Datasheet TLV320AIC23B-Q1 Datasheet (Texas Instruments)

TLV320AIC23B-Q1
Stereo Audio Codec, 8- to 96-kHz, With Integrated Headphone Amplifier
Data Manual
PRODUCTION DATA information is current as of publication date. Products conform to specifications per the terms of the Texas Instruments standard warranty. Production processing does not necessarily include testing of all parameters.
March 2004–Revised June 2012
TLV320AIC23B-Q1
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Stereo Audio Codec, 8- to 96-kHz, With Integrated Headphone
Amplifier
Check for Samples: TLV320AIC23B-Q1

1 1 Introduction

1.1 Features

1
• Qualified for Automotive Applications • Integrated Total Electret-Microphone Biasing
• High-Performance Stereo Codec – 90-dB SNR Multibit Sigma-Delta ADC (A-
weighted at 48 kHz)
– 100-dB SNR Multibit Sigma-Delta DAC (A-
weighted at 48 kHz)
– 1.42 V – 3.6 V Core Digital Supply:
Compatible With TI C54x DSP Core Voltages
– 2.7 V – 3.6 V Buffer and Analog Supply:
Compatible Both TI C54x DSP Buffer – Integrated Programmable Gain Amplifier Voltages
– 8-kHz – 96-kHz Sampling-Frequency Support
• Software Control Via TI McBSP-Compatible and Microphone Multiprotocol Serial Port
– 2-wire-Compatible and SPI-Compatible
Serial-Port Protocols
– Glueless Interface to TI McBSPs
• Audio-Data Input/Output Via TI McBSP­Compatible Programmable Audio Interface
– I2S-Compatible Interface Requiring Only One
McBSP for both ADC and DAC
– Standard I2S, MSB, or LSB Justified-Data
Transfers – 16/20/24/32-Bit Word Lengths – Audio Master/Slave Timing Capability
Optimized for TI DSPs (250/272 fs), USB
mode – Industry-Standard Master/Slave Support
Provided Also (256/384 fs), Normal mode – Glueless Interface to TI McBSPs
and Buffering Solution – Low-Noise MICBIAS pin at 3/4 AVDD for
Biasing of Electret Capsules
– Integrated Buffer Amplifier With Tunable
Fixed Gain of 1 to 5
– Additional Control-Register Selectable Buffer
Gain of 0 dB or 20 dB
• Stereo-Line Inputs
– Analog Bypass Path of Codec
• ADC Multiplexed Input for Stereo-Line Inputs
• Stereo-Line Outputs – Analog Stereo Mixer for DAC and Analog
Bypass Path
• Volume Control With Mute on Input and Output
• Highly Efficient Linear Headphone Amplifier – 30 mW into 32 Ω From a 3.3-V Analog Supply
Voltage
• Flexible Power Management Under Total Software Control
– 23-mW Power Consumption During Playback
Mode – Standby Power Consumption < 150 µW – Power-Down Power Consumption < 15 µW
• 28-Pin TSSOP (62 mm2Total Board Area)
• Ideally Suitable for Portable Solid-State Audio Players and Recorders

1.2 Description

The TLV320AIC23B-Q1 is a high-performance stereo audio codec with highly integrated analog functionality. The analog-to-digital converters (ADCs) and digital-to-analog converters (DACs) within the TLV320AIC23B-Q1 use multibit sigma-delta technology with integrated oversampling digital interpolation filters. Data-transfer word lengths of 16, 20, 24, and 32 bits, with sample rates from 8 kHz to 96 kHz, are supported. The ADC sigma-delta modulator features third-order multibit architecture with up to 90-dBA signal-to-noise ratio (SNR) at audio sampling rates up to 96 kHz, enabling high-fidelity audio recording in a compact, power-saving design. The DAC sigma-delta modulator features a second-order multibit architecture with up to 100-dBA SNR at audio sampling rates up to 96 kHz, enabling high-quality digital audio-playback capability, while consuming less than 23 mW during playback only. The TLV320AIC23B­Q1 is the ideal analog input/output (I/O) choice for portable digital audio-player and recorder applications,
1
Copyright © 2004–2012, Texas Instruments Incorporated
such as MP3 digital audio players.
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.
PRODUCTION DATA information is current as of publication date. Products conform to specifications per the terms of the Texas Instruments standard warranty. Production processing does not necessarily include testing of all parameters.
TLV320AIC23B-Q1
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Integrated analog features consist of stereo-line inputs with an analog bypass path, a stereo headphone amplifier, with analog volume control and mute, and a complete electret-microphone-capsule biasing and buffering solution. The headphone amplifier is capable of delivering 30 mW per channel into 32 Ω. The analog bypass path allows use of the stereo-line inputs and the headphone amplifier with analog volume control, while completely bypassing the codec, thus enabling further design flexibility, such as integrated FM tuners. A microphone bias-voltage output provides a low-noise current source for electret-capsule biasing. The AIC23B has an integrated adjustable microphone amplifier (gain adjustable from 1 to 5) and a programmable gain microphone amplifier (0 dB or 20 dB). The microphone signal can be mixed with the output signals if a sidetone is required.
While the TLV320AIC23B-Q1 supports the industry-standard oversampling rates of 256 fsand 384 fs, unique oversampling rates of 250 fsand 272 fsare provided, which optimize interface considerations in designs using TI C54x digital signal processors (DSPs) and universal serial bus (USB) data interfaces. A single 12-MHz crystal can supply clocking to the DSP, USB, and codec. The TLV320AIC23B-Q1 features an internal oscillator that, when connected to a 12-MHz external crystal, provides a system clock to the DSP and other peripherals at either 12 MHz or 6 MHz, using an internal clock buffer and selectable divider. Audio sample rates of 48 kHz and compact-disc (CD) standard 44.1 kHz are supported directly from a 12-MHz master clock with 250 fsand 272 fsoversampling rates.
Low power consumption and flexible power management allow selective shutdown of codec functions, thus extending battery life in portable applications. This design solution makes powerful portable stereo audio designs easily realizable in a cost-effective, space-saving total analog I/O solution: the TLV320AIC23B-Q1.
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Σ
Σ−∆
DAC
Σ
OSC
SDIN
SCLK
MODE
DVDD
BVDD
DGND
LRCIN
DIN
LRCOUT
DOUT
BCLK
B0486-01
AVDD
VMID
AGND
RLINEIN
LLINEIN
HPVDD
HPGND
RHPOUT
ROUT
LOUT
LHPOUT
XTI/MCLK
XTO
CLKOUT
1.0X
1.0X
VADC
VMID
50 kΩ
VDAC
Σ−∆
ADC
VMID
50 kΩ
10 kΩ
VADC
1.0X
1.5X
VDAC
MICBIAS
MICIN
Sidetone Mute
NOTE: MCLK, BCLK, and SCLK are all asynchronous to each other.
50 kΩ
DSPcodec
TLV320AIC23B
Mute,
0 dB, 20 dB
Bypass Mute
Line
Mute
2:1
MUX
Σ−∆
ADC
12 to 34.5 dB,
1.5-dB Steps
CS
Control
Interface
12 to 34 dB,
1.5-dB Steps
6 to 73 dB,
1-dB Steps
Σ−∆
DAC
Bypass Mute
6 to 73 dB,
1-dB Steps
Headphone
Driver
Headphone
Driver
CLKIN Divider
(1x, x)½
CLKOUT
Divider
(1x, x)½
Digital
Audio
Interface
Digital Filters
2:1
MUX
Line
Mute
TLV320AIC23B-Q1
SGLS240C –MARCH 2004–REVISED JUNE 2012

