• Stereo 1.29-W Class-D BTL 8-Ω Speaker Driver
With Direct Battery Connection
• 25 Built-In Processing Blocks (PRB_P1 –
PRB_P25) Providing Biquad Filters, DRC, and 3D
• Digital Sine-Wave Generator for Beeps and KeyClicks (PRB_P25)
• User-Programmable Biquad and FIR Filters
• Two Single-Ended Inputs With Mixing and Output
Level Control
• Stereo Headphone or Lineout and Class-D
Speaker Outputs Available
• Microphone Bias
• Headphone Detection
• Digital Mixing Capability
• Pin Control or Register Control for Digital-Playback
Volume-Control Settings
• Programmable PLL for Flexible Clock Generation
• I2S, Left-Justified, Right-Justified, DSP, and TDM
Audio Interfaces
• I2C Control With Register Auto-Increment
• Full Power-Down Control
• Power Supplies:
– Analog: 2.7 V–3.6 V
– Digital Core: 1.65 V–1.95 V
– Digital I/O: 1.1 V–3.6 V
– Class-D: 2.7 V–5.5 V (SPLVDD and SPRVDD ≥
AVDD)
• 5-mm × 5-mm 32-QFN Package
TLV320DAC3101
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
1.2Applications
•Portable Audio Devices
•Mobile Internet Devices
•e-Books
1.3Description
The TLV320DAC3101 device is a low-power,
highly integrated, high-performance DAC with
selectable digital audio processing blocks and
24-bit stereo playback.
The device integrates headphone drivers and
speaker drivers. The TLV320DAC3101 device
has a suite of built-in processing blocks for
digital audio processing. The digital audio data
format is programmable to work with popular
audio standard protocols (I2S, left-justified, and
right-justified) in master, slave, DSP, and TDM
modes. Bass boost, treble, or EQ is supported
by the programmable digital signal-processing
block. An on-chip PLL provides the high-speed
clock needed by the digital signal-processing
block. The volume level is controlled by either
pin control or by register control. The audio
functions are controlled using the I2C serial
bus.
TheTLV320DAC3101devicehasa
programmable digital sine-wave generator and
is available in a 32-pin QFN package.
Device Information
PART NUMBERPACKAGEBODY SIZE
TLV320DAC3101QFN (RHB)5.00 mm x 5.00 mm
(1) For more information, see , Mechanical, Packaging, and
Orderable Information.
(1)
1
An IMPORTANT NOTICE at the end of this data sheet addresses availability, warranty, changes, use in safety-critical applications,
intellectual property matters and other important disclaimers. PRODUCTION DATA.
This data manual is designed using PDF document-viewing features that allow quick access
to information. For example, performing a global search on page 0 / register 27 produces all
references to this page and register in a list. This makes it easy to traverse the list and find
all information related to a page and register. Note that the search string must be of the
indicated format. Also, this document includes document hyperlinks to allow the user to
quickly find a document reference. To come back to the original page, click the green left
arrow near the PDF page number at the bottom of the file. The hot-key for this function is altleft arrow on the keyboard. A different way to find information quickly is to use the PDF
NOTE
bookmarks.
Submit Documentation Feedback
Product Folder Links: TLV320DAC3101
TLV320DAC3101
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SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
2Revision History
NOTE: Page numbers for previous revisions may differ from page numbers in the current version.
Changes from Revision A (May 2012) to Revision BPage
•Added Device Information table, ESD Ratings table, Feature Description section, Application and
Implementation section, Power Supply Recommendations section, Layout section, Device and Documentation
Support section, and Mechanical, Packaging, and Orderable Information section. ......................................... 1
•Added Power-Supply Sequence section to the Device Initialization section ................................................ 19
•Changed Section 6.3.10.1.2 diagrams for PRB_P2/5/8/10/13/15/18/21/24/25 to reflect that the DRC_HPF filter
cannot be bypassed when the DRC is turned off .............................................................................. 26
•Added sequence for inserting a beep in the middle of an already-playing signal and note text following script in
the Key-Click Functionality With Digital Sine-Wave Generator (PRB_P25) section........................................ 42
•Changed PRB_Rx to PRB_Px in DAC Setup section.......................................................................... 48
•Changed text from: "the rising edge of the word clock..." To: "the rising edge of the word clock..." in the DSP Mode 60
•Changed DOSR note in Page 0 / Register 14 by switching multiple value for Filter Type A and Filter Type C........ 68
•Changed description in Page 0 / Register 14 to remove parameters for miniDSP......................................... 68
•Changed reset value to include all bits instead of just two (xx) ............................................................... 74
•Deleted reference to Dig_Mic_In in Page 0 / Register 54 table for bits D2-D1 ............................................. 75
•Changed values in Page 0 / Register 69 (0x45): DRC Control 2 ............................................................. 78
•Changed Page 0, Register 70, bit D3-D0 decay rate value for 0000 from DR = 1.5625e
•Switched D1 and D0 descriptions so that D1 is for SP and D0 is for HP in Page 1 / Register 30 table ................ 81
•Changed Page 1 / Register 40, D1 to reserved................................................................................. 84
•Changed Page 1 / Register 41, D1 to reserved................................................................................. 84
AIN113IAnalog input #1 routed to output mixer
AIN214IAnalog input #2 routed to output mixer
AVDD17–Analog power supply
AVSS16–Analog ground
BCLK7I/OAudio serial bit clock
DIN5IAudio serial data input
DVDD3–Digital power – digital core
DVSS18–Digital ground
GPIO132I/OGeneral-purpose input/output and multifunction pin
HPL27OLeft-channel headphone/line driver output
HPR30ORight-channel headphone/line driver output
HPVDD28–Headphone/line driver and PLL power
HPVSS29–Headphone/line driver and PLL ground
IOVDD2–Interface power
IOVSS1–Interface ground
MCLK8IExternal master clock
MICBIAS12–Microphone bias for external microphone
NC4, 15INo connecton
RESET31IDevice reset
SDL10I/OI2C control bus clock input
SDA9I/OI2C control bus data input
SPLM19OLeft-channel class-D speaker-driver inverting output
over operating free-air temperature range (unless otherwise noted)
AVDD to AVSS-0.33.9V
DVDD to DVSS-0.32.5V
HPVDD to HPVSS-0.33.9V
SPLVDD to SPLVSS-0.36V
SPRVDD to SPRVSS-0.36V
IOVDD to IOVSS-0.33.9V
Digital input voltageIVOSS – 0.3IVODD + 0.3V
Analog input voltageAVSS – 0.3AVDD + 0.3V
Operating temperature–4085°C
Junction temperature (TJ Max)105°C
Storage temperature, T
Power dissipation(TJMax - TA)/R
R
thermal impedance (with thermal pad soldered to board)35°C/W
θJA
(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.
stg
(1)
MINMAXUNIT
-55150°C
θJA
W
4.2ESD Ratings
VALUEUNIT
V
(ESD)
Electrostatic
discharge
Human-body model (HBM), per ANSI/ESDA/JEDEC JS-001
Charged-device model (CDM), per JEDEC specification JESD22-C101
(1)
(2)
±2000
±1000
V
(1) JEDEC document JEP155 states that 500-V HBM allows safe manufacturing with a standard ESD control process.
(2) JEDEC document JEP157 states that 250-V CDM allows safe manufacturing with a standard ESD control process.
4.3Recommended Operating Conditions
over operating free-air temperature range (unless otherwise noted)
MINNOMMAXUNIT
(1)
AVDD
Referenced to AVSS
DVDDReferenced to DVSS
HPVDDReferenced to HPVSS
SPLVDD
SPRVDD
(1)
(1)
Power-supply voltage
Referenced to SPLVSS
Referenced to SPRVSS
IOVDDReferenced to IOVSS
Speaker impedance
Resistance applied across class-D
output pins (BTL)
Headphone impedanceAC coupled to R
V
I
Analog audio full-scale input voltageAVDD = 3.3 V, single-ended0.707V
Stereo line output load impedanceAC coupled to R
(1) To minimize battery-current leakage, the SPLVDD and SPRVDD voltage levels must not be below the AVDD voltage level.
(2) All grounds on board are tied together, so they must not differ in voltage by more than 0.2-V maximum for any combination of ground
signals. By use of a wide trace or ground plane, ensure a low-impedance connection between HPVSS and DVSS.
(3) The maximum input frequency must be 50 MHz for any digital pin used as a general-purpose clock.
(1) Ratio of output level with 1-kHz full-scale sine-wave input, to the output level with the inputs 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 may
result 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) DAC to headphone-out PSRR measurement is calculated as PSRR = 20 × log(ΔV
This data was taken using 2-oz. (0,071-mm thick) trace and copper pad that is soldered to a JEDEC high-K, standard 4-layer
3-inch × 3-inch (7,62-cm × 7,62-cm) PCB.
Power Rating at 25°CDerating FactorPower Rating at 70°CPower Rating at 85°C
Note: All timing specifications are measured at characterization.
PARAMETER
f
SCL
t
HD;STA
t
LOW
t
HIGH
t
SU;STA
t
HD;DAT
t
SU;DAT
t
r
t
f
t
SU;STO
t
BUF
C
b
SCL clock frequency01000400kHz
Hold time (repeated) START condition.
After this period, the first clock pulse is
generated.
LOW period of the SCL clock4.71.3μs
HIGH period of the SCL clock40.6μs
Setup time for a repeated START
condition
Data hold time: for I2C bus devices03.4500.9μs
Data set-up time250100ns
SDA and SCL rise time100020 + 0.1C
SDA and SCL fall time30020 + 0.1C
Set-up time for STOP condition40.8μs
Bus free time between a STOP and
START condition
Capacitive load for each bus line400400pF
The device is a highly integrated stereo-audio DAC for portable computing, communication, and
entertainment applications. A register-based architecture eases integration with microprocessor-based
systems through standard serial-interface buses. This device supports the two-wire I2C bus interface
which provides full register access. All peripheral functions are controlled through these registers and the
onboard state machines.
The device consists of the following blocks:
•Stereo Audio DAC
•Dynamic Range Compressor (DRC)
•Digital sine-wave generator for clicks and beeps
•Stereo headphone and lineout amplifier
•Pin-controlled or register-controlled volume level
•Power-down de-pop and power-up soft start
•Analog inputs
•I2C control interface
•Power-down control block
Following a toggle of the RESET pin or a software reset, the device operates in the default mode. The I2C
interface is used to write to the control registers to configure the device.
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The I2C address assigned to the device is 001 1000. This device always operates in an I2C slave mode.
All registers are 8-bit, and all writable registers have read-back capability. The device auto-increments to
support sequential addressing and can be used with the I2C fast mode. When the device is reset, all
appropriate registers are updated by the host processor to configure the device as needed by the user.
The requires multiple power supply rails for operation. All the power rails must be powered up for the
device to operate at the fullest potention. The following is the recommended power-up sequencing for
proper operation:
1. Power up SPLVDD and SPRVDD
2. Power up IOVDD
3. Power up DVDD shortly after IOVDD
4. Power up AVDD and HPVDD
Although not necessary, if the system requires, during shutdown, remove the power supplies in the
The internal logic must be initialized to a known condition for proper device function. To initialize the
device to its default operating condition, the hardware reset pin (RESET) must be pulled low for at least 10
ns. For this initialization to work, both the IOVDD and DVDD supplies must be powered up. TI
recommends that while the DVDD supply powers up, the RESET pin is pulled low.
The device can also be reset via software reset. Writing a 1 into page 0 / register 1, bit D0 resets the
device.
6.3.3Device Start-Up Lockout Times
After the is initialized through hardware reset at power up or software reset, the internal memories are
initialized to default values. This initialization takes place within 1 ms after pulling the RESET signal high.
During this initialization phase, no register-read or register-write operation should be performed on DAC
coefficient buffers. Also, no block within the codec should be powered up during the initialization phase.
6.3.4PLL Start-Up
Whenever the PLL is powered up, a start-up delay of approximately of 10 ms occurs after the power-up
command of the PLL and before the clocks are available to the codec. This delay is to ensure stable
operation of the PLL and clock-divider logic.
6.3.5Power-Stage Reset
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The power-stage-only reset is used to reset the device after an overcurrent latching shutdown has
occurred. Using this reset re-enables the output stage without resetting all of the registers in the device.
Each of the four power stages has its own dedicated reset bit. The headphone power-stage reset is
performed by setting page 1 / register 31, bit D7 for HPL and by setting page 1 / register 31, bit D6 for
HPR. The speaker power-stage reset is performed by setting page 1 / register 32, bit D7 for SPLP and
SPLM, and by setting page 1 / register 32, bit D6 for SPRP and SPRM.
6.3.6Software Power Down
By default, all circuit blocks are powered down following a reset condition. Hardware power up of each
circuit block can be controlled by writing to the appropriate control register. This approach allows the
lowest power-supply current for the functionality required. However, when a block is powered down, all of
the register settings are maintained as long as power is still being applied to the device.
6.3.7Audio Analog I/O
The has a stereo audio DAC. The device supports a wide range of analog interfaces to support different
headsets and analog outputs. The has features to interface output drivers (8-Ω, 16-Ω, 32-Ω). A special
circuit has also been included in the to insert a short key-click sound into the stereo audio output. The keyclick sound is used to provide feedback to the user when a particular button is pressed or item is selected.
The specific sound of the keyclick can be adjusted by varying several register bits that control its
frequency, duration, and amplitude (see Section 6.3.10.7).
