AD ADAU1461WBCPZ-RL 24-bit audio codec with integrated pllnull Datasheet

SigmaDSP Stereo, Low Power, 96 kHz,
24-Bit Audio Codec with Integrated PLL
ADAU1461
FEATURES
GENERAL DESCRIPTION
SigmaDSP 28-/56-bit, 50 MIPS digital audio processor
Fully programmable with SigmaStudio graphical tool
24-bit stereo audio ADC and DAC: >98 dB SNR
Sampling rates from 8 kHz to 96 kHz
Low power: 17 mW record, 18 mW playback, 48 kHz
6 analog input pins, configurable for single-ended or
differential inputs
Flexible analog input/output mixers
Stereo digital microphone input
Analog outputs: 2 differential stereo, 2 single-ended stereo,
1 mono headphone output driver
PLL supporting input clocks from 8 MHz to 27 MHz
Analog automatic level control (ALC)
Microphone bias reference voltage
Analog and digital I/O: 3.3 V
I2C and SPI control interfaces
Digital audio serial data I/O: stereo and time-division
multiplexing (TDM) modes
Software-controllable clickless mute
GPIO pins for digital controls and outputs
32-lead, 5 mm × 5 mm LFCSP
−40°C to +105°C operating temperature range
Qualified for automotive applications
The ADAU1461 is a low power, stereo audio codec with
integrated digital audio processing that supports stereo 48 kHz
record and playback at 35 mW from a 3.3 V analog supply. The
stereo audio ADCs and DACs support sample rates from 8 kHz
to 96 kHz as well as a digital volume control.
The SigmaDSP® core features 28-bit processing (56-bit double
precision). The processor allows system designers to compensate
for the real-world limitations of microphones, speakers, amplifiers,
and listening environments, resulting in a dramatic improvement
in the perceived audio quality through equalization, multiband
compression, limiting, and third-party branded algorithms.
The SigmaStudio™ graphical development tool is used to program
the ADAU1461. This software includes audio processing blocks
such as filters, dynamics processors, mixers, and low level DSP
functions for fast development of custom signal flows.
The record path includes an integrated microphone bias circuit
and six inputs. The inputs can be mixed and muxed before the
ADC, or they can be configured to bypass the ADC. The
ADAU1461 includes a stereo digital microphone input.
The ADAU1461 includes five high power output drivers (two
differential and three single-ended), supporting stereo headphones, an earpiece, or other output transducer. AC-coupled
or capless configurations are supported. Individual fine level
controls are supported on all analog outputs. The output mixer
stage allows for flexible routing of audio.
APPLICATIONS
Automotive head units
Automotive amplifiers
Navigation systems
Rear-seat entertainment systems
HP JACK
DETECTION
JACKDET/MICIN
AGND
AGND
AVDD
AVDD
DVDDOUT
DGND
IOVDD
CM
FUNCTIONAL BLOCK DIAGRAM
ADAU1461
REGULATOR
LAUX
LOUTP
LINN
INPUT
MIXERS
RINP
ALC
ADC
ADC
DAC
DIGITAL DIGITAL
FILTERS FILTERS
ADC
LOUTN
DAC
DAC
LHP
OUTPUT
MIXERS
MONOOUT
RHP
RINN
ROUTP
RAUX
ROUTN
DAC_SDATA/
GPIO0
MCLK
SERIAL DATA
INPUT/OUTPUT PORTS
LRCLK/
GPIO3
PLL
BCLK/
GPIO2
MICROPHONE
BIAS
ADC_SDATA/
GPIO1
MICBIAS
I2C/SPI
CONTROL PORT
ADDR0/ ADDR1/ SCL/ SDA/
CLATCH CDATA CCLK COUT
08914-001
LINP
Figure 1.
Rev. 0
Information furnished by Analog Devices is believed to be accurate and reliable. However, no
responsibility is assumed by Analog Devices for its use, nor for any infringements of patents or other
rights of third parties that may result from its use. Specifications subject to change without notice. No
license is granted by implication or otherwise under any patent or patent rights of Analog Devices.
Trademarks and registered trademarks are the property of their respective owners.
One Technology Way, P.O. Box 9106, Norwood, MA 02062-9106, U.S.A.
Tel: 781.329.4700
www.analog.com
Fax: 781.461.3113
©2010 Analog Devices, Inc. All rights reserved.
ADAU1461* PRODUCT PAGE QUICK LINKS
Last Content Update: 02/23/2017
COMPARABLE PARTS
DESIGN RESOURCES
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• ADAU1461 Material Declaration
• PCN-PDN Information
DOCUMENTATION
• Quality And Reliability
Application Notes
• Symbols and Footprints
• AN-1006: Using the EVAL-ADUSB2EBZ
• AN-951: Using Hardware Controls with SigmaDSP GPIO
Pins
DISCUSSIONS
View all ADAU1461 EngineerZone Discussions.
