Datasheet MX93032-M1 Datasheet (MXIC)

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MX93032
1.0 GENERAL DESCRIPTIONS
• MX93032-M1 is an engine chip on which digital an­swering machine (DAM) with full duple x 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. V oice 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 W aiting.
• FSK decoder for Calling Identity Delivery on CID T ype I and CID Type II
• Caller ID Type II can be turned on under speaker phone mode
• Auto detecting MCU mode and DSP mode
APPLICATION NOTE [M1 VERSION]
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MX93032
3.0 BLOCK DIAGRAM
DSP MODE
MCU MODE
Dis play
MX93032 -M1
Commands
FLASH
Keyp ad
MX93002
DAA
Spea ker
Microphone
Telephone Line
Display
MX93032-M1
Commands
FLASH
MCU
Commands
Keypad
MX93002
DA A
Speaker
Microphone
Telephone L ine
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MX93032
4.0 PIN CONFIGURATIONS
4.1 MCU MODE
EAD9 EAD10 EAD11 EAD12 EAD13 EAD14 GND VDD EAD15 NMI\ INT1\/SCKL PHILO CDR0 PHRD\ CMCK CFS CDX0 PHWR\ PHDB0 PHDB1
ED11 ED12 ED13
VDD
GND ED14 ED15
X1\VDD
X2\GND
A19 A18 A17
IPT8 FCE2\ FCE1\
RST\ EROM SDEN\ SDATA
A16
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
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
1234567891011121314151617181920212223242526272829
30
50 49 48 47 46 45 44 43 42 41 40 39 38 37 36 35 34 33 32 31
8079787776757473727170696867666564636261605958575655545352
51
81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100
MX93032-M1
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MX93032
4.2 DSP MODE
EAD9 EAD10 EAD11 EAD12 EAD13 EAD14 GND VDD FAD15 NMI\ INT1\ A16 CDR0 A15 CMCK CFS CDX0 OPT16PHDB0 PHDB1
ED11 ED12 ED13
VDD
GND ED14 ED15
X1\VDD X2\GND
A19 A18 A17
IPT8 FCE2\ FCE1\
RST\ EROM SDEN\ SDATA
OPT11
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
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
1234567891011121314151617181920212223242526272829
30
50 49 48 47 46 45 44 43 42 41 40 39 38 37 36 35 34 33 32 31
8079787776757473727170696867666564636261605958575655545352
51
81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100
MX93032-M1
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5.0 PIN DESCRIPTIONS
POWER/CLOCK/CONTR OL PINS :
SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
VD D 23, 43, 69, 84 5V power source GND 24, 44, 53, 70, 85 Ground X1/VDD 88 32.256MHz crystal input/Connect to VDD in single low X'tal mode X2/GND 8 9 32.256MHz crystal output/Connect to GND in single low X'tal mode RST\ IS 96 Po wer-on reset XF\ OA 14 External flag if UPMODX=1. This pin can be directly written by one
DSP instruction. Default inactive (5V output) HOLD\ IS 67 Hold DSP clock down and release bus EROM IS 97 Disable internal ROM, use external ROM only NMI\ IS 41 Non-maskable interrupt pin INT1\ IS 40 Interrupt pin X32O 13 32.768KHz crystal output X32I 12 32.768KHz crystal input FLLEN\ IS 54 1 : Dual X'tal mode
0 : Single low X'tal mode P/S IS 20 1:Parallel interf ace in MCU mode
0:Seriall interface in MCU mode PDN IS 1 6 0:POWER DOWN
CODEC INTERF ACE PINS :
SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
CFS OA 3 5 Codec frame sync, 8KHz (9.6KHz). Output low in power down mode. CMCK OA 36 Codec master clock, 1.536MHz. Output low in power down mode. CDX0 OA 3 4 Codec data transmit CDR0 IS 38 Codec data receive CDX1 OA 66 Codec data transmit CDR1 IS 15 Codec data receive SDEN\ OB 9 8 Serial data enable(for MXIC codec family) SD ATA OB 99 Serial data transmit(for MXIC codec family)
IPT : Input port
SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
IPT4-IPT5, IS 18-17,93 Input port IPT8 IPT0-IPT3 ISH 22-19 Input port with internal pull high resistor (R=30KW)
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OPT : Output port
SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
OPT0-OPT9 OB 2-11 Output to pin, all output values are registered and may be read back
when read by "IN" instruction.
OPT10-11 OB 1, 100 Output to pin, all output values are registered and may be read back
when read by "IN" instruction. Only available for DSP mode
MEMORY INTERF A CE PINS :
SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
EAD0-EAD15 OA/Z 61-55, 52-45, 42 External memory address bus. ED0-ED15 IT/OA/ZR 68, 71-83, 86-87 External memory data bus. With soft latch feed back current is 250
u
A EDCE\ OA/Z 6 5 External data chip enable EPCE\ OA/Z 6 4 External program chip enable ERD\ OA/Z 63 SRAM/ROM/IO external read EWR\ OA/Z 62 SRAM/ROM/IO external write A15 0A 37 (DSP MODE) address bus for FLASH
1 (MCU MODE)
A16 OA 39 (DSP MODE) address bus for FLASH
100 (MCU MODE) A17 OA 9 2 address bus for FLASH A18 OA 9 1 Address bus for FLASH A19 OA 9 0 Address bus for FLASH FCE1\, FCE2\ OB 95, 94 FLASH chip enable
u
P Parallel Interface (MCU MODE) SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
PHDB0-PHDB7 IS/OA/Z 25-32 Parallel data bus PHILO IS/OA/Z 39 Select High or low byte. 1: high byte , 0: low byte PHRD\ IS/OA/Z 37 HOST read enable PHWR\ IS/OA/Z 33 HOST write enable P A CK\ OA 14 Response is ready
uP Serial Interface (MCU MODE)
SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
SCLK IS 40 Synchronous clock SI OA/Z 25 HOST serial data in SO IS/Z 2 6 HOST Serial data out
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BIO: Bi-directional I/O (DSP MODE)
SYMBOL PIN TYPE PIN NUMBER DESCRIPTION
BIO7-BIO0 IT/OA 25-32 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
MUL TIPLEX PINS:
PIN NUMBER PIN NAME DSP MODE PIN NAME MCU MODE
25~32 BIO(7:0) Input/output port PHDB(7:0) Host data bus 39 A16 Flash address PHILO High low data select 37 A15 Flash address PHRB\ Host read 33 OPT16 Output port PHWR\ Host write 14 XF\ External flag PACK\ Acknowledge to host 100 OPT11 Output port A16 Flash address 1 OPT10 Output port A15 Flash address
PIN NUMBER PIN NAME FLLEN\=1 (Dual X'tal) PIN NAME FLLEN\=0 (Single X'tal)
88 X1 32.256MHz crystal input VDD P o wer V DD 89 X2 32.256MHz crystal output G N D Po wer ground
NOTE FLLEN\: pin 54
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6.0 FUNCTIONAL DESCRIPTIONS
HOST INTERF ACE (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 b y 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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MX93032
Synchronous Serial :
The protocol of synchronous serial interface will be interpreted by the timing chart given below .
16 PULSES
SCLK
SI
Tget
16 PULSES
SCLK
SO
Tput
0 R15 R14 R1 R0
C15
C14
C1 C0START
0
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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MX93032
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 consid-
ered as DSP mode. The configuration is used for mass production f or 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 CONFIGURA TION (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) m ust be set to high.
26K words
6K words
0000H
6800H
7FFFH
External high-speed EPROMs or SRAMs
BIOS+ALGORITHM
c
(address 0000H - 67
F
cont rol co de (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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MX93032
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 .
26K words
0000H
6800H
7FFFH
DSP in t e rnal RO M
BIOS+ALGORITHM
c
(address 0000H - 67
F
Ext er n al l ow -sp e ed EPRO M s
0000H
E000H
FFFFH
ex t e rnal voi ce prom p data address 1000H
ex t e rnal voi ce prom p program address 80
0
1000H
max. 8K words
8000H
co ntrol code (address E000H - F
F
This configuration can be used at the development stage or f or 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 .