6.3.8Digital Processing Low-Power Modes
The device can be tuned to minimize power dissipation, to maximize performance, or to an operating point
between the two extremes to best fit the application. The choice of processing blocks, PRB_P1 to
PRB_P25 for stereo playback, also influences the power consumption. In fact, the numerous processing
blocks have been implemented to offer a choice among configurations having a different balance of power
optimization and signal-processing capabilities.
•Analog outputs, class-D speaker driver and headphone and lineout driver, providing output capability
for the DAC, AIN1, AIN2 or a mix of the three
6.3.9.1MICBIAS
The device includes a microphone bias circuit that sources up to 4 mA of current and is programmable to
a 2-V, 2.5-V, or AVDD level. The level is controlled by writing to page 1 / register 46, bits D1–D0. Table 6-
10 lists this functionality.
D1D0FUNCTIONALITY
00MICBIAS output is powered down
01MICBIAS output is powered to 2 V
10MICBIAS output is powered to 2.5 V
11MICBIAS output is powered to AVDD
During normal operation, MICBIAS can be set to 2.5 V for better performance. However, based on the
model of the selected microphone, optimal performance can be obtained at another setting and therefore
the performance at a given setting must be verified.
The lowest current consumption occurs when MICBIAS is powered down. The next-lowest current
consumption occurs when MICBIAS is set at AVDD.
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Table 6-10. MICBIAS Settings
6.3.9.2Analog Inputs AIN1 and AIN2
AIN1 (pin 13) and AIN2 (pin 14) are inputs to the output mixer along with the DAC output. Page 1 /
register 35 provides control signals for determining the signals routed through the output mixer. The output
of the output mixer then can be attenuated or gained through the class-D and, or, headphone and lineout
drivers.
6.3.10 Audio DAC and Audio Analog Outputs
Each channel of the stereo audio DAC consists of a digital-audio processing block, a digital interpolation
filter, a digital delta-sigma modulator, and an analog reconstruction filter. This high oversampling ratio
(typically DOSR is between 32 and 128) exhibits good dynamic range by ensuring that the quantization
noise generated within the delta-sigma modulator stays outside of the audio frequency band. Audio analog
outputs include stereo headphone, or lineouts, and stereo class-D speaker outputs.
6.3.10.1 DAC
The stereo-audio DAC supports data rates from 8 kHz to 192 kHz. Each channel of the stereo audio-DAC
consists of a signal-processing engine with fixed processing blocks, a digital interpolation filter, a multibit
digital delta-sigma modulator, and an analog reconstruction filter. The DAC is designed to provide
enhanced performance at low sampling rates through increased oversampling and image filtering, thereby
keeping quantization noise generated within the delta-sigma modulator and signal images strongly
suppressed within the audio band to beyond 20 kHz. To handle multiple input rates and optimize power
dissipation and performance, the device allows the system designer to program the oversampling rates
over a wide range from 1 to 1024 by configuring page 0 / register 13 and page 0 / register 14. The system
designer can choose higher oversampling ratios for lower input data rates and lower oversampling ratios
for higher input data rates.
24
The DAC channel includes a built-in digital interpolation filter to generate oversampled data for the deltasigma modulator. The interpolation filter can be chosen from three different types, depending on required
frequency response, group delay, and sampling rate.
DAC power up is controlled by writing to page 0 / register 63, bit D7 for the left channel and bit D6 for the
right channel. The left-channel DAC clipping flag is provided as a read-only bit on page 0 / register 39, bit
D7. The right-channel DAC clipping flag is provided as a read-only bit on page 0 / register 39, bit D6.
6.3.10.1.1 DAC Processing Blocks
The device implements signal-processing capabilities and interpolation filtering through processing blocks.
These fixed processing blocks give users the choice of how much and what type of signal processing they
use and which interpolation filter is applied.
The choices among these processing blocks allow the system designer to balance power conservation
and signal-processing flexibility. Table 6-11 gives an overview of all available processing blocks of the
DAC channel and their properties. The resource-class column gives an approximate indication of power
consumption for the digital (DVDD) supply; however, based on the out-of-band noise spectrum, the analog
power consumption of the drivers (HPVDD) may differ.
The signal processing blocks available are:
•First-order IIR
•Scalable number of biquad filters
•3D effect
•Digital sine-wave (beep) generator
The processing blocks are tuned for common cases and can achieve high image rejection or low group
delay in combination with various signal-processing effects such as audio effects and frequency shaping.
The available first-order IIR and biquad filters have fully user-programmable coefficients.
6.3.10.1.2.11 Five Biquads, DRC, 3D, Beep Generator, Filter A
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
Figure 6-11. Signal Chain for PRB_P25
6.3.10.1.3 DAC User-Programmable Filters
Based on the selected processing block, different types and orders of digital filtering are available. Up to
six biquad sections are available for specific processing blocks.
The coefficients of the available filters are arranged as sequentially-indexed coefficients in two banks. If
adaptive filtering is chosen, the coefficient banks can be switched in real time.
When the DAC is running, the user-programmable filter coefficients are locked and cannot be accessed
for either read or write.
However, the device offers an adaptive filter mode as well. Setting page 8 / register 1, bit D2 = 1 turns on
double buffering of the coefficients. In this mode, filter coefficients are updated through the host and
activated without stopping and restarting the DAC which enables advanced adaptive filtering applications.
In the double-buffering scheme, all coefficients are stored in two buffers (buffers A and B). When the DAC
is running and the adaptive filtering mode is turned on, setting page 8 / register 1, bit D0 = 1 switches the
coefficient buffers at the next start of a sampling period. This bit is set back to 0 after the switch occurs. At
the same time, page 8 / register 1, bit D1 toggles.
The flag in page 8 / register 1, bit D1 indicates which of the two buffers is actually in use.
Page 8 / register 1, bit D1 = 0: buffer A is in use by the DAC processing block; bit D1 = 1: buffer B is in
use.
While the device is running, coefficient updates are always made to the buffer not in use by the DAC,
regardless of the buffer to which the coefficients have been written.
Yes0Buffer ABuffer A (Pages 8 and9)Buffer B (Pages 12
Yes0Buffer ABuffer B (Pages 12 and
Yes1Buffer BBuffer A (Pages 8 and9)Buffer A (Pages 8
Yes1Buffer BBuffer B (Pages 12 and
PAGE 8 / REGISTER 1, BIT D1
The user-programmable coefficients for the DAC processing blocks are defined on pages 8 and 9 for
buffer A and pages 12 and 13 for buffer B.
The coefficients of these filters are each 16-bit, 2s-complement format, occupying two consecutive 8-bit
registers in the register space. Specifically, the filter coefficients are in 1.15 (one dot 15) format with a
range from –1.0 (0x8000) to 0.999969482421875 (0x7FFF) as shown in Figure 6-12.
COEFFICIENT BUFFER IN
USE
WRITING TOUPDATES
and 13)
13)
13)
Buffer B (Pages 12
and 13)
and 9)
Buffer A (Pages 8
and 9)
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Figure 6-12. 1.15 2s-Complement Coefficient Format
6.3.10.1.3.1 First-Order IIR Section
The IIR is of first order and its transfer function is given by Equation 1.
The frequency response for the first-order IIR section with default coefficients is flat.
Figure 6-14. Frequency Response of Channel Interpolation Filter B
6.3.10.1.4.3 Interpolation Filter C
Filter C is specifically designed for the 192-ksps mode. The pass band extends up to 0.4 × f
(corresponds to 80 kHz), more than sufficient for audio applications.
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
S
PARAMETERCONDITIONVALUE (TYPICAL)UNIT
Filter-gain pass band0 … 0.35 f
Filter-gain stop band0.6… 1.4 f
Filter group delay13 / f
Figure 6-15. Frequency Response of DAC Interpolation Filter C
Table 6-17. Specification for DAC Interpolation Filter C
The DAC has a digital-volume control block which implements programmable gain. Each channel has an
independent volume control that can be varied from 24 dB to –63.5 dB in 0.5-dB steps. The left-channel
DAC volume is controlled by writing to page 0 / register 65, bits D7–D0. The right-channel DAC volume
can be controlled by writing to page 0 / register 66, bits D7–D0. DAC muting and setting up a master gain
control to control both channels occurs by writing to page 0 / register 64, bits D3–D0. The gain is
implemented with a soft-stepping algorithm, which only changes the actual volume by 0.125 dB per input
sample, either up or down, until the desired volume is reached. The rate of soft-stepping is slowed to one
step per two input samples by writing to page 0 / register 63, bits D1–D0. Note that the default source for
volume-control level settings is control by register writes (page 0 / register 65 and page 0 / register 66 to
control volume). Use of the VOL/MICDET pin to control the DAC volume is ignored until the volume
control source selected has been changed to pin control (page 0 / register 116, bit D7 = 1). This
functionality is shown in Figure 1-1.
During soft-stepping, the host does not receive a signal when the DAC has been completely muted. This
may be important if the host must mute the DAC before making a significant change, such as changing
sample rates. In order to help with this situation, the device provides a flag back to the host through a
read-only register, page 0 / register 38, bit D4 for the left channel and bit D0 for the right channel. This
information alerts the host when the part has completed the soft-stepping and the actual volume has
reached the desired volume level. The soft-stepping feature can be disabled by writing to page 0 /
register 63, bits D1–D0.
If soft-stepping is enabled, the CODEC_CLKIN signal must be kept active until the DAC power-up flag is
cleared. When this flag is cleared, the internal DAC soft-stepping process is complete, and
CODEC_CLKIN can be stopped if desired. (The analog volume control can be ramped down using an
internal oscillator.)
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6.3.10.3 Volume Control Pin
The volume-control pin is not enabled by default but is enabled by writing 1 to page 0 / register 116, bit
D7. The default DAC volume control uses software control of the volume, which occurs if page 0 /
register 116, bit D7 = 0. Soft-stepping the volume level is set up by writing to page 0 / register 63, bits
D1–D0.
When the volume-pin function is used, a 7-bit Vol ADC reads the voltage on the VOL/MICDET pin and
updates the digital volume control by overwriting the current value of the volume control. The new volume
setting which has been applied because of a change of voltage on the volume control pin is read on
page 0 / register 117, bits D6–D0. The 7-bit Vol ADC clock source is selected on page 0 / register 116, bit
D6. The update rate is programmed on page 0 / register 116, bits D2–D0 for this 7-bit SAR ADC.
Figure 6-16 shows the VOL/MICDET pin connection and functionality.
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
Table 6-18. VOL/MICDET Pin Gain Mapping (continued)
VOL/MICDET PIN SAR OUTPUTDIGITAL GAIN APPLIED
91–28 dB
::
125–62 dB
126–63 dB
127Mute
Figure 6-16. Digital Volume Controls for Beep Generator and DAC Play Data
As shown in Table 6-18, the VOL/MICDET pin has a range of volume control from 18 dB down to –63 dB,
and mute. However, if less maximum gain is required, then a smaller range of voltage must be applied to
the VOL/MICDET pin. Applying a smaller range of voltage occurs by increasing the value of R2 relative to
the value of (P1 + R1), so that more voltage is available at the bottom of P1. The circuit must also be
designed such that for the values of R1, R2, and P1 chosen, the maximum voltage (top of the
potentiometer) does not exceed AVDD/2 (see Figure 6-16). The recommended values for R1, R2, and P1
for several maximum gains are shown in Table 6-19.
Table 6-19. VOL/MICDET Pin Gain Scaling
R1
(kΩ)
252500 to 1.6518 to –63
33257.680.386 to 1.6423 to –63
Typical music signals are characterized by crest factors, the ratio of peak signal power to average signal
power, of 12 dB or more. To avoid audible distortions due to clipping of peak signals, the gain of the DAC
channel must be adjusted so as not to cause hard clipping of peak signals. As a result, during nominal
periods, the applied gain is low, causing the perception that the signal is not loud enough. To overcome
this problem, dynamic range conpression (DRC) in the continuously monitors the output of the DAC digital
volume control to detect its power level relative to 0 dBFS. When the power level is low, DRC increases
the input signal gain to make it sound louder. At the same time, if a peaking signal is detected, it
autonomously reduces the applied gain to avoid hard clipping. This results in sounds more pleasing to the
ear as well as sounding louder during nominal periods.
The DRC functionality in the is implemented by a combination of processing blocks in the DAC channel as
described in Section 6.3.10.1.2.
DRC can be disabled by writing to page 0 / register 68, bits D6–D5.
DRC typically works on the filtered version of the input signal. The input signals have no audio information
at dc and extremely low frequencies; however, they can significantly influence the energy estimation
function in the dynamic range compressor (the DRC). Also, most of the information about signal energy is
concentrated in the low-frequency region of the input signal.
To estimate the energy of the input signal, the signal is first fed to the DRC high-pass filter and then to the
DRC low-pass filter. These filters are implemented as first-order IIR filters given by
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(3)
(4)
The coefficients for these filters are 16 bits wide in 2s-complement format and are user-programmable
through register write as given in Table 6-20.
The default values of these coefficients implement a high-pass filter with a cutoff at 0.00166 × DAC_fS,
and a low-pass filter with a cutoff at 0.00033 × DAC_fS.
The output of the DRC high-pass filter is fed to the processing block selected for the DAC channel. The
absolute value of the DRC LPF filter is used for energy estimation within the DRC.
The gain in the DAC digital volume control is controlled by page 0 / register 65 and page 0 / register 66.
When the DRC is enabled, the applied gain is a function of the digital volume control register setting and
the output of the DRC.
36
The DRC parameters are described in sections that follow.
