TLV320AIC1110
SLAS359 – DECEMBER 2001
PCM CODEC
Capable of Driving 32 Ω Down to a 8-Ω
FEATURES
2.7-V to 3.3-V Operation
Designed for Analog and Digital Wireless
Speaker
Handsets and Telecommunications
Applications
Two Differential Microphone Inputs
Differential Earphone Outputs and One
Single-Ended Earphone Output
Earphone and Microphone Mute
Programmable Transmit, Receive, and
Sidetone Paths With Extended Gain and
Attenuation Ranges
Programmable for 15-Bit Linear Data or 8-Bit
Companded (µ-law and A-law) Mode
Supports PCM Clock Rates of 128 kHz and
2.048 MHz
Pulse Density Modulated (PDM) Buzzer
Output
On-Chip I2C Bus, Which Provides Simple,
Standard, Two-Wire Serial Interface With
Digital ICs
Dual-Tone Multifrequency (DTMF) and
Single-Tone Generator Capable of up to 8-kHz
Tone With Three Selectable Resolutions of
7.8125 Hz, 15.625 Hz, and 31.25 Hz
2-Channel Auxiliary Multiplexer (MUX) (Analog
Switch)
Programmable Power Down Modes
Pin Compatible to the TLV320AIC1103 and
TLV320AIC1109 Devices for TQFP Only
Available in a 32-Pin Thin Quad Flatpack
(TQFP) Package and MicroStar Junior BGA
APPLICATIONS
Digital Handset
Digital Headset
Cordless Phones
Digital PABX
Digital Voice Recording
DESCRIPTION
The TLV320AIC1110 provides extended gain and
attenuation flexibility for transmit, receive, and sidetone
paths. A differential earphone output is capable of
driving speaker loads as low as 8 Ω for use in speaker
phone applications. The single tone function on the
TLV320AIC1110 generates a single tone output of up to
8 kHz. The resolution of the DTMF tone is also
selectable to 7.8125 Hz, 15.625 Hz, or 31.25 Hz through
the interface control. The analog switch provides more
control capabilities for voice-band audio processor
(PCM codec).
This device contains circuits to protect its inputs and outputs against damage due to high static voltages or electrostatic fields. These
circuits have been qualified to protect this device against electrostatic discharges (ESD) of up to 2 kV according to MIL-STD-883C,
Method 3015; however, it is advised that precautions be taken to avoid application of any voltage higher than maximum-rated
voltages to these high-impedance circuits. During storage or handling, the device leads should be shorted together or the device
should be placed in conductive foam. In a circuit, unused inputs should always be connected to an appropriated logic voltage level,
preferably either VCC or ground. Specific guidelines for handling devices of this type are contained in the publication Guidelines for
Handling Electrostatic-Discharge-Sensitive (ESDS) Devices and Assemblies available from Texas Instruments.
Please be aware that an important notice concerning availability, standard warranty, and use in critical applications of
Texas Instruments semiconductor products and disclaimers thereto appears at the end of this data sheet.
†These options are available on some devices. Please see the table of comparison for the last two generations of PCM codecs.
MicroStar Junior is a trademark of Texas Instruments.
All other trademarks are the property of their respective owners.
Copyright 2001, Texas Instruments Incorporated
PRODUCTION DATA information is current as of publication date.
Products conform to specifications per the terms of Texas Instruments
standard warranty. Production processing does not necessarily include
testing of all parameters.
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1
TLV320AIC1110
SLAS359 – DECEMBER 2001
DESCRIPTION (Continued)
The PCM codec is an analog-digital interface for voice band signals designed with a combination of coders and
decoders (codecs) and filters. It is a low-power device with companding options and programming features, and
it meets the requirements for communication systems, including the cellular phone. The device operates in
either the 15-bit linear or 8-bit companded (µ-law or A-Law) mode, which is selectable through the I2C interface.
A coder, an analog-to-digital converter or ADC, digitizes the analog voice signal, and a decoder, a
digital-to-analog converter or DAC, converts the digital-voice signal to an analog output. The PCM codec
provides a companding option to overcome the bandwidth limitations of telephone networks without degrading
the sound quality. The human auditory system is a logarithmic system in which high amplitude signals require
less resolution than low amplitude signals. Therefore, an 8-bit code word with nonuniform quantization (µ-law
or A-law) has the same quality as 13-bit linear coding. The PCM codec provides better digital code words by
generating a 15-bit linear coding option.