Data Sheet
• ADAU1461: SigmaDSP Stereo, Low Power, 96 kHz, 24-Bit
Audio Codec with Integrated PLL Data Sheet
SOFTWARE AND SYSTEMS REQUIREMENTS
• ADAU1361 Sound CODEC Linux Driver
SAMPLE AND BUY
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TECHNICAL SUPPORT
• Firmware Loader for SigmaDSPs
Submit a technical question or find your regional support
number.
TOOLS AND SIMULATIONS
DOCUMENT FEEDBACK
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• ADAU1361/ADAU1761 IBIS Models
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ADAU1461
TABLE OF CONTENTS
Features .............................................................................................. 1
Playback Signal Path ...................................................................... 33
Applications ....................................................................................... 1
Output Signal Paths ................................................................... 33
General Description ......................................................................... 1
Headphone Output .................................................................... 34
Functional Block Diagram .............................................................. 1
Pop-and-Click Suppression ...................................................... 35
Revision History ............................................................................... 2
Line Outputs ............................................................................... 35
Specifications..................................................................................... 3
Control Ports ................................................................................... 36
Analog Performance Specifications, TA = 25°C ....................... 3
Burst Mode Writing and Reading ............................................ 36
Analog Performance Specifications, −40°C < TA < +105°C ... 5
I2C Port ........................................................................................ 36
Power Supply Specifications........................................................ 7
SPI Port ........................................................................................ 39
Digital Filters ................................................................................. 8
Serial Data Input/Output Ports .................................................... 40
Digital Input/Output Specifications........................................... 8
Applications Information .............................................................. 42
Digital Timing Specifications ..................................................... 9
Power Supply Bypass Capacitors .............................................. 42
Digital Timing Diagrams........................................................... 10
GSM Noise Filter ........................................................................ 42
Absolute Maximum Ratings.......................................................... 12
Grounding ................................................................................... 42
Thermal Resistance .................................................................... 12
Exposed Pad PCB Design ......................................................... 42
ESD Caution ................................................................................ 12
DSP Core ......................................................................................... 43
Pin Configuration and Function Descriptions ........................... 13
Signal Processing ........................................................................ 43
Typical Performance Characteristics ........................................... 15
Architecture ................................................................................ 43
System Block Diagrams ................................................................. 18
Program Counter ....................................................................... 43
Theory of Operation ...................................................................... 21
Features ........................................................................................ 43
Startup, Initialization, and Power ................................................. 22
Startup .......................................................................................... 43
Power-Up Sequence ................................................................... 22
Numeric Formats ....................................................................... 44
Power Reduction Modes............................................................ 22
Programming .............................................................................. 44
Digital Power Supply .................................................................. 22
Program RAM, Parameter RAM, and Data RAM ..................... 45
Input/Output Power Supply ...................................................... 22
Program RAM ............................................................................ 45
Clock Generation and Management ........................................ 22
Parameter RAM .......................................................................... 45
Clocking and Sampling Rates ....................................................... 24
Data RAM ................................................................................... 45
Core Clock ................................................................................... 24
Read/Write Data Formats ......................................................... 45
Sampling Rates ............................................................................ 25
Software Safeload ....................................................................... 46
PLL ............................................................................................... 25
Software Slew .............................................................................. 47
Record Signal Path.......................................................................... 27
General-Purpose Input/Output .................................................... 48
Input Signal Paths ....................................................................... 27
GPIO Pins Set from the Control Port...................................... 48
Analog-to-Digital Converters ................................................... 29
Control Registers ............................................................................ 49
Automatic Level Control (ALC) ................................................... 30
Control Register Details ............................................................ 50
ALC Parameters .......................................................................... 30
Outline Dimensions ....................................................................... 88
Noise Gate Function .................................................................. 31
Ordering Guide .......................................................................... 88
Automotive Products ................................................................. 88
REVISION HISTORY
6/10—Revision 0: Initial Version
Rev. 0 | Page 2 of 88
ADAU1461
SPECIFICATIONS
Supply voltage (AVDD) = 3.3 V, TA = 25°C, master clock = 12.288 MHz (48 kHz fS, 256 × fS mode), input sample rate = 48 kHz, measurement
bandwidth = 20 Hz to 20 kHz, word width = 24 bits, CLOAD (digital output) = 20 pF, ILOAD (digital output) = 2 mA, VIH = 2 V, VIL = 0.8 V,
unless otherwise noted. Performance of all channels is identical, exclusive of the interchannel gain mismatch and interchannel phase
deviation specifications.