DRC threshold represents the level of the DAC playback signal at which the gain compression becomes
active. The output of the digital volume control in the DAC is compared with the set threshold. The
threshold value is programmable by writing to page 0 / register 68, bits D4–D2. The threshold value can
be adjusted between –3 dBFS and –24 dBFS in steps of 3 dB. Keeping the DRC threshold value too high
may not leave enough time for the DRC block to detect peaking signals, and can cause excessive
distortion at the outputs. Keeping the DRC threshold value too low can limit the perceived loudness of the
output signal.
The recommended DRC threshold value is –24 dB.
When the output signal exceeds the set DRC threshold, the interrupt flag bits at page 0 / register 44,
bits D3–D2 are updated. These flag bits are sticky in nature, and are reset only after they are read back
by the user. The non-sticky versions of the interrupt flags are also available at page 0 / register 46,
bits D3–D2.
6.3.10.4.2 DRC Hysteresis
DRC hysteresis is programmable by writing to page 0 / register 68, bits D1–D0. These bits can be
programmed to represent values between 0 dB and 3 dB in steps of 1dB. DRC hysteresis provides a
programmable window around the programmed DRC threshold that must be exceeded for the disabled
DRC to become enabled, or the enabled DRC to become disabled. For example, if the DRC threshold is
set to –12 dBFS and the DRC hysteresis is set to 3 dB, then if the gain compression in the DRC is
inactive, the output of the DAC digital volume control must exceed –9 dBFS before gain compression due
to the DRC is activated. Similarly, when the gain compression in the DRC is active, the output of the DAC
digital volume control must fall below –15 dBFS for gain compression in the DRC to be deactivated. The
DRC hysteresis feature prevents the rapid activation and de-activation of gain compression in the DRC in
cases when the output of the DAC digital volume control rapidly fluctuates in a narrow region around the
programmed DRC threshold. By programming the DRC hysteresis as 0 dB, the hysteresis action is
disabled.
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
The recommended value of DRC hysteresis is 3 dB.
6.3.10.4.3 DRC Hold Time
DRC hold time is intended to slow the start of decay for a specified period of time in response to a
decrease in energy level. To minimize audible artifacts, TI recommends to set the DRC hold time to 0
through programming page 0 / register 69, bits D6–D3 = 0000.
6.3.10.4.4 DRC Attack Rate
When the output of the DAC digital volume control exceeds the programmed DRC threshold, the gain
applied in the DAC digital volume control is progressively reduced to avoid the signal from saturating the
channel. This process of reducing the applied gain is called attack. To avoid audible artifacts, the gain is
reduced slowly with a rate equaling the attack rate, programmable via page 0 / register 70, bits D7–D4.
Attack rates can be programmed from 4-dB gain change per sample period to 1.2207e–5-dB gain change
per sample period.
Attack rates should be programmed such that before the output of the DAC digital volume control can clip,
the input signal should be sufficiently attenuated. High attack rates can cause audible artifacts, and tooslow attack rates may not be able to prevent the input signal from clipping.
The recommended DRC attack rate value is 1.9531e–4 dB per sample period.
When the DRC detects a reduction in output signal swing beyond the programmed DRC threshold, the
DRC enters a decay state, where the applied gain in the digital-volume control is gradually increased to
programmed values. To avoid audible artifacts, the gain is slowly increased with a rate equal to the decay
rate programmed through page 0 / register 70, bits D3–D0. The decay rates can be programmed from
1.5625e–3 dB per sample period to 4.7683e–7 dB per sample period. If the decay rates are programmed
too high, then sudden gain changes can cause audible artifacts. However, if it is programmed too slow,
then the output may be perceived as too low for a long time after the peak signal has passed.
The recommended value of DRC decay rate is 2.4414e–5 dB per sample period.
6.3.10.4.6 Example Setup for DRC
•Digital Vol gain = 12 dB
•Threshold = –24 dB
•Hysteresis = 3 dB
•Hold time = 0 ms
•Attack rate = 1.9531e–4 dB per sample period
•Decay rate = 2.4414e–5 dB per sample period
Script
#Go to Page 0
w 30 00 00
#DAC => 12 db gain left
w 30 41 18
#DAC => 12 db gain right
w 30 42 18
#DAC => DRC Enabled for both channels, Threshold = -24 db, Hysteresis = 3 dB
w 30 44 7F
#DRC Hold = 0 ms, Rate of Changes of Gain = 0.5 dB/Fs'
w 30 45 00
#Attack Rate = 1.9531e-4 dB/Frame , DRC Decay Rate =2.4414e-5 dB/Frame
w 30 46 B6
#Go to Page 9
w 30 00 09
#DRC HPF
w 30 0E 7F AB 80 55 7F 56
#DRC LPF W 30 14 00 11 00 11 7F DE
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6.3.10.5 Headphone Detection
The device includes capability to monitor a headphone jack to determine if a plug has been inserted into
the jack. The device also includes the capability to detect a button press, even, for example, when starting
calls on mobile phones with headsets. Figure 6-17 shows the circuit configuration to enable this feature.
Headphone Detection is enabled by programming page 0 / register 67, bit D1. In order to avoid false
detections because of mechanical vibrations in headset jacks or microphone buttons, a debounce function
is provided for glitch rejection. For the case of headset insertion, a debounce function with a range of 32
ms to 512 ms is provided. This can be programmed through page 0 / register 67, bits D4–D2. For
improved button-press detection, the debounce function has a range of 8 ms to 32 ms by programming
page 0 / register 67, bits D1–D0.
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
Figure 6-17. Jack Connections for Headphone Detection
The device also provides feedback to the user through register-readable flags as well as an interrupt on
the I/O pins when a button press or a headset insertion or removal event is detected. The value in page 0
/ register 46, bits D5–D4 provides the instantaneous state of button press and headset insertion. Page 0 /
register 44, bit D5 is a sticky (latched) flag that is set when the button-press event is detected. Page 0 /
register 44, bit D4 is a sticky flag which is set when the headset insertion or removal event is detected.
These sticky flags are set by the event occurrence, and are reset only when read. This requires polling
page 0 / register 44. To avoid polling and the associated overhead, the device also provides an interrupt
feature, whereby events can trigger the INT1, the INT2, or both interrupts. These interrupt events can be
routed to one of the digital output pins. See Section 6.3.10.6 for details.
The device not only detects a headset insertion event, but also is able to distinguish between the different
headsets inserted, such as stereo headphones or cellular headphones. After the headset-detection event,
the user can read page 0 / register 67, bits D6–D5 to determine the type of headset inserted.
Table 6-21. Headphone Detection Block Registers
REGISTERDESCRIPTION
Page 0 / register 67, bit D1Headset-detection enable/disable
Page 0 / register 67, bits D4–D2Debounce programmability for headset detection
Page 0 / register 67, bits D1–D0Debounce programmability for button press
Page 0 / register 44, bit D5Sticky flag for button-press event
Page 0 / register 44, bit D4Sticky flag for headset-insertion or -removal event
Page 0 / register 46, bit D5Status flag for button-press event
Page 0 / register 46, bit D4Status flag for headset insertion and removal
Page 0 / register 67, bits D6–D5Flags for type of headset detected
The headset detection block requires AVDD to be powered. The headset detection feature in the device is
achieved with very low power overhead, requiring less than 20 μA of additional current from the AVDD
supply.
6.3.10.6 Interrupts
Some specific events in the device that can require host processor intervention are used to trigger
interrupts to the host processor. This avoids polling the status-flag registers continuously. The device has
two defined interrupts, INT1 and INT2, that are configured by programming page 0 / register 48 and page
0 / register 49. A user can configure interrupts INT1 and INT2 to be triggered by one or many events, such
as:
•Headset detection
•Button press
•DAC DRC signal exceeding threshold
•Overcurrent condition in headphone drivers and speaker drivers
•Data overflow in the DAC processing blocks and filters
Each of these INT1 and INT2 interrupts can be routed to output pin GPIO1. These interrupt signals can
either be configured as a single pulse or a series of pulses by programming page 0 / register 48, bit D0
and page 0 / register 49, bit D0. If the user configures the interrupts as a series of pulses, the events
trigger the start of pulses that stop when the flag registers in page 0 / registers 44, 45, and 50 are read by
the user to determine the cause of the interrupt.
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6.3.10.7 Key-Click Functionality With Digital Sine-Wave Generator (PRB_P25)
A special algorithm has been included in the digital signal processing block PRB_P25 for generating a
digital sine-wave signal that is sent to the DAC. The digital sine-wave generator is also referred to as the
beep generator in this document.
This functionality is intended for generating key-click sounds for user feedback. The sine-wave generator
is very flexible (see Table 6-22) and is completely register programmable. Programming page 0 / register
71 through page 0 / register 79 (8 bits each) completely controls the functionality of this generator and
allows for differentiating sounds.
The two registers used for programming the 16-bit sine-wave coefficient are page 0 / register 76 and
page 0 / register 77. The two registers used for programming the 16-bit cosine-wave coefficient are
page 0 / register 78 and page 0 / register 79. This coefficient resolution allows virtually any frequency of
sine wave in the audio band to be generated, up to fS/ 2.
The three registers used to control the length of the sine-burst waveform are page 0 / register 73 through
page 0 / register 75. The resolution (bit) in the registers of the sine-burst length is one sample time, so this
allows great control on the overall time of the sine-burst waveform. This 24-bit length timer supports
16 777 215 sample times. For example, if fSis set at 48 kHz, and the register value equals 96 000 d
(01 7700h), then the sine burst lasts exactly 2 seconds. The default settings for the tone generator, based
on using a sample rate of 48 kHz, are 1-kHz (approximately) sine wave, with a sine-burst length of five
cycles (5 ms).
Table 6-23. Example Beep-Generator Settings for a 1000-Hz Tone
BEEP FREQUENCYBEEP LENGTHSINECOSINESAMPLE RATE
Hz
(1)
1000
(1) These are the default settings.
(hex)
MSB
00EE10D87EE348 000
MID
(hex)
LSB
(hex)
MSB
(hex)
LSB
(hex)
MSB
(hex)
LSB
(hex)
Two registers are used to control the left sine-wave volume and the right sine-wave volume independently.
The 6-bit digital volume control used allows level control of 2 dB to –61 dB in 1-dB steps. The left-channel
volume is controlled by writing to page 0 / register 71, bits D5–D0. The right-channel volume is controlled
by writing to page 0, register 72, bits D5–D0. A master volume control that controls the left and right
channels of the beep generator are set up by writing to page 0 / register 72, bits D7–D6. The default
volume control setting is 2 dB, which provides the maximum tone-generator output level.
For generating other tones, the three tone-generator coefficients are found by running the following script
using MATLAB™ :
Sine = dec2hex(round(sin(2*π*Fin/Fs)*2^15))
Cosine = dec2hex(round(cos(2*π*Fin/Fs)*2^15))
Beep Length = dec2hex(floor(Fs*Cycle/Fin))
where,
Fin = Beep frequency desired
Fs = Sample rate
Cycle = Number of beep (sine wave) cycles that are required
dec2hex = Decimal to hexadecimal conversion function
Hz
NOTES:
1. Fin must be less than Fs / 4.
2. For the sine and cosine values, if the number of bits is less than the full 16-bit value, then the unused
MSBs must be written as 0s.
3. For the beep-length values, if number of bits is less than the full 24-bit value, then the unused MSBs
must be written as 0s.
Following the beep-volume control is a digital mixer that mixes in a playback data stream whose level has
already been set by the DAC volume control. Therefore, once the key-click volume level is set, the keyclick volume is not affected by the DAC volume control, which is the main control available to the end
user. Figure 1-1 shows this functionality.
Following the DAC, the signal can be further scaled by the analog output volume control and poweramplifier level control.
To insert a beep in the middle of an already-playing signal over DAC, use the following sequence.
Before the beep is desired, program the desired beep frequency, volume, and length in the configuration
registers. When a beep is desired, use the example configuration script.
w 30 00 00# change to Page 0
w 30 40 0C# mute DACs
f 30 26 xxx1xxx1# wait for DAC gain flag to be set
w 30 0B 02# power down NDAC divider
w 30 47 80# enable beep generator with left channel volume = 0dB, volume level could
w 30 0B 82# power up NDAC divider, in this specific example NDAC = 2, but NDAC could
w 30 40 00# un-mute DAC to resume playing audio
# be different as per requirement
# be different value as per overall setup
Note that in this scheme the audio signal on the DAC is temporarily muted to enable beep generation.
Because powering down of NDAC clock divider is required, do not use the DAC_CLK or DAC_MOD_CLK
for generation of I2S clocks.
6.3.10.8 Programming DAC Digital Filter Coefficients
The digital filter coefficients must be programmed through the I2C interface. All digital filtering for the DAC
signal path must be loaded into the RAM before the DAC is powered on. Note that default ALLPASS filter
coefficients for programmable biquads are located in boot ROM. The boot ROM automatically loads the
default values into the RAM following a hardware reset (toggling the RESET pin) or after a software reset.
After resetting the device, loading boot ROM coefficients into the digital filters requires 100 μs of
programming time. During this time, reading or writing to page 8 through page 15 for updating DAC filter
coefficient values is not permitted. The DAC should not be powered up until after all of the DAC
configurations have been done by the system microprocessor.
6.3.10.9 Updating DAC Digital Filter Coefficients During PLAY
When it is required to update the DAC digital filter coefficients or beep generator during play, care must be
taken to avoid click and pop noise or even a possible oscillation noise. These artifacts can occur if the
DAC coefficients are updated without following the proper update sequence. The correct sequence is
shown in Figure 6-18. The values for times listed in Figure 6-18 are conservative and should be used for
software purposes.
There is also an adaptive mode, in which DAC coefficients can be updated while the DAC is on. For
details, see Section 6.3.10.1.3.