1.3 Functional Block Diagram

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1
2
3
4
5
6
7
11
8
12
9
13
10
14
21
28
27
26
25
24
23
22
18
17
20
16
19
15
BVDD
CLKOUT
BCLK
DIN
LRCIN
DOUT
LRCOUT
HPGND
HPVDD
LOUT
LHPOUT
ROUT
RHPOUT
AVDD
DGND
DVDD
XTO
XTI/MCLK
SCLK
SDIN
MODE
MICIN
CS
MICBIAS
LLINEIN
VMID
RLINEIN
AGND
P0043-05
PW Package
(Top View)
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1.4 Pin Assignments

1.5 Ordering Information

T
A
40°C to 85°C TSSOP 2000 TLV320AIC23BIPWRQ1 AIC23BIQ1
SGLS240C –MARCH 2004–REVISED JUNE 2012
PACKAGE REEL ORDERABLE PART NUMBER TOP-SIDE MARKING
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1.6 Pin Functions

Table 1-1. Pin Functions
PIN
NAME NO.
AGND 15 Analog supply return AVDD 14 Analog supply input. Voltage level is 3.3 V nominal.
BCLK 3 I/O BVDD 1 Buffer supply input. Voltage range is from 2.7 V to 3.6 V. CLKOUT 2 O
CS 21 I For 2-wire control mode this input defines the seventh bit in the device address field. See Section 3.1 for
DIN 4 I I2S format serial data input to the sigma-delta stereo DAC DGND 28 Digital supply return DOUT 6 O I2S format serial data output from the sigma-delta stereo ADC DVDD 27 Digital supply input. Voltage range is 1.4 V to 3.6 V. HPGND 11 Analog headphone amplifier supply return HPVDD 8 Analog headphone amplifier supply input. Voltage level is 3.3 V nominal.
LHPOUT 9 O
LLINEIN 20 I LOUT 12 O Left stereo mixer-channel line output. Nominal output level is 1.0 VRMS. LRCIN 5 I/O
LRCOUT 7 I/O
MICBIAS 17 O
MICIN 18 I MODE 22 I Serial-interface-mode input. See Section 3.1 for details.
NC Not Used—No internal connection RHPOUT 10 O
RLINEIN 19 I ROUT 13 O Right stereo mixer-channel line output. Nominal output level is 1.0 VRMS. SCLK 24 I
SDIN 23 I
VMID 16 I XTI/MCLK 25 I Crystal or external-clock input. Used for derivation of all internal clocks on the AIC23B. XTO 26 O
I/O DESCRIPTION
I2S serial-bit clock. In audio master mode, the AIC23B generates this signal and sends it to the DSP. In audio slave mode, the signal is generated by the DSP.
Clock output. This is a buffered version of the XTI input and is available in 1X or 1/2X frequencies of XTI. Bit 07 in the sample rate control register controls frequency selection.
Control port input latch/address select. For SPI control mode this input acts as the data latch control. details.
Left stereo mixer-channel amplified headphone output. Nominal 0-dB output level is 1 VRMS. Gain of –73 dB to 6 dB is provided in 1-dB steps.
Left stereo-line input channel. Nominal 0-dB input level is 1 VRMS. Gain of –34.5 dB to 12 dB is provided in 1.5-dB steps.
I2S DAC-word clock signal. In audio master mode, the AIC23B generates this framing signal and sends it to the DSP. In audio slave mode, the signal is generated by the DSP.
I2S ADC-word clock signal. In audio master mode, the AIC23B generates this framing signal and sends it to the DSP. In audio slave mode, the signal is generated by the DSP.
Buffered low-noise-voltage output suitable for electret-microphone-capsule biasing. Voltage level is 3/4 AVDD nominal.
Buffered amplifier input suitable for use with electret-microphone capsules. Without external resistors a default gain of 5 is provided. See Section 2.3.1.2 for details.
Right stereo mixer-channel amplified headphone output. Nominal 0-dB output level is 1 VRMS. Gain of
73 dB to 6 dB is provided in 1-dB steps. Right stereo-line input channel. Nominal 0-dB input level is 1 VRMS. Gain of –34.5 dB to 12 dB is
provided in 1.5-dB steps.
Control-port serial-data clock. For SPI and 2-wire control modes this is the serial-clock input. See Section 3.1 for details.
Control-port serial-data input. For SPI and 2-wire control modes this is the serial-data input and also is used to select the control protocol after reset. See Section 3.1 for details.
Midrail voltage decoupling input. 10-μF and 0.1-μF capacitors should be connected in parallel to this terminal for noise filtering. Voltage level is 1/2 AVDD nominal.
Crystal output. Connect to external crystal for applications where the AIC23B is the audio timing master. Not used in applications where external clock source is used.
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2 Electrical Specifications

2.1 Absolute Maximum Ratings

over operating free-air temperature range (unless otherwise noted)
Supply voltage range Analog supply return to digital supply return AGND to DGND –0.3 to 3 .63 V
Input voltage range, all input signals
Case temperature for 10 seconds 240 °C Operating free-air temperature range: Industrial, T Storage temperature range, T
(1) 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.
(2) DVDDmay not exceed BVDD0.3 V; BVDDmay not exceed AVDD0.3 V or HPVDD0.3.
(2)
stg

2.2 Recommended Operating Conditions

over operating free-air temperature range (unless otherwise noted)
DD
DD
(1)
DD
(1)
(1)
A
Analog supply voltage, AVDD, HPV Digital buffer supply voltage, BV Digital core supply voltage, DV Analog input voltage, full scale 0 dB (AVDD= 3.3 V) 1 V Stereo-line output load resistance 10 k Headphone-amplifier output load resistance 0 CLKOUT digital output load capacitance 20 pF All other digital output load capacitance 10 pF Stereo-line output load capacitance 50 pF XTI master clock Input 18.43 MHz ADC or DAC conversion rate 96 kHz Operating free-air temperature, T
(1) Digital voltage values are with respect to DGND; analog voltage values are with respect to AGND.
(1)
VALUE UNIT
AVDDto AGND, DVDDto DGND, BVDDto DGND, HPVDDto HPGND
Digital –0.3 to DV Analog –0.3 to AV
A
–0.3 to 3.63 V
DD DD
–40 to 85 °C
–65 to 150 °C
MIN NOM MAX UNIT
2.7 3.3 3.6 V
2.7 3.3 3.6 V
1.42 1.5 3.6 V
Industrial –40 85 °C
V V
RMS
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2.3 Electrical Characteristics

over recommended operating conditions, AVDD, HPVDD, BVDD= 3.3 V, DVDD= 1.5 V, slave mode, XTI/MCLK = 256 fs, fs= 48 kHz (unless otherwise stated)

