Datasheet TLV320AIC23PW, TLV320AIC23IPW, TLV320AIC23IGQE, TLV320AIC23GQE Datasheet (Texas Instruments)

TLV320AIC23

Stereo Audio CODEC,
8Ć to 96ĆkHz, With Integrated Headphone Amplifier

Data Manual

July 2001 Digital Audio Products

SLWS106C

IMPORTANT NOTICE

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Contents

Section Title Page

1 Introduction 1–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.1 Features 1–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.2 Functional Block Diagram 1–3. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.3 Terminal Assignments 1–4. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.4 Ordering Information 1–4. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

1.5 Terminal Functions 1–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2 Specifications 2–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.1 Absolute Maximum Ratings Over Operating Free-Air

Temperature Range 2–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.2 Recommended Operating Conditions 2–1. . . . . . . . . . . . . . . . . . . . . . . . . .

2.3 Electrical Characteristics Over Recommended Operating

Conditions 2–2. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.3.1 ADC 2–2. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.3.2 DAC 2–3. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.3.3 Analog Line Input to Line Output 2–3. . . . . . . . . . . . . . . . . . . . . .

2.3.4 Stereo Headphone Output 2–4. . . . . . . . . . . . . . . . . . . . . . . . . . .

2.3.5 Analog Reference Levels 2–4. . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.3.6 Digital I/O 2–4. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.3.7 Supply Current 2–4. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.4 Digital-Interface Timing 2–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.4.1 Audio Interface (Master Mode) 2–5. . . . . . . . . . . . . . . . . . . . . . .

2.4.2 Audio Interface (Slave-Mode) 2–6. . . . . . . . . . . . . . . . . . . . . . . .

2.4.3 Three-Wire Control Interface 2–7. . . . . . . . . . . . . . . . . . . . . . . . .

2.4.4 Two-Wire Control Interface 2–7. . . . . . . . . . . . . . . . . . . . . . . . . . .

3 How to Use the AIC23 3–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.1 Control Interfaces 3–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.1.1 SPI 3–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.1.2 I2C 3–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.1.3 Register Map 3–2. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2 Analog Interface 3–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2.1 Line Inputs 3–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2.2 Microphone Input 3–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2.3 Line Outputs 3–6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2.4 Headphone Output 3–6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2.5 Analog Bypass Mode 3–7. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.2.6 Sidetone Insertion 3–7. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

iii

3.3 Digital Audio Interface 3–7. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.3.1 Digital Audio-Interface Modes 3–7. . . . . . . . . . . . . . . . . . . . . . . .

3.3.2 Audio Sampling Rates 3–9. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3.3.3 Digital Filter Characteristics 3–11. . . . . . . . . . . . . . . . . . . . . . . . . .

A Mechanical Data A–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

iv

List of Illustrations

Figure Title Page

2–1 System-Clock Timing Requirements 2–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2–2 Master-Mode Timing Requirements 2–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2–3 Slave-Mode Timing Requirements 2–6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2–4 Three-Wire Control Interface Timing Requirements 2–7. . . . . . . . . . . . . . . . . . . .

2–5 Two-Wire Control Interface Timing Requirements 2–7. . . . . . . . . . . . . . . . . . . . .

3–1 SPI Timing 3–1. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3–2 2-Wire I2C Compatible Timing 3–2. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3–3 Analog Line Input Circuit 3–5. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3–4 Microphone Input Circuit 3–6. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3–5 Right-Justified Mode Timing 3–7. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3–6 Left-Justified Mode Timing 3–8. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2

3–7I

3–8 DSP Mode Timing 3–8. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3–9 Digital De-Emphasis Filter Response – 441 kHz Sampling 3–12. . . . . . . . . . . . .

3–10 Digital De-Emphasis Filter Response – 48 kHz Sampling 3–12. . . . . . . . . . . . .

3–11 ADC Digital Filter Response I: TI DSP and Normal Modes

3–12 ADC Digital Filter Ripple I: TI DSP and Normal Modes

3–13 ADC Digital Filter Response II: TI DSP Mode Only 3–14. . . . . . . . . . . . . . . . . . .

3–14 ADC Digital Filter Ripple II: TI DSP Mode Only 3–14. . . . . . . . . . . . . . . . . . . . . .

3–15 ADC Digital Filter Response III: TI DSP and Normal Modes

3–16 ADC Digital Filter Ripple III: TI DSP and Normal Modes 3–15. . . . . . . . . . . . . . .

3–17 ADC Digital Filter Response IV: TI DSP Mode Only 3–16. . . . . . . . . . . . . . . . . .

3–18 ADC Digital Filter Ripple IV: TI DSP Mode Only 3–16. . . . . . . . . . . . . . . . . . . . . .

3–19 DAC Digital Filter Response I: TI DSP and Normal Modes 3–17. . . . . . . . . . . .

3–20 DAC Digital Filter Ripple I: TI DSP and Normal Modes 3–17. . . . . . . . . . . . . . . .

3–21 DAC Digital Filter Response II: TI DSP Mode Only 3–18. . . . . . . . . . . . . . . . . . .

3–22 DAC Digital Filter Ripple II: TI DSP Mode Only 3–18. . . . . . . . . . . . . . . . . . . . . .

3–23 DAC Digital Filter Response III: TI DSP and Normal Modes 3–19. . . . . . . . . . .