Figure 6-18. Example Flow For Updating DAC Digital Filter Coefficients During Play
6.3.10.10 Digital Mixing and Routing
The has four digital mixing blocks. Each mixer can provide either mixing or multiplexing of the digital audio
data. This arrangement of digital mixers allows independent volume control for both the playback data and
the key-click sound. The first set of mixers can be used to make monaural signals from left and right audio
data, or they can even be used to swap channels to the DAC. This function is accomplished by selecting
left audio data for the right DAC input, and right data for the left DAC input. The second set of mixers
provides mixing of the audio data stream and the key-click sound. The digital routing can be configured by
writing to page 0 / register 63, bits D5–D4 for the left channel and bits D3–D2 for the right channel.
Because the key-click function uses the digital signal processing block, the CODEC_CLKIN, DAC, analog
volume control, and output driver must be powered on for the key-click sound to occur.
The has the capability to route the DAC output to either the headphone or the speaker output. If desirable,
both output drivers can operate at the same time while playing at different volume levels. The provides
various digital routing capabilities, allowing digital mixing or even channel swapping in the digital domain.
All analog outputs other than the selected ones can be powered down for optimal power consumption.
6.3.10.11.1 Analog Output Volume Control
The output volume control fine tunes the level of the mixer amplifier signal supplied to the headphone
driver or the speaker driver. This architecture supports separate and concurrent volume levels for each of
the four output drivers. This volume control is also used as part of the output pop-noise reduction scheme.
This feature is available even if the DAC is powered down.
6.3.10.11.2 Headphone Analog-Output Volume Control
For the headphone outputs, the analog volume control has a range from 0 dB to –78 dB in 0.5-dB steps
for most of the useful range plus mute, which is shown in Table 6-24. This volume control includes softstepping logic. Routing the left-channel DAC output signal to the left-channel analog volume control occurs
by writing to page 1 / register 35, bit D6. Routing the right-channel DAC output signal to the right-channel
analog volume control occurs by writing to page 1 / register 35, bit D2.
Changing the left-channel analog volume for the headphone is controlled by writing to page 1 / register 36,
bits D6–D0. Changing the right-channel analog volume for the headphone is controlled by writing to
page 1 / register 37, bits D6–D0. Routing the signal from the output of the left-channel analog volume
control to the input of the left-channel headphone power amplifier occurs by writing to page 1 / register 36,
bit D7. Routing the signal from the output of the right-channel analog volume control to the input of the
right-channel headphone power amplifier occurs by writing to page 1 / register 37, bit D7.
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The analog volume-control soft-stepping time is based on the setting in page 0 / register 63, bits D1–D0.
6.3.10.11.3 Class-D Speaker Analog Output Volume Control
For the speaker outputs, the analog volume control has a range from 0 dB to –78 dB in 0.5-dB steps for
most of the useful range plus mute, as seen in Table 6-24. The implementation includes soft-stepping
logic.
Routing the left-channel DAC output signal to the left-channel analog volume control is done by writing to
page 1 / register 35, bit D6. Routing the right-channel DAC output signal to the right-channel analog
volume control is done by writing to page 1 / register 35, bit D2. Changing the left-channel analog volume
for the speaker is controlled by writing to page 1 / register 38, bits D6–D0. Changing the right-channel
analog volume for the speaker is controlled by writing to page 1 / register 39, bits D6–D0.
Routing the signal from the output of the left-channel analog volume control to the input of the left-channel
speaker amplifier is done by writing to page 1 / register 38, bit D7. Routing the signal from the output of
the right-channel analog volume control to the input of the right-channel speaker amplifier is done by
writing to page 1 / register 39, bit D7.
The analog volume-control soft-stepping time is based on the setting in page 0 / register 63, bits D1–D0.
6.3.10.12 Analog Outputs
Various analog routings are supported for playback. All the options can be conveniently viewed on the
functional block diagram, Figure 1-1.
6.3.10.12.1 Headphone Drivers
The device features a stereo headphone driver (HPL and HPR) that delivers up to 30 mW per channel, at
3.3-V supply voltage, into a 16-Ω load. The headphones are used in a single-ended configuration where
an ac-coupling capacitor (dc-blocking) is connected between the device output pins and the headphones.
The headphone driver also supports 32-Ω and 10-kΩ loads without changing any control register settings.
The headphone drivers can be configured to optimize the power consumption in the lineout-drive mode by
writing 11 to page 1 / register 44, bits D2–D1.
The output common mode of the headphone and lineout drivers is programmed to 1.35 V, 1.5 V, 1.65 V,
or 1.8 V by setting page 1 / register 31, bits D4–D3. Set the common-mode voltage to ≤ AVDD / 2.
The left headphone driver can be powered on by writing to page 1 / register 31, bit D7. The right
headphone driver can be powered on by writing to page 1 / register 31, bit D6. The left-output driver gain
can be controlled by writing to page 1 / register 40, bits D6–D3, and it can be muted by writing to page 1 /
register 40, bit D2. The right-output driver gain can be controlled by writing to page 1 / register 41, bits
D6–D3, and it can be muted by writing to page 1 / register 41, bit D2.
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The device has a short-circuit protection feature for the headphone drivers, which is always enabled to
provide protection. The output condition of the headphone driver during short circuit is programmed by
writing to page 1 / register 31, bit D1. If D1 = 0 when a short circuit is detected, the device limits the
maximum current to the load. If D1 = 1 when a short circuit is detected, the device powers down the
output driver. The default condition for headphones is the current-limiting mode. In case of a short circuit
on either channel, the output is disabled and a status flag is provided as read-only bits on page 1 /
register 31, bit D0. If shutdown mode is enabled, then as soon as the short circuit is detected, page 1 /
register 31, bit D7 (for HPL) or page 1 / register 31, bit D6, or both (for HPR) clear automatically. Next, the
device requires a reset to re-enable the output stage. Resetting occurs in two ways. First, the device
master reset can be used, which requires either toggling the RESET pin or using the software reset. If
master reset is used, it resets all of the registers. Second, a dedicated headphone power-stage reset can
also be used to re-enable the output stage, and that keeps all of the other device settings. The headphone
power stage reset occurs by setting page 1 / register 31, bit D7 for HPL and by setting page 1 /
register 31, bit D6 for HPR. If the fault condition has been removed, then the device returns to normal
operation. If the fault is still present, then another shutdown occurs. Repeated resetting (more than three
times) is not recommended, as this could lead to overheating.
6.3.10.12.2 Speaker Drivers
The device has an integrated class-D stereo speaker driver (SPLP/SPLM and SPRP/SPRM) capable of
driving an 8-Ω differential load. The speaker driver can be powered directly from the battery supply (2.7 V
to 5.5 V) on the SPLVDD and SPRVDD pins; however, the voltage (including spike voltage) must be
limited below the absolute-maximum voltage of 6 V.
The speaker driver is capable of supplying 400 mW per channel with a 3.6-V power supply. Through the
use of digital mixing, the device can connect one or both digital audio playback data channels to either
speaker driver; this also allows digital channel swapping if needed.
46
The left class-D speaker driver can be powered on by writing to page 1 / register 32, bit D7. The right
class-D speaker driver can be powered on by writing to page 1 / register 32, bit D6. The left-output driver
gain can be controlled by writing to page 1 / register 42, bits D4–D3, and it can be muted by writing to
page 1 / register 42, bit D2. The right-output driver gain can be controlled by writing to page 1 / register
43, bits D4–D3, and it can be muted by writing to page 1 / register 43, bit D2.
The device has a short-circuit protection feature for the speaker drivers that is always enabled to provide
protection. If the output is shorted, the output stage shuts down on the overcurrent condition. (Current
limiting is not an available option for the higher-current speaker driver output stage.) In case of a short
circuit on either channel, the output is disabled and a status flag is provided as a read-only bit on page 1 /
register 32, bit D0.
If shutdown occurs because of an overcurrent condition, then the device requires a reset to re-enable the
output stage. Resetting occurs in two ways. First, the device master reset can be used, which requires
either toggling the RESET pin or using the software reset. If master reset is used, it resets all of the
registers. Second, a dedicated speaker power-stage reset can be used that keeps all of the other device
settings. The speaker power-stage reset occurs by setting page 1 / register 32, bit D7 for SPLP and SPLM
and by setting page 1 / register 32, bit D6 for SPRP and SPRM. If the fault condition has been removed,
then the device returns to normal operation. If the fault is still present, then another shutdown occurs.
Repeated resetting (more than three times) is not recommended as this could lead to overheating.
To minimize battery current leakage, the SPLVDD and SPRVDD voltage levels must not be less
than the AVDD voltage level.
The device has a thermal protection (OTP) feature for the speaker drivers which is always enabled to
provide protection. If the device overheats, then the output stops switching. When the device cools down,
the device resumes switching. An overtemperature status flag is provided as a read-only bit on page 0 /
register 3, bit D1. The OTP feature is for self-protection of the device. If die temperature can be controlled
at the system or board level, then overtemperature does not occur.
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
6.3.10.13 Audio-Output Stage-Power Configurations
After the device has been configured (following a RESET) and the circuitry has been powered up, the
audio output stage can be powered up and powered down by register control.
These functions soft-start automatically. By using these register controls, it is possible to control these four
output-stage configuratios independently.
See Table 6-25 for register control of audio output stage power configurations.
AUDIO OUTPUT PINSDESIRED FUNCTIONPAGE 1 / REGISTER, BIT VALUES
HPL
HPR
SPLP / SPLM
SPRP / SPRM
Power down HPL driverPage 1 / register 31, bit D7 = 0
Power up HPL driverPage 1 / register 31, bit D7 = 1
Power down HPR driverPage 1 / register 31, bit D6 = 0
Power up HPR driverPage 1 / register 31, bit D6 = 1
Power down left class-D driversPage 1 / register 32, bit D7 = 0
Power up left class-D driversPage 1 / register 32, bit D7 = 1
Power down right class-D driversPage 1 / register 32, bit D6 = 0
Power up right class-D driversPage 1 / register 32, bit D6 = 1
6.3.10.14 DAC Setup
The following paragraphs are intended to guide a user through the steps necessary to configure the .
Step 1
The system clock source (master clock) and the targeted DAC sampling frequency must be identified.
Depending on the targeted performance, the decimation filter type (A, B, or C) and DOSR value can be
determined:
•Filter A should be used for 48-kHz high-performance operation; DOSR must be a multiple of 8.
•Filter B should be used for up to 96-kHz operations; DOSR must be a multiple of 4.
•Filter C should be used for up to 192-kHz operations; DOSR must be a multiple of 2.
In all cases, DOSR is limited in its range by the following condition:
2.8 MHz < DOSR × DAC_fS< 6.2 MHz
Based on the identified filter type and the required signal-processing capabilities, the appropriate
processing block can be determined from the list of available processing blocks (PRB_P1 to PRB_P25).
Based on the available master clock, the chosen DOSR and the targeted sampling rate, the clock-divider
values NDAC and MDAC can be determined. If necessary, the internal PLL can add a large degree of
flexibility.
In summary, CODEC_CLKIN (derived directly from the system clock source or from the internal PLL)
divided by MDAC, NDAC, and DOSR must be equal to the DAC sampling rate, DAC_fS. The
CODEC_CLKIN clock signal is shared with the DAC clock-generation block.
CODEC_CLKIN = NDAC × MDAC × DOSR × DAC_f
To a large degree, NDAC and MDAC can be chosen independently in the range of 1 to 128. In general,
NDAC should be as large as possible as long as the following condition can still be met:
MDAC × DOSR / 32 ≥ RC
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S
RC is a function of the chosen processing block and is listed in Table 6-11.
The common-mode voltage setting of the device is determined by the available analog power supply.
At this point, the following device-specific parameters are known: PRB_Px, DOSR, NDAC, MDAC, input
and output common-mode values. If the PLL is used, the PLL parameters P, J, D, and R are determined
as well.
Step 2
Setting up the device via register programming:
The following list gives an example sequence of items that must be executed in the time between
powering the device up and reading data from the device. Note that there are other valid sequences,
depending on which features are used.
1. Define starting point:
a. Power up applicable external power supplies
b. Set register page to 0
c. Initiate SW reset
2. Program clock settings
a. Program PLL clock dividers P, J, D, and R (if PLL is used)
b. Power up PLL (if PLL is used)
c. Program and power up NDAC
d. Program and power up MDAC
e. Program OSR value
f. Program I2S word length if required (16, 20, 24, or 32 bits)
g. Program the processing block to be used
h. Micellaneous page 0 controls
3. Program analog blocks
a. Set register page to 1
b. Program common-mode voltage
c. Program headphone-specific de-pop settings (in case headphone driver is used)
d. Program routing of DAC output to the output amplifier (headphone/lineout or speaker)
e. Unmute and set gain of output drivers
f. Power up output drivers
4. Apply waiting time determined by the de-pop settings and the soft-stepping settings of the driver gain,
or poll page 1 / register 63
5. Power up DAC
a. Set register page to 0
b. Power up DAC channels and set digital gain
c. Unmute digital volume control
A detailed example can be found in Section 6.3.10.15.
6.3.10.15 Example Register Setup to Play Digital Data Through DAC and Headphone/Speaker Outputs
A typical EVM I2C register control script follows to show how to set up the in playback mode with fS= 44.1
kHz and MCLK = 11.2896 MHz.