The human voice is effective from a frequency range of 300 Hz to 3300 Hz in telephony applications. In order
to eliminate unwanted signals, the PCM codec design has two types of filters that operate in both the transmit
and receive path. A low-pass filter attenuates the signals over 4 kHz. A selectable high-pass filter cleans up the
signals under 100 Hz. This reduces noise that may have coupled in from 50/60-Hz power cables. The high-pass
filter is bypassed by selecting the corresponding register bit.
The PCM codec has many programming features that are controlled using a 2-wire standard serial I2C interface.
This allows the device to interface with many digital ICs such as a DSP or a microprocessor. The device has
seven registers: power control, mode control, transmit PGA, receive PGA, high DTMF, low DTMF, and auxiliary
mode control. Some of the programmable features that can be controlled by I2C interface include:
Transmit amplifier gain
Receive amplifier gain
Sidetone gain
Volume control
Earphone control
PLL power control
Microphone selection
Transmit channel high-pass filter control
Receive channel high-pass filter control
Companding options and selection control
PCM loopback
DTMF control
Pulse density modulated control
The PCM codec is also capable of generating its own internal clocks from a 2.048-MHz master clock input.
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TLV320AIC1110
SLAS359 – DECEMBER 2001
PLLVSS
VSS
MCLK
RESET
PWRUPSEL
BUZZCON
PCMSYN
PCMCLK
PBS PACKAGE
(TOP VIEW)
24 23 22 21 20 19 18 17
25
16
26
15
14
13
PCMO
PCMI
DVSS
DVDD
12 SCL
11 SDA
10 MUXOUT2
9 MUXOUT1
27
28
29
30
31
32
1
2
3
4 5 6 7
8
MBIAS
MIC1P
MIC1N
MIC2P
MIC2N
REXT
MUXIN
AVSS
PLLVDD
EARVSS
EAR1ON
EARVDD
EAR1OP
EARVSS
EAR2O
AVDD
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3
TLV320AIC1110
SLAS359 – DECEMBER 2001
EAR1OP
EARVDD
EAR1ON
EARVSS
PLLVDD
PLLVSS
2
3
4
5
6
7
8
9
B
NC
NC
NC
NC
NC
NC
NC
VSS
C
NC
NC
NC
NC
NC
NC
MCLK
D
NC
NC
NC
NC
NC
NC
NC
RESET
E
NC
NC
NC
NC
NC
NC
NC
PWRUPSEL
F
NC
NC
NC
NC
NC
NC
NC
G
NC
NC
NC
NC
NC
NC
NC
H
NC
NC
NC
NC
NC
NC
NC
EAR2O
EARVSS
MicroStar Junior (GQE) PACKAGE
(TOP VIEW)
1
REXT
MUXIN
J
4
BUZZCON
PCMSYN
PCMCLK
PCMO
MUXOUT1
AVSS
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PCMI
MIC2N
DVSS
MIC2P
DVDD
MIC1N
SCL
MIC1P
SDA
MBIAS
A
MUXOUT2
AVDD
PCMOUT
PCMSYN
PCMCLK
RX Vol
Control
g = –18 dB
to
0 dB
MIC1P
PCM
Interface
MIC1N
MIC
Amplifier
2
g=6
or
18 dB
MIC
Amplifier
1
g=
23.5 dB
Analog
Modulator
Voice
0 dB
or
6 dB
RX Filter
and PGA
g = – 6 dB
to
+6 dB
TX Filter
and PGA
g = –10 dB
to
0 dB
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Sidetone
g = –24 db
to
–12 dB
MIC2P
Ear
Amp1
EAR1OP
EAR1ON
Digital
Modulator
and Filter
DTMF
GAIN
MUX
DTMF
–12 to
12 dB
in 6dB
Steps
MIC2N
functional block diagram
PCMIN
Ear
Amp2
EAR2O
Control Bus
DTMF
OUT
IN
Buzzer
Control
MUX
OUT
BUZZCON
Generator
2
I C
I/F
REF
PLL
Power and RESET
TLV320AIC1110
SLAS359 – DECEMBER 2001
PWRUPSEL
V SS
AVDD
AV SS
DVDD
DV SS
PLLVDD
PLLV SS
EARVDD
EARV SS
RESET
MCLK
REXT
MBIAS
SDATA
SCLK
5
TLV320AIC1110
SLAS359 – DECEMBER 2001
detailed description
power on/reset
The power for the various digital and analog circuits is separated to improve the noise performance of the
device. An external reset must be applied to the active low RESET terminal to assure reset upon power on and
to bring the device to an operational state. After the initial power-on sequence, the device can be functionally
powered up and powered down by writing to the power control register through the I2C interface. The device
has a pin-selectable power up in the default mode option. The hardwired pin-selectable PWRUPSEL function
allows the PCM codec to power up in the default mode and to be used without a microcontroller.