3–24 DAC Digital Filter Ripple III: TI DSP and Normal Modes 3–19. . . . . . . . . . . . . . .

3–25 DAC Digital Filter Response IV: TI DSP Mode Only 3–20. . . . . . . . . . . . . . . . . .

3–26 DAC Digital Filter Ripple IV: TI DSP Mode Only 3–20. . . . . . . . . . . . . . . . . . . . . .

S Mode Timing 3–8. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

(Group Delay = 12 Output Samples) 3–13. . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

(Group Delay = 20 Output Samples) 3–13. . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

(Group Delay = 3 Output Samples) 3–15. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

v

vi

1 Introduction

The TLV320AIC23 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 TLV320AIC23 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
TL V320AIC23 is the ideal analog input/output (I/O) choice for portable digital audio-player and recorder applications,
such as MP3 digital audio players.

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 AIC23 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 TL V320AIC23 supports the industry-standard oversampling rates of 256 f
rates of 250 f

and 272 fs are provided, which optimize interface considerations in designs using TI C54x digital signal

s

processors (DSPs) and universal serial bus (USB) data interfaces. A single 12-MHz crystal can supply clocking to
the DSP, USB, and codec. The TLV320AIC23 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 f

and 272 fs oversampling rates.

s

Low power consumption and flexible power management allow selective shutdown of codec functions, thus
extending battery life in portable applications. This design solution, coupled with the industry’s smallest package, the
TI proprietary MicroStar Junior using only 25 mm

2

of board area, makes powerful portable stereo audio designs

easily realizable in a cost-effective, space-saving total analog I/O solution: the TLV320AIC23.

and 384 fs, unique oversampling

s

1.1 Features

• 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 Voltages
– 8-kHz – 96-kHz Sampling-Frequency Support

• Software Control Via TI McBSP-Compatible Multiprotocol Serial Port

– I2C-Compatible and SPI-Compatible Serial-Port Protocols
– Glueless Interface to TI McBSPs

• Audio-Data Input/Output Via TI McBSP-Compatible Programmable Audio Interface

2

S-Compatible Interface Requiring Only One McBSP for both ADC and DAC

– I
– Standard I
– 16/20/24/32-Bit Word Lengths

MicroStar Junior is a trademark of Texas Instruments.

2

S, MSB, or LSB Justified-Data Transfers

1–1

– Audio Master/Slave Timing Capability Optimized for TI DSPs (250/272 fs), USB mode
– Industry-Standard Master/Slave Support Provided Also (256/384 f

), Normal mode

s

– Glueless Interface to TI McBSPs

• Integrated Total Electret-Microphone Biasing 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

– Integrated Programmable Gain Amplifier
– Analog Bypass Path of Codec

• ADC Multiplexed Input for Stereo-Line Inputs and Microphone

• Stereo-Line Outputs

– Analog Stereo Mixer for DAC and Analog Bypass Path

• Analog Volume Control With Mute

• 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

• Industry’s Smallest Package: 32-Pin TI Proprietary MicroStar Junior

2

Total Board Area

– 25 mm
– 28-Pin TSSOP Also Is Available (62 mm

2

Total Board Area)

• Ideally Suitable for Portable Solid-State Audio Players and Recorders

1–2

1.2 Functional Block Diagram

AVDD

VMID

AGND

MICBIAS

RLINEIN

MICIN

LLINEIN

HPVDD
HPGND

RHPOUT

50 kΩ

50 kΩ

12 to –34.5 dB,

10 kΩ

VMID

12 to –34 dB,

Headphone

Driver

1.0X

1.0X

1.0X

1.5X

1.5 dB Steps

50 kΩ

1.5 dB Steps
6 to –73 dB,

1 dB Steps

VADC

VDAC

VMID

Bypass
Mute

Line

Mute

Line

Mute

Bypass
Mute

Mute,

0 dB, 20 dB

2:1

MUX

Σ

DSPcodec

TLV320AIC23

2:1

MUX

VADC

Side Tone
Mute

Σ–∆

ADC

Σ–∆

ADC

Σ–∆

DAC

Control

Interface

Digital
Filters

CS
SDIN
SCLK
MODE

DVDD
BVDD

DGND

ROUT

LOUT

LHPOUT

Headphone

Driver

XTI/MCLK

XTO

CLKOUT

NOTE: MCLK, BCLK, and SCLK are all asynchronous to each other.

6 to –73 dB,

1 dB Steps

(1x, 1/2x)

OSC

Σ

CLKIN

Divider

CLKOUT

Divider

(1x, 1/2x)

VDAC

Σ–∆

DAC

Digital

Audio

Interface

LRCIN
DIN
LRCOUT
DOUT

BCLK

1–3

1.3 Terminal Assignments

GQE PACKAGE

(TOP VIEW)

NC

DIN

BCLK

CLKOUT

BVDD

DGND

DVDD

XTO

25 24 23 22 21 20 19 18 17

NC

LRCIN

DOUT

LRCOUT

HPVDD

LHPOUT

RHPOUT

HPGND

26
27
28
29
30
31
32

123456789

NC

LOUT

NC – No internal connection

BVDD

CLKOUT

BCLK

DIN

LRCIN

DOUT

LRCOUT

HPVDD

LHPOUT

RHPOUT

HPGND

LOUT

ROUT

AVDD

1
2
3
4
5
6
7
8
9
10
11
12
13
14

ROUT

AVDD

PW PACKAGE

(TOP VIEW)