# Key: w 30 XX YY ==> write to I2C address 0x30, to register 0xXX, data 0xYY
## ==> comment delimiter
#
# The following list gives an example sequence of items that must be executed in the time
# between powering the # device up and reading data from the device. Note that there are
# other valid sequences depending on which features are used.
# 1. Define starting point:
#(a) Power up applicable external hardware power supplies
#(b) Set register page to 0
#
w 30 00 00
#
#(c) Initiate SW reset (PLL is powered off as part of reset)
#
w 30 01 01
#
# 2. Program clock settings
#(a) Program PLL clock dividers P, J, D, R (if PLL is used)
#
# PLL_clkin = MCLK,codec_clkin = PLL_CLK
w 30 04 03
# J = 8
w 30 06 08
# D = 0000, D(13:8) = 0, D(7:0) = 0
w 30 07 00 00
#
#(b) Power up PLL (if PLL is used)
# PLL Power up, P = 1, R = 1
#
w 30 05 91
#
#(c) Program and power up NDAC
#
# NDAC is powered up and set to 8
w 30 0B 88
#
#(d) Program and power up MDAC
#
# MDAC is powered up and set to 2
w 30 0C 82
#
#(e) Program OSR value
#
# DOSR = 128, DOSR(9:8) = 0, DOSR(7:0) = 128
w 30 0D 00 80
#
#(f) Program I2S word length if required (16, 20, 24, 32 bits)
#and master mode (BCLK and WCLK are outputs)
#
# mode is i2s, wordlength is 16, slave mode
w 30 1B 00
#(g) Program the processing block to be used
#
# Select Processing Block PRB_P11
w 30 3C 0B
w 30 00 08
w 30 01 04
w 30 00 00
#
#(h) Miscellaneous page 0 controls
#
# DAC => volume control thru pin disable
w 30 74 00
# 3. Program analog blocks
#
#(a) Set register page to 1
#
w 30 00 01
#
#(b) Program common-mode voltage (defalut = 1.35 V)
#
w 30 1F 04
#
#(c) Program headphone-specific depop settings (in case headphone driver is used)
#
# De-pop, Power on = 800 ms, Step time = 4 ms
w 30 21 4E
#
#(d) Program routing of DAC output to the output amplifier (headphone/lineout or speaker)
#
# LDAC routed to HPL out, RDAC routed to HPR out
w 30 23 44
#
#(e) Unmute and set gain of output driver
#
# Unmute HPL, set gain = 0 db
w 30 28 06
# Unmute HPR, set gain = 0 dB
w 30 29 06
# Unmute Class-D, set gain = 18 dB
w 30 2A 1C
#
#(f) Power up output drivers
#
# HPL and HPR powered up
w 30 1F C2
# Power-up Class-D driver
w 30 20 86
# Enable HPL output analog volume, set = -9 dB
w 30 24 92
# Enable HPR output analog volume, set = -9 dB
w 30 25 92
# Enable Class-D output analog volume, set = -9 dB
w 30 26 92
#
# 4. Apply waiting time determined by the de-pop settings and the soft-stepping settings
#of the driver gain or poll page 1 / register 63
#
# 5. Power up DAC
#(a) Set register page to 0
#
w 30 00 00
#
#(b) Power up DAC channels and set digital gain
#
# Powerup DAC left and right channels (soft step enabled)
w 30 3F D4
#
# DAC Left gain = -22 dB
w 30 41 D4
# DAC Right gain = -22 dB
w 30 42 D4
#
#(c) Unmute digital volume control
#
# Unmute DAC left and right channels
w 30 40 00
6.3.11 CLOCK Generation and PLL
The device supports a wide range of options for generating clocks for the DAC section as well as interface
and other control blocks as shown in Figure 6-19. The clocks for the DAC require a source reference
clock. This clock is provided on a variety of device pins, such as the MCLK, BCLK, or GPIO1 pins. The
source reference clock for the codec is chosen by programming the CODEC_CLKIN value on page 0 /
register 4, bits D1–D0. The CODEC_CLKIN is then routed through highly-flexible clock dividers shown in
Figure 6-19 to generate the various clocks required for the DAC. In the event that the desired audio clocks
cannot be generated from the reference clocks on MCLK, BCLK, or GPIO1, the device also provides the
option of using the on-chip PLL which supports a wide range of fractional multiplication values to generate
the required clocks. Starting from CODEC_CLKIN, the device provides several programmable clock
dividers to help achieve a variety of sampling rates for the DAC.
The DAC modulator is clocked by DAC_MOD_CLK. For proper power-up operation of the DAC channel,
DAC_MOD_CLK must be enabled by configuring the NDAC and MDAC clock dividers (page 0 /
register 11, bit D7 = 1 and page 0 / register 12, bit D7 = 1). When the DAC channel is powered down, the
device internally initiates a power-down sequence for proper shutdown. During this shutdown sequence,
the NDAC and MDAC dividers must not be powered down, or else a proper low-power shutdown may not
take place. The user can read back the power-status flag at page 0 / register 37, bit D7 and page 0 /
register 37, bit D3. When both of the flags indicate power-down, the MDAC divider may be powered down,
followed by the NDAC divider.
In general, for proper operation, all the root clock dividers must power down only after the child clock
dividers have powered down.
The device also has options for routing some of the internal clocks to the GPIO1 pin to be used as
general-purpose clocks in the system. The feature is shown in Figure 6-21.
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
In the mode when the device is configured to drive the BCLK pin (page 0 / register 27, bit D3 = 1), the
device is driven as the divided value of BDIV_CLKIN. The division value is programmed in page 0 /
register 30, bits D6–D0 from 1 to 128. The BDIV_CLKIN is configurable to be one of DAC_CLK (DAC
processing clock) or DAC_MOD_CLK by configuring the BDIV_CLKIN multiplexer in page 0 / register 29,
bits D1–D0. Additionally, a general-purpose clock can be driven out on GPIO1.
This clock can be a divided-down version of CDIV_CLKIN. The value of this clock divider can be
programmed from 1 to 128 by writing to page 0 / register 26, bits D6–D0. CDIV_CLKIN can also be
programmed as one of the clocks among the list shown in Figure 6-21. This is controlled by programming
the multiplexer in page 0 / register 25, bits D2–D0.
For lower power consumption, the best process is to derive the internal audio processing clocks using the
simple dividers. When the input MCLK or other source clock is not an integer multiple of the audio
processing clocks then using the on-board PLL is necessary. The fractional PLL generates an internal
master clock that produces the processing clocks required by the DAC. The programmability of this PLL
allows operation from a wide variety of clocks that may be available in the system.
The PLL input supports clocks varying from 512 kHz to 20 MHz and is register-programmable to enable
generation of the required sampling rates with fine resolution. The PLL turns on by writing to page 0 /
register 5, bit D7. When the PLL is enabled, the PLL output clock, PLL_CLK, is given by Equation 6.
The PLL turns on through page 0 / register 5, bit D7. The variable P is programmed through page 0 /
register 5, bits D6–D4. The variable R is programmed through page 0 / register 5, bits D3–D0. The
variable J is programmed through page 0 / register 6, bits D5–D0. The variable D is 14 bits and is
programmed into two registers. The MSB portion is programmed through page 0 / register 7, bits D5–D0,
and the LSB portion is programmed thrugh page 0 / register 8, bits D7–D0. For proper update of the Ddivider value, page 0 / register 7 must be programmed first, followed immediately by page 0 / register 8.
The new value of D does not take effect unless the write to page 0 / register 8 is complete.
When the PLL is enabled, the following conditions must be satisfied:
•When the PLL is enabled and D = 0, the following conditions must be satisfied for PLL_CLKIN:
(7)
80 MHz ≤ (PLL_CLKIN × J.D. × R / P) ≤ 110 MHz
4 ≤ R × J ≤ 259
•When the PLL is enabled and D ≠ 0, the following conditions must be satisfied for PLL_CLKIN:
The PLL can power up independently from the DAC block, and can also be used as a general-purpose
PLL by routing the PLL output to the GPIO output. After powering up the PLL, PLL_CLK is available
typically after 10 ms.
The clocks for the codec and various signal processing blocks, CODEC_CLKIN, are generated from the
MCLK input, BCLK input, GPIO input, or PLL_CLK (page 0 / register 4, bits D1–D0).
If CODEC_CLKIN is derived from the PLL, then the PLL must be powered up first and powered down last.
Table 6-28 lists several example cases of typical PLL_CLKIN rates and how to program the PLL to
achieve a sample rate fSof either 44.1 kHz or 48 kHz.
The internal clock runs nominally at 8.2 MHz. This is used for various internal timing intervals, de-bounce
logic, and interrupts. The MCLK divider must be set in such a way that the divider output is approximately
1 MHz for the timers to be closer to the programmed value.
Used for de-bounce time for
headset detection logic,
various power up timers and
for generation of interrupts
TLV320DAC3101
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Figure 6-22. Interval Timer Clock Selection
6.3.13 Digital Audio and Control Interface
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
6.3.13.1 Digital Audio Interface
Audio data is transferred between the host processor and the device through the digital audio data, serial
interface, or audio bus. The audio bus on this device is very flexible, including left- or right-justified data
options, support for I2S or DSP protocols, programmable data length options, a TDM mode for
multichannel operation, very flexible master and slave configurability for each bus-clock line, and the
ability to communicate with multiple devices within a system directly.
The audio bus of the device can be configured for left-justified or right-justified, I2S, DSP, or TDM modes
of operation, where communication with standard telephony interfaces is supported within the TDM mode.
These modes are all MSB-first, with data width programmable as 16, 20, 24, or 32 bits by configuring
page 0 / register 27, bits D5–D4. In addition, the word clock and bit clock can be independently configured
in either master or slave mode, for flexible connectivity to a wide variety of processors. The word clock
defines the beginning of a frame, and can be programmed as either a pulse or a square-wave signal. The
frequency of this clock corresponds to the DAC sampling frequency.
The bit clock is used to clock-in and clock-out the digital audio data across the serial bus. When in master
mode, this signal can be programmed to generate variable clock pulses by controlling the bit-clock divider
in page 0 / register 30 (see Figure 6-19). The number of bit-clock pulses in a frame can require adjustment
to accommodate various word lengths as well as to support the case when multiple s share the same
audio bus.
The device also includes a feature to offset the position of start-of-data transfer with respect to the word
clock. This offset is controlled in terms of number of bit-clocks and can be programmed in page 0 / register
28.
The device also has the feature of inverting the polarity of the bit clock used for transferring the audio data
as compared to the default clock polarity used. This feature can be used independently of the mode of
audio interface chosen. This can be configured through page 0 / register 29, bit D3.
By default, when the word clocks and bit clocks are generated by the device, these clocks are active only
when the DACis powered up within the device. This is done to save power. However, it also supports a
feature when both the word clocks and bit clocks can be active even when the codec in the device is
powered down. This is useful when using the TDM mode with multiple codecs on the same bus, or when
word clocks or bit clocks are used in the system as general-purpose clocks.
The audio interface of the can enter the right-justified mode by programming page 0 / register 27, bits
D7–D6 = 10. In right-justified mode, the LSB of the left channel is valid on the rising edge of the bit clock
preceding the falling edge of the word clock. Similarly, the LSB of the right channel is valid on the rising
edge of the bit clock preceding the rising edge of the word clock.
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Figure 6-23. Timing Diagram for Right-Justified Mode
For the right-justified mode, the number of bit clocks per frame should be greater-than or equal-to twice
the programmed word length of the data.
6.3.13.1.2 Left-Justified Mode
The audio interface of the can enter the left-justified mode by programming page 0 / register 27, bits
D7–D6 = 11. In left-justified mode, the MSB of the right channel is valid on the rising edge of the bit clock
following the falling edge of the word clock. Similarly, the MSB of the left channel is valid on the rising
edge of the bit clock following the rising edge of the word clock.
Figure 6-24. Timing Diagram for Left-Justified Mode
Figure 6-25. Timing Diagram for Left-Justified Mode With Offset = 1
Figure 6-26. Timing Diagram for Left-Justified Mode With Offset = 0 and Inverted Bit Clock
For the left-justified mode, the number of bit clocks per frame should be greater-than or equal-to twice the
programmed word length of the data. Also, the programmed offset value should be less than the number
of bit clocks per frame by at least the programmed word length of the data.
6.3.13.1.3 I2S Mode
The audio interface of the device enters I2S mode by programming page 0 / register 27, bits D7–D6 = to
00. In I2S mode, the MSB of the left channel is valid on the second rising edge of the bit clock after the
falling edge of the word clock. Similarly, the MSB of the right channel is valid on the second rising edge of
the bit clock after the rising edge of the word clock.
Figure 6-28. Timing Diagram for I2S Mode With Offset = 2
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Figure 6-29. Timing Diagram for I2S Mode With Offset = 0 and Bit Clock Inverted
For I2S mode, the number of bit clocks per channel should be greater-than or equal-to the programmed
word length of the data. Also, the programmed offset value should be less than the number of bit clocks
per frame by at least the programmed word length of the data.
6.3.13.1.4 DSP Mode
The audio interface of the can enter DSP mode by programming page 0 / register 27, bits D7–D6 = 01. In
DSP mode, the rising edge of the word clock starts the data transfer with the left-channel data first and
immediately followed by the right-channel data. Each data bit is valid on the falling edge of the bit clock.
Figure 6-31. Timing Diagram for DSP Mode With Offset = 1
Figure 6-32. Timing Diagram for DSP Mode With Offset = 0 and Bit Clock Inverted
For the DSP mode, the number of bit clocks per frame should be greater-than or equal-to twice the
programmed word length of the data. Also, the programmed offset value should be less than the number
of bit clocks per frame by at least the programmed word length of the data.
