speed EPROMs 0000H 1000H external voice promp data address 1000H external voice promp program address 800
8000H
FFFFH
This configuration is used for production. To set the voice prompt configuration, see the MX93032-M1 Mode 7.
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MX93032-M1 The MX93032-M1 provides DSP embedded functional modules, including speech compression/decompression, telephone line signal processing, ROM voice prompt, Macronix MX29F16A Flash management, time keeping and full duplex speaker phone. In MCU mode, the MX93032-M1 service is requested via a command call with 16 bits parameter sent by Micro controller, and answered 16 bits sent by DSP. In DSP mode, user's system control program can be masked-in the MX93032. The MX93032-M1 supports 16 modes of service and responses useful to operational information about timing, FLASH utilization, detected DTMF, etc. Hereafter, HOST means "external micro-controller" for MCU mode, and it means "system control program" for DSP mode. coming and outgoing messages that are labeled as Msg ID 1 to 127. Although ICM could range from Msg 1 to 127, Msg IDs 127 to 121 are the only places for outgoing messages (OGM) storage so that the deletion and recovery of old OGM are easier to maintain. The Msg IDs of incoming messages are labeled chronologically and rearranged automatically after recording or deleting. ICM with Msg ID 1 is the oldest recorded incoming message. Each message can be recorded by either 4.8K or 12.8K compression rate depending on system programmer's assignment. During recording, the MX93032-M1 also monitors the telephone line signal to detect the presence of DTMF, Call Progress Tone and Continuous Tone and responses to the control program. The HOST can stop recording, and delete the last n*400ms/200ms from the memory using a Record command with assigned Tail Cut factor. The MX93032-M1 can generate a desired tone during recording voice message, which allows the application such as two-way recording. During recording, Record Pause function is provided. The recording voice signal, received through the CODEC input, is transmitted back to the CODEC output by the MX93032-M1 service commands.
FLASH MANAGEMENT The MX93032-M1 uses MXIC's MX29F16A FLASH as storage device for message and/or voice prompt data. Up to two MX29F16A can be supported by the MX93032M1 service command. The MX29F16A is a 16M-bit FLASH memory organized as 1M word x 16. It includes 16 sectors (each sector is 64K words) and some of these sixteen sectors are bad. To manage the MX29F16A, the MX93032-M1 will detect and then give up bad sectors and only utilize the good sectors. For the purpose of storage efficiency, each good sector is further divided into 32 blocks (each block contains 2K words). Each block can store speech data of about 6.8 seconds for 4.8kbps compression algorithm and 2.55 seconds for 12.8kbps compression algorithm.
OUTGOING MESSAGE (OGM) RECORD The MX93032-M1 provides flexible recording ways for outgoing message. The HOST can select to delete the old OGM before a new one is recorded or to abort the new OGM record without losing the old OGM.
NEW/OLD MESSAGE For the control program to manipulate incoming messages more handily, the MX93032-M1 keeps an internal record of the ICM to identify the new/old status of messages. Before playback, the status of ICM is considered as "new". All incoming messages that have been played could change their status to "old" by issuing Real Del bit on Delete command. After Real Del is issued, all new/ old ICM Msg IDs will be rearranged. The functions, Play New or Delete All Old messages , are provided by the MX93032-M1 in the command set.
MESSAGE RECORDING AND STORAGE The MX93032-M1 provides high quality speech compression techniques for recording and playback functions. It can permit over 15 minutes or 40 minutes of speech storage in each 16M bit of MXIC's MX29F16A FLASH device based on 12.8K or 4.8K compression rate, respectively. The MX93032-M1 supports up to 127 variable length in-
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MESSAGE PLAYBACK Random access for playback of any new message or message with specified Msg ID is supported by the MX93032-M1. During playback, the MX93032-M1 also monitors the telephone line signals, and Play Pause function is provided. The MX93032-M1 is able to fast-forward the playback of a recorded message, and the playing speed will be up to 33 %. sages, the HOST should set the corresponding mailbox number first with Personal Mail command.
TIME KEEPING There is a timer clock in the MX93032-M1. The HOST can issue Set Time and Get Time commands to get time services. The clock also supports system to stamp time to the recorded incoming message.
MESSAGE DELETION DTMF DETECTION The MX93032-M1 provides two ways to delete recorded ICMs. One is the two-step method - the combination of Del Mark and Real Del on Play and Delete command, and the other is the immediate method - Delete command. The two-step method enables the control program to handle the deletion more flexibly, while the immediate method deletes a specified message directly. For remote control operation of the answering machine, the MX93032-M1 monitors the incoming telephone line signal to detect the presence of DTMF signal.
TONE GENERATION Up to double tones can be generated by the MX93032M1. The levels and frequencies are programmable by the HOST.
VOICE PROMPT For high-quality voice prompt playback, the MX93032M1 utilizes the built-in speech decoder and external EPROM/ROM or FLASH to store the vocabulary. The MX93032-M1 also supports up to a 32K-word external EPROM/ROM that allows minimum 1.8 minutes of voice prompt storage. If voice prompt data is stored in FLASH, space of maximum 60K words (minimum 3.4 minutes) is permitted. The external (upper part) voice prompt is also allowed. But if FLASH voice prompt is enabled, the external voice prompt will be disabled. The vocabulary supports up to 255 phrases. Given a phrase number in the Voice Prompt command from the HOST, the MX93032M1 outputs the corresponding utterances to the Codec interface. The starting address of external voice prompt data could be located at data address 1000H, or program address 8000H. During playback of the voice prompt, the MX93032-M1 monitors the telephone line signals.
CALL PROGRESS TONE DETECTION The MX93032-M1 supports the detection of call progress tone within the band of 300 - 640 Hz for the incoming telephone line signal. The tone detector is implemented with a bandpass filter. Indication of the presence of call progress tone is responded to the control program, and could be used to terminate recording.
NEAR END ECHO CANCELLATION The MX93032-M1 implements a near end echo canceller in software to enhance the detection of DTMF and Call Progress Tone during Playback and Voice Prompt mode.
FSK Decoder PERSONAL MAIL The MX93032-M1 supports up to 8 mailboxes that may be used for personal mail application. Each mailbox has its own ICM/MEMO Msg ID. The total Msg.ID is 127 for all of mailbox. Before recording, playing or deleting mesAll of the services, Calling Number Delivery (CND), Calling Name Delivery (CNAM) and Calling Identity Delivery on Call Waiting (CIDCW), exploit the Frequency Shift Keying (FSK) method to transmit the desired data from the SPCS to CPE. Depending on the specific service,
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data transmission may occur with the CPE in an onhook state or in an off-hook state. The specifications for data transmission are given on SR-TSV-002474 which are classified into three layers, physical layer, data linker layer and message assemble layer. The MX93032-M1 only takes the responsibility to decode physical layer. For Data link layer (Check Sum) and Message Assemble layer (the meaningful data), system should take care. The performance test associated with physical layer based on SR-3004 will be given on Appendix A. Note: CPE: Custom Premises Equipment SPCS: Stored Program Controlled Switching System on the both ends, but also prevent feedback instability (singing) from occurring. Moreover, the MX93032-M1 provides automatic gain control (AGC) for microphone channel which helps the outgoing voice to keep at a desired level especially for a moving talker, and speaker volume control which allows the user to adjust the speaker volume for his need. In addition, DTMF/Tone generations and call-progress tone detection are also provided.
CAS Detection For Calling Identity Delivery on Call Waiting, the MX93032M1 will monitor the presence of CPE alerting signal according to the specification given on Bellcore Special Report SR-TSV-002476.
VOX DETECTION (DIGITAL VOICE ACTIVATED RECORDING) The speech detection is implemented on the MX93032M1 to indicate whether the recording signal level is above the VOX threshold. If the control program sets the VOX Record bit on Initial command, the recording of the speech would start when the voice activity is detected. The HOST has to set VOX threshold by issuing command Mode 13. Hexadecimal numbers for VOX threshold are listed in Table 6.
SPEAKERPHONE The speaker system provided by the MX93032-M1 contains two adaptive filters which are called as acoustical echo canceller (AEC) and line echo canceller (LEC). The purpose of AEC is used to cancel the acoustical echo between microphone and speaker which is caused by the sound reflection within space. A LEC cancels the echo resulted by the 4 to 2 wire telephone line connection and impedance mismatches between customer premises and the central office. When speakerphone is activated, the algorithms associated with AEC and LEC not only reduce the echo heard
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DAA CONSIDERATION FOR SPEAKER PHONE It should be emphasized that DAA circuit plays an important role on improving the performance of speakerphone. This sub section gives some guides on how the DAA circuit may be "tuned". A diagram of DAA circuit must be given for simplifying the flowing interpretation.
SPK DRV CODEC1
MX93032-M1 CODEC2
LINE GAIN 4 - 2 WIRE COUPLING LINE DRV
MIC GAIN ACOUSTIC COUPLING
LINE COUPLING
Referring to figure given above, we have 4 analog amplifiers which are 1.Microphone Pre-Gain (MIC GAIN) 2.Line Out Gain (LINE DRV) 3.Line in Pre-Gain (LINE GAIN) 4.Speaker amplifier (SPK DRV) Obviously, the 4 amplifiers together with the Line Out volume, Speaker Volume, receive attenuator, transmit attenuator, two volume compensators, acoustical coupling function Hac(f) and line coupling function Hlin(f) compose a closed loop system. The AEC and LEC will provide additional attenuation to the loop gain one they converge. To have a better performance on speaker phone, we strongly recommend that analog gain of DAA must satisfy the following two criteria: 1.As the near end (Microphone side) speaks only, the level at Codec 2 should fulfill that, "codec 2 input must be less than Codec 2 output due to electrical echo". This meaning can be expressed as LINE_DRV * Hlin(f) * LINE_GAIN < 1 2.As the far end (Line Side) speaks only, the level at Codec 1 should fulfill that, "Codec 1 output must be greater than Codec 1 input resulting from acoustical echo". This meaning can be expressed as SPK_DRV * Hac(f) * MIC_GAIN < 1 By considering the criterion 1, it should pay more attention that minimum echo should be reached by improving the telephone line interface ( that is, reducing Hlin(f)) and not by reducing LINE_GAIN due to that the MX93032-M1 must obtain the high enough signal to process. For the same reason for criterion 2, the low echo should be achieved by good acoustical and mechanical decoupling rather than by decreasing MIC_GAIN. The ultimate target for hardware design is to create one system which not only satisfies two criteria given above, but also provides a desirable volume to both speaker and telephone.
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QUERY AND REPORT The MX93032-M1 is capable of reporting to the control program about the query of the current status , such as the number of recorded messages, the number of new messages, the number of old messages, the available recording time of FLASH, the time of recorded message, and the FLASH good/bad check.
POWER DOWN With the power of backup batteries, the MX93032-M1 is capable of keeping messages and data stored in the FLASH during power failure. In power down mode, the MX93032-M1 will run at a lower clock rate to reduce power consumption.