VMID

AGND

28
27
26
25
24
23
22
21
20
19
18
17
16
15

MICIN

MICBIAS

DGND
DVDD
XTO
XTI/MCLK
SCLK
SDIN
MODE
CS
LLINEIN
RLINEIN
MICIN
MICBIAS
VMID
AGND

NC

16
15
14
13
12
11
10

XTI/MCLK
SCLK

SDIN
MODE
CS
LLINEIN
RLINEIN

1.4 Ordering Information

T

A

–10°C to 70°C TLV320AIC23GQE TLV320AIC23PW
–40°C to 85°C TLV320AIC23IGQE TLV320AIC23IPW

1–4

PACKAGE

32-Pin

MicroStar Junior GQE

28-Pin

TSSOP PW

1.5 Terminal Functions

NAME

I/O

TERMINAL

NO.

GQE PW

AGND 5 15 Analog supply return
AVDD 4 14 Analog supply input. Voltage level is 3.3 V nominal.
BCLK 23 3 I/O I2S serial-bit clock. In audio master mode, the AIC23 generates this signal and sends it to the DSP. In

BVDD 21 1 Buffer supply input. V oltage range is from 2.7 V to 3.6 V.
CLKOUT 22 2 O Clock output. This is a buffered version of the XTI input and is available in 1X or 1/2X frequencies of XTI.

CS 12 21 I Control port input latch/address select. For SPI control mode this input acts as the data latch control. For

DIN 24 4 I I2S format serial data input to the sigma-delta stereo DAC
DGND 20 28 Digital supply return
DOUT 27 6 O I2S format serial data output from the sigma-delta stereo ADC
DVDD 19 27 Digital supply input. Voltage range is 3.3 V nominal.
HPGND 32 11 Analog headphone amplifier supply return
HPVDD 29 8 Analog headphone amplifier supply input. Voltage level is 3.3 V nominal.
LHPOUT 30 9 O Left stereo mixer-channel amplified headphone output. Nominal 0-dB output level is 1 V

LLINEIN 11 20 I Left stereo-line input channel. Nominal 0-dB input level is 1 V

LOUT 2 12 O Left stereo mixer-channel line output. Nominal output level is 1.0 V
LRCIN 26 5 I/O I2S DAC-word clock signal. In audio master mode, the AIC23 generates this framing signal and sends it

LRCOUT 28 7 I/O I2S ADC-word clock signal. In audio master mode, the AIC23 generates this framing signal and sends it

MICBIAS 7 17 O Buffered low-noise-voltage output suitable for electret-microphone-capsule biasing. Voltage level is 3/4

MICIN 8 18 I Buffered amplifier input suitable for use with electret-microphone capsules. Without external resistors a

MODE 13 22 I Serial-interface-mode input. See Section 3.1 for details.
NC 1, 9

17, 25

RHPOUT 31 10 O Right stereo mixer-channel amplified headphone output. Nominal 0-dB output level is 1 V

RLINEIN 10 19 I Right stereo-line input channel. Nominal 0-dB input level is 1 V

ROUT 3 13 O Right stereo mixer-channel line output. Nominal output level is 1.0 V
SCLK 15 24 I Control-port serial-data clock. For SPI and I2C control modes this is the serial-clock input. See Section

SDIN 14 23 I Control-port serial-data input. For SPI and I2C control modes this is the serial-data input and also is used

VMID 6 16 I Midrail voltage decoupling input. 10-µF and 0.1-µF capacitors should be connected in parallel to this

XTI/MCLK 16 25 I Crystal or external-clock input. Used for derivation of all internal clocks on the AIC23.
XTO 18 26 O Crystal output. Connect to external crystal for applications where the AIC23 is the audio timing master.

audio slave mode, the signal is generated by the DSP.

Bit 07 in the sample rate control register controls frequency selection.

I2C control mode this input defines the seventh bit in the device address field. See Section 3.1 for details.

dB to 6 dB is provided in 1-dB steps.

in 1.5-dB steps.

to the DSP. In audio slave mode, the signal is generated by the DSP.

to the DSP. In audio slave mode, the signal is generated by the DSP.

AVDD nominal.

default gain of 5 is provided. See Section 2.3.1.2 for details.

Not Used—No internal connection

–73 dB to 6 dB is provided in 1-dB steps.

in 1.5-dB steps.

3.1 for details.

to select the control protocol after reset. See Section 3.1 for details.

terminal for noise filtering. Voltage level is 1/2 AVDD nominal.

Not used in applications where external clock source is used.

DESCRIPTION

RMS

. Gain of –34.5 dB to 12 dB is provided

RMS

.

RMS

RMS

. Gain of –34.5 dB to 12 dB is provided

RMS

.

RMS

. Gain of –73

. Gain of

1–5

1–6

2 Specifications

2.1 Absolute Maximum Ratings Over Operating Free-Air Temperature Range (unless
otherwise noted)

Supply voltage range, AVDD to AGND, DVDD to DGND, BVDD to DGND, HPVDD to HPGND

(see Note 1) –0.3 V to + 3.63 V. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Analog supply return to digital supply return, AGND to DGND –0.3 V to + 3 .63 V. . . . . . . . . . . . . . . . . . . . . . .

Input voltage range, all input signals: Digital –0.3 V to DV

Case temperature for 10 seconds 240°C. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Operating free-air temperature range, T
Storage temperature range, T

†

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.