6.3.13.2 Primary and Secondary Digital Audio Interface Selection
The audio serial interface on the has extensive I/O control to allow communication with two independent
processors for audio data. The processors can communicate with the device one at a time. This feature is
enabled by register programming of the various pin selections. shows the primary and secondary audio
interface selection and registers. Figure 6-33 is a high-level diagram showing the general signal flow and
multiplexing for the primary and secondary audio interfaces.
The control interface supports the I2C communication protocol.
6.3.13.3.1 I2C Control Mode
The supports the I2C control protocol, and responds to the I2C address of 0011 000. I2C is a two-wire,
open-drain interface supporting multiple devices and masters on a single bus. Devices on the I2C bus only
drive the bus lines LOW by connecting them to ground; they never drive the bus lines HIGH. Instead, the
bus wires are pulled HIGH by pullup resistors, so the bus wires are HIGH when no device is driving them
LOW. This way, two devices cannot conflict; if two devices drive the bus simultaneously, there is no driver
contention.
Communication on the I2C bus always takes place between two devices, one acting as the master and the
other acting as the slave. Both masters and slaves can read and write, but slaves can only do so under
the direction of the master. Some I2C devices can act as masters or slaves, but the can only act as a
slave device.
An I2C bus consists of two lines, SDA and SCL. SDA carries data, and the SCL signal provides the clock.
All data is transmitted across the I2C bus in groups of eight bits. To send a bit on the I2C bus, the SDA line
is driven to the appropriate level while SCL is LOW (a LOW on SDA indicates the bit is zero, while a HIGH
indicates the bit is one).
Once the SDA line has settled, the SCL line is brought HIGH, then LOW. This pulse on the SCL line
clocks the SDA bit into the receiver shift register.
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The I2C bus is bidirectional: the SDA line is used both for transmitting and receiving data. When a master
reads from a slave, the slave drives the data line; when a master sends to a slave, the master drives the
data line.
Most of the time the bus is idle, no communication is taking place, and both lines are HIGH. When
communication is taking place, the bus is active. Only master devices can start communication on the bus.
Generally, the data line is only allowed to change state while the clock line is LOW. If the data line
changes state while the clock line is HIGH, it is either a START condition or the counterpart, a STOP
condition. A START condition is when the clock line is HIGH and the data line goes from HIGH to LOW. A
STOP condition is when the clock line is HIGH and the data line goes from LOW to HIGH.
After the master issues a START condition, it sends a byte that selects the slave device for
communication. This byte is called the address byte. Each device on an I2C bus has a unique 7-bit
address to which it responds. (Slaves can also have 10-bit addresses; see the I2C specification for
details.) The master sends an address in the address byte, together with a bit that indicates whether it is
to read from or write to the slave device.
Every byte transmitted on the I2C bus, whether it is address or data, is acknowledged with an
acknowledge bit. When a master has finished sending a byte (eight data bits) to a slave, it stops driving
SDA and waits for the slave to acknowledge the byte. The slave acknowledges the byte by pulling SDA
LOW. The master then sends a clock pulse to clock the acknowledge bit. Similarly, when a master has
finished reading a byte, it pulls SDA LOW to acknowledge this to the slave. It then sends a clock pulse to
clock the bit. (Remember that the master always drives the clock line.)
A not-acknowledge is performed by simply leaving SDA HIGH during an acknowledge cycle. If a device is
not present on the bus, and the master attempts to address the device, the master receives a notacknowledge because no device is present at that address to pull the line LOW.
64
When a master has finished communicating with a slave, it may issue a STOP condition. When a STOP
condition is issued, the bus becomes idle again. A master may also issue another START condition. When
a START condition is issued while the bus is active, it is called a repeated START condition.
The can also respond to and acknowledge a general call, which consists of the master issuing a command
with a slave address byte of 00h. This feature is disabled by default, but can be enabled through page 0 /
register 34, bit D5.
In the case of an I2C register write, if the master does not issue a STOP condition, then the device enters
auto-increment mode. So in the next eight clocks, the data on SDA is treated as data for the next
incremental register.
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
Figure 6-34. I2C Write
Figure 6-35. I2C Read
Similarly, in the case of an I2C register read, after the device has sent out the 8-bit data from the
addressed register, if the master issues a ACKNOWLEDGE, the slave takes over control of the SDA bus
and transmits for the next eight clocks the data of the next incremental register.
6.4Register Map
6.4.1Register Map
All features on this device are addressed using the I2C bus. All of the writable registers can be read back.
However, some registers contain status information or data, and are only available for reading.
The device contains several pages of 8-bit registers, and each page can contain up to 128 registers. The
register pages are divided up based on functional blocks for this device. The pages defined for the device
are 0, 1, 3, 8–9, 12–13 (DAC coefficient pages). Page 0 is the default home page after RESET. Page
control occurs by writing a new page value into register 0 of the current page.
The control registers for the device are described in detail as follows. All registers are 8 bits in width, with
D7 referring to the most-significant bit of each register, and D0 referring to the least-significant bit.
Pages 0, 1, 3, 8–9, and 12–13 are available for use. All other pages and registers are reserved. Do not
read from or write to reserved pages and registers. Also, do not write other than the reset values for the
reserved bits and read-only bits of non-reserved registers; otherwise, device functionality failure can occur.
Note that the page and register numbers are shown in decimal format. For use in microcode,
these decimal values may need to be converted to hexadecimal format. For convenience,
the register numbers are shown in both formats, whereas the page numbers are shown only
in decimal format.
NOTE
PAGE NUMBERDESCRIPTION
0Page 0 is the default page on power up. Configuration for serial interface, digital I/O, and other circuitry.
D1–D0R/W0000: DAC-channel volume-control soft-stepping is enabled for one step per sample period.
RESET
VALUE
DESCRIPTION
1: Left-channel DAC is powered up.
1: Right-channel DAC is powered up.
01: Left-channel DAC data path = left data
10: Left-channel DAC data path = right data
11: Left-channel DAC data path = left-channel and right-channel data [(L + R) / 2]
01: Right-channel DAC data path = right data
10: Right-channel DAC data path = left data
11: Right-channel DAC data path = left-channel and right-channel data [(L + R) / 2]
01: DAC-channel volume-control soft-stepping is enabled for one step per two sample periods.
10: DAC-channel volume-control soft-stepping is disabled.
11: Reserved. Do not write this sequence to these bits.
D7–D4R/W0000Reserved. Write only zeros to these bits.
D3R/W10: Left-channel DAC not muted
D2R/W10: Right-channel DAC not muted
D1–D0R/W0000: Left and right channels have independent volume control.
(1) When DRC is enabled, left and right channel volume controls are always independent. Program bits D1–D0 to 00.
RESET
VALUE
DESCRIPTION
1: Left-channel DAC muted
1: Right-channel DAC muted
(1)
01: Left-channel volume control Is the programmed value of right-channel volume control.
10: Right-channel volume control is the programmed value of left-channel volume control.
11: Same as 00
Table 6-76. Page 0 / Register 65 (0x41): DAC Left Volume Control
READ/
BIT
WRITE
D7–D0R/W0000 0000Left DAC Channel Digital Volume Control Setting
RESET
VALUE
DESCRIPTION
0111 1111–0011 0001: Reserved. Do not use
0011 0000: Digital volume control = 24 dB
0010 1111: Digital volume control = 23.5 dB
0010 1110: Digital volume control = 23 dB
...
0000 0001: Digital volume control = 0.5 dB
0000 0000: Digital volume control = 0 dB
1111 1111: Digital volume control = –0.5 dB
...
1000 0010: Digital volume control = –63 dB
1000 0001: Digital volume control = –63.5 dB
1000 0000: Reserved.
Table 6-77. Page 0 / Register 66 (0x42): DAC Right Volume Control
READ/
BIT
WRITE
D7–D0R/W0000 0000Right DAC Channel Digital Volume Control Setting
RESET
VALUE
DESCRIPTION
0111 1111–0011 0001: Reserved. Do not use
0011 0000: Digital volume control = 24 dB
0010 1111: Digital volume control = 23.5 dB
0010 1110: Digital volume control = 23 dB
...
0000 0001: Digital volume control = 0.5 dB
0000 0000: Digital volume control = 0 dB
1111 1111: Digital volume control = –0.5 dB
...
1000 0010: = –63 dB
1000 0001: = –63.5 dB
1000 0000: Reserved.
D4–D2R/W000Debounce Programming for Glitch Rejection During Headset Detection
D1–D0R/W00Debounce programming for glitch rejection during headset button-press detection
(1) Note that these times are generated using the 1 MHz reference clock which is defined in Page 3 / Register 16.
RESET
VALUE
DESCRIPTION
1: Headset detection enabled
01: Headset without microphone is detected
10: Reserved
11: Headset with microphone is detected
(1)
000: 16 ms (sampled with 2-ms clock)
001: 32 ms (sampled with 4-ms clock)
010: 64 ms (sampled with 8-ms clock)
011: 128 ms (sampled with 16-ms clock)
100: 256 ms (sampled with 32-ms clock)
101: 512 ms (sampled with 64-ms clock)
110: Reserved
111: Reserved
00: 0 ms
01: 8 ms (sampled with 1-ms clock)
10: 16 ms (sampled with 2-ms clock)
11: 32 ms (sampled with 4-ms clock)
DR0Reserved. Write only the reset value to these bits.
D6–D3R/W0111DRC Hold Time
D2-D0R000Reserved. Write only the reset value to these bits.
RESET
VALUE
DESCRIPTION
0000: DRC Hold Disabled
0001: DRC Hold Time = 32 DAC Word Clocks
0010: DRC Hold Time = 64 DAC Word Clocks
0011: DRC Hold Time = 128 DAC Word Clocks
0100: DRC Hold Time = 256 DAC Word Clocks
0101: DRC Hold Time = 512 DAC Word Clocks
0110: DRC Hold Time = 1024 DAC Word Clocks
0111: DRC Hold Time = 2048 DAC Word Clocks
1000: DRC Hold Time = 4096 DAC Word Clocks
1001: DRC Hold Time = 8192 DAC Word Clocks
1010: DRC Hold Time = 16 384 DAC Word Clocks
1011: DRC Hold Time = 32 768 DAC Word Clocks
1100: DRC Hold Time = 65 536 DAC Word Clocks
1101: DRC Hold Time = 98 304 DAC Word Clocks
1110: DRC Hold Time = 131 072 DAC Word Clocks
1111: DRC Hold Time = 163 840 DAC Word Clocks
D7–D4R/W00000000: DRC attack rate = 4 dB per DAC Word Clock
D3–D0R/W0000Decay Rate is defined as DR / 2
RESET
VALUE
DESCRIPTION
0001: DRC attack rate = 2 dB per DAC word clock
0010: DRC attack rate = 1 dB per DAC word clock
...
1110: DRC attack rate = 2.4414e–5 dB per DAC word clock
1111: DRC attack rate = 1.2207e–5 dB per DAC word clock
[bits D3-D0 value]
0000: DRC decay rate (DR) = 0.015625 dB per DAC Word Clock
dB per DAC Word Clock, where DR = 0.015625 dB
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0001: DRC decay rate = DR / 2 dB per DAC Word Clock
0010: DRC decay rate = DR / 22dB per DAC Word Clock
0011: DRC decay rate = DR / 23dB per DAC Word Clock
0100: DRC decay rate = DR / 24dB per DAC Word Clock
0101: DRC decay rate = DR / 25dB per DAC Word Clock
0110: DRC decay rate = DR / 26dB per DAC Word Clock
0111: DRC decay rate = DR / 27dB per DAC Word Clock
1000: DRC decay rate = DR / 28dB per DAC Word Clock
1001: DRC decay rate = DR / 29dB per DAC Word Clock
1010: DRC decay rate = DR / 210dB per DAC Word Clock
1011: DRC decay rate = DR / 211dB per DAC Word Clock
1100: DRC decay rate = DR / 212dB per DAC Word Clock
1101: DRC decay rate = DR / 213dB per DAC Word Clock
1110: DRC decay rate = DR / 214dB per DAC Word Clock
1111: DRC decay rate = DR / 215dB per DAC Word Clock
1: Beep generator is enabled (self-clearing based on beep duration).
00 0001: Left-channel beep volume control = 1 dB
00 0010: Left-channel beep volume control = 0 dB
00 0011: Left-channel beep volume control = –1 dB
...
11 1110: Left-channel beep volume control = –60 dB
11 1111: Left-channel beep volume control = –61 dB
D7–D6R/W0000: Left and right channels have independent beep volume control.
D5–D0R/W00 000000 0000: Right-channel beep volume control = 2 dB
RESET
VALUE
DESCRIPTION
01: Left-channel beep volume control is the programmed value of right-channel beep volume control.
10: Right-channel beep volume control is the programmed value of left-channel beep volume control.
11: Same as 00
00 0001: Right-channel beep volume control = 1 dB
00 0010: Right-channel beep volume control = 0 dB
00 0011: Right-channel beep volume control = –1 dB
...
11 1110: Right-channel beep volume control = –60 dB
11 1111: Right-channel beep volume control = –61 dB
1.37 kHz
Note: These values are based on a nominal oscillator
frequency of 8.2 MHz. The values scale to the actual
oscillator frequency.
Table 6-93. Page 0 / Register 117 (0x75): VOL/MICDET-Pin Gain
READ/
BIT
WRITE
D7R/W0Reserved. Write only zero to this bit.
D6–D0RXXX XXXX000 0000: Gain applied by pin volume control = 18 dB
RESET
VALUE
DESCRIPTION
000 0001: Gain applied by pin volume control = 17.5 dB
000 0010: Gain applied by pin volume control = 17 dB
...
010 0011: Gain applied by pin volume control = 0.5 dB
010 0100: Gain applied by pin volume control = 0 dB
010 0101: Gain applied by pin volume control = –0.5 dB
...