2.3.1 ADC

2.3.1.1 Line Input to ADC
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
Input signal level (0 dB) 1 V
Signal-to-noise ratio, A-weighted, 0-dB gain
(1)(2)
Dynamic range, A-weighted, –60-dB full-scale input
Total harmonic distortion, –1-dB input, 0-dB gain dB
Power supply rejection ratio 1 kHz, 100 mV
fs= 48 kHz (3.3 V) 85 90 fs= 48 kHz (2.7 V) 90 AVDD= 3.3 V 85 90
(2)
AVDD= 2.7 V 90 AVDD= 3.3 V –80 AVDD= 2.7 V 80
pp
50 dB ADC channel separation 1 kHz input tone 90 dB Programmable gain 1 kHz input tone, R
< 50 –34.5 12 dB
SOURCE
Programmable gain step size Monotonic 1.5 dB Mute attenuation 0 dB, 1 kHz input tone 80 dB
Input resistance k
12 dB input gain 10 20 0 dB input gain 28 35
Input capacitance 10 pF
(1) Ratio of output level with 1-kHz full-scale input, to the output level with the input short circuited, measured A-weighted over a 20-Hz to
20-kHz bandwidth using an audio analyzer.
(2) All performance measurements done with 20-kHz low-pass filter and, where noted, A-weighted filter. Failure to use such a filter results in
higher THD + N and lower SNR and dynamic range readings than shown in the Electrical Characteristics. The low-pass filter removes out-of-band noise, which, although not audible, may affect dynamic specification values.
RMS
dB
dB
2.3.1.2 Microphone Input to ADC
0-dB Gain, fs= 8 kHz (40-Ksource impedance, see Section 1.3)
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
Input signal level (0 dB) 1 V
Signal-to-noise ratio, A-weighted, 0-dB gain
(1)(2)
Dynamic range, A-weighted, –60-dB full-scale input
Total harmonic distortion, –1-dB input, 0-dB gain dB
Power supply rejection ratio 1 kHz, 100 mV Programmable gain boost 1 kHz input tone, R Microphone-path gain MICBOOST = 0, R
AVDD= 3.3 V 77 85 AVDD= 2.7 V 84 AVDD= 3.3 V 77 85
(2)
AVDD= 2.7 V 84 AVDD= 3.3 V –60 AVDD= 2.7 V –60
pp
< 50 20 dB
SOURCE
< 50 14 dB
SOURCE
50 dB
Mute attenuation 0 dB, 1 kHz input tone 60 80 dB Input resistance 8 14 k Input capacitance 10 pF
(1) Ratio of output level with 1-kHz full-scale input, to the output level with the input short circuited, measured A-weighted over a 20-Hz to
20-kHz bandwidth using an audio analyzer.
(2) All performance measurements done with 20-kHz low-pass filter and, where noted, A-weighted filter. Failure to use such a filter results in
higher THD + N and lower SNR and dynamic range readings than shown in the Electrical Characteristics. The low-pass filter removes out-of-band noise, which, although not audible, may affect dynamic specification values.
RMS
dB
dB
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2.3.1.3 Microphone Bias
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
Bias voltage 3/4 AVDD− 100 m 3/4 AV
3/4 AVDD+ 100 m V
DD
Bias-current source 3 mA Output noise voltage 1 kHz to 20 kHz 25 nV/Hz

2.3.2 DAC

Line output, load = 10 k, 50 pF
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
0-dB full-scale output voltage (FFFFFF) 1 V
Signal-to-noise ratio, A-weighted, 0-dB gain
Dynamic range, A-weighted
(2)
(1)(2)(3)
Total harmonic distortion
Power supply rejection ratio 1 kHz, 100 mV
AVDD= 3.3 V fs= 48kHz 90 100 AVDD= 2.7 V fs= 48 kHz 100 AVDD= 3.3 V 85 90 AVDD= 2.7 V TBD
AVDD= 3.3 V dB
AVDD= 2.7 V dB
1 kHz, 0 dB –88 –80 1 kHz, –3 dB –92 –86 1 kHz, 0 dB –85 1 kHz, –3 dB –88
pp
50 dB
DAC channel separation 100 dB
(1) Ratio of output level with 1-kHz full-scale input, to the output level with the input short circuited, measured A-weighted over a 20-Hz to
20-kHz bandwidth using an audio analyzer
(2) All performance measurements done with 20-kHz low-pass filter and, where noted, A-weighted filter. Failure to use such a filter results in
higher THD + N and lower SNR and dynamic range readings than shown in the Electrical Characteristics. The low-pass filter removes out-of-band noise, which, although not audible, may affect dynamic specification values.
(3) Ratio of output level with 1-kHz full-scale input, to the output level with all zeros into the digital input, measured A-weighted over a 20-Hz
to 20-kHz bandwidth.
RMS
dB
dB

2.3.3 Analog Line Input to Line Output (Bypass)

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
0-dB full-scale output voltage 1 V
AVDD= 3.3 V 90 95
Signal-to-noise ratio, A-weighted, 0-dB gain
Total harmonic distortion
Power supply rejection ratio 1 kHz, 100 mV
(1)(2)
AVDD= 2.7 V 95
AVDD= 3.3 V dB
AVDD= 2.7 V dB
1 kHz, 0 dB –86 –80 1 kHz, –3 dB –92 –86 1 kHz, 0 dB –86 1 kHz, –3 dB –92
pp
50 dB DAC channel separation (left to right) 1 kHz, 0 dB 80 dB
(1) Ratio of output level with 1-kHz full-scale input, to the output level with the input short circuited, measured A-weighted over a 20-Hz to
20-kHz bandwidth using an audio analyzer.
(2) All performance measurements done with 20-kHz low-pass filter and, where noted, A-weighted filter. Failure to use such a filter results in
higher THD + N and lower SNR and dynamic range readings than shown in the Electrical Characteristics. The low-pass filter removes out-of-band noise, which, although not audible, may affect dynamic specification values.
RMS
dB
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2.3.4 Stereo Headphone Output

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
0-dB full-scale output voltage 1 VRMS
Maximum output power, P
Signal-to-noise ratio, A-weighted
O
(1)
Total harmonic distortion %
Power supply rejection ratio 1 kHz, 100 mV
RL = 32 Ω 30 RL = 16 40
mW
AVDD= 3.3 V 88 97 dB AVDD= 3.3 V,
1 kHz output
pp
PO = 10 mW 0.1 PO = 20 mW 1
50 dB Programmable gain 1 kHz output 73 6 dB Programmable-gain step size 1 dB Mute attenuation 1 kHz output 80 dB
(1) All performance measurements done with 20-kHz low-pass filter and, where noted, A-weighted filter. Failure to use such a filter results in
higher THD + N and lower SNR and dynamic range readings than shown in the Electrical Characteristics. The low-pass filter removes out-of-band noise, which, although not audible, may affect dynamic specification values.

2.3.5 Analog Reference Levels

PARAMETER MIN TYP MAX UNIT
Reference voltage AVDD/2 50 mV AVDD/2 + 50 mV V Divider resistance 40 50 60 k

2.3.6 Digital I/O

PARAMETER MIN TYP MAX UNIT
V
IL
V
IH
V
OL
V
OH
Input low level 0.3 × BV Input high level 0.7 × BV
DD
Output low level 0.1 × BV Output high level 0.9 × BV
DD
DD
DD

2.3.7 Supply Current

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
Record and playback (all active) 20 24 26 Record and playback (osc, clk, and MIC output powered down) 16 18 20 Line playback only 6 7.5 9
Total supply current, No input signal I
Record only 11 13.5 15
TOT
Analog bypass (line in to line out) 4 4.5 6 Power down, DVDD= 1.5 V, Oscillator enabled 0.8 1.5 3
AV
DD
= BVDD= HPVDD= 3.3 V
Oscillator disabled 0.01