reference
A precision band gap reference voltage is generated internally and supplies all required voltage references to
operate the transmit and receive channels. The reference system also supplies bias voltage for use with an
electret microphone at terminal MBIAS. An external precision resistor is required for reference current setting
at terminal REXT.
I2C control interface
The I2C interface is a two-wire bidirectional serial interface. The I2C interface controls the PCM codec by writing
data to seven control registers:
Power control
Mode control
Transmit PGA and sidetone control
Receive PGA gain and volume control
DTMF routing
Tone selection control
Auxiliary control
There are two power-up modes which may be selected at the PWRUPSEL terminal: (1) The PWRUPSEL state
(VDD at terminal 20) causes the device to power up in the default mode when power is applied. Without an I2C
interface or controlling device, the programmable functions are fixed at the default gain levels, and functions
such as the sidetone and DTMF are not accessible. (2) The PWRUPSEL state (ground at terminal 20) causes
the device to go to a power-down state when power is applied. In this mode, an I2C interface is required to power
up the device.
phase-locked loop (PLL)
The phase-lock loop generates the internal clock frequency required for digital filters and modulators by phase
locking to 2.048-MHz master clock input.
PCM interface
The PCM interface transmits and receives data at the PCMO and PCMI terminals respectively. The data is
transmitted or received at the PCMCLK speed once every PCMSYN cycle. The PCMCLK can be tied directly
to the 128-kHz or 2.048-MHz master clock (MCLK). The PCMSYN can be driven by an external source or
derived from the master clock and used as an interrupt to the host controller.
microphone amplifiers
The microphone input is a switchable interface for two differential microphone inputs. The first stage is a
low-noise differential amplifier that provides a gain of 23.5 dB. The second-stage amplifier has a selectable gain
of 6 dB or 18 dB.
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TLV320AIC1110
SLAS359 – DECEMBER 2001
detailed description (continued)
analog modulator
The transmit channel modulator is a third-order sigma-delta design.
transmit filter and PGA
The transmit filter is a digital filter designed to meet CCITT G.714 requirements. The device operates either in
the 15-bit linear or 8-bit companded µ-law or in the A-law mode, which is selectable through the I2C interface.
The transmit PGA defaults to 0 dB.
sidetone
A portion of the transmitted audio is attenuated and fed back to the receive channel through the sidetone path.
The sidetone path defaults to the mute condition. The default gain of -12 dB is set in the sidetone control register.
The sidetone path can be enabled by writing to the power control register.
receive volume control
The receive volume control block acts as an attenuator with a range of –18 dB to 0 dB in 2-dB steps for control
of the receive channel volume. The receive volume control gain defaults to 0 dB.
receive filter and PGA
The receive filter is a digital filter that meets CCITT G.714 requirements with a high-pass filter that is selectable
through the I2C interface. The device operates either in the 15-bit linear or the 8-bit µ-law or the A-law
companded mode, which is selectable through the I2C interface. The gain defaults to –4 dB, representing a
3-dBm level for a 32-Ω load impedance and the corresponding digital full scale PCMI code.
digital modulator and filter
The second-order digital modulator and filter convert the received digital PCM data to the analog output required
by the earphone interface.
earphone amplifiers
The analog signal can be routed to either of two earphone amplifiers, one with differential output (EAR1ON and
EAR1OP) and one with single-ended output (EAR2O). Clicks and pops are suppressed for EAR1 differential
output only.
tone generator
The tone generator provides generation of standard DTMF tones which are output to (1) the buzzer driver, as
a PDM signal, (2) the receive path DAC for outputting through the earphone, or (3) as PCMO data. The integer
value is loaded into one of two 8-bit registers, the high-tone register (04), or the low-tone register (05) (see the
Register Map Addressing section). The tone output is 2 dB higher when applied to the high tone register (04).