CRYSTAL CONNECTION The MX93032-M1 allows two ways of crystal connections : 1. Normal approach: use both high crystal (32.256 MHz) and low crystal (32768 Hz). To enable this mode, the FLLEN\ (54) pin must be connected to VDD. In this mode, the DSP high clock is generated from the high crystal directly, and the low crystal is mainly used in power down mode. 2. FLL (Frequency-Locked-Loop) approach: use only low crystal (32768 Hz). To enable this mode, the FLLEN\ (54) pin and X2 (89) pin must be connected to ground and the X1 (88) pin to VDD. In this mode, the low crystal is to generate all the DSP system clocks and the high crystal can be saved to reduce the overall system cost.
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7.0 OPERATIONAL DESCRIPTIONS
When the system is powered up, the RST\ pulse will make the MX93032-M1 to run programs from address 0. After check sum of internal ROM codes checked and some essential initialization of DSP are finished. In MCU mode, the MX93032-M1 is ready to poll the command issued by the HOST and indicate the HOST by setting PACK\ or SI\ to be low. The HOST should poll the indication sent by the MX93032-M1 and read out the pseudo response first. Following some necessary initialization given below must be set by HOST . 1. Set telephone data configuration 2. FLASH initialization depending on the FLASH good/bad check 3. Codec initialization if necessary 4. Setting Voice prompt configuration 5. Setting Silence threshold 6. Setting VOX level
8.0 MODES OF OPERATION
The MX93032-M1 supports 16 modes of operations, which are listed as follows : Command Mode 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 Mode Name Idle Mode Record Play Message Status DTMF Generator Line Monitor Delete Message Set Current Time/Voice Prompt/Silence Threshold Get Current Time Initial and Test Memory Get Record Message Information Voice Prompt Speaker Phone Personal Mail Telephone Data CAS Detection
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IDLE : This command is allowable to be issued within any other operation which is processing. As soon as the MX93032-M1 receives Idle command, the original operation will be stopped automatically.
RECORD When Record command is received from the HOST, the MX93032-M1 performs speech compression, stores the message into FLASH, and monitors telephone line. If the VOX Record bit had been set to 1 on Initial and Test Memory command, recording of the speech will start upon the voice activity detected, else the MX93032-M1 starts recording immediately after Record command is received. Clearing Marked Record (bit 8-10) to 0, if it is recording ICM. When recording OGM, the HOST should set Marked Record (bit 8-10) to desired number other than 0 as OGM ID. Setting bit 5 (Annc Del) to 1 on Initial and Test Memory command, the HOST can choose to delete the old OGM before a new one is recorded. If that bit is cleared to 0, the new OGM recording can be aborted without losing the old OGM by issuing Mark Fail (bit 11) on Record command. During recording, the HOST can pause the recording by setting Pause (bit 7) to 1 and resume recording by clearing that bit. Line monitoring will continue during pause. The HOST can stop recording by setting Stop (bit 6) to 1, and delete the last n*400ms/200ms from the memory by using a Record command with assigned Tail Cut factor (bit 0-5). While system memory is full, MX93032-M1 will stop message recording, show the status in response word (bit 7, Mfull). During Record mode, the HOST can instruct the MX93032-M1 to generate a tone by issuing the DTMF Generator command. Setting Stop bit on DTMF Generator command will stop tone generating. The MX93032-M1 will terminate Record mode when Stop bit on Record command is set to 1. Note: The MX93032-M1 supports up to 127 incoming and outgoing messages recording which are labeled as Msg ID 1-127. The Msg IDs of incoming message are labeled chronologically and rearranged automatically after recording. Although ICM could range from Msg ID 1 to 127, Msg IDs 127 to 121 are the only places for outgoing message storage. So, it is strongly suggested that the HOST should keep an upper bound of the number of ICM. (For example, upper bound of ICM = 127 - maximum OGM number that will be applied in user's specification.)
PLAY When Play command is received from the HOST, the MX93032-M1 performs message playback and line monitoring. The Msg ID (bit 0-6) is set by the HOST to specify which message to play. To playback new message (non-played one), set Play New (bit 10) to 1, then the MX93032-M1 will search to play the non-played message with specified Msg ID. During playing, the HOST can pause playback by setting Pause (bit 8) to 1 and resume playing from the same point by clearing that bit. Line monitoring will continue during pause. To fast playback a recorded message, set Fast Play (bit 11) to 1 and the playing speed will be up to 33%.
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To delete and stop the playing message, the HOST can set Del Mark (bit 7) on Play command. After being tagged a "Del Mark" the marked message is not really deleted until a Real Del (Delete Message mode bit 8) is set. The Real Del command not only deletes the messages that have been tagged a "Del Mark" but also changes the status of those ICMs that have been played to old. The ICMs that have not been played before are regarded as "new". After Real Del is issued, all new/old ICM Msg IDs will be rearranged. For the HOST, it is used to issue the Real Del after the user terminates the manner of playback. The response word contains the information of tones detected from the line, playing time spent (bit 8-15) and End of play flag (bit 6), which indicates the end of the current message. When the Play End bit is 1, the MX93032-M1 stops playback. The MX93032-M1 will terminate Play mode when Stop bit or Del Mark bit is set to 1 on Play command.
MESSAGE STATUS In this mode, there are 6 request codes for the HOST to get message information; 1. 2. 3. 4. 5. Get the number of personal recorded messages (request code = 0) Get the number of personal new messages (request code = 1) Get the number of personal old messages (request code = 2) Get the available recording time with unit of second (request code = 3) Get the number of total recorded messages (request code = 8)
The result will be reported in the response word.
DTMF GENERATOR In this mode, up to double tones can be generated by the MX93032-M1. The tone levels and frequencies are programmable and controlled by the HOST. The DTMF Generator command defines the output gains in bit 0-3 for the 1st tone, and bit 4-7 for the 2nd tone. Each tone can be programmed from 3 dB (gain code 0000) to -25 dB (gain code 1110) with 2 dB resolution. For DSP mode, the frequencies are defined in the variables BUF1 and BUF2. For MCU mode, three steps of commands should be issued, first, DTMF sent and signal Gain should be set, following the corresponding two frequencies should be specified. The frequency code and its frequency represented is according to the following relationship : frequency code = desired frequency (Hz) * 8.19 f(n) = 32767 * cos (2*pi*f_code*n/(8000*8.19)) where f_code is the input frequency code, n is sequence index and 8000 is the sampling rate For single tone, Codec outputs the signal s(n) : s(n) = gain * f(n) For dual tone, Codec outputs the signal s(n) : s(n) = 0.5 * (gain1*f1(n) + gain2*f2(n)) To generate a single tone, the gain code of the other tone should be set to 1111. The MX93032-M1 starts tone generation only after receiving the start command. The MX93032-M1 will terminate DTMF Generator when Stop bit is set to 1 on DTMF Generator command.
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LINE MONITOR In this mode the MX93032-M1 monitors the telephone line signal for the detection of DTMF, Call Progress Tone, VOX, and/or Continuous Tone. The MX93032-M1 keeps monitoring the line until Stop bit is set to 1. The MX93032-M1 returns to the HOST a response word that reflects the flag of Call Progress Tone, VOX, Continuous Tone and the index of DTMF signal that has been detected : INDEX CAS 0 0 0 0 0 0 0 0 0 DTMF 0000 0001 0010 0011 0100 0101 0110 0111 1000 DTMF CODE No Tone 1 2 3 4 5 6 7 8 INDEX CAS 0 0 0 0 0 0 0 1 1 DTMF CODE DTMF 1001 1010 1011 1100 1101 1110 1111 1111 0000 9 * 0 # A B C D CAS TONE
The MX93032-M1 will terminate Line Monitor mode when Stop bit is set to 1 on Line Monitor command.
DELETE MESSAGE This mode provides an immediate way to delete messages as follows; 1. To delete a specified message, set its Msg ID in bit 0-6 and clear bit 11, 10 and 7 to zero. 2. To delete a specified new ICM, set Del New Msg (bit 10) to 1, give the New Msg ID in bit 0-6, clear bit 11, 7 to zero. 3. To delete all old ICM, set Del All Old (bit 7) to 1 and clear bit 11, 10 to zero. (Bit 0-6 is ignored) 4. To erase the "Del Mark" from a specified ICM ("Del Mark" is a tag set on Play command for deletion), set Del Mark Clr (bit 11) to 1, give the Msg ID in bit 0-6 and clear bit 7 to zero.
SET CURRENT TIME/VOICE PROMPT/SILENCE THRESHOLD In this mode the HOST can set the current time to the timer clock of the MX93032-M1. The time information includes second, minute, hour and week. The timer in the MX93032-M1 will be used to stamp time and data to the recorded message. 1. To set second, put the data in bit 0-7 and use request code (bit 8-11) 0000. 2. To set minute, put the data in bit 0-7 and use request code (bit 8-11) 0001. 3. To set hour, put the data in bit 0-7 and use request code (bit 8-11) 0010. 4. To set week, put the data in bit 0-7 and use request code (bit 8-11) 0011. 5. To set voice prompt configuration, put the selection data in bit 0-1 and use request code (bit 8-11) 0110. 6. To set silence threshold level, put the data in bit 0-3 and use request code (bit 8-11) 0111. 7. To set output port OPT0-7, put data in bit0-7(Request Code=0100) 8. To set codec MX93002 registers in bit10-8 and put data in bit0-7(Request Code=1xxx) 9. To write link index and data in bit0-7(Request Code=0101), refer to mode10 to specify Msg. ID.
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GET CURRENT TIME In this mode the HOST can get the current time from the timer clock of the MX93032-M1. The time information includes second, minute, hour and week. 1. To get second, use request code (bit 8-11) 0000. 2. To get minute, use request code (bit 8-11) 0001. 3. To get hour, use request code (bit 8-11) 0010. 4. To get week, use request code (bit 8-11) 0011. The requested data will be reported in the response word.
INITIAL AND TEST MEMORY In this mode there are 9 functions for the control program to define; 1. FLASH Re-Initialization The HOST can re-initialize the FLASH by setting FLASH Re-Init (bit 0) to 1. After receiving this command, the MX93032-M1 will erase all the messages and data stored in the FLASH and then re-format the FLASH to establish the management structure. The response word reported to the control program contains the information of FLASH bit size (bit 0-3) , FLASH address size (bit 4-7), FLASH good/bad status (bit 8) and FLASH good rate (bit 9-15). After reinitialization, all messages and data in the FLASH are lost. 2. FLASH Initialization or Error Correction The HOST can perform FLASH first-time initialization or error correction by setting FLASH Init (bit 1) to 1. First-time initialization is to format a new FLASH and then establish the management structure, while error correction is to correct the problems that may result from power loss during operation. This function is normally used after first powerup of the system. The response word reported to the control program contains the information of FLASH bit size (bit 0-3) , FLASH address size (bit 4-7), FLASH good/bad status (bit 8) and FLASH good rate (bit 9-15). 3. FLASH Garbage Collection Due to the characteristics of MX29F16A FLASH and the management structure, when messages are deleted, the data regions occupied by them will not be released to be available immediately. To free out such regions, the MX93032M1 provides a two-step mechanism. First is to check the FLASH garbage collection recommendation status by Message Status command and, if garbage collection is recommended, then to issue bit 2 (Garbage Collect) on this command to perform the corresponding garbage collection actions. 4. Line On/Off The HOST should set Line On/Off (bit 3) to 1 if the system is logged in from telephone line, and clear that bit to 0 if in local keypad operation. Failing to set this bit properly will degrade the performance of echo cancellation. 5. VOX On/Off Record If the HOST turns on the VOX Record function (by setting bit 4 to 1) on this command, speech recording will start upon the voice activity detected, else recording will start immediately after Record command is issued.