NOTE 1: DVDD may not exceed BVDD + 0.3V; BVDD may not exceed AVDD + 0.3V or HPVDD + 0.3.

†

+ 0.3 V. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

DD
DD

+ 0.3 V. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

stg

Analog –0.3 V to AV

A

–10°C to 70°C. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

–65°C to 150°C. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2.2 Recommended Operating Conditions

MIN NOM MAX UNIT

Analog supply voltage, A VDD, HPVDD (see Note 2) 2.7 3.3 3.6 V
Digital buffer supply voltage, BVDD (see Note 2) 2.7 3.3 3.6 V
Digital core supply voltage, DVDD (see Note 2) 1.42 1.5 3.6 V
Analog input voltage, full scale – 0dB (AVDD = 3.3 V) 1 V
Stereo-line output load resistance 10 kΩ
Headphone-amplifier output load resistance 0 Ω
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

NOTE 2: Digital voltage values are with respect to DGND; analog voltage values are with respect to AGND.

A

–10 70 °C

RMS

2–1

2.3 Electrical Characteristics Over Recommended Operating Conditions, AVDD,

g,g,g(

dB

yg,g, (

dB

Total harmonic distortion

input, 0-dB gain

dB

Input resistance

kΩ

Signal-to-noise ratio, A-weighted, 0-dB gain (see Notes 3 and 4)

dB

Dynamic range, A-weighted

full-scale input (see Note 4)

dB

Total harmonic distortion

input, 0-dB gain

dB

HPV

, BVDD = 3.3 V, DVDD = 1.5 V, Slave Mode, XTI/MCLK = 256fs, fs = 48 kHz

DD

(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 (see Notes 3

and 4)
Dynamic range, A-weighted, –60-dB full-scale input (see

Note 4)

, –1-dB

Power supply rejection ratio 1 kHz, 100 mV
ADC channel separation 1 kHz input tone 90 dB
Programmable gain 1 kHz input tone, R
Programmable gain step size Monotonic 1.5 dB
Mute attenuation 0 dB, 1 kHz input tone 80 dB

p

Input capacitance 10 pF

NOTES: 3. 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.

4. 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.

p

AVDD = 3.3 V 85 90
AVDD = 2.7 V 90
AVDD = 3.3 V –80
AVDD = 2.7 V 80

12 dB Input gain 10 20
0 dB input gain 30 35

fs = 48 kHz (3.3 V) 85 90
fs = 48 kHz (2.7 V) 90

pp

SOURCE

< 50 Ω –34.5 12 dB

50 dB

RMS

2.3.1.2 Microphone Input to ADC, 0-dB Gain, fs = 8 kHz (40-KΩ Source Impedance, see Section 1.2,
Functional Block Diagram)

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT

Input signal level (0 dB) 1.0 V

AVDD = 3.3 V 80 85
AVDD = 2.7 V 84

, –60-dB

, –1-dB

Power supply rejection ratio 1 kHz, 100 mV
Programmable gain boost 1 kHz input tone, R
Microphone-path gain MICBOOST = 0, R
Mute attenuation 0 dB, 1 kHz input tone 60 80 dB
Input resistance 8 14 kΩ
Input capacitance 10 pF

NOTES: 3. 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.

4. 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.

p

p

AVDD = 3.3 V 80 85
AVDD = 2.7 V 84
AVDD = 3.3 V –60
AVDD = 2.7 V –60

pp

SOURCE

SOURCE

< 50 Ω 20 dB

< 50 Ω 14 dB

50 dB

RMS

2–2

2.3.1.3 Microphone Bias

Signal-to-noise ratio, A-weighted, 0-dB gain (see Notes 3, 4, and 5)

dB

Dynamic range, A-weighted (see Note 4)

dB

AV

3.3 V

dB

Total harmonic distortion

AV

2.7 V

dB

Signal-to-noise ratio, A-weighted, 0-dB gain (see Notes 3 and 4)

dB

AV

3.3 V

dB

Total harmonic distortion

AV

2.7 V

dB

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT

Bias voltage 3/4 AVDD – 100 m 3/4 AVDD 3/4 AVDD + 100 m V
Bias-current source 3 mA
Output noise voltage 1 kHz to 20 kHz 25 nV/√Hz

2.3.2 DAC

2.3.2.1 Line Output, Load = 10 kΩ, 50 pF

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT

0-dB full-scale output voltage (FFFFFF) 1.0 V

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

=

DD

=

DD

Power supply rejection ratio 1 kHz, 100 mV
DAC channel separation 100 dB

NOTES: 3. 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.

4. 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.

5. 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.

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

RMS

2.3.3 Analog Line Input to Line Output (Bypass)

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT

0-dB full-scale output voltage 1.0 V

AVDD = 3.3 V 90 95
AVDD = 2.7 V 95

=

DD

=

DD

Power supply rejection ratio 1 kHz, 100 mV
DAC channel separation (left to right) 1 kHz, 0 dB 80 dB

NOTES: 3. 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.

4. 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.

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

RMS

2–3

2.3.4 Stereo Headphone Output

mW

Total harmonic distortion

DD

,

%

No in ut signal

Power down

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT

0-dB full-scale output voltage 1.0 V
Maximum output power, P

Signal-to-noise ratio, A-weighted (see Note 4) AVDD = 3.3 V 90 97 dB

Power supply rejection ratio 1 kHz, 100 mV
Programmable gain 1 kHz output –73 6 dB
Programmable-gain step size 1 dB
Mute attenuation 1 kHz output 80 dB

NOTE 4: 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.

