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.
Literature Number: SGLS240C
March 2004–Revised June 2012
TLV320AIC23B-Q1
www.ti.com
SGLS240C –MARCH 2004–REVISED JUNE 2012
Stereo Audio Codec, 8- to 96-kHz, With Integrated Headphone
Amplifier
Check for Samples: TLV320AIC23B-Q1
11 Introduction
1.1Features
1
• Qualified for Automotive Applications• Integrated Total Electret-Microphone Biasing
Compatible Both TI C54x DSP Buffer– Integrated Programmable Gain Amplifier
Voltages
– 8-kHz – 96-kHz Sampling-Frequency Support
• Software Control Via TI McBSP-Compatibleand 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
1.2Description
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,
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
www.ti.com
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.
AGND15Analog supply return
AVDD14Analog supply input. Voltage level is 3.3 V nominal.
BCLK3I/O
BVDD1Buffer supply input. Voltage range is from 2.7 V to 3.6 V.
CLKOUT2O
CS21IFor 2-wire control mode this input defines the seventh bit in the device address field. See Section 3.1 for
DIN4II2S format serial data input to the sigma-delta stereo DAC
DGND28Digital supply return
DOUT6OI2S format serial data output from the sigma-delta stereo ADC
DVDD27Digital supply input. Voltage range is 1.4 V to 3.6 V.
HPGND11Analog headphone amplifier supply return
HPVDD8Analog headphone amplifier supply input. Voltage level is 3.3 V nominal.
LHPOUT9O
LLINEIN20I
LOUT12OLeft stereo mixer-channel line output. Nominal output level is 1.0 VRMS.
LRCIN5I/O
LRCOUT7I/O
MICBIAS17O
MICIN18I
MODE22ISerial-interface-mode input. See Section 3.1 for details.
NCNot Used—No internal connection
RHPOUT10O
RLINEIN19I
ROUT13ORight stereo mixer-channel line output. Nominal output level is 1.0 VRMS.
SCLK24I
SDIN23I
VMID16I
XTI/MCLK25ICrystal or external-clock input. Used for derivation of all internal clocks on the AIC23B.
XTO26O
I/ODESCRIPTION
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.
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 .63V
Input voltage range, all input signals
Case temperature for 10 seconds240°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 OperatingConditions 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
2.2Recommended 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)1V
Stereo-line output load resistance10kΩ
Headphone-amplifier output load resistance0Ω
CLKOUT digital output load capacitance20pF
All other digital output load capacitance10pF
Stereo-line output load capacitance50pF
XTI master clock Input18.43MHz
ADC or DAC conversion rate96kHz
Operating free-air temperature, T
(1) Digital voltage values are with respect to DGND; analog voltage values are with respect to AGND.
(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
2.3.1.2Microphone Input to ADC
0-dB Gain, fs= 8 kHz (40-KΩ source impedance, see Section 1.3)
(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.
(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
50dB
DAC channel separation (left to right)1 kHz, 0 dB80dB
(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.
(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.
tpd(2)Propagation delay, LRCIN/LRCOUT010ns
tpd(3)Propagation delay, DOUT010ns
tsu(1)Setup time, DIN10ns
th(1)Hold time, DIN10ns
Figure 2-2. Master-Mode Timing Requirements
2.4.2Audio Interface (Slave-Mode)
PARAMETERMINTYPMAX UNIT
tw(3)High20
tw(4)Low20
tc(2)Clock period, BCLK50ns
tpd(4)Propagation delay, DOUT010ns
tsu(2)Setup time, DIN10ns
th(2)Hold time, DIN10ns
tsu(3)Setup time, LRCIN10ns
th(3)Hold time, LRCIN10ns
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
TLV320AIC23B-Q1
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SGLS240C –MARCH 2004–REVISED JUNE 2012
2.4.42-Wire Control Interface
PARAMETERMINTYPMAX UNIT
tw(9)High1.3µs
tw(10)Low600ns
f(sf)Clock frequency, SCLK0400kHz
th(5)Hold time (start condition)600ns
tsu(6)Setup time (start condition)600ns
th(6)Data hold time900ns
tsu(7)Data setup time100ns
trRise time, SDIN, SCLK300ns
tfFall time, SDIN, SCLK300ns
tsu(8)Setup time (stop condition)600ns
tspPulse width of spikes suppressed by input filter050ns
Clock pulse duration, SCLK
Figure 2-5. 2-Wire Control Interface Timing Requirements
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.
MODEINTERFACE
02-wire
1SPI
3.1.1SPI
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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3.1.22-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
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
3.1.3Register 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).
ADDRESSREGISTER
0000000Left line input channel volume control
0000001Right line input channel volume control
0000010Left channel headphone volume control
0000011Right channel headphone volume control
0000100Analog audio path control
0000101Digital audio path control
0000110Power down control
0000111Digital audio interface format
0001000Sample rate control
0001001Digital interface activation
0001111Reset register
Table 3-1. Left Line Input Channel Volume Control (Address: 0000000)
MSMaser/slave mode0 = Slave1 = Master
LRSWAPDAC left/right swap0 = Disabled1 = Enabled
LRPDAC left/right phase0 = 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 length00 = 16 bit01 = 20 bit10 = 24 bit11 = 32 bit
FOR[1:0]Data format11 = 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
XReserved
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.
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.
3.2.2Microphone 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
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.
3.2.3Line Outputs
The TLV320AIC23B-Q1 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 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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3.2.4Headphone Output
The TLV320AIC23B-Q1 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 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).
3.2.5Analog 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
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.
3.3Digital Audio Interface
3.3.1Digital 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
3.3.1.1Right-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).
3.3.1.2Left-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).
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
3.3.1.4DSP 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).
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)
CLKINClock input divider0 = MCLK1 = MCLK/2
CLKOUTClock output divider0 = MCLK1 = MCLK/2
SR[3:0]Sampling rate control (see Sections 3.3.2.1 AND 3.3.2.2)
BOSRBase oversampling rate
USB mode:0 = 250 f
Normal mode:0 = 256 f
USB/NormalClock mode select:0 = Normal1 = USB
XReserved
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.
3.3.2.1USB-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)SR3SR2SR1SR0BOSR
9696301110
88.288.2211111
4848000000
44.144.1110001
3232001100
8.0218.021110111
88000110
488000010
44.18.021110011
848000100
8.02144.1110101
(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
3.3.2.2Normal-Mode Sampling Rates
In normal mode, the following ADC and DAC sampling rates, depending on the MCLK frequency, are
available:
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).
6PAIC23BIPWRG4Q1ACTIVETSSOPPW282000RoHS & GreenNIPDAULevel-1-260C-UNLIM-40 to 85AIC23BIQ1
TLV320AIC23BIPWRQ1ACTIVETSSOPPW282000RoHS & GreenNIPDAULevel-1-260C-UNLIM-40 to 85AIC23BIQ1
(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.
Important Information and Disclaimer:The information provided on this page represents TI's knowledge and belief as of the date that it is provided. TI bases its knowledge and belief on information
provided by third parties, and makes no representation or warranty as to the accuracy of such information. Efforts are underway to better integrate information from third parties. TI has taken and
continues to take reasonable steps to provide representative and accurate information but may not have conducted destructive testing or chemical analysis on incoming materials and chemicals.
TI and TI suppliers consider certain information to be proprietary, and thus CAS numbers and other limited information may not be available for release.
In no event shall TI's liability arising out of such information exceed the total purchase price of the TI part(s) at issue in this document sold by TI to Customer on an annual basis.
Samples
Addendum-Page 1
PACKAGE OPTION ADDENDUM
www.ti.com
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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