2.4 Digital-Interface Timing

PARAMETER MIN TYP MAX UNIT
tw(1) High 18 ns tw(2) Low 18 tc(1) System-clock period, MCLK/XTI 54 ns
tpd(1) Propagation delay, CLKOUT 0 10 ns
System-clock pulse duration, MCLK/XTI
Duty cycle, MCLK/XTI 40/60 60/40 %
V V V V
mA
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BCLK
LRCIN
DIN
DOUT
LRCOUT
t
su(1)
t
h(1)
t
pd(2)
t
pd(3)
T0548-01
t
pd(1)
MCLK/XTI
CLKOUT
CLKOUT
(Div 2)
t
c(1)
t
w(1)
t
w(2)
T0547-01
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Figure 2-1. System-Clock Timing Requirements

2.4.1 Audio Interface (Master Mode)

PARAMETER MIN TYP MAX UNIT
tpd(2) Propagation delay, LRCIN/LRCOUT 0 10 ns tpd(3) Propagation delay, DOUT 0 10 ns tsu(1) Setup time, DIN 10 ns th(1) Hold time, DIN 10 ns
Figure 2-2. Master-Mode Timing Requirements

2.4.2 Audio Interface (Slave-Mode)

PARAMETER MIN TYP MAX UNIT
tw(3) High 20 tw(4) Low 20 tc(2) Clock period, BCLK 50 ns tpd(4) Propagation delay, DOUT 0 10 ns tsu(2) Setup time, DIN 10 ns th(2) Hold time, DIN 10 ns tsu(3) Setup time, LRCIN 10 ns th(3) Hold time, LRCIN 10 ns
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Pulse duration, BCLK ns
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LSB
t
w(8)
t
w(5)
t
w(6)
t
c(3)
t
su(5)
t
su(4)
t
h(4)
CS
SCLK
DIN
T0550-01
BCLK
LRCIN
DIN
DOUT
LRCOUT
t
c(2)
t
w(4)
t
w(3)
t
su(2)
t
h(3)
t
h(2)
t
su(3)
t
pd(2)
T0549-01
TLV320AIC23B-Q1
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Figure 2-3. Slave-Mode Timing Requirements

2.4.3 3-Wire Control Interface (SDIN)

PARAMETER MIN TYP MAX UNIT
tw(5) High 20 tw(6) Low 20 tc(3) Clock period, SCLK 80 ns tsu(4) Clock rising edge to CS rising edge, SCLK 60 ns tsu(5) Setup time, SDIN to SCLK 20 ns th(4) Hold time, SCLK to SDIN 20 ns tw(7) High 20 tw(8) Low 20
Clock pulse duration, SCLK ns
Pulse duration, CS ns
12 Electrical Specifications Copyright © 2004–2012, Texas Instruments Incorporated
Figure 2-4. 3-Wire Control Interface Timing Requirements
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SCLK
DIN
t
w(9)
t
w(10)
t
sp
t
h(5)
t
h(6)
t
su(7)
t
su(8)
T0551-01
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2.4.4 2-Wire Control Interface

PARAMETER MIN TYP MAX UNIT
tw(9) High 1.3 µs tw(10) Low 600 ns f(sf) Clock frequency, SCLK 0 400 kHz th(5) Hold time (start condition) 600 ns tsu(6) Setup time (start condition) 600 ns th(6) Data hold time 900 ns tsu(7) Data setup time 100 ns tr Rise time, SDIN, SCLK 300 ns tf Fall time, SDIN, SCLK 300 ns tsu(8) Setup time (stop condition) 600 ns tsp Pulse width of spikes suppressed by input filter 0 50 ns
Clock pulse duration, SCLK
Figure 2-5. 2-Wire Control Interface Timing Requirements
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MSB LSB
CS
B15
B14
B13
B12 B11
B10 B9 B8
B7
B6
B5
B4
B3
B2 B1
B0
SCLK
SDIN
T0552-01
TLV320AIC23B-Q1
SGLS240C –MARCH 2004–REVISED JUNE 2012

3 How to Use the TLV320AIC23B-Q1

3.1 Control Interfaces

The TLV320AIC23B-Q1 has many programmable features. The control interface is used to program the registers of the device. The control interface complies with SPI (3-wire operation) and 2-wire operation specifications. The state of the MODE terminal selects the control interface type. The MODE pin must be hardwired to the required level.
MODE INTERFACE
0 2-wire 1 SPI

3.1.1 SPI

In SPI mode, SDIN carries the serial data, SCLK is the serial clock and CS latches the data word into the TLV320AIC23B-Q1. The interface is compatible with microcontrollers and DSPs with an SPI interface.
A control word consists of 16 bits, starting with the MSB. The data bits are latched on the rising edge of SCLK. A rising edge on CS after the 16th rising clock edge latches the data word into the AIC (see
Figure 3-1).
The control word is divided into two parts. The first part is the address block, the second part is the data block:
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3.1.2 2-Wire

In 2-wire mode, the data transfer uses SDIN for the serial data and SCLK for the serial clock. The start condition is a falling edge on SDIN while SCLK is high. The seven bits following the start condition determine which device on the 2-wire bus receives the data. R/W determines the direction of the data transfer. The TLV320AIC23B-Q1 is a write only device and responds only if R/W is 0. The device operates only as a slave device whose address is selected by setting the state of the CS pin as follows.
B[15:9] Control Address Bits B[8:0] Control Data Bits
Figure 3-1. SPI Timing
CS STATE ADDRESS
(Default = 0)
0 0011010 1 0011011
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ADDR R/W ACK
Start Stop
1
7
8 9
1
8 9
1
8 9
B15–B8
B7–B0
ACK
ACK
SCLK
SDI
T0553-01
TLV320AIC23B-Q1
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The device that recognizes the address responds by pulling SDIN low during the ninth clock cycle, acknowledging the data transfer. The control follows in the next two eight-bit blocks. The stop condition after the data transfer is a rising edge on SDIN when SCLK is high (see Figure 3-2).
The 16-bit control word is divided into two parts. The first part is the address block, the second part is the data block:
SGLS240C –MARCH 2004–REVISED JUNE 2012
B[15:9] Control Address Bits B[8:0] Control Data Bits
Figure 3-2. 2-Wire Compatible Timing