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6. OGM Deletion By setting bit 5 (Annc Del) to 1 on this command, the HOST can choose to delete the old OGM before recording a new one. If that bit is cleared to 0, the old OGM is not deleted until the new OGM recording is completed. With this option, the new OGM recording can be aborted without losing the old OGM by issuing Mark Fail (bit 11) on Record command. 7. New/Old Select By setting New/Old Select (bit 6), the HOST can decide which condition a new message will be changed to an old one. There are two options. When this bit is set to 1, the new message that has been played (even not played end) is regarded as an old message. If this bit is cleared to 0, the new message that has been played end is regarded as an old one. 8. Tail Cut Unit By setting Tail Cut Unit (bit 7) to 1, the HOST can choose the tail cut unit length to be 200 ms. If this bit is cleared to 0, the unit length is 400 ms. 9. Old Mark Set By setting Old Mark Set (bit 8) to 1, the HOST can disable that new messages change to be old ones even though they have been played (end). If this bit is cleared to 0, new to old change will be enabled and the change criterion is according to the setting of New/Old Select bit (bit 6).
GET RECORD MESSAGE INFORMATION In this mode, there are 10 request codes for the control program to get the related information about a specified message. Given the Msg ID (bit 0-6) and New Msg flag (bit 7), the MX93032-M1 can provide message recording time, recording length, attribute and "Del Mark" status according to the Request Code (bit 8-11) as follows : 1. Get the recording time - second. (Request Code = 0000) 2. Get the recording time - minute. (Request Code = 0001) 3. Get the recording time - hour. (Request Code = 0010) 4. Get the recording time - week. (Request Code = 0011) 5. Get the recording length (sec.). (Request Code = 0100) 6. Get the timer status. (Request Code = 0101). The response word is 1 if the MX93032-M1 timer has been set before with Set Current Time command. Otherwise, the response word is 0. 7. Get the message ICM/MEMO attribute. (Request code = 0110). To show the specified message is of MEMO type (response word = 1) or ICM type (response word = 0). 8. Get the Del Mark status (Request Code = 0111). To check if the specified message has been tagged a Del Mark (response word = 1) or not (response word = 0). 9. Get the message new/old status. (Request Code = 1000). To check if the specified message has been tagged an Old Mark (response word = 1) or not (response word = 0). A new message will be tagged an Old Mark when it has been played/played end and changes to old messages as Real Del command is issued. 10.Get the OGM existing status (Request Code = 1111). To check if the specified OGM exists (response word = 1) or not (response word = 0). Msg ID could only range from 127 to 121. 11.To get link index data(Request Code=1001),refer to mode7 12.To set the Msg. ID for reading/writing Link-index-data.(Request Code=1010) All the results are reported in the response word.
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VOICE PROMPT In this mode the MX93032-M1 plays back a speech segment that the corresponding phrase number is given in command bit 0-7. Up to 255 speech segments, previously stored in voice prompt ROM or Flash memory, are provided by the MX93032-M1 and pronounced using the speech decompression algorithm. The HOST can pause the playback by setting Pause (bit 8) to 1 or resume from pause by clearing Pause bit to 0. Fast forward playback of voice prompt is also provided by setting Fast Play (bit 11) to 1. During Voice Prompt mode, the MX93032-M1 keeps monitoring the line signal and sends the results in response word to the control program. The MX93032-M1 will terminate Voice Prompt mode when Stop bit is set to 1 on Voice Prompt command.
SPEAKERPHONE Regarding to speakerphone related function is comprised of three types which are parameter setting, DTMF/Tone generation and free run. Two Codecs are necessary for speakerphone operation. One Codec labeled as Codec 1 is used for microphone and speaker, and the other labeled as Codec 2 is used for telephone line input and output. In this mode, there are seven functions whose detail will be given as follows: 1. Free run which performs both filtering operation and updating operation on LEC and AEC one time for each Codec interrupt triggered (Request Code = 0x0000) 2. Setting the gain for DTMF/Tone generation (Request Code = 0x0001) 3. Generating DTMF/Tone according to pre-defined table (Request Code = 0x0010) 4. Setting both the echo return-loss on speaker side (ERL_AEC) and echo return loss on line side (ERL_LEC) (Request Code = 0x0100) 5. The T/R and R/T ratio are used to decide which side is speaking by comparing the transmitting and receiving energy.(Request Code=0x0110) 6. Set the volume gains to control the speaker out and line out signal levels(Request Code=0x1000) 7. The loop attenuation is to control the total receive and transmit attenuators (Request Code = 0x1001) 8. setting the tone table which require 3 steps : Index set, Freq_Lo set and Freq_Hi set.
PERSONAL MAIL Five functions are provided in this command as follows : 1. Set the personal mailbox number In this mode the HOST can select one out of eight mailboxes (setting in bit 0-2) for recording messages, playing recorded messages and retrieving message time. Each mailbox manages its own messages and Msg IDs. Make sure to set the desired mailbox before issuing the message related commands such as Record, Play, Delete Message and Get Record Message Information. 2. Set the recording message attribute Before recording a message, the HOST can set the attribute of that message to be MEMO or ICM. Clear bit 3 to zero for recording a message as an ICM or set bit 3 to one for recording a message as a MEMO. 3. Set the digital loop back control The HOST can enable or disable the DSP digital loop back path in Record or Line Monitor operation. By setting Loop Back bit to 1, the loop back path is disabled and the signal received from the codec input will not be transmitted back
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to the codec output. When this bit is cleared to 0, loop back is enabled and the received signal will be transmitted back to the codec output. The default status is the latter. 4. Set the output level The HOST can set the level difference between the output (playback) speech and the corresponding input (recordedin) speech. By setting the Output Level bit to 1, the output speech level is equal to the input speech level, and clearing this bit to 0, the output speech level is about 5dB less than the input speech level. The default status is the latter. 5. Set the codec data format The HOST can select one of two Codec data formats by setting the Codec Format bit. When Codec Format is cleared to 0, 8-bit m-law data format is selected, and set to 1, 16-bit linear data format is enabled. At present, MXIC's Codec family provide both m-law and linear formats, and the latter is suggested because of the better speech quality. The default format is 8-bit m-law. 6. Set Voice Activity In this mode the HOST can set the VOX threshold in bit 0-11. The hexadecimal numbers for VOX level are listed in Table 6. 7. Set Telephone Data Configuration This command should be issued before Aflash initialization.
CAS Detection The MX93032-M1 provides 2 output modes for CAS detection, which are continuous response and one time response. For continuous response, the MX93032-M1 will respond CAS detected after 45 5ms since CAS occurs in CPE. Regarding to response one time, the information of CAS detected will be responded after both the CAS is disappeared on the line and the duration for CAS appearance between 45ms to 95 ms.
FSK Detection The output of FSK detector can be divided into two mode which are Raw data mode and Cooked data buffer. The buffer length for Raw data mode and Cooked Data mode are 8 bytes and 60 bytes long, respectively. For Raw data mode, the data ready bit will be set on one bytes data ,in which start bit and stop bit are not included, received correctly. If the host's speed can not catch up the FSK baudrate, buffer overflow will occur. Therefore, when the serial interface is adopted, we do not recommend to select Raw data mode to receive FSK data. Cooked data mode concerned, the data ready bit will be set after all of the data, in which checksum is included, received. During the operation for Cooked data mode, the second data byte following the mark data will be considered as the total data length, the other data will not be recognized by MX93032-M1.
LINKAGE BETWEEN CID DATA AND VOICE MESSAGE 1. Set the link index which can be also considered as CID data number (Request code = 0x7500 | link index) 2. Get the link index (Request code = 0xA900 | msg_no) 3. write the link index specified by 0x7500 into message (Request code = 0xAA00 | msg_no)
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TELEPHONE DATA In this mode the HOST can write/read telephone or user-assigned data to/from the FLASH with the sequence of telephone data command as follows : 1. Write Telephone Data The first command defines the telephone record ID (bit 0-7) and set bit 11, 10 to 0, 0. Following command specifies the one-byte data to be stored under that ID. While the current record ID is full, the MX93032-M1 BIOS will show the status in response word (bit 0, Full). The MX93032-M1 BIOS will terminate Telephone Data mode when Stop (bit 9) is set to 1 on Telephone Data command. 2. Read Telephone Data The first command defines the telephone record ID (bit 0-7) and set bit 11, 10 to 0, 1. The telephone data will be put on the response word. Following command puts the one-byte data on the response word to the control program. The response word contains the byte-data (bit 0-7) and Read End (bit 8). When the Read End bit is 1, the byte-data is invalid. The MX93032-M1 BIOS will terminate Telephone Data mode when Stop (bit 9) is set to 1 on Telephone Data command. If host want to write/read more than 2 Tel. Record IDs at the same time, Before writing/reading an new Tel. ID, host should set STOP bit to 1 for the current Tel. ID.
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9.0 COMMAND FORMAT AND PROTOCAL
Mode 0. IDLE MODE
0000000000000000
(15-12)
(11-1) 00000000000
0 STOP
CONF
0000
(15-12)
(11-1) 0000000000000000
0
RESP
Mode 1. RECORD MODE
(15-12) (11) MARK FAIL (10-8) MARKED RECORD (7) (6) (5-0) TAIL CUT
CONF
0001
PAUSE STOP
: 0 ~ 63 units. One unit length can be 400ms or 200ms (set by Mode 9). = 1 : stop recording. = 0 : continue recording. = 1 : pause recording. MARKED RECORD = 000 : record a general message. = 001 : record OGM1 and store it in message 127. = 010 : record OGM2 and store it in message 126. = 011 : record OGM3 and store it in message 125. = 100 : record OGM4 and store it in message 124. = 101 : record OGM5 and store it in message 123. = 110 : record OGM6 and store it in message 122. = 111 : record OGM7 and store it in message 121. MARK FAIL =1 : give up and stop recording of the new OGMx and recover the original OGMx. If Mode 9 ANNC DEL=1, this bit will be ignored. When MARKED RECORD = 000, do not set this bit.