O

RL = 32 Ω 30
RL = 16 Ω 40

AV

= 3.3 V,

1 kHz output

PO = 10 mW 0.1
PO = 20 mW 1.0

pp

50 dB

RMS

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 kΩ

2.3.6 Digital I/O

V

Input low level 0.3 × BV

IL

V

Input high level 0.7 × BV

IH

V

Output low level 0.1 × BV

OL

V

Output high level 0.9 × BV

OH

2.3.7 Supply Current

Total supply current,

I

TOT

p

PARAMETER MIN TYP MAX UNIT

DD

DD

DD

DD

PARAMETER MIN TYP MAX UNIT

Record and playback (all active) 23
Record and playback (osc, clk, and MIC output powered down) 18
Line playback only 7
Record only 13
Analog bypass (line in to line out) 4

Oscillator enabled 1.5
Oscillator disabled 0.01

mA

V
V
V
V

2–4

2.4 Digital-Interface Timing

System-clock pulse duration, MCLK/XTI

ns

t

w(1)

t

w(2)

t

System-clock period, MCLK/XTI 54 ns

c(1)

Duty cycle, MCLK/XTI 40/60% 60/40%

t

Propagation delay, CLKOUT 0 10 ns

pd(1)

p

PARAMETER MIN TYP MAX UNIT

High 18
Low 18

t

c(1)

t

w(1)

MCLK/XTI

CLKOUT

CLKOUT

(Div 2)

Figure 2–1. System-Clock Timing Requirements

2.4.1 Audio Interface (Master Mode)

PARAMETER MIN TYP MAX UNIT

t

Propagation delay, LRCIN/LRCOUT 0 10 ns

pd(2)

t

Propagation delay, DOUT 0 10 ns

pd(3)

t

Setup time, DIN 10 ns

su(1)

t

Hold time, DIN 10 ns

h(1)

BCLK

t

pd(2)

LRCIN

LRCOUT

t

pd(3)

t

w(2)

t

pd(1)

DOUT

DIN

t

su(1)

t

h(1)

Figure 2–2. Master-Mode Timing Requirements

2–5

2.4.2 Audio Interface (Slave-Mode)

Pulse duration, BCLK

ns

PARAMETER MIN TYP MAX UNIT

t

w(3)

t

w(4)

t

Clock period, BCLK 50 ns

c(2)

t

Propagation delay, DOUT 0 10 ns

pd(4)

t

Setup time, DIN 10 ns

su(2)

t

Hold time, DIN 10 ns

h(2)

t

Setup time, LRCIN 10 ns

su(3)

t

Hold time, LRCIN 10 ns

h(3)

High 20
Low 20

t

c(2)

BCLK

LRCIN

LRCOUT

DIN

DOUT

t

w(4)

t

w(3)

t

su(2)

t

t

pd(2)

h(2)

Figure 2–3. Slave-Mode Timing Requirements

t

h(3)

t

su(3)

2–6

2.4.3 Three-Wire Control Interface (SDIN)

Clock pulse duration, SCLK

ns

Pulse duration, CS

ns

Clock pulse duration, SCLK

PARAMETER MIN TYP MAX UNIT

t

w(5)

t

w(6)

t

c(3)

t

su(4)

t

su(5)

t

h(4)

t

w(7)

t

w(8)

p

Clock period, SCLK 80 ns
Clock rising edge to CS rising edge, SCLK 60 ns
Setup time, SDIN to SCLK 20 ns
Hold time, SCLK to SDIN 20 ns

CS

SCLK

High 20
Low 20

High 20
Low 20

t

c(3)

t

w(5)

t

w(6)

t

w(8)

t

su(4)

t

h(4)

LSB

DIN

t

su(5)

Figure 2–4. Three-Wire Control Interface Timing Requirements

2.4.4 Two-Wire Control Interface (I2C)

PARAMETER MIN TYP MAX UNIT

t

w(9)

t

w(10)

f(sf) Clock frequency, SCLK 0 400 kHz
t

h(5)

t

su(6)

t

h(6)

t

su(7)

t

r

t

f

t

su(8)

p

Hold time (start condition) 600 ns
Setup time (start condition) 600 ns
Data hold time 900 ns
Data setup time 100 ns
Rise time, SDIN, SCLK 300 ns
Fall time, SDIN, SCLK 300 ns
Setup time (stop condition) 600 ns

t

w(9)

SCLK

High 1.3 µs
Low 600 ns

t

w(10)

DIN

t

h(5)

t

h(6)

t

su(7)

Figure 2–5. Two-Wire Control Interface Timing Requirements

t

su(8)

2–7

2–8

3 How to Use the TLV320AIC23

3.1 Control Interfaces

The TLV320AIC23 has many programmable features. The control interface is used to program the registers of the
device. The control interface complies with SPI (three-wire operation) and I

2

C (two-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 I2C
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
TLV320AIC23. 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:

B[15:9] Control Address Bits
B[8:0] Control Data Bits

CS

SCLK

SDIN

B15 B14 B13 B12 B11 B10 B9 B8 B7 B6 B5 B4 B3 B2 B1 B0

MSB LSB

Figure 3–1. SPI Timing

3.1.2 I2C

In I2C 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

I

C bus receives the data. R/W determines the direction of the data transfer. The TL V320AIC23 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.