3.1.3 Register Map

The TLV320AIC23B-Q1 has the following set of registers, which are used to program the modes of operation. To minimize corruption and potential noise injection due to improper sequencing, program the registers while the device is powered down (Register 0x06, value 0x80). After the registers are programmed, power on the device (Register 0x06, value 0x28).
ADDRESS REGISTER
0000000 Left line input channel volume control 0000001 Right line input channel volume control 0000010 Left channel headphone volume control 0000011 Right channel headphone volume control 0000100 Analog audio path control 0000101 Digital audio path control 0000110 Power down control 0000111 Digital audio interface format 0001000 Sample rate control 0001001 Digital interface activation 0001111 Reset register
Table 3-1. Left Line Input Channel Volume Control (Address: 0000000)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function LRS LIM X X LIV4 LIV3 LIV2 LIV1 LIV0 Default 0 1 0 0 1 0 1 1 1
LRS Left/right line simultaneous volume/mute update
Simultaneous update 0 = Disabled 1 = Enabled LIM Left line input mute 0 = Normal 1 = Muted LIV[4:0] Left line input volume control (10111 = 0 dB default)
11111 = 12 dB down to 00000 = –34.5 dB in 1.5-dB steps X Reserved
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Table 3-2. Right Line Input Channel Volume Control (Address: 0000001)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function RLS RIM X X RIV4 RIV3 RIV2 RIV1 RIV0 Default 0 1 0 0 1 0 1 1 1
RLS Right/left line simultaneous volume/mute update
Simultaneous update 0 = Disabled 1 = Enabled RIM Right line input mute 0 = Normal 1 = Muted RIV[4:0] Right line input volume control (10111 = 0 dB default)
11111 = 12 dB down to 00000 = –34.5 dB in 1.5-dB steps X Reserved
Table 3-3. Left Channel Headphone Volume Control (Address: 0000010)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function LRS LZC LHV6 LHV5 LHV4 LHV3 LHV2 LHV1 LHV0 Default 0 1 1 1 1 1 0 0 1
LRS Left/right headphone channel simultaneous volume/mute update
Simultaneous update 0 = Disabled 1 = Enabled LZC Left channel zero-cross detect
Zero-cross detect 0 = Off 1 = On LHV[6:0] Left Headphone volume control (1111001 = 0 dB default)
1111111 = 6 dB, 79 steps between 6 dB and 73 dB (mute), 0110000 = 73 dB (mute), anything below 0110000 does nothing
you are still muted
Table 3-4. Right Channel Headphone Volume Control (Address: 0000011)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function RLS RZC RHV6 RHV5 RHV4 RHV3 RHV2 RHV1 RHV0 Default 0 1 1 1 1 1 0 0 1
LRS Right/left headphone channel simultaneous volume/mute update
Simultaneous update 0 = Disabled 1 = Enabled
RZC Right channel zero-cross detect
Zero-cross detect 0 = Off 1 = On
RHV[6:0] Right headphone volume control (1111001 = 0 dB default)
1111111 = 6 dB, 79 steps between 6 dB and 73 dB (mute), 0110000 = 73 dB (mute), any thing below 0110000 does nothing you are still muted
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Table 3-5. Analog Audio Path Control (Address: 0000100)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function STA2 STA1 STA0 STE DAC BYP INSEL MICM MICB Default 0 0 0 0 1 1 0 1 0
STA[2:0] and STE
STE STA2 STA1 STA0 ADDED SIDETONE
1 1 X X 0 dB 1 0 0 0 -6 dB 1 0 0 1 -9 dB 1 0 1 0 -12 dB 1 0 1 1 -18 dB 0 X X X Disabled
DAC DAC select 0 = DAC off 1 = DAC selected BYP Bypass 0 = Disabled 1 = Enabled INSEL Input select for ADC 0 = Line 1 = Microphone MICM Microphone mute 0 = Normal 1 = Muted MICB Microphone boost 0 = dB 1 = 20 dB X Reserved
Table 3-6. Digital Audio Path Control (Address: 0000101)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function X X X X X DACM DEEMP1 DEEMP0 ADCHP Default 0 0 0 0 0 0 1 0 0
DACM DAC soft mute 0 = Disabled 1 = Enabled DEEMP[1:0] De-emphasis control 00 = Disabled 01 = 32 kHz 10 = 44.1 kHz 11 = 48 kHz ADCHP ADC high-pass filter 1 = Disabled 0 = Enabled X Reserved
Table 3-7. Power Down Control (Address: 0000110)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function X OFF CLK OSC OUT DAC ADC MIC LINE Default 0 0 0 0 0 0 1 1 1
OFF Device power 0 = On 1 = Off CLK Clock 0 = On 1 = Off OSC Oscillator 0 = On 1 = Off OUT Outputs 0 = On 1 = Off DAC DAC 0 = On 1 = Off ADC ADC 0 = On 1 = Off MIC Microphone input 0 = On 1 = Off LINE Line input 0 = On 1 = Off X Reserved
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Table 3-8. Digital Audio Interface Format (Address: 0000111)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function X X MS LRSWAP LRP IWL1 IWL0 FOR1 FOR0 Default 0 0 0 0 0 0 0 0 1
MS Maser/slave mode 0 = Slave 1 = Master LRSWAP DAC left/right swap 0 = Disabled 1 = Enabled LRP DAC left/right phase 0 = Right channel on, LRCIN high
1 = Right channel on, LRCIN low DSP mode 1 = MSB is available on second BCLK rising edge after LRCIN rising edge
0 = MSB is available on first BCLK rising edge after LRCIN rising edge IWL[1:0] Input bit length 00 = 16 bit 01 = 20 bit 10 = 24 bit 11 = 32 bit FOR[1:0] Data format 11 = DSP format, frame sync followed by two data words
10 = I2S format, MSB first, left – 1 aligned
01 = MSB first, left aligned
00 = MSB first, right aligned X Reserved NOTES: 1. In master mode, the TLV320AIC23B-Q1 supplies the BCLK, LRCOUT, and LRCIN. In slave mode, BCLK, LRCOUT, and
LRCIN are supplied to the TLV320AIC23B-Q1.
2. In normal mode, BCLK = MCLK/4 for all sample rates except for 88.2 kHz and 96 kHz. For 88.2 kHz and 96 kHz sample rate, BCLK = MCLK.
3. In USB mode, bit BCLK = MCLK
Table 3-9. Sample Rate Control (Address: 0001000)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function X CLKOUT CLKIN SR3 SR2 SR1 SR0 BOSR USB/Norma
Default 0 0 0 1 0 0 0 0 0 CLKIN Clock input divider 0 = MCLK 1 = MCLK/2
CLKOUT Clock output divider 0 = MCLK 1 = MCLK/2 SR[3:0] Sampling rate control (see Section 3.3.2.1 and Section 3.3.2.2) BOSR Base oversampling rate
USB mode: 0 = 250 f
Normal mode: 0 = 256 f USB/Normal Clock mode select: 0 = Normal 1 = USB X Reserved
1 = 272 f
s
1 = 384 f
s
s s
l
Table 3-10. Digital Interface Activation (Address: 0001001)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function X RES RES X X X X X ACT Default 0 0 0 0 0 0 0 0 0
ACT Activate interface 0 = Inactive 1 = Active X Reserved
Table 3-11. Reset Register (Address: 0001111)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function RES RES RES RES RES RES RES RES RES Default 0 0 0 0 0 0 0 0 0
RES Write 000000000 to this register triggers reset
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50 kΩ
10 kΩ
VMID
0 dB/20 dB
To ADC
MICIN
S0527-01
R2
R1
C1
C2 +
CDIN LINEIN
S0526-01
AGND
Where:
R1 = 5 kΩ R2 = 5 kΩ C1 = 47 pF C2 = 470 nF
TLV320AIC23B-Q1
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3.2 Analog Interface

3.2.1 Line Inputs

The TLV320AIC23B-Q1 has line inputs for the left and the right audio channels (RLINEIN and LLINEIN). Both line inputs have independently programmable volume controls and mutes. Active and passive filters for the two channels prevent high frequencies from folding back into the audio band.
The line-input gain is logarithmically adjustable from 12 dB to –34.5 dB in 1.5-dB steps. The ADC full­scale range is 1 V AVDD. To avoid distortions, it is important not to exceed the full-scale range.
The gain is independently programmable on both left and right line-inputs. To reduce the number of software write cycles required. Both channels can be locked to the same value by setting the RLS and LRS bits (see Section 3.1.3).
The line inputs are biased internally to VMID. When the line inputs are muted or the device is set to standby mode, the line inputs are kept biased to VMID using special antithump circuitry. This reduces audible clicks that otherwise might be heard when reactivating the inputs.
For interfacing to a CD system, the line input should be scaled to 1 V shown in Figure 3-3.
SGLS240C –MARCH 2004–REVISED JUNE 2012
at AVDD= 3.3 V. The full-scale range tracks linearly with analog supply voltage
RMS
to avoid clipping, using the circuit
RMS
R1 and R2 divide the input signal by two, reducing the 2 V the AIC23B inputs. The C1 filters high-frequency noise, and C2 removes any dc component from the signal.