26K words
6 K words
00 00H
6800H
7FFFH
DSP internal ROM
BIOS+ALGORITHM
c
(addres s 0000H - 67
F
cont rol co de (addres s 6800H - 7F
0000H
8000H
FFF FH
External low-speed EPROMs
exter nal voice promp data address 1000H
exter nal voice promp program address 80
0
1000H
This configuration is used for production. T o set the v oice prompt configuration, see the MX93032-M1 Mode 7.
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MX93032
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" f or MCU mode, and it means "system control program" for DSP mode.
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. T o 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.
MESSAGE RECORDING AND ST ORAGE
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-
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 T one 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.
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.
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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 P ause func­tion is provided. The MX93032-M1 is ab le to fast-forw ard the playback of a recorded message, and the playing speed will be up to 33 %.
MESSAGE DELETION
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 enab les the control program to handle the deletion more flexibly, while the immediate method deletes a specified message directly .
VOICE PR OMPT
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 v ocabulary sup­ports up to 255 phrases. Giv en a phrase number in the V oice 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.
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 mes-
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 cloc k also supports system to stamp time to the recorded incoming message.
DTMF DETECTION
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 GENERA TION
Up to double tones can be generated by the MX93032­M1. The levels and frequencies are programmable by the HOST .
CALL PROGRESS T ONE 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 CANCELLA TION
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
All 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 tr ansmit 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 f or data transmission are given on SR-TSV-002474 which are classified into three layers, ph ysical lay er, data link er 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
CAS Detection
For Calling Identity Deliv ery 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 prog r am sets the VO X Record bit on Initial command, the recording of the speech would start when the voice activity is detected. The HOST has to set V O X threshold by issuing command Mode 13. Hexadecimal numbers f or V OX threshold are listed in T able
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
on the both ends, but also prevent feedback instability (singing) from occurring. Moreove r, the MX93032-M1 pro­vides automatic gain control (AGC) for microphone chan­nel which helps the outgoing voice to keep at a desired level especially f or a moving talk er, and speak er 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.
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DAA CONSIDERA TION FOR SPEAKER PHONE
It should be emphasized that DAA circuit pla ys an important role on improving the performance of speakerphone. This sub section gives some guides on how the D AA circuit may be "tuned". A diagram of D AA circuit must be given for simplifying the flowing interpretation.
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 pro vide additional attenuation to the loop gain one they con verge .
To have a better performance on speaker phone, w e strongly recommend that analog gain of D AA must satisfy the following two criteria:
1.As the near end (Microphone side) speaks only , the lev el at Codec 2 should fulfill that, "codec 2 input must be less than Codec 2 output due to electrical echo". This meaning can be e xpressed 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.
CODE C1
MX93032-M1
SPK DR V
MIC
GAIN
AC OU STIC COUP L ING
CODE C2
LINE
GAIN
LINE
DR V
4 - 2 WIRE COUPLING
LINE
COUPLING
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QUER Y 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 DO WN
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.
CRYST AL 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). T o enab le this mode, the FLLEN\ (54) pin must be connected to VDD. In this mode, the DSP high clock is gener ated from the high crystal directly, and the low crystal is mainly used in power down mode.
2. FLL (F requency-Lock ed-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 b y the MX93032-M1 and read out the pseudo response first. F ollowing 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 V oice prompt configuration
5. Setting Silence threshold
6. Setting VO X level
8.0 MODES OF OPERATION
The MX93032-M1 supports 16 modes of operations, which are listed as follows :
Command Mode Mode Name
0 Idle Mode 1 Record 2 Play 3 Message Status 4 DTMF Generator 5 Line Monitor 6 Delete Message 7 Set Current Time/Voice Prompt/Silence Threshold 8 Get Current Time 9 Initial and Test Memory 10 Get Record Message Information 11 V oice Prompt 12 Speaker Phone 13 Personal Mail 14 T elephone Data 15 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 V O X 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 Memor y command, the HOST can choose to delete the old OGM before a new one is recorded. If that bit is cleared to 0, the ne w 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 b y 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.)
PLA Y
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 pla yed to old. The ICMs that ha ve not been play ed 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 Pla y 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 ST ATUS
In this mode, there are 6 request codes for the HOST to get message information;
1. Get the number of personal recorded messages (request code = 0)
2. Get the number of personal new messages (request code = 1)
3. Get the number of personal old messages (request code = 2)
4. Get the available recording time with unit of second (request code = 3)
5. Get the number of total recorded messages (request code = 8) The result will be reported in the response word.
DTMF GENERA TOR
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 Gener ator 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 f ollowing 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 star ts 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, V O X, 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, VO X, Continuous Tone and the index of DTMF signal that has been detected :
INDEX DTMF CODE INDEX DTMF CODE CAS DTMF CAS DTMF 0 0000 No T one 0 1001 9 0 0001 1 0 1010 * 0 0010 2 0 1011 0 0 0011 3 0 1100 # 0 0100 4 0 1101 A 0 0101 5 0 1110 B 0 0110 6 0 1111 C 0 0111 7 1 1111 D 0 1000 8 1 0000 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. T o delete a specified ne w ICM, set Del New Msg (bit 10) to 1, give the New Msg ID in bit 0-6, clear bit 11, 7 to z ero.
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. T o erase the "Del Mark" from a specified ICM ("Del Mark" is a tag set on Play command f or 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/V OICE PROMPT/SILENCE THRESHOLD
In this mode the HOST can set the current time to the timer clock of the MX93032-M1. The time inf ormation 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. T o set output port OPT0-7, put data in bit0-7(Request Code=0100)
8. T o 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 infor mation includes second, minute, hour and week.
1. T o get second, use request code (bit 8-11) 0000.
2. To get minute, use request code (bit 8-11) 0001.
3. T o get hour , use request code (bit 8-11) 0010.
4. T o get w eek, use request code (bit 8-11) 0011.
The requested data will be reported in the response word.
INITIAL AND TEST MEMOR Y
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 pow er loss during operation. This function is normally used after first power­up of the system. The response word reported to the control program contains the inf ormation 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 av ailable 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. V O X On/Off Record
If the HOST turns on the V OX Record function (b y 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 INFORMA TION
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 w ord is 0.
7. Get the message ICM/MEMO attribute. (Request code = 0110). T o sho w 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.T o get link index data(Request Code=1001),refer to mode7
12.T o 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 R OM or Flash memory , are provided by the MX93032-M1 and pronounced using the speech decompression algorithm. The HOST can pause the playbac k by setting Pause (bit 8) to 1 or resume from pause b y clearing Pause bit to 0. Fast forward playback of v oice prompt is also provided by setting F ast 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 V oice 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/T one generation (Request Code = 0x0001)
3. Generating DTMF/T one 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, F req_Lo set and F req_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 bef ore issuing the message related commands such as Record, Pla y, 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 receiv ed 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 F ormat 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 f ormats, 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 le vel are listed in T ab le 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 f or Ra w data mode and Cooked Data mode are 8 b ytes and 60 bytes long, respectiv ely. 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 ov erflow 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 D A T A AND V OICE 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 DA T A
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 T elephone 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 ne w T el. ID, host should set STOP bit to 1 f or the current T el. ID.
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9.0 COMMAND FORMAT AND PROTOCAL
Mode 0. IDLE MODE
(15-12)
0000
CONF
00000000000
0000000000000000
0000000000000000
STOP
(11-1)
0
(15-12)
RESP
(11-1)
0
Mode 1. RECORD MODE
(15-12) (11) (10-8) (7) (6)
0001
MARK
FAIL
MARKED RECORD
P A USE STOP
CONF
TAIL CUT
(5-0)
TAIL CUT : 0 ~ 63 units. One unit length can be 400ms or 200ms (set by Mode 9). ST OP = 1 : stop recording. PAU SE = 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.
