• 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.