CS STATE

(Default = 0)

0 0011010
1 0011011

ADDRESS

3–1

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:

B[15:9] Control Address Bits
B[8:0] Control Data Bits

Start Stop

SCLK

SDI

1

7891

ADDR R/W ACK B15 – B8 ACK B7 – B0 ACK

89

18

9

Figure 3–2. 2-Wire I2C Compatible Timing

3.1.3 Register Map

The TLV320AIC23 has the following set of registers, which are used to program the modes of operation.

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

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

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

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 down to 0000000 = –73 dB in 1-dB steps
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

RLS 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 down to 0000000 = –73 dB in 1-dB steps
Analog Audio Path Control (Address: 0000100)

BIT D8 D7 D6 D5 D4 D3 D2 D1 D0

Function X STA1 STA0 STE DAC BYP INSEL MICM MICB

Default 0 0 0 0 1 0 0 1 0

ST A[1:0] Sidetone attenuation 00 = –6 dB 01 = –9 dB 10 = –12 dB 11 = –15 dB
STE Sidetone enable 0 = Disabled 1 = Enabled
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=OdB 1 = 20dB

X Reserved

3–3

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 0 = Disabled 1 = Enabled
X Reserved

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 1 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

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 Master/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 2nd BCLK rising edge after LRCIN rising edge

0 = MSB is available on 1st 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

2

10 = I

S format, MSB first, left – 1 aligned
01 = MSB first, left aligned
00 = MSB first, right aligned

X Reserved

NOTES: 1. In Master mode, the TLV320AIC23 supplies the BCLK, LRCOUT, and LRCIN. In Slave mode, BCLK, LRCOUT, and LRCIN are

supplied to the TL V320AIC23.

2. In master 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.

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

3–4

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

s
s

1 = 272 f
1 = 384 f

s
s

USB/Normal Clock mode select: 0 = Normal 1 = USB
X Reserved

Digital Interface Activation (Address: 0001001)

BIT D8 D7 D6 D5 D4 D3 D2 D1 D0

Function X X X X X X X X ACT

Default 0 0 0 0 0 0 0 0 1

ACT Activate interface 0 = Inactive 1 = Active
X Reserved

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

3.2 Analog Interface

3.2.1 Line Inputs

The TL V320AIC23 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 full-scale range
is 1.0 V
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
in Figure 3-3.

at A VDD = 3.3 V . The full-scale range tracks linearly with analog supply voltage AVDD. T o avoid distortions,

RMS

to avoid clipping, using the circuit shown

RMS

Where:

R1 = 5 kΩ
R2 = 5 kΩ
C1 = 47 pF
C2 = 470 nF

CDIN LINEIN

AGND

R1

C2 +

R

2

C1

Figure 3–3. Analog Line Input Circuit

R1 and R2 divide the input signal by two, reducing the 2 V

from the CD player to the nominal 1 V

RMS

of the AIC23

RMS

inputs. 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.

3–5

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
For example, R

MIC

) in series with MICIN, the gain of the first stage can be adjusted by G1 = 50 k/(10 k + R

MIC

MIC

= 40 k gives a gain of 0 dB. The second stage has a software programmable gain of 0 dB or 20

dB (see Section 3.1.3).

50 kΩ

).

MICIN

10 kΩ

VMID

To ADC

0 dB/20 dB

Figure 3–4. Microphone Input Circuit

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 TLV320AIC23 has two low-impedance line outputs (LLINEOUT and RLINEOUT) capable of driving line loads
with 10-kΩ and 50-pF impedances.

The DAC full-scale output voltage is 1.0 V
supply voltage AV

The DAC is connected to the line outputs via a low-pass filter that removes out-of-band

DD.

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.

at A VDD = 3.3 V. The full-scale range tracks linearly with the analog

RMS

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).

3.2.4 Headphone Output

The TL V320AIC23 has stereo headphone outputs (LHPOUT and RHPOUT), and is designed to drive 16-Ω or 32-Ω
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 time-out, 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).

3–6

3.2.5 Analog Bypass Mode

The TLV320AIC23 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 1.0V

at AVDD=3.3V to avoid clipping and distortion. This amplitude tracks linearly with AVDD.

rms

3.2.6 Sidetone Insertion

The TL V320AIC23 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, or –1 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 TL V320AIC23 supports four audio-interface modes.

• Right justified

• Left justified

2

• I

S 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.

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).

LRCIN/

LRCOUT

BCLK

DIN/

DOUT

Left Channel Right Channel

n n–1 01 n–1n

MSB LSB

1/fs

1 00

Figure 3–5. Right-Justified Mode Timing

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)

3–7

LRCIN/

LRCOUT

BCLK

DIN/

DOUT

Left Channel Right Channel

n n–1 01 n–1n

MSB LSB

1/fs

1 0 n

Figure 3–6. Left-Justified Mode Timing

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).

LRCIN/

LRCOUT

BCLK

1BCLK

DIN/

DOUT

Left Channel Right Channel

n n–1 01 n–1n

MSB LSB

1/fs

1 0

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).