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(15-9) (8) (7) (5)(6) (4)
RECORD TIME (sec) CAS MFULL
VOX
TONE
CONT TONE
RESP
DTMF
(3-0)
DTMF : the detected DTMF value during recording. CONT TONE = 1 : continuous tone is found. T O NE = 1 : call progress tone is found. VOX = 1 : the input speech po wer le v el is under the VOX LEVEL (set b y Mode 12). MFULL = 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 f ollowing table .
DTMF Reference T able:
INDEX DTMF CODE INDEX DTMF CODE CAS DTMF CAS DTMF 0 0000 No Tone 0 1001 9 0 0001 1 0 1010 * 0 0010 2 0 1011 0 0 0011 3 0 1100 # 0 0100 4 0 1101 A 0 0101 5 0 1110 B 0 0110 6 0 1111 C 0 0111 7 1 1111 D 0 1000 8 1 0000 CAS TONE
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Mode 2. PLAY MODE
(15-12) (11) (10) (9) (8) (7)
0010
FAST PLA Y
PLA Y NEW
P AUSE STOPSTOP
CONF
MSG ID
(6-0)
(15-9) (8) (7) (5)(6) (4)
PLA Y TIME (sec) CAS 0
PLA Y
END
TONE
0
RESP
DTMF
(3-0)
MSG ID = 1 ~ 127 can be assigned for playing. DEL MARK = 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. PAU SE = 0 : continue playing.
= 1 : pause playing. ST O P = 1 : stop playing. PLAY NEW = 0 : the MSG ID is related to total messages.
= 1 : the MSG ID is related to new messages. F AST PLAY = 0 : normal play speed.
= 1 : play speed will be up to 33%.
DTMF and T ONE : see RECORD MODE. PLA Y 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. MESSA GE ST ATUS MODE
(15-12)
0011
CONF
00000000000
DATA
REQUEST CODE
(11-4)
(15-0)
(3-0)
RESP
REQUEST CODE:
REQUEST CODE STA TUS D A TA
0000 number of personal total messages 0 ~ 127 0001 number of personal new messages 0 ~ 127 0010 number of personal old messages 0 ~ 127 0011 available time 0 ~ 65535 0101 garbage collection recommendation 0 or 1 1000 number of total messages 0 ~ 127
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Mode 4. DTMF GENERA T OR MODE
For DSP mode:
GAIN1, GAIN2 : 0 ~ 14 denote 3dB ~ -25dB (2dB/one step), 15 denotes no related signal. T O =0 OUTPUT CHANNEL CODEC 0.
=1 OUTPUT CHANNEL CODEC1. ST OP = 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 B UF2 (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.
(15-12) (11) (10) 9 8 (7-4)
0100
ST ART
STOP
0 GAIN2TO
CONF
GAIN1
(3-0)
0000000000000000
(15-0)
RESP
For MCU mode:
(15-12) (11) (10) 9 8 (7-4)
0100
ST ART
STOP
0 GAIN2TO
CONF
GAIN1
(3-0)
GAIN1, GAIN2 : 0 ~ 14 denote 3dB ~ -25dB (2dB/one step), 15 denotes no related signal. T O =0 OUTPUT CHANNEL CODEC 0.
=1 OUTPUT CHANNEL CODEC1. ST OP = 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 F requency 2 is put in the f ollowing 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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Frequency 1
(15-0)
CONF
(15-0)
RESP
0000000000000000
0000000000000000
Frequency 2
(15-0)
CONF
(15-0)
RESP
0000000000000000
(15-0)
RESP
(15-12) (11-8) (7) (6-1)
0101
0000
PAUSE 000000
CONF
STOP
(0)
(15-9) (8) (7) (6) (5) (4)
0000000
0CAS
VOX
CONT TONE
TONE
RESP
DTMF
(3-0)
Mode 5. LINE MONIT OR MODE
ST OP = 0 : start and continue the line monitor mode.
= 1 : stop the line monitor mode.
DTMF, CONT TONE, TONE, PAUSE, V O X and CAS: see RECORD MODE.
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Mode 6. DELETE MESSA GE MODE
MSG ID : 1 ~ 127. DEL ALL OLD = 1 : delete all old messages. (in this case, the MSG ID field is ignored) REAL DEL = 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) (10) (9) (8) (7)
0110
DEL
MARK
CLR
DEL NEW MSG
DEL ALL OLD
REAL
DEL
0
CONF
MSG ID
(6-0)
(15-12)
0110
RESP
000000000000
(11-0)
Mode 7. SET CURRENT TIME/VOICE PR OMPT/SILENCE THRESHOLD MODE
(15-12)
0111
RESP
000000000000
(11-0)
(15-12)
0110
CONF
REQUEST CODE DATA
(11-8) (7-0)
REQUEST CODE:
TIME REQUEST CODE D ATA
SECOND 0000 0 ~ 59 MINUTE 0001 0 ~ 59 HOUR 0010 0 ~ 23 WEEK 0011 0 ~ 6 SET OUTPUT PORT 0100 00 ~ FF H SET LINK INDEX 0101 00 ~ FF H VOICE PROMPT CONFIG 0110 0 ~ 7 SILENCE THRESHOLD 0111 0 ~ F H SEND MX93002 CMD 93993000(A) CMD 1xxx 00 ~ FF H
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Request Code = 0110 (set voice prompt configuration),
(15-12)
0110
CONF
0110
00000
FVOP
SELECT
EVOP
SELECT
IVOP
SELECT
(11-8) (7-3) (2) (1) (0)
IVOP SELECT (internal voice prompt setting),
= 0 : internal voice prompt not selected. = 1 : internal voice prompt selected and test.
EVOP SELECT (e xternal 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)
00000000000
RESP
EVOP
ADDR
EVOP
STATUS
IVOP
STATUS
(4) (3) (2) (1) (0)
EVOP
UNIT
FVOP
STATUS
IVOP STA TUS (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 STA TUS (FLASH voice prompt test status),
= 0 : test fail.
= 1 : test ok. (when EVOP STATUS = 1), EV OP 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) (7-4) (3-0)
0111 0111 0000 SILENCE LEVEL
CONF
(15-0)
0000000000000000
RESP
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 Le vel.
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Mode 8. GET CURRENT TIME MODE
REQUEST CODE, 0000 ~ 0011 : see Mode 7.
Mode 9. INITIAL AND TEST MEMORY MODE
(15-12) (11-8) (7-0)
1000 REQUEST CODE 00000000
CONF
(15-0)
DATA
RESP
(15-12) (11-9) (8) (7) (6) (5) (4) (3) (2) (1) (0)
1001 000
OLD
MARK
SET
TAIL CUT UNIT
NEW/
OLD
SELECT
ANNC
DEL
VOX
ON/OFF
RECORD
LINE
ON/ OFF
GARBAGE
COLLECT
FLASH
INIT
FLASH
RE-INIT
CONF
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. (Bef ore 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.
VO X ON/OFF RECORD
= 0 : record the starting silence. = 1 : not record the starting silence which power le vel is under VO X 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) (7-4) (3-0)
FLASH ADDR SIZE FLASH BIT SIZEFLASH GOOD RATE
FLASH
G/B
RESP
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 FLASH ADDR SIZE CODE FLASH BIT SIZE 0000 0 M 0000 0 0101 1 M 1111 16 0110 2 M
Mode 10. GET RECORD MESSA GE INFORMA TION MODE
(15-12) (11-8)
(7)
(6-0)
NEW MSG
MSG ID1010
REQUEST CODE
CONF
MSG ID : 1 ~ 127. NEW MSG = 0 : the MSG ID is related to total messages.
= 1 : the MSG ID is related to new messages.