TAIL CUT STOP PAUSE
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(15-9) (8) CAS (7) MFULL (6) VOX (5) TONE (4) CONT TONE (3-0) DTMF
RESP
RECORD TIME (sec)
DTMF CONT TONE TONE VOX MFULL
: the detected DTMF value during recording. = 1 : continuous tone is found. = 1 : call progress tone is found. = 1 : the input speech power level is under the VOX LEVEL (set by Mode 12). = 1 : ARAM is full and no further recording is possible. In this case, one more STOP command is needed to completely stop the recording action. RECORD TIME : recording time until now, 0 ~ 127 sec. CAS : CAS detection for CIDCW (detail see mode 15 ) If CAS tone and DTMF D are existed at the same system, the Host should Check if bit8(CAS) == 1 and bit3-0(DTMF) =0000, and then that's a real CAS Tone. Refer to the following table.
DTMF Reference Table: INDEX CAS 0 0 0 0 0 0 0 0 0 DTMF CODE DTMF 0000 0001 0010 0011 0100 0101 0110 0111 1000 No Tone 1 2 3 4 5 6 7 8 INDEX CAS 0 0 0 0 0 0 0 1 1 DTMF CODE DTMF 1001 1010 1011 1100 1101 1110 1111 1111 0000 9 * 0 # A B C D CAS TONE
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Mode 2. PLAY MODE
(15-12) (11) FAST PLAY (10) PLAY NEW (9) STOP (8) (7) (6-0) MSG ID
CONF
0010
PAUSE STOP
MSG ID DEL MARK PAUSE STOP PLAY NEW FAST PLAY
= 1 ~ 127 can be assigned for playing. = 1 : stop playing and mark the current message with a "Del Mark". This message will not be deleted until the Mode 6 REAL DEL command is done. = 0 : continue playing. = 1 : pause playing. = 1 : stop playing. = 0 : the MSG ID is related to total messages. = 1 : the MSG ID is related to new messages. = 0 : normal play speed. = 1 : play speed will be up to 33%.
(15-9)
(8) CAS
(7) 0
(6) PLAY END
(5) TONE
(4) 0
(3-0) DTMF
RESP
PLAY TIME (sec)
DTMF and TONE : see RECORD MODE. PLAY END = 1 : the current message is played end. PLAY TIME : playing time until now, 0 ~ 239 sec. CAS : CAS detection for CIDCW (detail see mode 15 and mode1)
Mode 3. MESSAGE STATUS MODE
(15-12) (11-4) 00000000000 (3-0)
REQUEST CODE
CONF
0011
(15-0)
RESP
DATA
REQUEST CODE: REQUEST CODE 0000 0001 0010 0011 0101 1000
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STATUS number of personal total messages number of personal new messages number of personal old messages available time garbage collection recommendation number of total messages
DATA 0 ~ 127 0 ~ 127 0 ~ 127 0 ~ 65535 0 or 1 0 ~ 127
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Mode 4. DTMF GENERATOR MODE For DSP mode:
(15-12) (11) START (10) STOP 9 TO 8 0 (7-4) GAIN2 (3-0) GAIN1
CONF
0100
GAIN1, GAIN2 : 0 ~ 14 denote 3dB ~ -25dB (2dB/one step), 15 denotes no related signal. TO =0 OUTPUT CHANNEL CODEC 0. =1 OUTPUT CHANNEL CODEC1. STOP = 1 : stop the DTMF generation. START = 1 : start the DTMF generation. When this bit is set, GAIN1 and GAIN2 must be also set and Frequency 1 is put in BUF1 (data address 7) and Frequency 2 in BUF2 (data address 8). 16-bit Frequency = target frequency * 8.192 Note : For single-tone signals (GAIN1 = 15 or GAIN2 = 15), the output level is according to GAIN1 or GAIN2. But for dualtone signals, the output level is equal to half of sum of two individual signals. Doing this can avoid saturation of output signals.
(15-0)
RESP
0000000000000000
For MCU mode:
(15-12) (11) START (10) STOP 9 TO 8 0 (7-4) GAIN2 (3-0) GAIN1
CONF
0100
GAIN1, GAIN2 : 0 ~ 14 denote 3dB ~ -25dB (2dB/one step), 15 denotes no related signal. TO =0 OUTPUT CHANNEL CODEC 0. =1 OUTPUT CHANNEL CODEC1. STOP = 1 : stop the DTMF generation. START = 1 : start the DTMF generation. When this bit is set, GAIN1 and GAIN2 must be also set and Frequency 1 and Frequency 2 is put in the following command. 16-bit Frequency = target frequency * 8.192 Note : For single-tone signals (GAIN1=15 or GAIN2=15), the output level is according to GAIN1 or GAIN2. But for dual-tone signals, the output level is equal to half of sum of two individual signals. Doing this can avoid saturation of output signals.
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(15-0)
RESP
0000000000000000
(15-0)
CONF
Frequency 1
(15-0)
RESP
0000000000000000
(15-0)
CONF
Frequency 2
(15-0)
RESP
0000000000000000
Mode 5. LINE MONITOR MODE
(15-12) (11-8) 0000 (7) PAUSE (6-1) 000000 (0) STOP
CONF
0101
STOP = 0 : start and continue the line monitor mode. = 1 : stop the line monitor mode.
(15-9) (8) CAS (7) 0 (6) VOX (5) TONE (4) CONT TONE (3-0) DTMF
RESP
0000000
DTMF, CONT TONE, TONE, PAUSE, VOX and CAS: see RECORD MODE.
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Mode 6. DELETE MESSAGE MODE
(15-12) (11)
DEL MARK CLR
(10)
DEL NEW MSG
(9) 0
(8) REAL DEL
(7)
DEL ALL OLD
(6-0) MSG ID
CONF
0110
MSG ID DEL ALL OLD REAL DEL
: 1 ~ 127. = 1 : delete all old messages. (in this case, the MSG ID field is ignored) = 1 : delete the messages that have "Del Mark" and change the new messages that have been played to be old messages. (in this case, the MSG ID filed is ignored) DEL NEW MSG = 0 : the MSG ID is related to total messages. = 1 : the MSG ID is related to new messages. DEL MARK CLR= 1 : clear the message "Del Mark" (set by Mode 2 DEL MARK=1).
(15-12) (11-0) 000000000000
RESP
0110
Mode 7. SET CURRENT TIME/VOICE PROMPT/SILENCE THRESHOLD MODE
(15-12) (11-8) REQUEST CODE (7-0) DATA
CONF
0110
(15-12)
(11-0) 000000000000
RESP
0111
REQUEST CODE: TIME SECOND MINUTE HOUR WEEK SET OUTPUT PORT SET LINK INDEX VOICE PROMPT CONFIG SILENCE THRESHOLD SEND MX93002 CMD 93993000(A) CMD
REQUEST CODE 0000 0001 0010 0011 0100 0101 0110 0111 1xxx
DATA 0 ~ 59 0 ~ 59 0 ~ 23 0~6 00 ~ FF H 00 ~ FF H 0~7 0~FH 00 ~ FF H
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Request Code = 0110 (set voice prompt configuration),
(15-12) (11-8) 0110 (7-3) 00000 (2) FVOP SELECT (1) EVOP SELECT (0) IVOP SELECT
CONF
0110
IVOP SELECT (internal voice prompt setting), = 0 : internal voice prompt not selected. = 1 : internal voice prompt selected and test. EVOP SELECT (external voice prompt setting), = 0 : external voice prompt not selected. = 1 : external voice prompt selected and test. FVOP SELECT (FLASH voice prompt setting), = 0 : FLASH voice prompt not selected. = 1 : FLASH voice prompt selected and test.
(15-5) (4) FVOP STATUS (3) EVOP UNIT (2) EVOP ADDR (1) (0)
RESP
00000000000
EVOP IVOP STATUS STATUS
IVOP STATUS (internal voice prompt test status), = 0 : test fail. = 1 : test ok. EVOP STATUS (external voice prompt test status), = 0 : test fail. = 1 : test ok. FVOP STATUS (FLASH voice prompt test status), = 0 : test fail. = 1 : test ok. (when EVOP STATUS = 1), EVOP ADDR = 0 : external voice prompt at data space 1000H. = 1 : external voice prompt at program space 8000H. EVOP UNIT = 0 : external voice prompt in byte unit. = 1 : external voice prompt in word unit. Request Code = 0111 (set silence threshold level - used by the silence management algorithm),
(15-12) (11-8) 0111 (7-4) 0000 (3-0) SILENCE LEVEL
CONF
0111
SILENCE LEVEL : 0 ~ 15, the larger the level is, the more speech will be recognized as silence. 0 denotes no silence management. Please refer to Table 7 for Silence Threshold Level.
(15-0)
RESP
0000000000000000
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Mode 8. GET CURRENT TIME MODE
(15-12) (11-8) REQUEST CODE (7-0) 00000000
CONF
1000
(15-0)
RESP
DATA
REQUEST CODE, 0000 ~ 0011 : see Mode 7.
Mode 9. INITIAL AND TEST MEMORY MODE
(15-12) (11-9) 000 (8)
OLD MARK SET
(7)
(6)
(5)
(4)
(3)
LINE ON/ OFF
(2)
(1)
(0)
CONF
1001
TAIL NEW/ ANNC VOX CUT OLD DEL ON/OFF UNIT SELECT RECORD
GARBAGE FLASH FLASH RE-INIT COLLECT INIT
FLASH RE-INIT = 1 : re-initialize the FLASH. The information of FLASH size and FLASH good rate will be responded in RESP. (Before this command is issued, Tel. Data Config., BIOS Mode 13 Sub-mode 3, should be set properly.) FLASH INIT = 1 : do FLASH first-time initialization or error correction after power loss. The information of FLASH size and FLASH good rate will be responded in RESP. (Before this command is issued, Tel. Data Config., BIOS Mode 13 Sub-mode 3, should be set properly.) GARBAGE COLLECT = 1 : perform FLASH garbage collection. After garbage collection, more deleted (but not released) data regions will be free out for further use. LINE ON/OFF = 0 : not in remote line operation. = 1 : in remote line operation. VOX ON/OFF RECORD = 0 : record the starting silence. = 1 : not record the starting silence which power level is under VOX LEVEL. ANNC DEL = 0 : delete the old OGMx after the new OGMx is recorded. = 1 : delete the old OGMx before the new OGMx is recorded. NEW/OLD SELECT (set the judgment point of message new/old attributes) = 0 : one message that had been played end is viewed as an old message. = 1 : one message that has been played (even not played end) is viewed as an old message. TAIL CUT UNIT (set the tail cut unit) = 0 : the tail cut unit is 400 ms. = 1 : the tail cut unit is 200 ms. OLD MARK SET= 0 : enable setting old mark. = 1 : disable setting old mark. New messages will not change to be old even they have been played (end). In this case, NEW/OLD SELECT will be ignored.