3.2.2 Microphone Input

MICIN is a high-impedance, low-capacitance input that is compatible with a wide range of microphones. It has a programmable volume control and a mute function. Active and passive filters prevent high frequencies from folding back into the audio band.
The MICIN signal path has two gain stages. The first stage has a nominal gain of G1 = 50 k/10 k = 5. By adding an external resistor (R 50 k/(10 k + R programmable gain of 0 dB or 20 dB (see Section 3.1.3).
). For example, R
MIC
Figure 3-3. Analog Line Input Circuit
from the CD player to the nominal 1 V
RMS
) in series with MICIN, the gain of the first stage can be adjusted by G1 =
MIC
= 40 k gives a gain of 0 dB. The second stage has a software
MIC
RMS
of
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Figure 3-4. Microphone Input Circuit
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The microphone input is biased internally to VMID. When the line inputs are muted, the MICIN input is kept biased to VMID using special antithump circuitry. This reduces audible clicks that may otherwise be heard when reactivating the input.
The MICBIAS output provides a low-noise reference voltage suitable for biasing electret type microphones and the associated external resistor biasing network. The maximum source current capability is 3 mA. This limits the smallest value of external biasing resistors that safely can be used.
The MICBIAS output is not active in standby mode.

3.2.3 Line Outputs

The TLV320AIC23B-Q1 has two low-impedance line outputs (LLINEOUT and RLINEOUT) capable of driving line loads with 10-kand 50-pF impedances.
The DAC full-scale output voltage is 1 V analog supply voltage AVDD. The DAC is connected to the line outputs via a low-pass filter that removes out-of-band components. No further external filtering is required in most applications.
The DAC outputs, line inputs, and the microphone signal are summed into the line outputs. These sources can be switched off independently. For example, in bypass mode, the line inputs are routed to the line outputs, bypassing the ADC and the DAC. If sidetone is enabled, the microphone signal is routed to both line outputs via a four-step programmable attenuation circuit.
The line outputs are muted by either muting the DAC (analog) or soft muting (digital) and disabling the bypass and sidetone paths (see Section 3.1.3).
at AVDD= 3.3 V. The full-scale range tracks linearly with the
RMS
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3.2.4 Headphone Output

The TLV320AIC23B-Q1 has stereo headphone outputs (LHPOUT and RHPOUT), and is designed to drive 16-or 32-headphones. The headphone output includes a high-quality volume control and mute function.
The headphone volume is logarithmically adjustable from 6 dB to –73 dB in 1-dB steps. Writing 000000 to the volume-control registers (see Section 3.1.3) mutes the headphone output. When the headphone output is muted or the device is placed in standby mode, the dc voltage is maintained at the outputs to prevent audible clicks.
A zero-cross detection circuit is provided under the control of the LZC and RZC bits. If this circuit is enabled, the volume-control values are updated only when the input signal to the gain stage is close to the analog ground level.
This minimizes audible clicks as the volume is changed or the device is muted. This circuit has no time­out, so, if only dc levels are being applied to the gain stage input of more than 20 mV, the gain is not updated.
The gain is independently programmable on the left and right channels. Both channels can be locked to the same value by setting the RLS and LRS bits (see Section 3.1.3).

3.2.5 Analog Bypass Mode

The TLV320AIC23B-Q1 includes a bypass mode in which the analog line inputs are directly routed to the analog line outputs, bypassing the ADC and DAC. This is enabled by selecting the bypass bit in the analog audio path control register (see Section 3.1.3).
For a true bypass mode, the output from the DAC and the sidetone should be disabled. The line input and headphone output volume controls and mutes are still operational in bypass mode. Therefore the line inputs, DAC output, and microphone input can be summed together. The maximum signal at any point in the bypass path must be no greater than 1 V
at AVDD= 3.3 V to avoid clipping and distortion. This
RMS
amplitude tracks linearly with AVDD.
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n
n−1
0
1 n−1
n
1/fs
1
00
MSB LSB
LRCIN/
LRCOUT
BCLK
DIN/
DOUT
Left Channel Right Channel
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3.2.6 Sidetone Insertion

The TLV320AIC23B-Q1 has a sidetone insertion made where the microphone input is routed to the line and headphone outputs. This is useful for telephony and headset applications. The attenuation of the sidetone signal may be set to 6 dB, 9 dB, 12 dB, 15 dB, or 0 dB, by software selection (see
Section 3.1.3). If this mode is used to sum the microphone input with the DAC output and line inputs, care
must be taken not to exceed signal level to avoid clipping and distortion.

3.3 Digital Audio Interface

3.3.1 Digital Audio-Interface Modes

The TLV320AIC23B-Q1 supports four audio-interface modes.
Right justified
Left justified
I2S mode
DSP mode The four modes are MSB first and operate with a variable word width between 16 to 32 bits (except right-
justified mode, which does not support 32 bits). The digital audio interface consists of clock signal BCLK, data signals DIN and DOUT, and
synchronization signals LRCIN and LRCOUT. BCLK is an output in master mode and an input in slave mode.
SGLS240C –MARCH 2004–REVISED JUNE 2012
3.3.1.1 Right-Justified Mode
In right-justified mode, the LSB is available on the rising edge of BCLK, preceding a falling edge on LRCIN or LRCOUT (see Figure 3-5).
3.3.1.2 Left-Justified Mode
In left-justified mode, the MSB is available on the rising edge of BCLK, following a rising edge on LRCIN or LRCOUT (see Figure 3-6).
Figure 3-5. Right-Justified Mode Timing
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n
n−1
0
1 n−1
n
1
0
MSB LSB
LRCIN/
LRCOUT
BCLK
DIN/
DOUT
Left Channel Right Channel
MSB LSB
T0557-01
n
n−1
0
1 n−1
n
1/fs
1
0
MSB LSB
LRCIN/
LRCOUT
BCLK
DIN/
DOUT
Left Channel Right Channel
1BCLK
T0556-01
n
n−1
0
1 n−1
n
1/fs
1
0
MSB LSB
LRCIN/
LRCOUT
BCLK
DIN/
DOUT
Left Channel Right Channel
n
T0555-01
TLV320AIC23B-Q1
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3.3.1.3 I2S Mode
In I2S mode, the MSB is available on the second rising edge of BCLK, after the falling edge on LRCIN or LRCOUT (see Figure 3-7).
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Figure 3-6. Left-Justified Mode Timing
Figure 3-7. I2S Mode Timing
3.3.1.4 DSP Mode
The DSP mode is compatible with the McBSP ports of TI DSPs. LRCIN and LRCOUT must be connected to the Frame Sync signal of the McBSP. A falling edge on LRCIN or LRCOUT starts the data transfer. The left-channel data consists of the first data word, which is immediately followed by the right channel data word (see Figure 3-8). Input word length is defined by the IWL register. Figure 3-8 shows LRP = 1 (default LRP = 0).
Figure 3-8. DSP Mode Timing
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3.3.2 Audio Sampling Rates