INFORMA TION REQUEST CODE DAT A
SECOND 0000 0000 ~ 003BH MINUTE 0001 0000 ~ 003BH HOUR 0010 0000 ~ 0017H WEEK 0011 0000 ~ 0006H RECORD LENGTH 0100 BIT 15 - BIT 0 TIMER ST ATUS 0101 0 = default timer ; 1 = timer has set ICM/MEMO 0110 0 = ICM record ; 1 = MEMO record DEL MARK ST ATUS 0111 0 = no del mark ; 1 = has del mark MSG NEW/OLD ST ATUS 1000 0 = no old mark ; 1 = has old mark GET LINK INDEX 1001 00~FFH WRITE LINK INDEX (specified by 75XX) 1010 MSG ID=1~127 OGM EXISTING ST ATUS 1111 0 = OGM not exist ; 1 = OGM exist
(15-0)
DATA
RESP
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Mode 11. V OICE PR OMPT MODE
(15-12) (11) (10) (9) (8)
1011
FAST PLA Y
P A USE
STOP
0
CONF
PHRASE NUMBER
(7-0)
(15-9)
(8) (7) (6) (5) (4)
PLA Y TIME(sec)
CAS
PLA Y
END
TONE
0
0
RESP
DTMF
(3-0)
PHRASE NUMBER : 1 ~ 255. PAUSE, ST OP and FAST PLAY : see PLAY MODE.
DTMF, T ONE, PLAY END CAS and Play TIME : see PLAY MODE.
Mode 12. SPEAKERPHONE MODE
(15-12) (11-8) (7-0)
1100 REQUEST CODE
PARAMETERS
CONF
REQUEST CODE FUNCTION
0000 Free Run 0001 Gain Setting For T one 0010 Generate T one 0011 Reserved 0100 Set ERL 0101 Reserved 0110 Set T/R & R/T ratios 0111 Set T one T able * 1000 Set Speaker volume & line out gain 1001 Set Loop attenuation 1111 Read Internal Parameter (Reserved) Others Reserved
* : Not available
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Request Code = 0000 (Free Run)
(15-12) (11-8) (4-0)(5)(6)(7)
1100 0000
STOP
LINE
MUTE
SPK
MUTE
00000
CONF
ST OP = 0 : turn on speakerphone
= 1 : turn off speaker phone
LINE MUTE = 1 : mute line out
= 0 : normal line out
SPK MUTE = 1 : mute loudspeaker
= 0 : normal speaker out
(15-9) (2-0)(4-3)(7-6)(8) (5)
0000000
CAS
00
00TONE
LOOP
A TTENU ATION
RESP
T O N E = 1 : Call Prog ress Tone is detected Loop Attenuation : see Set Loop Attenuation CAS : CAS detection (see mode 15)
Request Code = 0001 (Set Gain for Tone)
(15-3) (2-0)
0000000000000
LOOP
A TTENU A TION
RESP
(15-12)
(11-8)
(7-4)
(3-0)
1100
0001
GAIN-LO, GAIN-HI:0~14 denote 0dB~-14dB; 15 denotes mute
GAIN-LO GAIN-HI
CONF
GAIN TABLE:
GAIN 01234567 dB 0dB -1dB -2dB -3dB -4dB -5dB -6dB -7dB GAIN 8 9 A B C D E F dB -8dB -9dB -10dB -11dB -12dB -13dB -14dB MUTE
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Request Code = 0010 (Send DTMF or Single Tone)
START = 1 : start to generate Tone according to the index assigned
= 0 : stop generating Tone
D or S = 0 : generate Single Tone (This bit is valid under bit 4 set)
= 1 : generate Dual Tone
(15-12)
(4-0)(7) (6) (5)
(11-8)
1100 0010
ST ART
0
D S
INDEX
CONF
(15-9) (8) (7-3) (2-0)
0000000 CAS 00000
LOOP
ATTENUATION
RESP
INDEX TABLE:
Index Value Tone Index Value Tone
00000 DTMF_0 10000 00001 DTMF_1 10001 00010 DTMF_2 10010 00011 DTMF_3 10011 00100 DTMF_4 10100 00101 DTMF_5 10101 00110 DTMF_6 10110 00111 DTMF_7 10111 01000 DTMF_8 11000 01001 DTMF_9 11001 01010 DTMF_* 11010 01011 DTMF_# 11011 01100 DTMF_A 11100 01101 DTMF_B 11101 01110 DTMF_C 11110 01111 DTMF_D 11111
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)
(15-3) (2-0)
0000000000000
LOOP
A TTENU A TION
RESP
(15-12)
(11-8)
(7-4)
(3-0)
1100
0100
ERL_ACE:echo return loss occurred on speaker side ERL_LEC:echo return loss occurred on line side
ERL_AEC ERL_LEC
CONF
ERL_AEC & ERL_LEC : Range from 0 to 30 with 2 dB/step
0000 00001 0010 0011 0100 0101 0110 0111 0 dB 2 dB 4 dB 6 dB 8 dB 10 dB 12 dB 14 dB 1000 1001 1010 1011 1100 1101 1110 1111 16 dB 18 dB 20 dB 22 dB 24 dB 26 dB 28 dB 30 dB
Note: The def ault value of ERL_LEC and ERL_AEC are 0 dB.
Request Code = 0110 (Set T/R & R/T ratio)
(15-3) (2-0)
0000000000000
LOOP
A TTENU ATION
RESP
(15-12)
(11-8)
(5-3)(7-6)
(2-0)
1100
0110
00
R/T
RA TIO
T/R
RA TIO
CONF
T/R & R/T RATIO
000 001 010 011 100 101 110 111 3 dB 6 dB 9 dB 12 dB 15 dB Reserved Reserved Reserved
Note : 12 dB is default value
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Request Code = 0111 (Set T one T able)
(15-0)
0000000000000000
RESP
(15-0)
0000000000000000
RESP
(15-12)
(11-8)
(7-4)
(3-0)
1100
Index: Range is from 0 to 15 (see Request Code=0010)
0110
0000
INDEX
CONF
(15-12)
(11-8)
(7-0)
1100
0111
FREQ_LO
CONF
(15-0)
0000000000000000
FREQ_HI and FREQ_LO will compose a 16-bit frequency, which is equal to "target Frequency *8.192".
RESP
(15-12)
(11-8)
(7-0)
1100
0111
FREQ_LO
CONF
Request Code = 1000 (Set SPK_GAIN & LINE _GAIN)
(15-3)
0000000000000
RESP
(15-12)
(11-8) (7-4) (3-0)
(2-0)
1100
LINE_GAIN:line out volume gain SPK_GAIN:speaker out volume gain
1000
LINE_GAIN
SPK_GAIN
LOOP
ATTENUATION
CONF
SPK_GAIN & LINE_GAIN : Range from -20 to 20
0000 00001 0010 0011 0100 0101 0110 0111
-20dB -16 dB -12 dB -8 dB -4 dB 0 dB 2 dB 4 dB 1000 1001 1010 1011 1100 1101 1110 1111 6 dB 8 dB 10 dB 12 dB 14 dB 16 dB 18 dB 20 dB
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Request Code = 1001 (Set Loop Attenuation)
(15-3)
0000000000000
RESP
(15-12)
(11-8) (7-3) (2-0)
(2-0)
1100
1001
00000
LOOP
A TTENU A TION
LOOP
A TTENU A TION
CONF
Loop Attenuation (CONF) : Range from 0 to 28 with 4dB/Step
000 001 010 011 100 101 110 111
0.dB 4 dB 8 dB 12 dB 16 dB 20 dB 24 dB 28 dB
Full Duplex Smooth Note:8dB is default value Half Duplex
Loop Attenuation (RESP) :
000 001 010 011 100 101 110 111
0.dB 4 dB 8 dB 12 dB 16 dB 20 dB 24 dB >28 dB
Request Code = 1111 (Read Internal Parameters)
(15-0)
RESP
(15-12)
(11-8)
(7-0)
1100
1111
Parameter Type
Parameter
CONF
Parameter Type Description
00000000 AEC_SNR1 00000001 LEC_SNR1 00000010 AEC_STATE 00000011 LEC_STATE 00000100 AEC_AERLE 00000101 LEC_AERLE 00000110 ACC_ATT 00000111 LEC_ATT 00001000 TR_MODE
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Read CODEC Register (Reference 93002)
(15-0)
RESP
(15-12)
(11-8) (7)
(6-0)
1100
1111
1
Parameter Type
Parameter
CONF
CODEC Register Description
0000000 Register 0 0000001 Register 1 0000010 Register 2 0000011 Register 3 0000100 Register 4 0000101 Register 5 0000110 Register 6 0000111 Register 7
Mode 13. PERSONAL MAIL MODE
(15-12)
(11-8) (7-5) (3)(4)
(2-0)
1101
0000 000
LOOP BACK
MEMO /ICM
PERSONAL
MAIL
CONF
PERSONAL MAIL : 0 ~ 7, total 8 personal mailbo xes. 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 : disab le loop back. Then in RECORD and LINE MONIT OR 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) (6)(7) (3)(5-3)
(2)
(0)
1101
0001
CAS SPK
0
COMPRESS FORMAT
SILENCE RATE
OUTPUT
LEVEL
CODEC
FORMAT
CONF
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 FORMA T (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 RA TE
(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 FORMA T (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
Set T elephone Data Configuration,
(15-12)
(11-8) (7-5) (4-2)
(0)(1)
1101
0011 000
TEL RECORD
NUMBER
01
CONF
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
(15-0)
RESP
000000000000000
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V oice Activity Mode,
(15-12)
(11-8) (7-6) (5-0)
1101
0111 00
VOX Level
CONF
VO X LEVEL : see Table 6.