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(15-9) (8) FLASH G/B (7-4) FLASH ADDR SIZE (3-0) FLASH BIT SIZE
RESP
FLASH GOOD RATE
FLASH BIT SIZE : the detected FLASH data bit number in one address access. FLASH ADDR SIZE : the detected FLASH address space. FLASH G/B (a conclusion of FLASH initialization or error correction), = 0 : FLASH is bad. = 1 : FLASH is good. FLASH GOOD RATE : 0 ~ 100 %. It is significant when FLASH G/B = 1. CODE 0000 0101 0110 FLASH ADDR SIZE 0M 1M 2M CODE 0000 1111 FLASH BIT SIZE 0 16
Mode 10. GET RECORD MESSAGE INFORMATION MODE
(15-12) (11-8) REQUEST CODE (7) NEW MSG (6-0) MSG ID
CONF
1010
MSG ID NEW MSG
: 1 ~ 127. = 0 : the MSG ID is related to total messages. = 1 : the MSG ID is related to new messages. REQUEST CODE 0000 0001 0010 0011 0100 0101 0110 0111 1000 1001 1010 1111
(15-0)
INFORMATION SECOND MINUTE HOUR WEEK RECORD LENGTH TIMER STATUS ICM/MEMO DEL MARK STATUS MSG NEW/OLD STATUS GET LINK INDEX WRITE LINK INDEX (specified by 75XX) OGM EXISTING STATUS
DATA 0000 ~ 003BH 0000 ~ 003BH 0000 ~ 0017H 0000 ~ 0006H BIT 15 - BIT 0 0 = default timer ; 1 = timer has set 0 = ICM record ; 1 = MEMO record 0 = no del mark ; 1 = has del mark 0 = no old mark ; 1 = has old mark 00~FFH MSG ID=1~127 0 = OGM not exist ; 1 = OGM exist
RESP
DATA
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Mode 11. VOICE PROMPT MODE
(15-12) (11)
FAST PLAY
(10) 0
(9) STOP
(8)
PAUSE
(7-0) PHRASE NUMBER
CONF
1011
PHRASE NUMBER : 1 ~ 255. PAUSE, STOP and FAST PLAY : see PLAY MODE.
(15-9) (8) CAS (7) 0 (6) PLAY END (5) TONE (4) 0 (3-0) DTMF
RESP
PLAY TIME(sec)
DTMF, TONE, PLAY END CAS and Play TIME : see PLAY MODE.
Mode 12. SPEAKERPHONE MODE
(15-12) (11-8) REQUEST CODE (7-0) PARAMETERS
CONF
1100
REQUEST CODE 0000 0001 0010 0011 0100 0101 0110 0111 1000 1001 1111 Others * : Not available
FUNCTION Free Run Gain Setting For Tone Generate Tone Reserved Set ERL Reserved Set T/R & R/T ratios Set Tone Table * Set Speaker volume & line out gain Set Loop attenuation Read Internal Parameter (Reserved) Reserved
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Request Code = 0000 (Free Run)
(15-12) (11-8) 0000 (7) STOP (6) LINE MUTE (5) SPK MUTE (4-0) 00000
CONF
1100
STOP LINE MUTE SPK MUTE
= 0 : turn on speakerphone = 1 : turn off speaker phone = 1 : mute line out = 0 : normal line out = 1 : mute loudspeaker = 0 : normal speaker out
(15-9) (8) CAS (7-6) 00 (5) TONE (4-3) 00 (2-0) LOOP ATTENUATION
RESP
0000000
TONE = 1 : Call Progress Tone is detected Loop Attenuation : see Set Loop Attenuation CAS : CAS detection (see mode 15)
Request Code = 0001 (Set Gain for Tone)
(15-12)
(11-8) 0001
(7-4) GAIN-LO
(3-0) GAIN-HI
CONF
1100
GAIN-LO, GAIN-HI:0~14 denote 0dB~-14dB; 15 denotes mute
(15-3)
(2-0) LOOP ATTENUATION
RESP
0000000000000
GAIN TABLE: GAIN 0 dB 0dB GAIN 8 dB -8dB
1 -1dB 9 -9dB
2 -2dB A -10dB
3 -3dB B -11dB
4 -4dB C -12dB
5 -5dB D -13dB
6 -6dB E -14dB
7 -7dB F MUTE
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Request Code = 0010 (Send DTMF or Single Tone)
(15-12) (11-8) 0010 (7) START (6) 0 (5) D S (4-0) INDEX
CONF
1100
START D or S
= 1 : start to generate Tone according to the index assigned = 0 : stop generating Tone = 0 : generate Single Tone (This bit is valid under bit 4 set) = 1 : generate Dual Tone
(15-9) (8) CAS (7-3) 00000 (2-0) LOOP ATTENUATION
RESP
0000000
INDEX TABLE: Index Value 00000 00001 00010 00011 00100 00101 00110 00111 01000 01001 01010 01011 01100 01101 01110 01111
Tone DTMF_0 DTMF_1 DTMF_2 DTMF_3 DTMF_4 DTMF_5 DTMF_6 DTMF_7 DTMF_8 DTMF_9 DTMF_* DTMF_# DTMF_A DTMF_B DTMF_C DTMF_D
Index Value 10000 10001 10010 10011 10100 10101 10110 10111 11000 11001 11010 11011 11100 11101 11110 11111
Tone
Note : 1. The tone frequency whose index is specified form 10000 to 11111 can be stored into the ROM space with starting address specified by the label "TONE_TAB". At the same time, the label "TONE_TAB" should be declared as global. 2. The legal index values for dual tone are just from 10000 to 10111. For dual tone mode, it will catch the frequency specified by both the index and index+8.
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Request Code = 0100 (Set ERL)
(15-12) (11-8) 0100 (7-4) ERL_AEC (3-0) ERL_LEC
CONF
1100
ERL_ACE:echo return loss occurred on speaker side ERL_LEC:echo return loss occurred on line side
(15-3)
(2-0) LOOP ATTENUATION
RESP
0000000000000
ERL_AEC & ERL_LEC : Range from 0 to 30 with 2 dB/step 0000 0 dB 1000 16 dB 00001 2 dB 1001 18 dB 0010 4 dB 1010 20 dB 0011 6 dB 1011 22 dB 0100 8 dB 1100 24 dB 0101 10 dB 1101 26 dB 0110 12 dB 1110 28 dB 0111 14 dB 1111 30 dB
Note: The default value of ERL_LEC and ERL_AEC are 0 dB.
Request Code = 0110 (Set T/R & R/T ratio)
(15-12) (11-8) 0110 (7-6) 00 (5-3) R/T RATIO (2-0) T/R RATIO
CONF
1100
(15-3)
(2-0) LOOP ATTENUATION
RESP
0000000000000
T/R & R/T RATIO 000 001 3 dB 6 dB Note : 12 dB is default value
010 9 dB
011 12 dB
100 15 dB
101 Reserved
110 Reserved
111 Reserved
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Request Code = 0111 (Set Tone Table)
(15-12) (11-8) 0110 (7-4) 0000 (3-0) INDEX
CONF
1100
Index: Range is from 0 to 15 (see Request Code=0010) (15-0)
RESP
0000000000000000
(15-12)
(11-8) 0111
(7-0) FREQ_LO
CONF
1100
(15-0)
RESP
0000000000000000
(15-12)
(11-8) 0111
(7-0) FREQ_LO
CONF
1100
FREQ_HI and FREQ_LO will compose a 16-bit frequency, which is equal to "target Frequency *8.192". (15-0)
RESP
0000000000000000
Request Code = 1000 (Set SPK_GAIN & LINE _GAIN)
(15-12) (11-8) 1000 (7-4) LINE_GAIN (3-0) SPK_GAIN
CONF
1100
LINE_GAIN:line out volume gain SPK_GAIN:speaker out volume gain (15-3) (2-0) LOOP ATTENUATION
RESP
0000000000000
SPK_GAIN & LINE_GAIN : Range from -20 to 20 0000 -20dB 1000 6 dB 00001 -16 dB 1001 8 dB 0010 -12 dB 1010 10 dB 0011 -8 dB 1011 12 dB 0100 -4 dB 1100 14 dB 0101 0 dB 1101 16 dB 0110 2 dB 1110 18 dB 0111 4 dB 1111 20 dB
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Request Code = 1001 (Set Loop Attenuation)
(15-12) (11-8) 1001 (7-3) 00000 (2-0) LOOP ATTENUATION
CONF
1100
(15-3)
(2-0) LOOP ATTENUATION
RESP
0000000000000
Loop Attenuation (CONF) : Range from 0 to 28 with 4dB/Step 000 001 010 011 100 0.dB 4 dB 8 dB 12 dB 16 dB Full Duplex Note:8dB is default value
101 20 dB
110 24 dB
111 28 dB Smooth Half Duplex
Loop Attenuation (RESP) : 000 0.dB 001 4 dB 010 8 dB 011 12 dB 100 16 dB 101 20 dB 110 24 dB 111 >28 dB
Request Code = 1111 (Read Internal Parameters)
(15-12) (11-8) 1111 (7-0) Parameter Type
CONF
1100
(15-0)
RESP
Parameter
Parameter Type 00000000 00000001 00000010 00000011 00000100 00000101 00000110 00000111 00001000
Description AEC_SNR1 LEC_SNR1 AEC_STATE LEC_STATE AEC_AERLE LEC_AERLE ACC_ATT LEC_ATT TR_MODE
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Read CODEC Register (Reference 93002)
(15-12) (11-8) 1111 (7) 1 (6-0) Parameter Type
CONF
1100
(15-0)
RESP
Parameter
CODEC Register 0000000 0000001 0000010 0000011 0000100 0000101 0000110 0000111
Description Register 0 Register 1 Register 2 Register 3 Register 4 Register 5 Register 6 Register 7
Mode 13. PERSONAL MAIL MODE
(15-12) (11-8) 0000 (7-5) 000 (4) LOOP BACK (3) MEMO /ICM (2-0) PERSONAL MAIL
CONF
1101
PERSONAL MAIL : 0 ~ 7, total 8 personal mailboxes. The MSG IDs of PLAY MODE, DELETE MESSAGE MODE and GET RECORD MESSAGE INFORMATION MODE are all limited to the personal mailbox specified by this command. MEMO/ICM = 0 : set the attribute of the recorded message to be ICM. = 1 : set the attribute of the recorded message to be MEMO. LOOP BACK = 0 : enable loop back. = 1 : disable loop back. Then in RECORD and LINE MONITOR modes, the codec-in data will not be put on the codec-out path.
(15-0)
RESP
000000000000000
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Set Some Status,
(15-12) (11-8) 0001 (7) CAS SPK (6) 0 (5-3) (3) (2) (0)
CONF
1101
COMPRESS SILENCE CODEC OUTPUT RATE FORMAT LEVEL FORMAT
OUTPUT LEVEL (set the relation of the output speech level and the corresponding input speech level) = 0 : the output level is 5dB less than the input level (default setting). = 1 : the output level (playback) is equal to the input level (recording). CODEC FORMAT (set the Codec data format) = 0 : 8-bit m-law data format is selected. = 1 : 16-bit linear data format is selected. (Note: Macronix MX93000 Codec family provide this mode.) SILENCE RATE (set the silence storage rate, it is significant only when COMPRESS FORMAT = 000) = 0 : silence storage rate is 0.8kbps. (default setting) = 1 : silence storage rate is 4.8kbps. COMPRESS FORMAT (select the speech compression algorithm for message recording) = 000 : CELP 4.8kbps with silence compression. = 001 : CELP 12.8kbps with silence suppression. = 010 - 111 : reserved. CAS SPK = 0 : turn off detecting CAS within speaker phone = 1 : turn on detecting CAS within speaker phone
(15-0)
RESP
000000000000000
Set Telephone Data Configuration,
(15-12) (11-8) 0011 (7-5) 000 (4-2) TEL RECORD NUMBER (1) 0 (0) 1
CONF
1101
TEL RECORD NUMBER : set the number of total telephone data records, = 000 : 0 record (default setting). = 001 : 63 records. = 010 : 126 records. = 011 : 189 records. = 100 : 252 records.