The high DTMF tones must be applied to the high-tone register, and the low DTMF tones to the low-tone register.
The tone signals can be generated with three different resolutions at ∆F= 7.8125 Hz, 15.625 Hz, and 31.250 Hz.
The resolution option can be selected by setting the register (06).
analog mux
The analog switch can be used to source an analog signal to two different loads. The output can be reselected
by setting the auxiliary register (06).
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7
TLV320AIC1110
SLAS359 – DECEMBER 2001
detailed description (continued)
DTMF gain MUX
The DTMF gain MUX selects the signal path and applies the appropriate gain setting. Therefore the device is
either in tone mode or in voice mode. When set in the voice mode, the gain is controlled by the auxiliary register
and is set to 0 dB or 6 dB. When set in the tone mode, the gain is from –12 dB to 12 dB in 6-dB steps which
is set by the volume control register. The gain setting is controlled by the RXPGA register. This will not create
any control contention since the device is working in one mode at a time.
Terminal Functions
TERMINAL†
NAME
NO.
I/O
DESCRIPTION
µBGA
TQFP
AVDD
AVSS
A1
32
I
Analog positive power supply
J1
8
I
Analog negative power supply (use for ground connection)
BUZZCON
F9
19
O
Buzzer output, a pulse-density modulated signal to apply to external buzzer driver
DVDD
J6
13
I
Digital positive power supply
DVSS
J7
14
I
Digital negative power supply
EAR1ON
A6
27
O
Earphone 1 amplifier output (–)
EAR1OP
A4
29
O
Earphone 1 amplifier output (+)
EAR2O
A2
31
O
Earphone 2 amplifier output
EARVDD
EARVSS
A5
28
I
Analog positive power supply for the earphone amplifiers
A3, A7
30, 26
I
Analog negative power supply for the earphone amplifiers
MBIAS
B1
1
O
Microphone bias supply output, no decoupling capacitors
MCLK
C9
22
I
Master system clock input (2.048 MHz, digital)
MIC1P
C1
2
I
MIC1 input (+)
MIC1N
D1
3
I
MIC1 input (–)
MIC2P
E1
4
I
MIC2 input (+)
MIC2N
F1
5
I
MIC2 input (–)
MUXIN
H1
7
I
Analog MUX input
MUXOUT1
J2
9
I
Analog MUX output
MUXOUT2
J3
10
I
Analog MUX output
PCMI
J8
15
I
Receive PCM input
PCMO
J9
16
O
Transmit PCM output
PCMSYN
G9
18
I
PCM frame sync
PCMCLK
H9
17
I
PCM data clock
PLLVSS
PLLVDD
A9
24
I
PLL negative power supply
A8
25
I
PLL digital power supply
PWRUPSEL
E9
20
I
Selects the power-up default mode
REXT
G1
6
I/O
Internal reference current setting terminal (use precision 100-kΩ resistor and no filtering capacitors)
RESET
D9
21
I
SCL
J5
12
I
Active low reset
I2C-bus serial clock. This input is used to synchronize the data transfer from and to the PCM codec.
SDA
J4
11
I/O
I2C-bus serial address/data input/output. This is a bidirectional terminal used to transfer register
control addresses and data into and out of the codec. It is an open-drain terminal and therefore
requires a pullup resistor to VDD (typical 10 kΩ for 100 kHz).
VSS
B9
23
I
Ground return for bandgap internal reference (use for ground connection)
† All MicroStar Junior BGA pins that are not mentioned have no internal connection.
8
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TLV320AIC1110
SLAS359 – DECEMBER 2001
absolute maximum ratings over operating free-air temperature range (unless otherwise noted)†
Supply voltage range, AVDD, DVDD, PLLVDD, EARVDD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . –0.5 V to 3.6 V
Output voltage range, VO . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . –0.5 V to 3.6 V
Input voltage range, VI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . –0.5 V to 3.6 V
Continuous total power dissipation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . See Dissipation Rating Table
Operating free air temperature range (industrial temperature) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . – 40°C to 85°C
Storage temperature range, testing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . – 65°C to 150°C
Lead temperature 1,6 mm from case for 10 seconds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 260°C
† Stresses beyond those listed under “absolute maximum ratings” may cause permanent damage to the device. These are stress ratings only, and
functional operation of the device at these or any other conditions beyond those indicated under “recommended operating conditions” is not
implied. Exposure to absolute-maximum-rated conditions for extended periods may affect device reliability.