Mode 14. TELEPHONE D A T A MODE
WRITE TELEPHONE DA TA
(15-12)
(11-10) (9) (8)
(7-0)
1110
TEL RECORD ID:1~255, each telephone record has maximum 60 bytes. STOP =1:stop telephone data writing.
FULL=1: no free telephone record is available.
00 STOP 0
TEL RECORD ID
CONF
(C1)
(15-12)
(11-10) (0)
(9-1)
1110
00
000000000
FULL
RESP
(R1)
(15-12)
(11-10) (9) (8)
(7-0)
1110
DATA : the telephone data to write. STOP =1:stop telephone data writing.
FULL =1:current telephone record is full.
00 STOP 0
DATA
CONF
(C2)
(15-12)
(11-10) (0)
(9-1)
1110
00
000000000
FULL
RESP
(R2)
(15-0)
RESP
000000000000000
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READ TELEPHONE DATA
(15-12)
(11-10) (9) (8)
(7-0)
1110
01 STOP 0
TEL RECORD ID
CONF
TEL RECORD ID : 1 ~ 255. ST OP = 1 : stop telephone data reading.
(15-12)
(11-10) (9) (8)
(7-0)
1110
00 0
READ END
DATA
RESP
D AT A : the read out telephone data. READ END = 1 : end of the current telephone record and the f ollowed DA TA is invalid.
Mode 15. CALLER _ID MODE
(15-12)
(11-9)
(8-0)
1111
REQUEST
CODE
P ARAMETER
CONF
REQUEST CODE FUNCTION
00 0 Demodulate FSK 10 0 Set Response method of CAS detected 101 Set CAS sensitivity 00 1 ACK DTMF A 01 0 ACK DTMF D
Request Code = 000 (Demodulate FSK T o RA W DATA)
(15-12)
(11-8) (8) (7-2)
(0)(1)
1111
000
0
000000
TYPE START\
CONF
TYPE = 0 : CID TYPE I 200 bits channel seizure & 10 bits mark signal
= 1 : CID TYPE II 10 bits mark signal
START\ = 0 : Start / Continue to Demodulate
= 1 : Stop Demodulating
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(15) (14) (13)
(12-8)
(7-0)
CD ERR DR
00000
FSK DATA BYTE
RESP
C D = 0 : Carrier is not detected (FSK signal do not occur on line)
= 1 : Carrier is detected (FSK signal takes place on line)
ER R = 0 : Data buffer is not overflow
= 1 : Data buffer is overflow
DR = 0 : FSK data byte is not ready
= 1 : FSK data byte is ready
Note : The length for RAW Data buff er is 8 b ytes long.
Request Code = 000 (Demodulate FSK T o COOKED DATA)
(15-12)
(11-8) (8) (7-2)
(0)(1)
1111
000
1
000000
TYPE START\
CONF
TYPE = 0 : CID TYPE I 200 bits channel seizure & 10 bits mark signal
= 1 : CID TYPE II 10 bits mark signal
START\ = 0 : Start / Continue to Demodulate
= 1 : Stop Demodulating
(15) (14) (13)
(12-8)
(7-0)
CD ERR DR
00000
FSK DATA BYTE
RESP
C D = 0 : Carrier is not detected (FSK signal do not occur on line)
= 1 : Carrier is detected (FSK signal takes place on line)
ER R = 0 : Data buffer is not overflow
= 1 : Data buffer is overflow
DR = 0 : FSK data byte is not ready
= 1 : FSK data byte is ready
Note: 1) The second b yte data receiv ed by FSK decoder will be considered as the total data length.
2) The first b yte sent out are the total length +1 in which the chec ksum is included.
3) The length f or cook ed data buff er is 60 b ytes long.
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(15-12)
(11-9) (8-1)
(0)
1111
000
00000000
CAS
TYPE
CONF
Request Code = 100 (Set Response method of CAS detected)
CAS TYPE = 0 : Continuous Response. Response CAS behavior fidelity (Delay about 55ms)
= 1 : Response One Time. Response CAS detected after CAS is off
55 ms
15 ms
CAS SIGNAL :
Continuous Resp
Resp One Time
Request Code = 101 (Set CAS Sensitivity)
(15-0)
RESP
000000000000000
(15-12)
(11-9)
(8-3)
(2-0)
1111
101
000000
CAS_L VL
CONF
(15-0)
RESP
000000000000000
01234567
Sensitivity -32 -34 -3 6 -38 -40 -42 -44 -46
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Request Code = 001 or 010 (ACK DTMF A or A CK DTMF D)
(15-12)
(11-9)
(7-4)(8)
(3-0)
1111
ACK
TO
GAIN2
GAIN1
CONF
(15-0)
RESP
000000000000000
GAIN1, GAIN2 : 0~14 denote 3dB ~ -25dB (2dB/one step), 15 denotes no related signal. T O = 0 OUTPUT CHANNEL CODEC 0.
= 1 OUTPUT CHANNEL CODEC 1.
ACK =001 ACK DTMF A.
=010 ACK DTMF D.