101 1001: Gain applied by pin volume control = –26.5 dB
101 1010: Gain applied by pin volume control = –27 dB
101 1011: Gain applied by pin volume control = –28 dB
...
111 1101: Gain applied by pin volume control = –62 dB
111 1110: Gain applied by pin volume control = –63 dB
111 1111: Reserved.
D2R/W1Reserved. Write only 1 to this bit.
D1R/W00: If short-circuit protection is enabled for headphone driver and short circuit detected, device limits the
D0R00: Short circuit is not detected on the headphone driver.
RESET
VALUE
DESCRIPTION
1: HPL output driver is powered up.
1: HPR output driver is powered up.
01: Output common-mode voltage = 1.5 V
10: Output common-mode voltage = 1.65 V
11: Output common-mode voltage = 1.8 V
maximum current to the load.
1: If short-circuit protection is enabled for headphone driver and short circuit detected, device powers
down the output driver.
1: Short circuit is detected on the headphone driver.
Table 6-100. Page 1 / Register 33 (0x21): HP Output Drivers POP Removal Settings
READ/
BIT
WRITE
D7R/W00: If the power down sequence is activated by device software, power down using page 1 / register 46,
D6–D3R/W01110000: Driver power-on time = 0 μs
D2–D1R/W1100: Driver ramp-up step time = 0 ms
D0R/W00: Weakly driven output common-mode voltage is generated from resistor divider of the AVDD supply.
RESET
VALUE
DESCRIPTION
││ bit D7, then power down the DAC simultaneously with the HP and SP amplifiers.
1: If the power down sequence is activated by device software, power down using page 1 / register 46,
││ bit D7, then power down DAC only after HP and SP amplifiers are completely powered down. This is to
││ optimize power-down POP.
0001: Driver power-on time = 15.3 μs
0010: Driver power-on time = 153 μs
0011: Driver power-on time = 1.53 ms
0100: Driver power-on time = 15.3 ms
0101: Driver power-on time = 76.2 ms
0110: Driver power-on time = 153 ms
0111: Driver power-on time = 304 ms
1000: Driver power-on time = 610 ms
1001: Driver power-on time = 1.22 s
1010: Driver power-on time = 3.04 s
1011: Driver power-on time = 6.1 s
1100–1111: Reserved. Do not write these sequences to these bits.
Note: These values are based on typical oscillator frequency of 8.2 MHz. Scale according to the actual
oscillator frequency.
01: Driver ramp-up step time = 0.98 ms
10: Driver ramp-up step time = 1.95 ms
11: Driver ramp-up step time = 3.9 ms
Note: These values are based on typical oscillator frequency of 8.2 MHz. Scale according to the actual
oscillator frequency.
1: Reserved
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Table 6-101. Page 1 / Register 34 (0x22): Output Driver PGA Ramp-Down Period Control
READ/
BIT
WRITE
D7R/W0Reserved. Write only the reset value to this bit.
D6–D4R/W000Speaker power-up wait time (duration based on using internal oscillator)
D3–D0R/W0000Reserved. Write only the reset value to these bits.
RESET
VALUE
DESCRIPTION
000: Wait time = 0 ms
001: Wait time = 3.04 ms
010: Wait time = 7.62 ms
011: Wait time = 12.2 ms
100: Wait time = 15.3 ms
101: Wait time = 19.8 ms
110: Wait time = 24.4 ms
111: Wait time = 30.5 ms
Note: These values are based on typical oscillator frequency of 8.2 MHz. Scale according to the actual
oscillator frequency.
0001: HPL driver PGA = 1 dB
0010: HPL driver PGA = 2 dB
...
1000: HPL driver PGA = 8 dB
1001: HPL driver PGA = 9 dB
1010–1111: Reserved. Do not write these sequences to these bits.
D1R/W1Reserved. Write only '1' to this bit.
D0R00: Not all programmed gains to HPR have been applied yet.
RESET
VALUE
DESCRIPTION
0001: HPR driver PGA = 1 dB
0010: HPR driver PGA = 2 dB
...
1000: HPR driver PGA = 8 dB
1001: HPR driver PGA = 9 dB
1010–1111: Reserved. Do not write these sequences to these bits.
D7–D5R/W000Reserved. Write only zeros to these bits.
D4–D3R/W0000: Left-channel class-D driver output stage gain = 6 dB
D2R/W00: Left-channel class-D driver is muted.
D1R/W0Reserved. Write only zero to this bit.
D0R00: Not all programmed gains to the Left-channel class-D driver have been applied yet.
RESET
VALUE
DESCRIPTION
01: Left-channel class-D driver output stage gain = 12 dB
10: Left-channel class-D driver output stage gain = 18 dB
11: Left-channel class-D driver output stage gain = 24 dB
1: Left-channel class-D driver is not muted.
1: All programmed gains to the Left-channel class-D driver have been applied.
D7–D5R/W000Reserved. Write only zeros to these bits.
D4–D3R/W0000: Right-channel class-D driver output stage gain = 6 dB
D2R/W00: Right-channel class-D driver is muted.
D1R/W0Reserved. Write only zero to this bit.
D0R00: Not all programmed gains to right-channel class-D driver have been applied yet.
RESET
VALUE
DESCRIPTION
01: Right-channel class-D driver output stage gain = 12 dB
10: Right-channel class-D driver output stage gain = 18 dB
11: Right-channel class-D driver output stage gain = 24 dB
1: Right-channel class-D driver is not muted.
1: All programmed gains to right-channel class-D driver have been applied.
Table 6-111. Page 1 / Register 44 (0x2C): HP Driver Control
READ/
BIT
WRITE
D7–D5R/W000Debounce time for the headset short-circuit detection
D4–D3R/W0000: Default mode for the DAC
D2R/W00: HPL output driver is programmed as headphone driver.
D1R/W00: HPR output driver is programmed as headphone driver.
D0R/W0Reserved. Write only zero to this bit.
(1) The clock used for the debounce has a clock period = debounce duration / 8.
RESET
VALUE
DESCRIPTION
(1)
000: Debounce time =
001: Debounce time =
010: Debounce time =
011: Debounce time =
100: Debounce time =
101: Debounce time =
110: Debounce time =
111: Debounce time =
01: DAC performance increased by increasing the current
10: Reserved
11: DAC performance increased further by increasing the current again
1: HPL output driver is programmed as lineout driver.
1: HPR output driver is programmed as lineout driver.
MCLK/DIV (Page 3 /
register 16) = 1-MHz
Source
0 μs
8 μs
16 μs
32 μs
64 μs
128 μs
256 μs
512 μs
SLAS666B –JANUARY 2010–REVISED OCTOBER 2018
Internal Oscillator Source
0 μs
7.8 μs
15.6 μs
31.2 μs
62.4 μs
124.9 μs
250 μs
500 μs
Note: These values are based on a nominal oscillator
frequency of 8.2 MHz. The values scale to the actual
oscillator frequency.
The remaining page-8 registers are either reserved registers or are used for setting coefficients for the
various filters in the . Reserved registers must not be written to.
The filter coefficient registers are arranged in pairs, with two adjacent 8-bit registers containing the 16-bit
coefficient for a single filter. The 16-bit integer contained in the MSB and LSB registers for a coefficient
are interpreted as a 2s-complement integer, with possible values ranging from –32 768 to 32 767. When
programming any coefficient value for a filter, the MSB register must always be written first, immediately
followed by the LSB register. Even if only the MSB or LSB portion of the coefficient changes, both
registers must be written in this sequence. is a list of the page-8 registers, excepting the previously
described register 0.
0: Adaptive filtering disabled in DAC processing block
1: Adaptive filtering enabled in DAC processing block
0: In adaptive filter mode, DAC processing block accesses DAC coefficient Buffer A and the external
control interface accesses DAC coefficient Buffer B
1: In adaptive filter mode, DAC processing block accesses DAC coefficient Buffer B and the external
control interface accesses DAC coefficient Buffer A
0: DAC coefficient buffers are not switched at the next frame boundary.
1: DAC coefficient buffers are switched at the next frame boundary, if adaptive filtering mode is enabled.
This bit self-clears on switching.
Table 6-120. Page-8 DAC Buffer A Registers
REGISTER
NUMBER
2 (0x02)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad A
3 (0x03)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad A
4 (0x04)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad A
5 (0x05)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad A
6 (0x06)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad A
7 (0x07)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad A
8 (0x08)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad A
9 (0x09)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad A
10 (0x0A)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad A
11 (0x0B)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad A
12 (0x0C)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad B
13 (0x0D)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad B
14 (0x0E)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad B
15 (0x0F)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad B
16 (0x10)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad B
17 (0x11)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad B
18 (0x12)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad B
19 (0x13)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad B
20 (0x14)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad B
21 (0x15)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad B
22 (0x16)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad C
23 (0x17)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad C
24 (0x18)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad C
25 (0x19)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad C
26 (0x1A)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad C
27 (0x1B)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad C
28 (0x1C)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad C
29 (0x1D)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad C
30 (0x1E)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad C
31 (0x1F)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad C
32 (0x20)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad D
33 (0x21)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad D
34 (0x22)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad D
35 (0x23)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad D
Table 6-120. Page-8 DAC Buffer A Registers (continued)
REGISTER
NUMBER
36 (0x24)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad D
37 (0x25)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad D
38 (0x26)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad D
39 (0x27)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad D
40 (0x28)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad D
41 (0x29)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad D
42 (0x2A)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad E
43 (0x2B)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad E
44 (0x2C)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad E
45 (0x2D)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad E
46 (0x2E)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad E
47 (0x2F)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad E
48 (0x30)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad E
49 (0x31)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad E
50 (0x32)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad E
51 (0x33)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad E
52 (0x34)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad F
53 (0x35)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad F
54 (0x36)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad F
55 (0x37)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad F
56 (0x38)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad F
57 (0x39)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad F
58 (0x3A)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad F
59 (0x3B)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad F
60 (0x3C)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad F
61 (0x3D)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad F
62 (0x3E)0000 0000Reserved
63 (0x3F)0000 0000Reserved
64 (0x40)0000 00008 MSBs of 3D PGA gain for PRB_P23, PRB_P24 and PRB_P25
65 (0x41)0000 00008 LSBs of 3D PGA gain for PRB_P23, PRB_P24 and PRB_P25
66 (0x42)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad A
67 (0x43)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad A
68 (0x44)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad A
69 (0x45)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad A
70 (0x46)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad A
71 (0x47)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad A
72 (0x48)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad A
73 (0x49)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad A
74 (0x4A)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad A
75 (0x4B)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad A
76 (0x4C)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad B
77 (0x4D)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad B
78 (0x4E)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad B
79 (0x4F)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad B
80 (0x50)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad B
81 (0x51)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad B
82 (0x52)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad B
Table 6-120. Page-8 DAC Buffer A Registers (continued)
REGISTER
NUMBER
83 (0x53)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad B
84 (0x54)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad B
85 (0x55)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad B
86 (0x56)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad C
87 (0x57)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad C
88 (0x58)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad C
89 (0x59)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad C
90 (0x5A)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad C
91 (0x5B)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad C
92 (0x5C)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad C
93 (0x5D)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad C
94 (0x5E)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad C
95 (0x5F)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad C
96 (0x60)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad D
97 (0x61)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad D
98 (0x62)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad D
99 (0x63)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad D
100 (0x64)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad D
101 (0x65)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad D
102 (0x66)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad D
103 (0x67)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad D
104 (0x68)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad D
105 (0x69)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad D
106 (0x6A)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad E
107 (0x6B)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad E
108 (0x6C)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad E
109 (0x6D)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad E
110 (0x6E)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad E
111 (0x6F)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad E
112 (0x70)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad E
113 (0x71)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad E
114 (0x72)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad E
115 (0x73)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad E
116 (0x74)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad F
117 (0x75)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad F
118 (0x76)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad F
119 (0x77)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad F
120 (0x78)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad F
121 (0x79)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad F
122 (0x7A)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad F
123 (0x7B)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad F
124 (0x7C)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad F
125 (0x7D)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad F
The remaining page-9 registers are either reserved registers or are used for setting coefficients for the
various filters in the . Reserved registers must not be written to.
The filter-coefficient registers are arranged in pairs, with two adjacent 8-bit registers containing the 16-bit
coefficient for a single filter. The 16-bit integer contained in the MSB and LSB registers for a coefficient
are interpreted as a 2s-complement integer, with possible values ranging from –32 768 to 32 767. When
programming any coefficient value for a filter, the MSB register must always be written first, immediately
followed by the LSB register. Even if only the MSB or LSB portion of the coefficient changes, both
registers must be written in this sequence. is a list of the page-9 registers, excepting the previously
described register 0.
Table 6-122. Page-9 DAC Buffer A Registers
REGISTER
NUMBER
1 (0x01)XXXX XXXXReserved. Do not write to this register.