(15-0)
RESP
000000000000000
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Voice Activity Mode,
(15-12) (11-8) 0111 (7-6) 00 (5-0) VOX Level
CONF
1101
VOX LEVEL : see Table 6.
(15-0)
RESP
000000000000000
Mode 14. TELEPHONE DATA MODE WRITE TELEPHONE DATA
(15-12) (11-10) 00 (9) STOP (8) 0 (7-0) TEL RECORD ID
CONF (C1)
1110
TEL RECORD ID:1~255, each telephone record has maximum 60 bytes. STOP =1:stop telephone data writing. (15-12) (11-10) 00 (9-1) 000000000 (0) FULL
RESP (R1)
1110
FULL=1: no free telephone record is available.
(15-12)
(11-10) 00
(9) STOP
(8) 0
(7-0) DATA
CONF (C2)
DATA STOP
1110
: the telephone data to write. =1:stop telephone data writing.
(15-12)
(11-10) 00
(9-1) 000000000
(0) FULL
RESP (R2)
FULL
1110
=1:current telephone record is full.
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READ TELEPHONE DATA
(15-12) (11-10) 01 (9) STOP (8) 0 (7-0) TEL RECORD ID
CONF
1110
TEL RECORD ID : 1 ~ 255. STOP = 1 : stop telephone data reading.
(15-12)
(11-10) 00
(9) 0
(8) READ END
(7-0) DATA
RESP
1110
DATA READ END
: the read out telephone data. = 1 : end of the current telephone record and the followed DATA is invalid.
Mode 15. CALLER _ID MODE
(15-12) (11-9) REQUEST CODE (8-0) PARAMETER
CONF
1111
REQUEST CODE 000 100 101 001 010
FUNCTION Demodulate FSK Set Response method of CAS detected Set CAS sensitivity ACK DTMF A ACK DTMF D
Request Code = 000 (Demodulate FSK To RAW DATA)
(15-12) (11-8) 000 (8) 0 (7-2) 000000 (1) TYPE (0) START\
CONF
1111
TYPE START\
= 0 : CID TYPE I 200 bits channel seizure & 10 bits mark signal = 1 : CID TYPE II 10 bits mark signal = 0 : Start / Continue to Demodulate = 1 : Stop Demodulating
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(15) (14) ERR (13) DR (7-0) FSK DATA BYTE
(12-8) 00000
RESP
CD
CD ERR DR
= 0 : Carrier is not detected (FSK signal do not occur on line) = 1 : Carrier is detected (FSK signal takes place on line) = 0 : Data buffer is not overflow = 1 : Data buffer is overflow = 0 : FSK data byte is not ready = 1 : FSK data byte is ready
Note : The length for RAW Data buffer is 8 bytes long.
Request Code = 000 (Demodulate FSK To COOKED DATA)
(15-12) (11-8) 000 (8) 1 (7-2) 000000 (1) TYPE (0) START\
CONF
1111
TYPE START\
= 0 : CID TYPE I 200 bits channel seizure & 10 bits mark signal = 1 : CID TYPE II 10 bits mark signal = 0 : Start / Continue to Demodulate = 1 : Stop Demodulating
(15) (14) ERR (13) DR (7-0) FSK DATA BYTE
(12-8) 00000
RESP
CD
CD ERR DR
= 0 : Carrier is not detected (FSK signal do not occur on line) = 1 : Carrier is detected (FSK signal takes place on line) = 0 : Data buffer is not overflow = 1 : Data buffer is overflow = 0 : FSK data byte is not ready = 1 : FSK data byte is ready 1) The second byte data received by FSK decoder will be considered as the total data length. 2) The first byte sent out are the total length +1 in which the checksum is included. 3) The length for cooked data buffer is 60 bytes long.
Note:
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Request Code = 100 (Set Response method of CAS detected)
(15-12) (11-9) 000 (8-1) 00000000 (0) CAS TYPE
CONF
1111
CAS TYPE
= 0 : Continuous Response. Response CAS behavior fidelity (Delay about 55ms) = 1 : Response One Time. Response CAS detected after CAS is off
CAS SIGNAL :
55 ms
Continuous Resp
15 ms
Resp One Time
(15-0)
RESP
000000000000000
Request Code = 101 (Set CAS Sensitivity)
(15-12) (11-9) 101 (8-3) 000000 (2-0) CAS_LVL
CONF
1111
(15-0)
RESP
000000000000000
Sensitivity
0 -32
1 -34
2 -36
3 -38
4 -40
5 -42
6 -44
7 -46
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Request Code = 001 or 010 (ACK DTMF A or ACK DTMF D)
(15-12)
(11-9) ACK
(8) TO
(7-4) GAIN2
(3-0) GAIN1
CONF
1111
GAIN1, GAIN2 : 0~14 denote 3dB ~ -25dB (2dB/one step), 15 denotes no related signal. TO = 0 OUTPUT CHANNEL CODEC 0. = 1 OUTPUT CHANNEL CODEC 1. ACK =001 ACK DTMF A. =010 ACK DTMF D.
(15-0)
RESP
000000000000000
TABLE 1. PERFORMANCE TEST FOR TONE DETECTION
(noiseless) Power (dB) p2p (volt) 280 Hz 300 Hz 400 Hz 500 Hz 600 Hz 640 Hz 670 Hz 480+620 Hz 350+440 Hz 0 5 x ok ok ok ok ok x ok ok -3 3.54 x ok ok ok ok ok x ok ok -6 2.5 x ok ok ok ok ok x ok ok -10 1.58 x ok ok ok ok ok x ok ok -20 .5 x ok ok ok ok ok x ok ok -30 .158 x ok ok ok ok ok x ok ok -33 .112 x ok ok ok ok ok x x x -36 .079 x x ok ok ok x x x x -39 .056 x x x x x x x x x
* 0 dB is defined as the Vrms of maximum codec input ** The maximum peak-to-peak voltage (P2P) of Codec input is assumed to be 5V
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SNR (dB) 300 Hz 400 Hz 500 Hz 640 Hz 480+620 Hz 350+440 Hz 40 50 / 40 40 / 40 40 / 40 45 / 40 40 / 40 45 / 40 25 detect time delay / off time delay (ms) 55 / 40 40 / 40 40 / 40 45 / 40 45 / 40 45 / 40 12 65 / 40 45 / 45 45 / 45 60 / 35 50 / 40 55 / 40
* 0 dB is defined as the Vrms of maximum codec input ** the detect/off time delay is defined as follows
INPUT
RESPONSE
detect time delay
off time delay
TABLE 2. RESOLUTION TEST FOR DUAL TONE (FREQUENCY DETECTABLE RANGE)
RANGE / POWER (300 + F2 Hz) (640 + F2 Hz)
-4 dB F2 > 320 Hz F2 < 620 Hz
-25 dB F2 > 320 Hz F2 < 620 Hz
TABLE 3. THD DISTORTION TEST FOR DUAL TONE (DISTORTION TOLERANCE)
INPUT / POWER (300 + 320 Hz) (640 + 620 Hz) -4 dB > 10 dB > 10 dB -25 dB > 10 dB > 10 dB
* 0 dB is defined as the Vrms of maximum codec input
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TABLE 4. CALL PROGRESS TONE DETECTOR CHARACTERISTICS
PARAMETER Detection level * Rejection level Frequency range Noise tolerance THD tolerance Rejection frequency range Minimum signal duration accept Maximum response time * 0 dB is defined as the Vrms of maximum codec input MIN -30 300 TYP UNIT dB -37 dB 640 Hz -12 dB -10 dB > 670 Hz MAX
< 280 Hz or 50 ms 50 ms
TABLE 5. DTMF CHARACTERISTICS
PARAMETER DTMF signal level for detection * DTMF twist (high/low tone) DTMF noise tolerance DTMF tone duration accept DTMF interdigit pause accept DTMF frequency deviation accept limit DTMF frequency deviation reject limit Tone generator level Tone generator frequency accuracy Tone generator level accuracy MIN -43 TYP MAX 0 UNIT dB dB dB ms ms % % dB % dB
9
-10 50 50
1.5 3
-25 +3
0.1 0.5
* 0 dB is defined as the Vrms of maximum codec input
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TABLE 6. VOX THRESHOLD
dB -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 -40 -41 -42 -43 -44 -45 -46 -47 -48 -49 -50 -51 -52 -53 -54 -55 -56 -57 -58 -59 -60 -61 -62 VOX_LEVEL 0 1 2 3 4 5 6 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 40 41 42 43 44 45 46 47 48 49 50 51 52 53
0 dB is defined as the Vrms of maximum Codec input.
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TABLE 7. SILENCE THRESHOLD LEVEL
SILENCE LEVEL 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 dB -58 -55 -52 -49 -46 -43 -40 -37 -34 -31 -28 -25 -22 -19 -16
0 dB : maximum input of sinusoidal wave
TABLE 8. ECHO CANCELLER PERFORMANCE *
Test Description Final Echo Return Loss Improvement (ERLI) (Singletalk mode) Maximum tail length Convergence Rate (Singletalk mode) 500 ms after initialization with cleared register and with near-end signal set to zero at initialization time Infinite Return Loss Convergence Return Echo Loss Improvement 500 ms after echo path is interrupted * test signal : band-limited white-noise (300-3400 Hz)
> 28 dB 12 ms > 28 dB
> 28 dB
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TABLE 9. MAXIMUM COMMAND RESPONSE TIME
COMMAND Record Mode : 1. start 2. normal 3. stop Play Mode : 1. start 2. normal 3. stop Message Status Mode : 1. get message number 2. get FLASH available time 3. get FLASH garbage collection recommendation status DTMF Generator Mode Line Monitor Mode Delete Message Mode : 1. delete one message 2. delete all old messages 3. real delete Set Current Time/Voice Prompt/Silence Threshold Mode Get Current Time Mode Initial and Test Memory Mode : 1. FLASH re-initialization 2. FLASH initialization/error correction 3. FLASH garbage collection 4. other misc. setting Get Record Message Information Mode Voice Prompt Mode : 1. start 2. normal 3. stop Voice Activity Mode Personal Mail Mode MAX RESP TIME (ms) 32 6 104 14 5 1 8 6 1 1 6 4300 5800 5500 1 1 2400 5800 150 1 10 6 5 1 1 1
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APPENDIX A : CID TEST REPORT BASED ON SR-3004
1. 1.1 Type 1 Testing Testing Items
A list of parameters to test based on SR-3004 are divided into two categories: Transmission tests and Message Format tests. Transmission tests vary electrical properties of signals to identify when and where CPE fails to detect. The Message Format tests vary the content and formatting of the data message. The following tests assumes that Transmission testing is independent of Message Format testing. The FSK mode provided on the MX93032-M1 is just responsible to decode the physical layer and data link layer. Therefore, Only transmission test associated with the MX93032-M1 will be given. For message format tests, it should be tested by "system manufacturers" depending on the their interesting region. 1.2 Transmission Testing Mark Frequency Space Frequency Twist Immunity Noise Immunity Baud Rate
1.2.1
Immunity to Frequency and Baud Rate Variation Testing Range 2100HZ ~ 2300HZ 1100HZ ~ 1300HZ 1150 baud ~ 1250 baud Desired Range 2178HZ ~ 2222HZ 1188HZ ~ 1212HZ 1188 baud ~ 1212 baud
Item SPACE MARK BAUD RATE
Testing Condition : Signal amplitude at -20dBm for per frequency SNR : 40 dB Signal : For the signal which meets the following each conditions will be send 100 times, the times CPE receive correctly will be recorded.