TABLE 1. PERFORMANCE TEST FOR TONE DETECTION
(noiseless) 0 -3 -6 -10 -20 -30 -33 -36 -39 Pow er (dB) p2p (volt) 5 3.54 2.5 1.58 .5 .158 .112 .079 .056 280 Hz x x x x x x x x x 300 Hz ok ok o k ok ok ok ok x x 400 Hz ok ok ok ok ok ok ok ok x 500 Hz ok ok ok ok ok ok ok ok x 600 Hz ok ok ok ok ok ok ok ok x 640 Hz ok ok ok ok ok ok ok x x 670 Hz x x x x x x x x x 480+620 Hz ok ok ok ok ok ok x x x 350+440 Hz ok ok ok ok ok ok x x x
* 0 dB is defined as the Vrms of maximum codec input ** The maxim um peak-to-peak voltage (P2P) of Codec input is assumed to be 5V
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SNR (dB) 40 25 12
detect time delay / off time delay (ms) 300 Hz 50 / 40 55 / 40 65 / 40 400 Hz 40 / 40 40 / 40 45 / 45 500 Hz 40 / 40 40 / 40 45 / 45 640 Hz 45 / 40 45 / 40 60 / 35 480+620 Hz 40 / 40 45 / 40 50 / 40 350+440 Hz 45 / 40 45 / 40 55 / 40
* 0 dB is defined as the Vrms of maximum codec input ** the detect/off time delay is defined as follows
detect time delay
off time delay
INPUT
RESPONSE
TABLE 2. RESOLUTION TEST FOR DUAL TONE (FREQUENCY DETECTABLE RANGE)
RANGE / POWER -4 dB -25 dB (300 + F2 Hz) F2 > 320 Hz F2 > 320 Hz (640 + F2 Hz) F2 < 620 Hz F2 < 620 Hz
TABLE 3. THD DISTORTION TEST FOR DUAL TONE (DISTORTION TOLERANCE)
INPUT / POWER -4 dB -25 dB (300 + 320 Hz) > 10 dB > 10 dB (640 + 620 Hz) > 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
P ARAMETER MIN TYP MAX UNIT
Detection level * -30 dB Rejection level -3 7 dB Frequency range 300 64 0 Hz
Noise tolerance -1 2 dB
THD tolerance -10 dB
Rejection frequency range < 280 Hz or > 670 Hz Minimum signal duration accept 50 ms
Maximum response time 50 ms * 0 dB is defined as the Vrms of maximum codec input
TABLE 5. DTMF CHARACTERISTICS
PARAMETER MIN TYP MAX UNIT DTMF signal level for detection * -4 3 0 dB DTMF twist (high/low tone) 9 dB DTMF noise tolerance -10 d B DTMF tone duration accept 50 ms DTMF interdigit pause accept 50 ms DTMF frequency deviation accept limit 1. 5 % DTMF frequency deviation reject limit 3 % T one generator le vel -2 5 +3 dB T one generator frequency accur acy 0.1 % T one generator le vel accuracy 0.5 dB
±
±
±
± ±
* 0 dB is defined as the Vrms of maximum codec input
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TABLE 6. VOX THRESHOLD
dB VOX_LEVEL
-9 0
-10 1
-11 2
-12 3
-13 4
-14 5
-15 6
-16 7
-17 8
-18 9
-19 10
-20 11
-21 12
-22 13
-23 14
-24 15
-25 16
-26 17
-27 18
-28 19
-29 20
-30 21
-31 22
-32 23
-33 24
-34 25
-35 26
-36 27
-37 28
-38 29
-39 30
-40 31
-41 32
-42 33
-43 34
-44 35
-45 36
-46 37
-47 38
-48 39
-49 40
-50 41
-51 42
-52 43
-53 44
-54 45
-55 46
-56 47
-57 48
-58 49
-59 50
-60 51
-61 52
-62 53
0 dB is defined as the Vrms of maximum Codec input.
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TABLE 7. SILENCE THRESHOLD LEVEL
SILENCE LEVEL dB 1 -58 2 -55 3 -52 4 -49 5 -46 6 -43 7 -40 8 -37 9 -34 10 -31 11 -28 12 -25 13 -22 14 -19 15 -16
0 dB : maximum input of sinusoidal wave
TABLE 8. ECHO CANCELLER PERFORMANCE *
T est Description Final Echo Return Loss Improvement (ERLI) (Singletalk mode) > 28 dB Maximum tail length 12 ms Convergence Rate (Singletalk mode) > 28 dB 500 ms after initialization with cleared register and with near-end signal set to zero at initialization time Infinite Return Loss Convergence > 28 dB Return Echo Loss Improvement 500 ms after echo path is interrupted
* test signal : band-limited white-noise (300-3400 Hz)
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TABLE 9. MAXIMUM COMMAND RESPONSE TIME
COMMAND MAX RESP TIME (ms) Record Mode :
1. start 32
2. normal 6
3. stop 10 4 Play Mode :
1. start 14
2. normal 5
3. stop 1 Message Status Mode :
1. get message number 8
2. get FLASH available time 6
3. get FLASH garbage collection recommendation status 1 DTMF Generator Mode 1 Line Monitor Mode 6 Delete Message Mode :
1. delete one message 4300
2. delete all old messages 5800
3. real delete 5500 Set Current Time/V oice Prompt/Silence Threshold Mode 1 Get Current Time Mode 1 Initial and Test Memory Mode :
1. FLASH re-initialization 2400
2. FLASH initialization/error correction 5800
3. FLASH garbage collection 15 0
4. other misc. setting 1 Get Record Message Information Mode 10 V oice Prompt Mode :
1. start 6
2. normal 5
3. stop 1 Voice Activity Mode 1 Personal Mail Mode 1
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APPENDIX A : CID TEST REPORT BASED ON SR-3004
1 . T ype 1 T esting
1.1 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 T ransmission testing is independent of Message F ormat 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 T ransmission T esting
Mark Frequency Space Frequency Twist Immunity Noise Immunity Baud Rate
1.2.1 Immunity to Frequency and Baud Rate V ariation
Item T esting Range Desired Range SP ACE 2100HZ ~ 2300HZ 2178HZ ~ 2222HZ MARK 1100HZ ~ 1300HZ 1188HZ ~ 1212HZ BAUD RATE 1150 baud ~ 1250 baud 1188 baud ~ 1212 baud
T esting Condition : Signal amplitude at -20dBm for per frequency SN R : 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.
2100 2150 2178 2222 2250 2300 1100 95 100 100 100 100 100 1150 100 100 100 100 100 100 1188 100 100 100 100 100 100 1212 100 100 100 100 100 100 1250 100 100 100 100 100 100 1300 100 100 100 100 100 80
Baud Rate = 1188 baud
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2100 2150 2178 2222 2250 2300 1100 100 100 100 100 100 100 1150 100 100 100 100 100 100 1188 100 100 100 100 100 100 1212 100 100 100 100 100 100 1250 100 100 100 100 100 100 1300 100 100 100 100 100 100
Baud Rate = 1212 baud
1.2.2 T wist Immunity Range +/- 10 dB
Fix Condition: Space Frequency : 2200 HZ Mark F requency : 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.