The TLV320AIC23B-Q1 can operate in master or slave clock mode. In the master mode, the TLV320AIC23B-Q1 clock and sampling rates are derived from a 12-MHz MCLK signal. This 12-MHz clock signal is compatible with the USB specification. The TLV320AIC23B-Q1 can be used directly in a USB system.
In the slave mode, an appropriate MCLK or crystal frequency and the sample rate control register settings control the TLV320AIC23B-Q1 clock and sampling rates.
The settings in the sample rate control register control the clock mode and sampling rates.
Table 3-12. Sample Rate Control (Address: 0001000)
BIT D8 D7 D6 D5 D4 D3 D2 D1 D0
Function X CLKOUT CLKIN SR3 SR2 SR1 SR0 BOSR USB/Normal Default 0 0 0 1 0 0 0 0 0
CLKIN Clock input divider 0 = MCLK 1 = MCLK/2 CLKOUT Clock output divider 0 = MCLK 1 = MCLK/2 SR[3:0] Sampling rate control (see Sections 3.3.2.1 AND 3.3.2.2) BOSR Base oversampling rate
USB mode: 0 = 250 f
Normal mode: 0 = 256 f USB/Normal Clock mode select: 0 = Normal 1 = USB X Reserved
1 = 272 f
s
1 = 384 f
s
s s
The clock circuit of the AIC23B has two internal dividers. The first, controlled by CLKIN, applies to the sampling-rate generator of the codec. The second, controlled by CLKOUT, applies only to the CLKOUT terminal. By setting CLKIN to 1, the entire codec is clocked with half the frequency, effectively dividing the resulting sampling rates by two. The following sampling-rate tables are based on CLKIN = MCLK.
3.3.2.1 USB-Mode Sampling Rates (MCLK = 12 MHz)
In the USB mode, the following ADC and DAC sampling rates are available:
SAMPLING RATE
ADC (kHz) DAC (kHz) SR3 SR2 SR1 SR0 BOSR
96 96 3 0 1 1 1 0
88.2 88.2 2 1 1 1 1 1 48 48 0 0 0 0 0 0
44.1 44.1 1 1 0 0 0 1 32 32 0 0 1 1 0 0
8.021 8.021 1 1 0 1 1 1 8 8 0 0 0 1 1 0
48 8 0 0 0 0 1 0
44.1 8.021 1 1 0 0 1 1 8 48 0 0 0 1 0 0
8.021 44.1 1 1 0 1 0 1
(1) The sampling rates are derived from the 12-MHz master clock. The available oversampling rates do not produce exactly 8-kHz, 44.1-
kHz, and 88.2-kHz sampling rates, but 8.021 kHz, 44.117 kHz, and 88.235 kHz, respectively. See Figure 3-9 through Figure 3-26 for filter responses.
(1)
FILTER TYPE
SAMPLING-RATE CONTROL SETTINGS
3.3.2.2 Normal-Mode Sampling Rates
In normal mode, the following ADC and DAC sampling rates, depending on the MCLK frequency, are available:
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MCLK = 12.288 MHz
SAMPLING RATE SAMPLING-RATE CONTROL SETTINGS
ADC (kHz) DAC (kHz) SR3 SR2 SR1 SR0 BOSR
FILTER TYPE
96 96 2 0 1 1 1 0 48 48 1 0 0 0 0 0 32 32 1 0 1 1 0 0
8 8 1 0 0 1 1 0
48 8 1 0 0 0 1 0
8 48 1 0 0 1 0 0
MCLK = 11.2896 MHz
SAMPLING RATE SAMPLING-RATE CONTROL SETTINGS
ADC (kHz) DAC (kHz) SR3 SR2 SR1 SR0 BOSR
FILTER TYPE
88.2 88.2 2 1 1 1 1 0
44.1 44.1 1 1 0 0 0 0
8.021 8.021 1 1 0 1 1 0
44.1 8.021 1 1 0 0 1 0
8.021 44.1 1 1 0 1 0 0
MCLK = 18.432 MHz
SAMPLING RATE SAMPLING-RATE CONTROL SETTINGS
ADC (kHz) DAC (kHz) SR3 SR2 SR1 SR0 BOSR
FILTER TYPE
96 96 2 0 1 1 1 1 48 48 1 0 0 0 0 1 32 32 1 0 1 1 0 1
8 8 1 0 0 1 1 1
48 8 1 0 0 0 1 1
8 48 1 0 0 1 0 1
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MCLK = 16.9344 MHz
SAMPLING RATE SAMPLING-RATE CONTROL SETTINGS
ADC (kHz) DAC (kHz) SR3 SR2 SR1 SR0 BOSR
FILTER TYPE
88.2 88.2 2 1 1 1 1 1
44.1 44.1 1 1 0 0 0 1
8.021 8.021 1 1 0 1 1 1
44.1 8.021 1 1 0 0 1 1
8.021 44.1 1 1 0 1 0 1