Baud Rate = 1188 baud 1100 1150 1188 1212 1250 1300 2100 95 100 100 100 100 100 2150 100 100 100 100 100 100 2178 100 100 100 100 100 100 2222 100 100 100 100 100 100 2250 100 100 100 100 100 100 2300 100 100 100 100 100 80
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Baud Rate = 1212 baud 2100 100 100 100 100 100 100 2150 100 100 100 100 100 100 2178 100 100 100 100 100 100 2222 100 100 100 100 100 100 2250 100 100 100 100 100 100 2300 100 100 100 100 100 100
1100 1150 1188 1212 1250 1300 1.2.2
Twist Immunity Range +/- 10 dB Fix Condition: Space Frequency : 2200 HZ Mark Frequency : 1200 HZ Baud Rate : 1200 Baud SNR : 40 dB Signal : For the signal which meets the following each conditions will be send 100 times, the times CPE receive correctly will be recorded. Positive Twist : (Mark > Space) Mark is fixed to -13 dBm and Space vary from -13 dBm to -48 dBm with step 5 dB -13 100 -18 100 -23 100 -28 0 -33 0 -38 0 -43 0 -48 0
Negative Twist : (Mark < Space) Space is Fixed to -13 dBm and Mark vary from -13dBm to -48 dBm with step 5 dB -13 100 1.3.4 -18 100 -23 100 -28 0 -33 0 -38 0 -43 0 -48 0
Noise Immunity: Fix Condition: Space Frequency : 2200 HZ Mark Frequency : 1200 HZ Baud Rate : 1200 Baud Signal Level : -20 dBm Signal : For the signal which meets the following each conditions will be send 100 times, the times CPE receive correctly will be recorded.
10 0
12 0
14 0
15 100
18 100
20 100
22 100
24 100
26 100
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28 100
30 100
32 100
34 100
36 100
38 100
40 100
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TYPE II TESTING The Type 2 CPE requires all of the same conformance test as Type 1 CPE in addition to "CAS detector performance" 2.1 CAS Detector Performance There are three types of tests to perform for CAS detector performance: Signal Recognition Tolerance ( without speech ) Talk off ( false detection due to speech ) Talk Down (missed detection due to interference from speech ) 2.1.1 Signal Recognition Tolerance Nominal 2130 +/- 2 Hz 2750 +/- 2 Hz -22 +/- 1 dBm 0 ~ 2 dB 80 +/- 1 ms Minimum 2119 Hz 2736 Hz -32 dBm 0 dB 75 ms Maximum 2141 Hz 2764 Hz -14 dBm 6 dB 85 ms
Parameter Lower Frequency Upper Frequency Power level Twist Duration
Various combinations of the above parameter values will be tested by sending 1000 CAS tones to the CPE at a rate not exceeding 2 per second. The various parameter combinations are broken into three groups described as follows. Each test combination requires 34 minutes to complete with 1000 burst at a 2 second interval. Group 1 consists of all the parameters at their nominal value. Group 2 sets each parameter ( one at a time) to its extreme values, while the remaining parameters remain at their nominal value. Group 3 sets all parameters to 90% of its extreme value. All of the possible parameter combinations are tested.
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Group 1 : Test No 1 Frequency ( Low , High) (2130,2750) Power Level (Low , High) (-22,-22) Duration 80 Correct Times 1000
Group 2 : Test No 1 2 3 4 5 6 7 8 9 10 11 12 Frequency ( Low , High) (2119,2750) (2141,2750) (2130,2736) (2130,2764) (2130,2750) (2130,2750) (2130,2750) (2130,275) (2130,2750) (2130,2750) (2130,2750) (2130,2750) Power Level (Low , High) (-22,-22) (-22,-22) (-22,-22) (-22,-22) (-14,-14) (-32,-32) (-14,-20) (-32,-26) (-20,-14) (-26,-32) (-22,-22) (-22,-22) Duration 80 80 80 80 80 80 80 80 80 80 75 85 Correct Times 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000
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Group 3 : Test No Frequency ( Low , High) 1 (2120,2737) 2 (2120,2737) 3 (2120,2737) 4 (2120,2737) 5 (2120,2737) 6 (2120,2737) 7 (2120,2737) 8 (2120,2737) 9 (2120,2737) 10 (2120,2737) 11 (2120,2737) 12 (2120,2737) 13 (2140,2737) 14 (2140,2737) 15 (2140,2737) 16 (2140,2737) 17 (2140,2737) 18 (2140,2737) 19 (2140,2737) 20 (2140,2737) 21 (2140,2737) 22 (2140,2737) 23 (2140,2737) 24 (2140,2737) 25 (2120,2763) 26 (2120,2763) 27 (2120,2763) 28 (2120,2763) 29 (2120,2763) 30 (2120,2763) 31 (2120,2763) 32 (2120,2763) 33 (2120,2763) 34 (2120,2763) 35 (2120,2763) 36 (2120,2763) 37 (2140,2763) 38 (2140,2763) 39 (2140,2763) 40 (2140,2763) 41 (2140,2763) 42 (2140,2763) 43 (2140,2763) 44 (2140,2763) 45 (2140,2763) 46 (2140,2763) 47 (2140,2763) 48 (2140,2763)
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Power Level (Low , High) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15) (-31,-31) (-31,-26) (-26,-31) (-15,-15) (-15,-20) (-20,-15)
58
Duration 76 76 76 76 76 76 84 84 84 84 84 84 76 76 76 76 76 76 84 84 84 84 84 84 76 76 76 76 76 76 84 84 84 84 84 84 76 76 76 76 76 76 84 84 84 84 84 84
Correct Times 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000 1000
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MX93032
CAS Level Reject Limit Test No 1 Frequency ( Low , High) (2120,2750) Power Level (Low , High) (-46,-46) Duration 80 Correct Times 0
2.1.1.2 Talk-down Test Row 1 2 3 4 5 6 Signal Level (dBm/Tone) -28 -28 -22 -26 -28 -22 Speech Level (dBm) -10 -13 -10 -16 -22 -19 SNR -18 -15 -12 -10 -6 -3 CAS Pulses Received 3608 4586 909 1073 1112 1124 CAS Pulses Sent 6000 6000 1125 1125 1125 1125 Derated Miss Rate (%) 19.933 11.7833 9.6 2.311 0.5778 0.0444
CAS LEVEL Speech -7 dBm -10 dBm -13 dBm -16 dBm -19 dBm -22 dBm -25 dBm -28 dBm -31 dBm -30 dBm 33.5166^ 25.3666^ 17.2166~ 11.0555~ 5.9556~ 1.4445~ 0.4~ 0* 0* -28 dBm 28.0833^ 19.933 11.7833 9.6 1.8778~ 0.5788 0.0444 0* 0* -26 dBm 22.6500^ 14.5000~ 10.3278~ 2.3111 1.0111~ 0.2222~ 0* 0* 0* -24 dBm 17.2166~ 11.0555~ 5.9556~ 1.4445~ 0.4~ 0* 0* 0* 0* -22 dBm 11.7833 9.60 1.8778~ 0.5778 0.0444 0* 0* 0* 0 -20 dBm 10.3278~ 2.3111 1.0111~ 0.2222~ 0* 0* 0* 0* 0* -18 dBm 5.9556~ 1.4445~ 0.4000~ 0* 0* 0* 0* 0* 0* -16 dBm 1.8778~ 0.5778 0.0444 0* 0* 0* 0* 0* 0*
in which the symbols "~" , "^" and "*" represent interpolated, extrapolated and clamped to minimum or maximum, respectively.