Positiv e Twist : (Mark > Space) Mark is fixed to -13 dBm and Space vary from -13 dBm to -48 dBm with step 5 dB
-13 -18 -23 -28 -33 -38 -43 -48 100 100 100 0 0 0 0 0
Negative Twist : (Mark < Space) Space is Fixed to -13 dBm and Mark vary from -13dBm to -48 dBm with step 5 dB
-13 -18 -23 -28 -33 -38 -43 -48 100 100 100 0 0 0 0 0
1.3.4 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 12 14 15 18 20 22 24 0 0 0 100 100 100 100 100
26 28 30 32 34 36 38 40 100 100 100 100 100 100 100 100
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TYPE II TESTING The Type 2 CPE requires all of the same conformance test as T ype 1 CPE in addition to "CAS detector perf ormance”
2.1 CAS Detector Perf ormance 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
Parameter Nominal Minimum Maximum Lower Frequency 2130 +/- 2 Hz 2119 Hz 2141 Hz Upper Frequency 2750 +/- 2 Hz 2736 Hz 2764 Hz Pow er le v el -22 +/- 1 dBm -32 dBm -14 dBm Twist 0 ~ 2 dB 0 dB 6 dB Duration 80 +/- 1 ms 75 ms 85 ms
Various combinations of the abov e parameter v alues 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 follow s. 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 :
T est No Frequency ( Low , High) Power Le vel (Low , High) Duration Correct Times 1 (2130,2750) (-22,-22) 80 1000
T est No Frequency ( Low , High) Power Lev el (Low , High) Duration Correct Times 1 (2119,2750) (-22,-22) 80 1000 2 (2141,2750) (-22,-22) 80 1000 3 (2130,2736) (-22,-22) 80 1000 4 (2130,2764) (-22,-22) 80 1000 5 (2130,2750) (-14,-14) 80 1000 6 (2130,2750) (-32,-32) 80 1000 7 (2130,2750) (-14,-20) 80 1000 8 (2130,275) (-32,-26) 80 1000 9 (2130,2750) (-20,-14) 80 1000 10 (2130,2750) (-26,-32) 80 1000 11 (2130,2750) (-22,-22) 75 1000 12 (2130,2750) (-22,-22) 85 1000
Group 2 :
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T est No Frequency ( Low , High) Power Le vel (Low , High) Duration Correct Times 1 (2120,2737) (-31,-31) 76 1000 2 (2120,2737) (-31,-26) 76 1000 3 (2120,2737) (-26,-31) 76 1000 4 (2120,2737) (-15,-15) 76 1000 5 (2120,2737) (-15,-20) 76 1000 6 (2120,2737) (-20,-15) 76 1000 7 (2120,2737) (-31,-31) 84 1000 8 (2120,2737) (-31,-26) 84 1000 9 (2120,2737) (-26,-31) 84 1000 10 (2120,2737) (-15,-15) 84 1000 11 (2120,2737) (-15,-20) 84 1000 12 (2120,2737) (-20,-15) 84 1000 13 (2140,2737) (-31,-31) 76 1000 14 (2140,2737) (-31,-26) 76 1000 15 (2140,2737) (-26,-31) 76 1000 16 (2140,2737) (-15,-15) 76 1000 17 (2140,2737) (-15,-20) 76 1000 18 (2140,2737) (-20,-15) 76 1000 19 (2140,2737) (-31,-31) 84 1000 20 (2140,2737) (-31,-26) 84 1000 21 (2140,2737) (-26,-31) 84 1000 22 (2140,2737) (-15,-15) 84 1000 23 (2140,2737) (-15,-20) 84 1000 24 (2140,2737) (-20,-15) 84 1000 25 (2120,2763) (-31,-31) 76 1000 26 (2120,2763) (-31,-26) 76 1000 27 (2120,2763) (-26,-31) 76 1000 28 (2120,2763) (-15,-15) 76 1000 29 (2120,2763) (-15,-20) 76 1000 30 (2120,2763) (-20,-15) 76 1000 31 (2120,2763) (-31,-31) 84 1000 32 (2120,2763) (-31,-26) 84 1000 33 (2120,2763) (-26,-31) 84 1000 34 (2120,2763) (-15,-15) 84 1000 35 (2120,2763) (-15,-20) 84 1000 36 (2120,2763) (-20,-15) 84 1000 37 (2140,2763) (-31,-31) 76 1000 38 (2140,2763) (-31,-26) 76 1000 39 (2140,2763) (-26,-31) 76 1000 40 (2140,2763) (-15,-15) 76 1000 41 (2140,2763) (-15,-20) 76 1000 42 (2140,2763) (-20,-15) 76 1000 43 (2140,2763) (-31,-31) 84 1000 44 (2140,2763) (-31,-26) 84 1000 45 (2140,2763) (-26,-31) 84 1000 46 (2140,2763) (-15,-15) 84 1000 47 (2140,2763) (-15,-20) 84 1000 48 (2140,2763) (-20,-15) 84 1000
Group 3 :
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T est No Frequency ( Low , High) Power Le vel (Low , High) Duration Correct Times 1 (2120,2750) (-46,-46) 80 0
CAS Level Reject Limit
2.1.1.2 Talk-down Test
Row Signal Level Speech Level SNR CAS Pulses CAS Pulses Derated Miss
(dBm/Tone) (dBm) Received Sent Rate (%) 1 -28 -10 -18 3608 6000 19.933 2 -28 -13 -15 4586 6000 11.7833 3 -22 -10 -12 909 1125 9.6 4 -26 -16 -10 1073 1125 2.311 5 -28 -22 -6 1112 1125 0.5778 6 -22 -19 -3 1124 1125 0.0444
CAS LEVEL
Speech -30 dBm -28 dBm -26 dBm -24 dBm -22 dBm -20 dBm -18 dBm -16 dBm
-7 dBm 33.5166^ 28.0833^ 22.6500^ 17.2166~ 11.7833 10.3278~ 5.9556~ 1.8778~
-10 dBm 25.3666^ 19.933 14.5000~ 11.0555~ 9.60 2.3111 1.4445~ 0.5778
-13 dBm 17.2166~ 11.7833 10.3278~ 5.9556~ 1.8778~ 1.0111~ 0.4000~ 0.0444
-16 dBm 11.0555~ 9.6 2.3111 1.4445~ 0.5778 0.2222~ 0* 0*
-19 dBm 5.9556~ 1.8778~ 1.0111~ 0.4~ 0.0444 0* 0* 0 *
-22 dBm 1.4445~ 0.5788 0.2222~ 0* 0* 0* 0* 0*
-25 dBm 0.4~ 0.0444 0* 0* 0* 0* 0* 0*
-28 dBm 0* 0 * 0* 0* 0* 0* 0* 0*
-31 dBm 0* 0 * 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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-30 -28 -26 -24 -22 -20 -18 -16 Sum
-7 0.000129 0.000172 0.000344 0.000731 0.000731 0.000731 0.000731 0.000731 0.0043
-10 0.000774 0.001032 0.002064 0.004386 0.004386 0.004386 0.004386 0.004386 0.0258
-13 0.002973 0.003964 0.007928 0.016847 0.016847 0.016847 0.016847 0.016847 0.0991
-16 0.006684 0.008912 0.017824 0.037876 0.037876 0.037876 0.037876 0.037876 0.2228
-19 0.0088 0.01184 0.02368 0.05032 0.05032 0.05032 0.05032 0.05032 0.296
-22 0.006684 0.008912 0.017824 0.037876 0.037876 0.037876 0.037876 0.037876 0.2228
-25 0.002973 0.003964 0.007928 0.016847 0.016847 0.016847 0.016847 0.016847 0.0991
-28 0.000774 0.001032 0.002064 0.004388 0.004386 0.004386 0.004386 0.004386 0.0258
-31 0.000129 0.000172 0.000344 0.000731 0.000731 0.000731 0.000731 0.000731 0.0043 Sum 0.03 0.04 0.08 0.17 0.17 0.17 0.17 0.17 1
Weighted average detection rate for all 72 level combinations given above
T alkdo wn Condition 1 -- the combination of average signal le vel (-22 dBm /tone) and av erage speech level (-19 dBm). At least 99.5% of the signals sent should be received correctly using derated missed rate. T alkdown Condition 2 -- the w eight 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 T alkdown Result:
Condition 1 (nominal) : 99.956 % ( Spec. >= 99.5% ) Condition 2 (worst 1 %) : 83.784 % ( Spec. >= 93 % ) Condition 3 (global) : 99.005 % ( Spec. >= 99.5%)
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Ro w Near End Hits No on Near Testing Time Far End Speech Hits No on Far Testing Time
Speech Level End Speech Level (dBm) End Speech
(dBm) 1 -10 8 96 2 -16 5 96 -16 5 96 3 -22 3 96 -22 3 96
2.1.1.3 Talk-off Test
Talkoff Test Raw Data:
F AR END SPEECH
-13 dBm^ -16 dBm -19 dBm~ -22 dBm -25 dBm^ -28 dBm^ -31 dBm^
-7 dBm^ 0.0729 0.0677 0.0625 0.0573 0.0521 0.0469 0.0417
-10 dBm 0.0729 0.0677 0.0625 0.0573 0.0521 0.0469 0.0417
-13 dBm~ 0.0651 0.0599 0.0547 0.0495 0.0443 0.0391 0.0339