LRCIN/

LRCOUT

BCLK

Left Channel Right Channel

DIN/

DOUT

n n–1 01 n–1n

MSB LSB MSB LSB

1 0

Figure 3–8. DSP Mode Timing

3–8

3.3.2 Audio Sampling Rates

SAMPLING RATE CONTROL SETTINGS

ADC

DAC

FILTER TYPE

The TLV320AIC23 can operate in master or slave clock mode. In the master mode, the TLV320AIC23 clock and
sampling rates are derived from a 12-MHz MCLK signal. This 12-MHz clock signal is compatible with the USB
specification. The TLV320AIC23 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 TL V320AIC23 clock and sampling rates.

The settings in the sample rate control register control the clock mode and sampling rates.
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/Nor-

Default 0 0 0 0 0 0 0 0 0

CLKOUT Clock output divider 0 = MCLK 1 = MCLK/2
CLKIN Clock input 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

s
s

1 = 272 f
1 = 384 f

s
s

USB/Normal Clock mode select: 0 = Normal 1 = USB
X Reserved

The clock circuit of the AIC23 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.

mal

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

(kHz)

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

†

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 Figures 3–17 through 3–34 for filter responses

†

(kHz)

-

SR3 SR2 SR1 SR0 BOSR

3–9

3.3.2.2 Normal-Mode Sampling Rates

SAMPLING RATE CONTROL SETTINGS

ADC

DAC

FILTER TYPE

SAMPLING RATE CONTROL SETTINGS

ADC

DAC

FILTER TYPE

SAMPLING RATE CONTROL SETTINGS

ADC

DAC

FILTER TYPE

SAMPLING RATE CONTROL SETTINGS

ADC

DAC

FILTER TYPE

In normal mode, the following ADC and DAC sampling rates, depending on the MCLK frequency, are available:

MCLK = 12.288 MHz

SAMPLING RATE

(kHz)

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

(kHz)

SR3 SR2 SR1 SR0 BOSR

-

MCLK = 11.2896 MHz

SAMPLING RATE

(kHz)

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

(kHz)

SR3 SR2 SR1 SR0 BOSR

-

MCLK = 18.432 MHz

SAMPLING RATE

-

(kHz)

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

(kHz)

SR3 SR2 SR1 SR0 BOSR

MCLK = 16.9344 MHz

SAMPLING RATE

(kHz)

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

(kHz)

SR3 SR2 SR1 SR0 BOSR

-

3–10

3.3.3 Digital Filter Characteristics

Corner frequency

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT

ADC Filter Characteristics ( TI DSP 250 fs Mode Operation )

Passband ±0.05 dB 0.416 f
Stopband –6 dB 0.5 f
Passband ripple ±0.05 dB
Stopband attenuation f > 0.584 f

ADC Filter Characteristics ( TI DSP 272 fs and Normal Mode Operation )

Passband ±0.05 dB 0.4535 f
Stopband –6 dB 0.5 f
Passband ripple ±0.05 dB
Stopband attenuation f > 0.5465 f

ADC High-Pass Filter Characteristics

–3 dB, fs = 44.1 kHz 3.7 Hz
–3 dB, fs = 48 kHz 4.0 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
Stopband –6 dB 0.5 f
Passband ripple ±0.03 dB
Stopband attenuation f > 0.584 f

DAC Filter Characteristics (44.1-kHz Sampling Rate)

Passband ±0.03 dB 0.4535 f
Stopband –6 dB 0.5 fs Hz
Passband ripple ±0.03 dB
Stopband attenuation f > 0.5465 f

s

s

s

s

s

s

–60 dB

s

s

–60 dB

s

s

–50 dB

s

–50 dB

Hz
Hz

Hz
Hz

Hz
Hz

Hz

3–11

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

0

–2

–4

–6

Filter Response – dB

–8

–10

0 0.1 0.2 0.3

Normalized Audio Sampling Frequency

0.4 0.5

Figure 3–9. Digital De-Emphasis Filter Response – 44.1 kHz Sampling

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

0

–2

–4

–6

Filter Response – dB

–8

–10

0 0.10 0.20 0.30

Normalized Audio Sampling Frequency

0.40 0.50

Figure 3–10. Digital De-Emphasis Filter Response – 48 kHz Sampling

3–12

Filter Response – dB

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

–70

–90

0 0.5 1 1.5

Normalized Audio Sampling Frequency

2 2.5 3

Figure 3–11. ADC Digital Filter Response I: TI DSP and Normal Modes

(Group Delay = 12 Output Samples)

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

0.10

0.08

0.06

0.04

0.02
0

–0.02
–0.04

Filter Response – dB

–0.06
–0.08

–0.10

0 0.05 0.1 0.15 0.2 0.25 0.3

Normalized Audio Sampling Frequency

0.35 0.4 0.45 0.5

Figure 3–12. ADC Digital Filter Ripple I: TI DSP and Normal Modes

(Group Delay = 20 Output Samples)

3–13

Filter Response – dB

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

–70

–90

0 0.5 1 1.5 2

Normalized Audio Sampling Frequency

Figure 3–13. ADC Digital Filter Response II: TI DSP Mode Only

FILTER RESPONSE

NORMALIZED AUDIO SAMPLING FREQUENCY

0.10

0.08

0.06

0.04

0.02
0

–0.02
–0.04

Filter Response – dB

–0.06
–0.08

–0.10

0 0.05 0.1 0.15 0.2 0.25 0.3

Normalized Audio Sampling Frequency

Figure 3–14. ADC Digital Filter Ripple II: TI DSP Mode Only

2.5 3

vs

0.35 0.4 0.45 0.5

3–14

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

Filter Response – dB

–70

–90

0 0.5 1 1.5

Normalized Audio Sampling Frequency

2 2.5 3

Figure 3–15. ADC Digital Filter Response III: TI DSP and Normal Modes

(Group Delay = 3 Output Samples)