DISSIPATION RATING TABLE
PACKAGE
TA ≤ 25°C
POWER RATING
DERATING FACTOR
ABOVE TA = 25°C
TA = 85°C
POWER RATING
TQFP
702 mW
7.2 mW/°C
270 mW
Low dissipation printed
circuit board (PCB)
MicroStar Junior BGA
660 mW
164 mW/°C
220 mW
Low dissipation PCB
MicroStar Junior BGA
2.75 W
36 mW/°C
917 mW
High dissipation PCB
COMMENTS
recommended operating conditions (see Notes 1 and 2)
MIN
Supply voltage, AVDD, DVDD, PLLVDD, EARVDD
NOM
2.7
High-level input voltage, VIH
MAX
UNIT
3.3
V
0.7 x VDD
V
Low-level input voltage, VIL
0.3 x VDD
Load impedance between EAR1OP and EAR1ON-RL
Load impedance for EAR2OP-RL
V
8 to 32
Ω
32
Ω
Operating free-air temperature, TA
– 40
85
C
NOTES: 1. To avoid possible damage and resulting reliability problems to these CMOS devices, follow the power-on initialization paragraph,
described in the Principles of Operation.
2. Voltages are with respect to AVSS, DVSS, PLLVSS , and EARVSS.
electrical characteristics over recommended ranges of supply voltage and free-air temperature (unless
otherwise noted)
supply current
PARAMETER
IDD
Supply current from VDD
TEST CONDITIONS
TYP
MAX
Operating, EAR1 selected, MicBias disabled
4.5
6
mA
Operating, EAR2 selected, MicBias disabled
4.5
6
mA
2
10
µA
10
30
µA
5
10
ms
Power down room temperature, VDD = 3 V, Reg 6 bit 7 = 1, MClk
not present (see Note 3)
Power down room temperature, VDD = 3 V, , Reg 6 bit 7 = 0,
MClk not present (see Note 3)
ton(i)
Power-up time from
power down
MIN
UNIT
NOTE 3: VIHMIN = VDD, VILMAX = VSS.
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9
TLV320AIC1110
SLAS359 – DECEMBER 2001
electrical characteristics over recommended ranges of supply voltage and free-air temperature (unless
otherwise noted) (continued)
digital interface
PARAMETER
TEST CONDITIONS
MAX
UNIT
High-level output voltage PCMO (BUZZCON)
IIH
IIL
High-level input current, any digital input
Ci
Input capacitance
Co
Output capacitance
20
pF
RL
Load impedance (BUZZCON)
5
kΩ
Low-level input current, any digital input
VDD = 3 V
VDD = 3 V
TYP
VOH
VOL
Low-level output voltage PCMO
IOH = – 3.2 mA,
IOL = 3.2 mA,
MIN
DVDD– 0.25
V
VI = VDD
VI = VSS
0.25
V
10
µA
10
µA
10
pF
microphone interface
PARAMETER
TEST CONDITIONS
VIO
IIB
Input offset voltage at MIC1N, MIC2N
Ci
Input capacitance at MIC1N, MIC2N
Vn
Microphone input referred noise, psophometrically weighted,
(C-message weighted is similar)
IOmax
V(mbias)
See Note 4
Input bias current at MIC1N, MIC2N
MIN
TYP
–5
– 300
MAX
UNIT
5
mV
300
nA
5
MIC amp 1 gain = 23.5 dB
MIC amp 2 gain = 0 dB
Output source current—MBIAS
3
4.7
µVrms
1.2
mA
2.5
2.65
V
60
100
kΩ
1
Microphone bias supply voltage (see Note 5)
2.3
MICMUTE
pF
– 80
Input impedance
Fully differential
35
dB
NOTES: 4. Measured while MIC1P and MIC1N are connected together. Less than 0.5-mV offset results in 0 value code on PCMOUT.