-16 dBm 0.0573 0.0521 0.0469 0.0417 0.0365 0.0313 0.0260
-19 dBm~ 0.0521 0.0469 0.0417 0.0365 0.0313 0.0260 0.0208
-22 dBm 0.0469 0.0417 0.0365 0.0313 0.0260 0.0208 0.0156
-25 dBm^ 0.0417 0.0365 0.0313 0.0260 0.0208 0.0156 0.0104
-28 dBm^ 0.0365 0.0313 0.0260 0.0208 0.0156 0.0104 0.0052
-31 dBm^ 0.0313 0.0260 0.0208 0.0156 0.0104 0.0052 0
N E A R
E N D
S P E E C H
~ interpolated ^ extrapolated
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-13 -16 -19 -22 -25 -28 -31 Sum
-7 3.07E-5 9.013E-5 0.0002528 0.0005126 0.0008063 0.000915 0.0016925 0.0043
-10 0.0001842 0.0005408 0.001517 0.0030754 0.0048375 0.0054902 0.0101549 0.0258
-13 0.0007076 0.0020771 0.0058271 0.0118127 0.0185813 0.0210885 0.0390058 0.0991
-16 0.0015908 0.0046699 0.0131006 0.0265578 0.041775 0.0474118 0.0876941 0.2228
-19 0.0021134 0.0062042 0.0174048 0.0352832 0.0555 0.0629888 0.1165056 0.296
-22 0.0015908 0.0046699 0.0131006 0.0265578 0.041775 0.0474118 0.0876941 0.2228
-25 0.0007076 0.0020771 0.0058271 0.0118127 0.0185813 0.0210885 0.0390058 0.0991
-28 0.0001842 0.0005408 0.001517 0.0030754 0.0048375 0.0054902 0.0101549 0.0258
-31 3.07E-5 9.013E-5 0.0002528 0.0005126 0.0008063 0.000915 0.0016925 0.0043 Sum 0.00714 0.02096 0.0588 0.1192 0.1875 0.2128 0.3935 1
Weighted average of all 63 combinations
Talkoff Condition 1 : the combination of a ver age near-end le v el (-19 dBm) and the av er age f ar end le v el (-28 dBm). This talkoff rate should be less than 1 in 45 hours T alkoff Condition 2 : the weighted a verage 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 a ver age 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 ( Spec. >= 45 hours) Condition 2 (worst 1 %) : 1 hit per 16.79 hours ( Spec. >= 10 hours) Condition 3 (global) : 1 hit per 35.73 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 w as connected directly to the Tip and Ring on which the far end speech and f ar 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 improv ed on adjusting the impedance match between CPE and SPCS adequately . According to our experience, if the quantity of the echo canceller can e xcess 6 dB, the performance of CAS detector embedded on 93032-M1 will meet the Bellcore specification for talkdo wn. 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 k eys: The power on def ault ke y function to demo D AM record/pla y/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 T est" and last f or 1 second in DAM mode to enter
CAS test mode. During this mode , the number of CAS signal receiv ed from Tip and Ring will be counted and represented on LCD display .
5. Caller ID TypeII Test mode: Press "CAS T est" within 1 second in D AM 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 D AM mode to perf orm FLAH Re-Initialization
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2. MA TRIX KEY DEFINITION:
Key P ad under D AM mode (Po wer on default mode)
ANN1 ON/OFF
PLA YANN2
ANN3
CODE/FF
TIME/ STOP
MEMO/ REV
SPK.Phone ON/OFF
Enc. Select
CAS T esT
Flash Re-lnit.
DEL
12 3A
45 6B
78 9C
0F ED
Key P ad in Speak er Phone mode :
12
54
7
Lout V ol +
0
Lout V ol
-
SPK.Phone OFF
Mute Lout/ Test Mode
Spk V ol +
Spk V ol
-
8
12
3 3
A
45 66B
78 99C
0F ED
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Ke y Pad in Speak er Phone Test mode :
Ke y P ad in CALLER ID Test mode :
Current Loop Att
Set Pre_PGA
Set AD1_PGA
Set LINE_DRV
Set SPK_DRV
Set AD2_PGA
Set ATT1
Set AEC_ERL
Exit Test Mode
Target Loop Att +
Target Loop Att -
Set T/R Ratio
Set R/T Ratio
Set LEC_ERL
12
3
A
45 6B
78 9C
0F ED
Forward Caller ID Data
STOP
Backward Caller ID Data
CALLER TYPE II/CAS TEST
R/C
DUMP CID
12
3
A
45 6B
78 9C
0F ED
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Key Pad Definitions under D AM/Caller ID mode:
NO FUNCTION PRESS METHOD 1 OGM1 PLAY Press within 2 sec
OGM1 RECORD AND PLA Y Press over 2 sec 2 Turn on DAM Press 3 Raw/Cook Data Select Press within 2 sec
Dump Caller ID Data Press over 2 sec 4 OGM2 PLAY Press within 2 sec
OGM2 RECORD AND PLA Y Press over 2 sec 5 PLAY ALL MESSAGE Press within 2 sec
PLAY NEW MESSAGE Press over 2 sec 6 7 8 Delete Current Message played Press 9 0 Sho w And Announce Pass word Press within 2 sec
Set Password Press over 2 sec
Forward (for playing message and setting password) Press A Flash Re-initialization Press B CAS T est Press C Enc. Selection Press E Record Memo Press over 2 sec
Rewind (for playing message and setting password) Press F Announce Time Stamp Press
Stop Press
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Key Pad Definitions under Speaker Phone Mode :
NO FUNCTION PRESS METHOD 1 GENERATE DTMF "1" Press 2 GENERATE DTMF "2" Press 3 GENERATE DTMF "3" Press 4 GENERATE DTMF "4" Press 5 GENERATE DTMF "5" Press 6 GENERATE DTMF "6" Press 7 GENERATE DTMF "7" Press 8 GENERATE DTMF "8" Press 9 GENERATE DTMF "9" Press 0 Increase Line Out V olume Press A Increase Speaker V olume Press B Decrease Speaker V olume Press C Mute Line Out Press within 2 sec
Enter Test Mode Press over 2 sec D T urn Off Speaker Phone Press E Decrease Line Out Volume Press F GENERA TE DTMF "0" Press
Key Pad Definitions under Speaker Phone Mode (T est Mode) :
NO FUNCTION PRESS METHOD 1 Show Current Loop Attenuation Press 2 SET LINE_DR V (MX93002 CODEC) Press 3 SET T/R RA TIO Press 4 5 SET AD2_PGA (MX93002 CODEC) Press 6 SET R/T RATIO Press 7 SET PRG_PGA (MX93002 CODEC) Press 8 SET_A TT1(MX93002 CODEC) Press 9 SEL LEC_ERL Press 0 SET AD1_PGA Press A TARGET LOOP ATTENUA TION UP Press B T ARGET LOOP ATTENUA TION DOWN Press C EXIT TEST MODE Press D E SET AEC_ERL Press F SET SPK_DRV(MX93002 CODEC) Press
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ERL_AEC & ERL_LEC : Range from 0 to 30 with 2 dB/step
Speaker Phone related BIOS parameters
0000 00001 0010 0011 0100 0101 0110 0111 0 dB 2 dB 4 dB 6 dB 8 dB 10 dB 12 dB 14 dB
1000 1001 1010 1011 1100 1101 1110 1111 16 dB 18 dB 20 dB 22 dB 24 dB 26 dB 28 dB 30 dB
Note: The def ault value of ERL_LEC and ERL_AEC are 0 dB.
T/R & R/T Ratio
000 001 010 011 100 101 110 111 3 dB 6 dB 9 dB 12 dB 15 dB Reserved Reserved Reserved
Note : 12 dB is default value
SPK_GAIN & LINE_GAIN : Range from -20 to 20
0000 00001 0010 0011 0100 0101 0110 0111
-20dB -16 dB -12 dB -8 dB -4 dB 0 dB 2 dB 4 dB
1000 1001 1010 1011 1100 1101 1110 1111 6 dB 8 dB 10 dB 12 dB 14 dB 16 dB 18 dB 20 dB
Loop Attenuation (CONF) : Range from 0 to 28 with 4dB/Step
000 001 010 011 100 101 110 111
0.dB 4 dB 8 dB 12 dB 16 dB 20 dB 24 dB 28 dB
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MX93032
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