3.3.3 Digital Filter Characteristics

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT ADC Filter Characteristics (TI DSP 250 fsMode Operation)
Passband ±0.05 dB 0.416 f Stopband –6 dB 0.5 f
s
s
Passband ripple ±0.05 dB Stopband attenuation f > 0.584 f
s
–60 dB
ADC Filter Characteristics (TI DSP 272 fsand Normal Mode Operation)
Passband ±0.05 dB 0.4535 f Stopband –6 dB 0.5 f
s
s
Passband ripple ±0.05 dB Stopband attenuation f > 0.5465 f
s
–60 dB
Hz Hz
Hz Hz
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−10
−8
−6
−4
−2
0
0 0.1 0.2 0.3 0.4 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G001
−10
−8
−6
−4
−2
0
0 0.1 0.2 0.3 0.4 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G002
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PARAMETER TEST CONDITIONS MIN TYP MAX UNIT ADC High-Pass Filter Characteristics
Corner frequency –3 dB, fs= 44.1 kHz 3.7 Hz
–3 dB, fs= 48 kHz 4 Hz –0.5 dB, fs= 44.1 kHz 10.4 Hz –0.5 dB, fs= 48 kHz 11.3 Hz –0.1 dB fs= 44.1 kHz 21.6 Hz –0.1 dB, fs= 48 kHz 23.5 Hz
DAC Filter Characteristics (48-kHz Sampling Rate)
Passband ±0.03 dB 0.416 f
s
Stopband –6 dB 0.5 f
s
Hz
Hz Passband ripple ±0.03 dB Stopband attenuation f > 0.584 f
s
–50 dB
DAC Filter Characteristics (44.1-kHz Sampling Rate)
Passband ±0.03 dB 0.4535 f
s
Stopband –6 dB 0.5 f
s
Hz
Hz Passband ripple ±0.03 dB Stopband attenuation f > 0.5465 f
s
–50 dB
SPACE SPACE SPACE SPACE SPACE SPACE
FILTER RESPONSE FILTER RESPONSE
vs vs
NORMALIZED AUDIO SAMPLING FREQUENCY NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-9. Digital De-Emphasis Filter Response Figure 3-10. Digital De-Emphasis Filter Response
44.1 kHz Sampling 48 kHz Sampling
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−0.1
−0.08
−0.06
−0.04
−0.02
0
0.02
0.04
0.06
0.08
0.1
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G004
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G003
TLV320AIC23B-Q1
SGLS240C –MARCH 2004–REVISED JUNE 2012
www.ti.com
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-11. ADC Digital Filter Response 0: USB Mode (Group Delay = 12 Output Samples)
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-12. ADC Digital Filter Ripple 0: USB (Group Delay = 20 Output Samples)
26 How to Use the TLV320AIC23B-Q1 Copyright © 2004–2012, Texas Instruments Incorporated
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−0.1
−0.1
−0.1
0
0
0
0
0
0.1
0.1
0.1
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G006
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G005
TLV320AIC23B-Q1
www.ti.com
SGLS240C –MARCH 2004–REVISED JUNE 2012
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-13. ADC Digital Filter Response 1: USB Mode Only
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-14. ADC Digital Filter Ripple 1: USB Mode Only
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−0.4
−0.3
−0.2
−0.1
0
0.1
0.2
0.3
0.4
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G008
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G007
TLV320AIC23B-Q1
SGLS240C –MARCH 2004–REVISED JUNE 2012
www.ti.com
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-15. ADC Digital Filter Response 2: USB Mode and Normal Modes (Group Delay = 3 Output
Samples)
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
28 How to Use the TLV320AIC23B-Q1 Copyright © 2004–2012, Texas Instruments Incorporated
Figure 3-16. ADC Digital Filter Ripple 2: USB Mode and Normal Modes
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−0.4
−0.3
−0.2
−0.1
0
0.1
0.2
0.3
0.4
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G010
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G009
TLV320AIC23B-Q1
www.ti.com
SGLS240C –MARCH 2004–REVISED JUNE 2012
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-17. ADC Digital Filter Response 3: USB Mode Only
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-18. ADC Digital Filter Ripple 3: USB Mode Only
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−0.1
−0.08
−0.06
−0.04
−0.02
0
0.02
0.04
0.06
0.08
0.1
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G012
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G011
TLV320AIC23B-Q1
SGLS240C –MARCH 2004–REVISED JUNE 2012
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SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-19. DAC Digital Filter Response 0: USB Mode
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-20. DAC Digital Filter Ripple 0: USB Mode
30 How to Use the TLV320AIC23B-Q1 Copyright © 2004–2012, Texas Instruments Incorporated
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−0.1
−0.08
−0.06
−0.04
−0.02
0
0.02
0.04
0.06
0.08
0.1
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G014
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G013
TLV320AIC23B-Q1
www.ti.com
SGLS240C –MARCH 2004–REVISED JUNE 2012
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-21. DAC Digital Filter Response 1: USB Mode Only
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-22. DAC Digital Filter Ripple 1: USB Mode Only
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−0.4
−0.3
−0.2
−0.1
0
0.1
0.2
0.3
0.4
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G016
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G015
TLV320AIC23B-Q1
SGLS240C –MARCH 2004–REVISED JUNE 2012
www.ti.com
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-23. DAC Digital Filter Response 2: USB Mode and Normal Modes
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-24. DAC Digital Filter Ripple 2: USB Mode and Normal Modes
32 How to Use the TLV320AIC23B-Q1 Copyright © 2004–2012, Texas Instruments Incorporated
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−0.4
−0.3
−0.2
−0.1
0
0.1
0.2
0.3
0.4
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
Normalized Audio Sampling Frequency
Filter Response (dB)
G018
−90
−70
−50
−30
−10
10
0 0.5 1 1.5 2 2.5 3
Normalized Audio Sampling Frequency
Filter Response (dB)
G017
TLV320AIC23B-Q1
www.ti.com
SGLS240C –MARCH 2004–REVISED JUNE 2012
SPACE SPACE SPACE
FILTER RESPONSE
vs
NORMALIZED AUDIO SAMPLING FREQUENCY
Figure 3-25. DAC Digital Filter Response 3: USB Mode Only
SPACE SPACE SPACE
FILTER RESPONSE
NORMALIZED AUDIO SAMPLING FREQUENCY
vs
Figure 3-26. DAC Digital Filter Ripple 3: USB Mode Only
The delay between the converter is a function of the sample rate. The group delays for the AIC23B are shown in the following table. Each delay is one LR clock (1/sample rate).
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Table 3-13. Group Delays
FILTER GROUP DELAY
DAC type 0 11 DAC type 1 18 DAC type 2 5 DAC type 3 5 ADC type 0 12 ADC type 1 20 ADC type 2 3 ADC type 3 6
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Revision History

NOTE: Page numbers for previous revisions may differ from page numbers in the current version.
Changes from Revision B (June 2008) to Revision C Page
Added sentences after: The TLV320AIC23B-Q1 has the following set of registers, which are used to
program the modes of operation, in the Register Map section. ......................................................... 15
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PACKAGE OPTION ADDENDUM
www.ti.com
10-Dec-2020
PACKAGING INFORMATION
Orderable Device Status
6PAIC23BIPWRG4Q1 ACTIVE TSSOP PW 28 2000 RoHS & Green NIPDAU Level-1-260C-UNLIM -40 to 85 AIC23BIQ1
TLV320AIC23BIPWRQ1 ACTIVE TSSOP PW 28 2000 RoHS & Green NIPDAU Level-1-260C-UNLIM -40 to 85 AIC23BIQ1
(1)
The marketing status values are defined as follows:
ACTIVE: Product device recommended for new designs. LIFEBUY: TI has announced that the device will be discontinued, and a lifetime-buy period is in effect. NRND: Not recommended for new designs. Device is in production to support existing customers, but TI does not recommend using this part in a new design. PREVIEW: Device has been announced but is not in production. Samples may or may not be available. OBSOLETE: TI has discontinued the production of the device.
Package Type Package
(1)
Drawing
Pins Package
Qty
Eco Plan
(2)
Lead finish/ Ball material
(6)
MSL Peak Temp
(3)
Op Temp (°C) Device Marking
(4/5)
(2)
RoHS: TI defines "RoHS" to mean semiconductor products that are compliant with the current EU RoHS requirements for all 10 RoHS substances, including the requirement that RoHS substance
do not exceed 0.1% by weight in homogeneous materials. Where designed to be soldered at high temperatures, "RoHS" products are suitable for use in specified lead-free processes. TI may reference these types of products as "Pb-Free".
RoHS Exempt: TI defines "RoHS Exempt" to mean products that contain lead but are compliant with EU RoHS pursuant to a specific EU RoHS exemption. Green: TI defines "Green" to mean the content of Chlorine (Cl) and Bromine (Br) based flame retardants meet JS709B low halogen requirements of <=1000ppm threshold. Antimony trioxide based
flame retardants must also meet the <=1000ppm threshold requirement.
(3)
MSL, Peak Temp. - The Moisture Sensitivity Level rating according to the JEDEC industry standard classifications, and peak solder temperature.
(4)
There may be additional marking, which relates to the logo, the lot trace code information, or the environmental category on the device.
(5)
Multiple Device Markings will be inside parentheses. Only one Device Marking contained in parentheses and separated by a "~" will appear on a device. If a line is indented then it is a continuation
of the previous line and the two combined represent the entire Device Marking for that device.
(6)
Lead finish/Ball material - Orderable Devices may have multiple material finish options. Finish options are separated by a vertical ruled line. Lead finish/Ball material values may wrap to two
lines if the finish value exceeds the maximum column width.
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Samples
Addendum-Page 1
PACKAGE OPTION ADDENDUM
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OTHER QUALIFIED VERSIONS OF TLV320AIC23B-Q1 :
Catalog: TLV320AIC23B
NOTE: Qualified Version Definitions:
Catalog - TI's standard catalog product
10-Dec-2020
Addendum-Page 2
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