2 (0x02)0111 1111Coefficient N0(15:8) for left DAC-programmable first-order IIR
3 (0x03)1111 1111Coefficient N0(7:0) for left DAC-programmable first-order IIR
4 (0x04)0000 0000Coefficient N1(15:8) for left DAC-programmable first-order IIR
5 (0x05)0000 0000Coefficient N1(7:0) for left DAC-programmable first-order IIR
6 (0x06)0000 0000Coefficient D1(15:8) for left DAC-programmable first-order IIR
7 (0x07)0000 0000Coefficient D1(7:0) for left DAC-programmable first-order IIR
8 (0x08)0111 1111Coefficient N0(15:8) for right DAC-programmable first-order IIR
9 (0x09)1111 1111Coefficient N0(7:0) for right DAC-programmable first-order IIR
10 (0x0A)0000 0000Coefficient N1(15:8) for right DAC-programmable first-order IIR
11 (0x0B)0000 0000Coefficient N1(7:0) for right DAC-programmable first-order IIR
12 (0x0C)0000 0000Coefficient D1(15:8) for right DAC-programmable first-order IIR
13 (0x0D)0000 0000Coefficient D1(7:0) for right DAC-programmable first-order IIR
14 (0x0E)0111 1111Coefficient N0(15:8) for DRC first-order high-pass filter
15 (0x0F)1111 0111Coefficient N0(7:0) for DRC first-order high-pass filter
16 (0x10)1000 0000Coefficient N1(15:8) for DRC first-order high-pass filter
17 (0x11)0000 1001Coefficient N1(7:0) for DRC first-order high-pass filter
18 (0x12)0111 1111Coefficient D1(15:8) for DRC first-order high-pass filter
19 (0x13)1110 1111Coefficient D1(7:0) for DRC first-order high-pass filter
20 (0x14)0000 0000Coefficient N0(15:8) for DRC first-order low-pass filter
21 (0x15)0001 0001Coefficient N0(7:0) for DRC first-order low-pass filter
22 (0x16)0000 0000Coefficient N1(15:8) for DRC first-order low-pass filter
23 (0x17)0001 0001Coefficient N1(7:0) for DRC first-order low-pass filter
24 (0x18)0111 1111Coefficient D1(15:8) for DRC first-order low-pass filter
25 (0x19)1101 1110Coefficient D1(7:0) for DRC first-order low-pass filter
The remaining page-12 registers are either reserved registers or are used for setting coefficients for the
various filters in the . Reserved registers should not be written to.
The filter coefficient registers are arranged in pairs, with two adjacent 8-bit registers containing the 16-bit
coefficient for a single filter. The 16-bit integer contained in the MSB and LSB registers for a coefficient
are interpreted as a 2s-complement integer, with possible values ranging from –32 768 to 32 767. When
programming any coefficient value for a filter, the MSB register should always be written first, immediately
followed by the LSB register. Even if only the MSB or LSB portion of the coefficient changes, both
registers should be written in this sequence. is a list of the page-12 registers, excepting the previously
described register 0.
Table 6-124. Page-12 AC Buffer B Registers
REGISTER
NUMBER
1 (0x01)0000 0000Reserved. Do not write to this register.
2 (0x02)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad A
3 (0x03)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad A
4 (0x04)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad A
5 (0x05)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad A
6 (0x06)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad A
7 (0x07)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad A
8 (0x08)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad A
9 (0x09)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad A
10 (0x0A)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad A
11 (0x0B)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad A
12 (0x0C)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad B
13 (0x0D)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad B
14 (0x0E)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad B
15 (0x0F)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad B
16 (0x10)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad B
17 (0x11)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad B
18 (0x12)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad B
19 (0x13)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad B
20 (0x14)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad B
21 (0x15)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad B
22 (0x16)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad C
23 (0x17)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad C
24 (0x18)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad C
25 (0x19)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad C
Table 6-124. Page-12 AC Buffer B Registers (continued)
REGISTER
NUMBER
26 (0x1A)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad C
27 (0x1B)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad C
28 (0x1C)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad C
29 (0x1D)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad C
30 (0x1E)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad C
31 (0x1F)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad C
32 (0x20)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad D
33 (0x21)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad D
34 (0x22)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad D
35 (0x23)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad D
36 (0x24)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad D
37 (0x25)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad D
38 (0x26)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad D
39 (0x27)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad D
40 (0x28)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad D
41 (0x29)0000 0000Coefficient D2(17:0) for left DAC-programmable biquad D
42 (0x2A)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad E
43 (0x2B)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad E
44 (0x2C)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad E
45 (0x2D)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad E
46 (0x2E)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad E
47 (0x2F)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad E
48 (0x30)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad E
49 (0x31)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad E
50 (0x32)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad E
51 (0x33)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad E
52 (0x34)0111 1111Coefficient N0(15:8) for left DAC-programmable biquad F
53 (0x35)1111 1111Coefficient N0(7:0) for left DAC-programmable biquad F
54 (0x36)0000 0000Coefficient N1(15:8) for left DAC-programmable biquad F
55 (0x37)0000 0000Coefficient N1(7:0) for left DAC-programmable biquad F
56 (0x38)0000 0000Coefficient N2(15:8) for left DAC-programmable biquad F
57 (0x39)0000 0000Coefficient N2(7:0) for left DAC-programmable biquad F
58 (0x3A)0000 0000Coefficient D1(15:8) for left DAC-programmable biquad F
59 (0x3B)0000 0000Coefficient D1(7:0) for left DAC-programmable biquad F
60 (0x3C)0000 0000Coefficient D2(15:8) for left DAC-programmable biquad F
61 (0x3D)0000 0000Coefficient D2(7:0) for left DAC-programmable biquad F
62 (0x3E)0000 0000Reserved
63 (0x3F)0000 0000Reserved
64 (0x40)0000 00008 MSBs 3D PGA gain for PRB_P23, PRB_P24 and PRB_P25
65 (0x41)0000 00008 LSBs 3D PGA gain for PRB_P23, PRB_P24 and PRB_P25
66 (0x42)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad A
67 (0x43)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad A
68 (0x44)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad A
69 (0x45)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad A
70 (0x46)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad A
71 (0x47)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad A
72 (0x48)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad A
Table 6-124. Page-12 AC Buffer B Registers (continued)
REGISTER
NUMBER
73 (0x49)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad A
74 (0x4A)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad A
75 (0x4B)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad A
76 (0x4C)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad B
77 (0x4D)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad B
78 (0x4E)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad B
79 (0x4F)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad B
80 (0x50)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad B
81 (0x51)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad B
82 (0x52)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad B
83 (0x53)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad B
84 (0x54)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad B
85 (0x55)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad B
86 (0x56)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad C
87 (0x57)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad C
88 (0x58)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad C
89 (0x59)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad C
90 (0x5A)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad C
91 (0x5B)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad C
92 (0x5C)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad C
93 (0x5D)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad C
94 (0x5E)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad C
95 (0x5F)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad C
96 (0x60)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad D
97 (0x61)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad D
98 (0x62)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad D
99 (0x63)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad D
100 (0x64)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad D
101 (0x65)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad D
102 (0x66)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad D
103 (0x67)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad D
104 (0x68)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad D
105 (0x69)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad D
106 (0x6A)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad E
107 (0x6B)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad E
108 (0x6C)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad E
109 (0x6D)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad E
110 (0x6E)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad E
111 (0x6F)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad E
112 (0x70)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad E
113 (0x71)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad E
114 (0x72)0000 0000Coefficient ND2(15:8) for right DAC-programmable biquad E
115 (0x73)0000 0000Coefficient ND2(7:0) for right DAC-programmable biquad E
116 (0x74)0111 1111Coefficient N0(15:8) for right DAC-programmable biquad F
117 (0x75)1111 1111Coefficient N0(7:0) for right DAC-programmable biquad F
118 (0x76)0000 0000Coefficient N1(15:8) for right DAC-programmable biquad F
119 (0x77)0000 0000Coefficient N1(7:0) for right DAC-programmable biquad F
Table 6-124. Page-12 AC Buffer B Registers (continued)
REGISTER
NUMBER
120 (0x78)0000 0000Coefficient N2(15:8) for right DAC-programmable biquad F
121 (0x79)0000 0000Coefficient N2(7:0) for right DAC-programmable biquad F
122 (0x7A)0000 0000Coefficient D1(15:8) for right DAC-programmable biquad F
123 (0x7B)0000 0000Coefficient D1(7:0) for right DAC-programmable biquad F
124 (0x7C)0000 0000Coefficient D2(15:8) for right DAC-programmable biquad F
125 (0x7D)0000 0000Coefficient D2(7:0) for right DAC-programmable biquad F
The remaining page-13 registers are either reserved registers or are used for setting coefficients for the
various filters in the . Reserved registers must not be written to.
The filter coefficient registers are arranged in pairs, with two adjacent 8-bit registers containing the 16-bit
coefficient for a single filter. The 16-bit integer contained in the MSB and LSB registers for a coefficient
are interpreted as a 2s-complement integer, with possible values ranging from –32 768 to 32 767. When
programming any coefficient value for a filter, the MSB register must always be written first, immediately
followed by the LSB register. Even if only the MSB or LSB portion of the coefficient changes, both
registers must be written in this sequence. is a list of the page-13 registers, excepting the previously
described register 0.
Table 6-126. Page-13 DAC Buffer B Registers
REGISTER
NUMBER
100000000Reserved. Do not write to this register.
2 (0x02)0111 1111Coefficient N0(15:8) for left DAC-programmable first-order IIR
3 (0x03)1111 1111Coefficient N0(7:0) for left DAC-programmable first-order IIR
4 (0x04)0000 0000Coefficient N1(15:8) for left DAC-programmable first-order IIR
5 (0x05)0000 0000Coefficient N1(7:0) for left DAC-programmable first-order IIR
6 (0x06)0000 0000Coefficient D1(15:8) for left DAC-programmable first-order IIR
7 (0x07)0000 0000Coefficient D1(7:0) for left DAC-programmable first-order IIR
8 (0x08)0111 1111Coefficient N0(15:8) for right DAC-programmable first-order IIR
9 (0x09)1111 1111Coefficient N0(7:0) for right DAC-programmable first-order IIR
10 (0x0A)0000 0000Coefficient N1(15:8) for right DAC-programmable first-order IIR
11 (0x0B)0000 0000Coefficient N1(7:0) for right DAC-programmable first-order IIR
12 (0x0C)0000 0000Coefficient D1(15:8) for right DAC-programmable first-order IIR
13 (0x0D)0000 0000Coefficient D1(7:0) for right DAC-programmable first-order IIR
14 (0x0E)0111 1111Coefficient N0(15:8) for DRC first-order high-pass filter
15 (0x0F)1111 0111Coefficient N0(7:0) for DRC first-order high-pass filter
16 (0x10)1000 0000Coefficient N1(15:8) for DRC first-order high-pass filter
17 (0x11)0000 1001Coefficient N1(7:0) for DRC first-order high-pass filter
18 (0x12)0111 1111Coefficient D1(15:8) for DRC first-order high-pass filter
Information in the following applications sections is not part of the TI component
specification, and TI does not warrant its accuracy or completeness. TI’s customers are
responsible for determining suitability of components for their purposes. Customers should
validate and test their design implementation to confirm system functionality.
7.1Application Information
This typical connection highlights the required external components and system level connections for
proper operation of the device in several popular use cases.
Each of these configurations can be realized using the Evaluation Modules (EVMs) for the device. These
flexible modules allow full evaluation of the device in the most common modes of operation. Any design
variation can be supported by TI through schematic and layout reviews. Visit http://e2e.ti.com for design
assistance and join the audio amplifier discussion forum for additional information.
For this design example, use the parameters listed in Table 7-1 as the input parameters.
DESIGN PARAMETEREXAMPLE VALUE
AVDD3.3 V
DVDD1.8 V
HPVDD3.3 V
IOVDD3.3 V
Maximum MICBIAS current4 mA
SPKVDD5 V
7.2.2Detailed Design Procedure
Using Figure 7-1 as a guide, integrate the hardware into the system.
Following the recommended component placement, schematic layout and routing given in Section 9,
integrate the device and its supporting components into the system PCB file.
Determining sample rate and master clock frequency is required since powering up the device as all
internal timing is derived from the master clock. Refer to Section 6.3.11 to get more information of how to
configure correctly the required clocks for the device.
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Table 7-1. Design Parameters
As the TLV320DAC3101 is designed for low-power applications, when powered up, the device has several
features powered down. A correct routing of the TLV320DAC3101 signals is achieved by a correct setting
of the device registers, powering up the required stages of the device and configuring the internal switches
to follow a desired route.
The TLV320DAC3101 has been designed to be extremely tolerant of power supply sequencing. However,
in some rare cases, unexpected conditions and behaviors can be attributed to power supply sequencing.
It is important to consider that the digital activity must be separated from the analog and speaker activity.
In order to separate the power supplies, the recommended power sequence is:
1. Speaker supplies
2. Digital supplies
3. Analog supplies
First, turn on the speaker supplies. Once they are stabilized, turn on the digital power supplies. Finally,
once the digital power supplies are stabilized, the analog power supplies must be turned on.
Also, TI recommends to add decoupling capacitors close to the power supplies pins (see Section 9 for
details). These capacitors will ensure that the power pins will be stable. Additionally, undesired effects
such pops will be avoided.
PCB design is made considering the application and the review is specific for each system requirements.
However, general considerations can optimize the system performance.
•The TLV320DAC3101 thermal pad must be connected to analog output driver ground using multiple
VIAS to minimize impedance between the device and ground.
•Analog and digital grounds must be separated to prevent possible digital noise form affecting the
analog performance of the board.
•The TLV320DAC3101 requires the decoupling capacitors to be placed as close as possible to the
device power supply terminals.