5. Not a JEDEC symbol.
10
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TLV320AIC1110
SLAS359 – DECEMBER 2001
electrical characteristics over recommended ranges of supply voltage and free-air temperature (unless
otherwise noted) (continued)
speaker interface
PARAMETER
TEST CONDITIONS
TYP
MAX
VDD = 2.7 V, fully differential, 8-Ω load,
3-dBm0 output, volume control = – 3 dB,
RXPGA = – 4 dB level
161
200
VDD = 2.7 V, fully differential, 16-Ω load,
3-dBm0 output, volume control = – 3 dB,
RXPGA = – 2 dB level
128
160
VDD = 2.7 V, fully differential, 32-Ω load,
3-dBm0 output, volume control = – 3 dB,
RXPGA = – 1 dB level
81
100
Earphone AMP2 output power (see Note 6)
VDD = 2.7 V, single-ended, 32-Ω load,
3-dBm0 output
10
12.5
mW
Output offset voltage at EAR1
Fully differential
mV
Earphone AMP1 output power (see Note 6)
VOO
IOmax
Maximum out
output
ut current for EAR1 (rms)
Maximum output current for EAR2 (rms)
MIN
±5
± 30
3-dBm0 input, 8-Ω load
141
178
3-dBm0 input, 16-Ω load
90
112
3-dBm0 input, 32-Ω load
50
63
17.7
22.1
3-dBm0 input
EARMUTE
– 80
UNIT
mW
mA
dB
NOTE 6: Maximum power is with a load impedance of –25%.
transmit gain and dynamic range, companded mode (µ-law or A-law) or linear mode selected, transmit slope
filter bypassed (see Notes 7 and 8)
PARAMETER
Transmit reference-signal level (0 dB)
Overload signal level (3 dBm0)
Overload-signal
Absolute gain error
TEST CONDITIONS
MAX
UNIT
Differential
87.5
mVpp
Differential, normal mode
124
Differential, extended mode
31.5
0-dBm0 input signal, VDD ±10%
MIC1N, MIC1P to PCMO at 3 dBm0 to –30 dBm0
G i error with
ith iinputt llevell relative
l ti tto gain
i att
Gain
–10 dBm0 MIC1N,
MIC1N MIC1P to PCMO
MIN
TYP
–1
1
– 0.5
0.5
MIC1N, MIC1P to PCMO at –31 dBm0 to –45 dBm0
–1
1
MIC1N, MIC1P to PCMO at –46 dBm0 to –55 dBm0
–1.2
1.2
mVpp
dB
dB
NOTES: 7. Unless otherwise noted, the analog input is 0 dB, 1020-Hz sine wave, where 0 dB is defined as the zero-reference point of the channel
under test.
8. The reference signal level, which is input to the transmit channel, is defined as a value 3 dB below the full-scale value of 88-mVrms.
transmit gain and dynamic range, companded mode (µ-law or A-law) or linear mode selected, transmit slope
filter enabled (see Notes 7 and 8)
PARAMETER
Transmit reference-signal level (0 dB)
Overload signal level (3 dBm0)
Overload-signal
Absolute gain error
MAX
UNIT
Differential
TEST CONDITIONS
87.5
mVpp
Differential, normal mode
124
Differential, extended mode
31.5
0-dBm0 input signal, VDD ±10%
MIC1N, MIC1P to PCMO at 3 dBm0 to – 30 dBm0
Gain
G
i error with
ith iinputt llevell relative
l ti tto gain
i att
–10 dBm0 MIC1N,
MIC1N MIC1P to PCMO
MIN
TYP
–1
1
– 0.5
0.5
MIC1N, MIC1P to PCMO at – 31 dBm0 to – 45 dBm0
–1
1
MIC1N, MIC1P to PCMO at – 46 dBm0 to – 55 dBm0
–1.2
1.2
mVpp
dB
dB
NOTES: 7. Unless otherwise noted, the analog input is 0 dB, 1020-Hz sine wave, where 0 dB is defined as the zero-reference point of the
channel under test.
8 The reference signal level, which is input to the transmit channel, is defined as a value 3 dB below the full-scale value of 88-mVrms.
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11
TLV320AIC1110
SLAS359 – DECEMBER 2001
electrical characteristics over recommended ranges of supply voltage and free-air temperature (unless
otherwise noted) (continued)
transmit filter transfer, companded mode (µ-law or A-law) or linear mode selected, transmit slope filter
bypassed (MCLK = 2.048 MHz)
PARAMETER
Gain
G
i relative
l ti tto iinputt signal
i
l gain
i att 1020 H
Hz, iinternal
t
l hi
high-pass
h
filter disabled
Gain relative to in
input
ut signal gain at 1020 Hz, internal high
high-pass
ass
filter enabled
TEST CONDITIONS
MIN
TYP
MAX
fMIC1 or fMIC2