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Weighted average detection rate for all 72 level combinations given above -30 0.000129 0.000774 0.002973 0.006684 0.0088 0.006684 0.002973 0.000774 0.000129 0.03 -28 0.000172 0.001032 0.003964 0.008912 0.01184 0.008912 0.003964 0.001032 0.000172 0.04 -26 0.000344 0.002064 0.007928 0.017824 0.02368 0.017824 0.007928 0.002064 0.000344 0.08 -24 0.000731 0.004386 0.016847 0.037876 0.05032 0.037876 0.016847 0.004388 0.000731 0.17 -22 0.000731 0.004386 0.016847 0.037876 0.05032 0.037876 0.016847 0.004386 0.000731 0.17 -20 0.000731 0.004386 0.016847 0.037876 0.05032 0.037876 0.016847 0.004386 0.000731 0.17 -18 0.000731 0.004386 0.016847 0.037876 0.05032 0.037876 0.016847 0.004386 0.000731 0.17 -16 0.000731 0.004386 0.016847 0.037876 0.05032 0.037876 0.016847 0.004386 0.000731 0.17 Sum 0.0043 0.0258 0.0991 0.2228 0.296 0.2228 0.0991 0.0258 0.0043 1
-7 -10 -13 -16 -19 -22 -25 -28 -31 Sum
Talkdown Condition 1 -- the combination of average signal level (-22 dBm /tone) and average speech level (-19 dBm). At least 99.5% of the signals sent should be received correctly using derated missed rate. Talkdown Condition 2 -- the weight average of the worst one percent of the 72 combinations. At least 93% of the signal sent should be received correctly using the derated missed rate. Talkdown Condition 3 -- the weighted average of all of the 72 combinations. At least 99.5% of the signal should be received correctly using derated missed rate.\
Statistical Talkdown Result: Condition 1 (nominal) Condition 2 (worst 1 %) Condition 3 (global)
: 99.956 % ( Spec. >= 99.5% ) : 83.784 % ( Spec. >= 93 % ) : 99.005 % ( Spec. >= 99.5%)
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2.1.1.3 Talk-off Test Talkoff Test Raw Data: Row Near End Speech Level (dBm) -10 -16 -22 Hits No on Near End Speech 8 5 3 Testing Time Far End Speech Level (dBm) 96 96 96 Hits No on Far Testing Time End Speech
1 2 3
-16 -22
5 3
96 96
N E A R E N D S P E E C H
-7 dBm^ -10 dBm -13 dBm~ -16 dBm -19 dBm~ -22 dBm -25 dBm^ -28 dBm^ -31 dBm^
FAR END SPEECH -13 dBm^ -16 dBm 0.0729 0.0677 0.0729 0.0677 0.0651 0.0599 0.0573 0.0521 0.0521 0.0469 0.0469 0.0417 0.0417 0.0365 0.0365 0.0313 0.0313 0.0260 ^ extrapolated
-19 dBm~ 0.0625 0.0625 0.0547 0.0469 0.0417 0.0365 0.0313 0.0260 0.0208
-22 dBm 0.0573 0.0573 0.0495 0.0417 0.0365 0.0313 0.0260 0.0208 0.0156
-25 dBm^ 0.0521 0.0521 0.0443 0.0365 0.0313 0.0260 0.0208 0.0156 0.0104
-28 dBm^ 0.0469 0.0469 0.0391 0.0313 0.0260 0.0208 0.0156 0.0104 0.0052
-31 dBm^ 0.0417 0.0417 0.0339 0.0260 0.0208 0.0156 0.0104 0.0052 0
~ interpolated
P/N:PM0689
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Weighted average of all 63 combinations -7 -10 -13 -16 -19 -22 -25 -28 -31 Sum -13 3.07E-5 0.0001842 0.0007076 0.0015908 0.0021134 0.0015908 0.0007076 0.0001842 3.07E-5 0.00714 -16 9.013E-5 0.0005408 0.0020771 0.0046699 0.0062042 0.0046699 0.0020771 0.0005408 9.013E-5 0.02096 -19 0.0002528 0.001517 0.0058271 0.0131006 0.0174048 0.0131006 0.0058271 0.001517 0.0002528 0.0588 -22 0.0005126 0.0030754 0.0118127 0.0265578 0.0352832 0.0265578 0.0118127 0.0030754 0.0005126 0.1192 -25 0.0008063 0.0048375 0.0185813 0.041775 0.0555 0.041775 0.0185813 0.0048375 0.0008063 0.1875 -28 0.000915 0.0054902 0.0210885 0.0474118 0.0629888 0.0474118 0.0210885 0.0054902 0.000915 0.2128 -31 0.0016925 0.0101549 0.0390058 0.0876941 0.1165056 0.0876941 0.0390058 0.0101549 0.0016925 0.3935 Sum 0.0043 0.0258 0.0991 0.2228 0.296 0.2228 0.0991 0.0258 0.0043 1
Talkoff Condition 1 : the combination of average near-end level (-19 dBm) and the average far end level (-28 dBm). This talkoff rate should be less than 1 in 45 hours Talkoff Condition 2 : the weighted average of the worst one percent of the 63 combinations. The talkoff rate should be less than 1 in 10 hours. Talkoff Condition 3 : the weighted average of all of the 63 combinations. The talk off rate should be less than 1 in 35 hours.
Tested Result : Condition 1 (nominal ) : 1 hit per 38.4 hours Condition 2 (worst 1 %) : 1 hit per 16.79 hours Condition 3 (global) : 1 hit per 35.73 hours ( Spec. >= 45 hours) ( Spec. >= 10 hours) ( Spec. >= 35 hours)
3. Recommendation: All of the test results corresponding to talkdown given above are obtained by configuring our demo board to "Adjunct Model". That is, our demo board was connected directly to the Tip and Ring on which the far end speech and far end CAS has been mixed. Therefore, there is no way to tell speech signal from the CAS on that testing configuration adopted by us. For practical application, Near end speech instead of far end speech should occur on talkdown test. If the mixed signal captured from tip and ring can be subtracted from the source signal captured from the near end speech with proper gain. Then the SNR of resultant signal will be highly improved on adjusting the impedance match between CPE and SPCS adequately. According to our experience, if the quantity of the echo canceller can excess 6 dB, the performance of CAS detector embedded on 93032-M1 will meet the Bellcore specification for talkdown. The feature of echo canceller will be inherently possessed by phone mode, there will be additional circuit to implement echo canceller for adjunct model.
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APPENDIX B : OPERATION GUIDE FOR THE 93032-M1 DEMO BOARD 1. INTRODUCTION: The evaluation board (EVB) is provided by Macronix is to demonstrate both Digital speakerphone, Caller ID and DAM functions. There are four parts in EVB: 1. Main Module with label "A001-BD-0038A" 2. DSP Module with label "A001-BD-0031B" 3. DAA Module with label "MSP-PCB-010A" 4. MCU Module with label "A001-BD-0040A" The key pad is arranged into six kinds of definitions: 1. DAM mode keys: The power on default key function to demo DAM record/play/etc functions. 2. Speakerphone mode keys: Press "spk phone on" in DAM mode to enter speakerphone mode. User can dial DTMF, set loudspeaker / lineout volume or mute line-out signal. 3. Speaker test mode keys : Press "test mode" key in speakerphone mode for 1 second to enter this mode which allows user to set all the speakerphone BIOS parameters through key pad. 4. CPE Alerting Signal (CAS) test : Press "CAS Test" and last for 1 second in DAM mode to enter CAS test mode. During this mode , the number of CAS signal received from Tip and Ring will be counted and represented on LCD display. 5. Caller ID TypeII Test mode: Press "CAS Test" within 1 second in DAM mode to come into Caller ID Type II Test mode. After coming into this mode, 93032-M1 will monitor line and detect the CAS signal presented on line. The DTMF will be sent out as the CAS was found, following the FSK will be decoded and showed on LCD. 6. Encoding mode selection : Press "Enc. Select" in DAM mode to switch the encoding mode either 4.8 k or 12.8 k. 7. Flash Re-initialization : Press "Flash Re-Init." in DAM mode to perform FLAH Re-Initialization
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2. MATRIX KEY DEFINITION:
Key Pad under DAM mode (Power on default mode)
ANN1 1 ANN2 4 ANN3 7 CODE/FF 0
ON/OFF 2 PLAY 5 DEL 8 TIME/ STOP F 9 MEMO/ REV E 6 3
Flash Re-lnit. A CAS TesT B Enc. Select C
SPK.Phone ON/OFF
D
Key Pad in Speaker Phone mode :
1 1 4 4 7 7 Lout Vol + 0
2 2 5 5 8 8 0 F
3 3 6 6 9 9 Lout Vol E
Spk Vol + A Spk Vol B
Mute Lout/ Test Mode
C
SPK.Phone OFF
D
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Key Pad in Speaker Phone Test mode :
Current Loop Att 1
Set LINE_DRV 2 Set AD2_PGA 5 Set ATT1 8 Set SPK_DRV F
Set T/R Ratio 3 Set R/T Ratio 6 Set LEC_ERL 9 Set AEC_ERL E
Target Loop Att + A Target Loop Att B
Exit Test Mode
4 Set Pre_PGA 7 Set AD1_PGA 0
C
D
Key Pad in CALLER ID Test mode :
R/C DUMP CID
1
2
3
A
CALLER TYPE II/CAS TEST
4
5
6
B
7
Forward Caller ID Data
8 STOP F
9
Backward Caller ID Data
C
0
E
D
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Key Pad Definitions under DAM/Caller ID mode:
NO 1 2 3 4 5 6 7 8 9 0
FUNCTION OGM1 PLAY OGM1 RECORD AND PLAY Turn on DAM Raw/Cook Data Select Dump Caller ID Data OGM2 PLAY OGM2 RECORD AND PLAY PLAY ALL MESSAGE PLAY NEW MESSAGE
PRESS METHOD Press within 2 sec Press over 2 sec Press Press within 2 sec Press over 2 sec Press within 2 sec Press over 2 sec Press within 2 sec Press over 2 sec
Delete Current Message played Show And Announce Password Set Password Forward (for playing message and setting password) Flash Re-initialization CAS Test Enc. Selection Record Memo Rewind (for playing message and setting password) Announce Time Stamp Stop
Press Press within 2 sec Press over 2 sec Press Press Press Press Press over 2 sec Press Press Press
A B C E F
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Key Pad Definitions under Speaker Phone Mode : NO 1 2 3 4 5 6 7 8 9 0 A B C D E F FUNCTION GENERATE DTMF "1" GENERATE DTMF "2" GENERATE DTMF "3" GENERATE DTMF "4" GENERATE DTMF "5" GENERATE DTMF "6" GENERATE DTMF "7" GENERATE DTMF "8" GENERATE DTMF "9" Increase Line Out Volume Increase Speaker Volume Decrease Speaker Volume Mute Line Out Enter Test Mode Turn Off Speaker Phone Decrease Line Out Volume GENERATE DTMF "0" PRESS METHOD Press Press Press Press Press Press Press Press Press Press Press Press Press within 2 sec Press over 2 sec Press Press Press
Key Pad Definitions under Speaker Phone Mode (Test Mode) : NO 1 2 3 4 5 6 7 8 9 0 A B C D E F FUNCTION Show Current Loop Attenuation SET LINE_DRV (MX93002 CODEC) SET T/R RATIO SET AD2_PGA (MX93002 CODEC) SET R/T RATIO SET PRG_PGA (MX93002 CODEC) SET_ATT1(MX93002 CODEC) SEL LEC_ERL SET AD1_PGA TARGET LOOP ATTENUATION UP TARGET LOOP ATTENUATION DOWN EXIT TEST MODE SET AEC_ERL SET SPK_DRV(MX93002 CODEC) PRESS METHOD Press Press Press Press Press Press Press Press Press Press Press Press Press Press
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Speaker Phone related BIOS parameters ERL_AEC & ERL_LEC 0000 0 dB 1000 16 dB 00001 2 dB 1001 18 dB : Range from 0 to 30 with 2 dB/step 0010 4 dB 1010 20 dB 0011 6 dB 1011 22 dB 0100 8 dB 1100 24 dB 0101 10 dB 1101 26 dB 0110 12 dB 1110 28 dB 0111 14 dB 1111 30 dB
Note: The default value of ERL_LEC and ERL_AEC are 0 dB.
T/R & R/T Ratio
000 3 dB
001 6 dB
010 9 dB
011 12 dB
100 15 dB
101 Reserved
110 Reserved
111 Reserved
Note : 12 dB is default value
SPK_GAIN & LINE_GAIN : Range from -20 to 20
0000 -20dB
00001 -16 dB
0010 -12 dB
0011 -8 dB
0100 -4 dB
0101 0 dB
0110 2 dB
0111 4 dB
1000 6 dB
1001 8 dB
1010 10 dB
1011 12 dB
1100 14 dB
1101 16 dB
1110 18 dB
1111 20 dB
Loop Attenuation (CONF)
: Range from 0 to 28 with 4dB/Step
000 0.dB
001 4 dB
010 8 dB
011 12 dB
100 16 dB
101 20 dB
110 24 dB
111 28 dB
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MACRONIX INTERNATIONAL CO., LTD.
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TEL:+886-3-578-8888 FAX:+886-3-578-8887
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http : //www.macronix.com
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