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

0.4

0.3

0.2

0.1
0

–0.1

Filter Response – dB

–0.2
–0.3

–0.4

0 0.05 0.1 0.15 0.2 0.25 0.3

Normalized Audio Sampling Frequency

0.35 0.4 0.45 0.5

Figure 3–16. ADC Digital Filter Ripple III: TI DSP and Normal Modes

3–15

Filter Response – dB

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

–70

–90

0 0.5 1 1.5

Normalized Audio Sampling Frequency

Figure 3–17. ADC Digital Filter Response IV: TI DSP Mode Only

FILTER RESPONSE

NORMALIZED AUDIO SAMPLING FREQUENCY

0.4

0.3

0.2

0.1
0

–0.1
–0.2

Filter Response – dB

–0.3
–0.4

0 0.05 0.10 0.15 0.20 0.25 0.30

Normalized Audio Sampling Frequency

Figure 3–18. ADC Digital Filter Ripple IV: TI DSP Mode Only

2 2.5 3

vs

0.35 0.40 0.45 0.50

3–16

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

Filter Response – dB

–70

–90

0 0.5 1 1.5

Normalized Audio Sampling Frequency

2 2.5 3

Figure 3–19. DAC Digital Filter Response I: TI DSP and Normal Modes

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

0.10

0.08

0.06

0.04

0.02
0

–0.02
–0.04

Filter Response – dB

–0.06
–0.08

–0.10

0 0.05 0.1 0.15 0.2 0.25 0.3

Normalized Audio Sampling Frequency

0.35 0.4 0.45 0.5

Figure 3–20. DAC Digital Filter Ripple I: TI DSP and Normal Modes

3–17

Filter Response – dB

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

–70

–90

0 0.5 1 1.5

Normalized Audio Sampling Frequency

Figure 3–21. DAC Digital Filter Response II: TI DSP Mode Only

FILTER RESPONSE

NORMALIZED AUDIO SAMPLING FREQUENCY

0.10

0.08

0.06

0.04

0.02
0

–0.02

–0.04

Filter Response – dB

–0.06
–0.08

–0.10

0 0.05 0.1 0.15 0.2 0.25 0.3

Normalized Audio Sampling Frequency

Figure 3–22. DAC Digital Filter Ripple II: TI DSP Mode Only

2 2.5 3

vs

0.35 0.4 0.45 0.5

3–18

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

Filter Response – dB

–70

–90

0 0.5 1 1.5

Normalized Audio Sampling Frequency

2 2.5 3

Figure 3–23. DAC Digital Filter Response III: TI DSP and Normal Modes

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

0.4

0.3

0.2

0.1
0

–0.1
–0.2

Filter Response – dB

–0.3
–0.4

0 0.05 0.1 0.15 0.2 0.25 0.3

Normalized Audio Sampling Frequency

0.35 0.4 0.45 0.5

Figure 3–24. DAC Digital Filter Ripple III: TI DSP and Normal Modes

3–19

Filter Response – dB

FILTER RESPONSE

vs

NORMALIZED AUDIO SAMPLING FREQUENCY

10

–10

–30

–50

–70

–90

0 0.5 1 1.5

Normalized Audio Sampling Frequency

Figure 3–25. DAC Digital Filter Response IV: TI DSP Mode Only

FILTER RESPONSE

NORMALIZED AUDIO SAMPLING FREQUENCY

0.4

0.3

0.2

0.1
0

–0.1

Filter Response – dB

–0.2
–0.3

–0.4

0 0.05 0.1 0.15 0.2 0.25 0.3

Normalized Audio Sampling Frequency

Figure 3–26. DAC Digital Filter Ripple IV: TI DSP Mode Only

2 2.5 3

vs

0.35 0.4 0.45 0.5

3–20

Appendix A

Mechanical Data

GQE (S-PBGA-N80) PLASTIC BALL GRID ARRAY

0,68
0,62

5,10
4,90

0,35
0,25

SQ

∅ 0,05

4,00 TYP

0,50

J
H

G
F
E
D
C
B
A

321

4

1,00 MAX

Seating Plane

M

0,21
0,11

0,08

98765

4200461/C 10/00

0,50

NOTES: A. All linear dimensions are in millimeters.

B. This drawing is subject to change without notice.
C. MicroStar Junior BGA configuration
D. Falls within JEDEC MO-225

MicroStar Junior is a trademark of Texas Instruments.

A–1

PW (R-PDSO-G**) PLASTIC SMALL-OUTLINE PACKAGE

14 PINS SHOWN

0,65

1,20 MAX

14

0,30

0,19

8

4,50
4,30

PINS **

7

0,15
0,05

8

1

A

DIM

M

0,10

6,60
6,20

Seating Plane

0,10

14

0,15 NOM

0°–8°

2016

Gage Plane

24

0,25

0,75
0,50

28

A MAX

A MIN

NOTES: A. All linear dimensions are in millimeters.

B. This drawing is subject to change without notice.
C. Body dimensions do not include mold flash or protrusion not to exceed 0,15.
D. Falls within JEDEC MO-153

A–2

3,10

2,90

5,10

4,90

5,10

4,90

6,60

6,40

7,90

7,70

9,80

9,60

4040064/F 01/97