ANALOG PERFORMANCE SPECIFICATIONS, TA = 25°C
IOVDD = 3.3 V ± 10%.
Table 1.
Parameter
ANALOG-TO-DIGITAL CONVERTERS
ADC Resolution
Digital Attenuation Step
Digital Attenuation Range
INPUT RESISTANCE
Single-Ended Line Input
PGA Inverting Inputs
PGA Noninverting Inputs
SINGLE-ENDED LINE INPUT
Full-Scale Input Voltage (0 dB)
Dynamic Range
With A-Weighted Filter (RMS)
No Filter (RMS)
Total Harmonic Distortion + Noise
Signal-to-Noise Ratio
With A-Weighted Filter (RMS)
No Filter (RMS)
Input Mixer Gain per Step
Mute Attenuation
Interchannel Gain Mismatch
Offset Error
Gain Error
Interchannel Isolation
Power Supply Rejection Ratio
PSEUDO-DIFFERENTIAL PGA INPUT
Full-Scale Input Voltage (0 dB)
Dynamic Range
With A-Weighted Filter (RMS)
No Filter (RMS)
Total Harmonic Distortion + Noise
Signal-to-Noise Ratio
With A-Weighted Filter (RMS)
No Filter (RMS)
PGA Boost Gain Error
Test Conditions/Comments
ADC performance excludes mixers
and PGA
All ADCs
Min
−12 dB gain
0 dB gain
6 dB gain
−12 dB gain
0 dB gain
35.25 dB gain
All gains
Typ
Max
Unit
24
0.375
95
Bits
dB
dB
80.4
21
10.5
84.5
53
1.7
105
kΩ
kΩ
kΩ
kΩ
kΩ
kΩ
kΩ
1.0 (2.83)
V rms (V p-p)
99
96
−90
−71
dB
dB
dB
3.07
−77
dB
dB
dB
dB
20 Hz to 20 kHz, −60 dB input
83.5
83
−1 dBFS
−12 dB to +6 dB range
LINPG[2:0], LINNG[2:0] = 000,
RINPG[2:0], RINNG[2:0] = 000,
MX1AUXG[2:0], MX2AUXG[2:0] = 000
2.89
−0.3
−5
−17
CM capacitor = 20 μF, 100 mV p-p @ 1 kHz
99
96
3
−85.5
+0.032
0
−12
68
67
+0.3
+5
−8
dB
mV
%
dB
dB
1.0 (2.83)
V rms (V p-p)
98
95
−89
−83
dB
dB
dB
+8
dB
dB
dB
20 Hz to 20 kHz, −60 dB input
94
91
−1 dBFS
20 dB gain setting (RDBOOST[1:0],
LDBOOST[1:0] = 10)
Rev. 0 | Page 3 of 88
−8
98
95
+0.4
ADAU1461
Parameter
Mute Attenuation
Interchannel Gain Mismatch
Offset Error
Gain Error
Interchannel Isolation
Common-Mode Rejection Ratio
FULL DIFFERENTIAL PGA INPUT
Full-Scale Input Voltage (0 dB)
Dynamic Range
With A-Weighted Filter (RMS)
No Filter (RMS)
Total Harmonic Distortion + Noise
Signal-to-Noise Ratio
With A-Weighted Filter (RMS)
No Filter (RMS)
PGA Boost Gain Error
Mute Attenuation
Interchannel Gain Mismatch
Offset Error
Gain Error
Interchannel Isolation
Common-Mode Rejection Ratio
MICROPHONE BIAS
Bias Voltage
0.65 × AVDD
0.90 × AVDD
Bias Current Source
Noise in the Signal Bandwidth
DIGITAL-TO-ANALOG CONVERTERS
DAC Resolution
Digital Attenuation Step
Digital Attenuation Range
DAC TO LINE OUTPUT
Full-Scale Output Voltage (0 dB)
Dynamic Range
Test Conditions/Comments
PGA muted
LDMUTE, RDMUTE = 0
RDBOOST[1:0], LDBOOST[1:0] = 00
Min
−0.6
−6
−24
100 mV rms, 1 kHz
100 mV rms, 20 kHz
Differential PGA inputs
−52
Typ
Max
Unit
−76
−87
−0.073
0
−14
83
−58
−48
−73
−82
+0.6
+6
−3
dB
dB
dB
mV
%
dB
dB
dB
−44
1.0 (2.83)
V rms (V p-p)
98
95
−78
−74
dB
dB
dB
+8
dB
dB
dB
20 Hz to 20 kHz, −60 dB input
94
91
−1 dBFS
20 dB gain setting (RDBOOST[1:0],
LDBOOST[1:0] = 10)
PGA muted
LDMUTE, RDMUTE = 0
RDBOOST[1:0], LDBOOST[1:0] = 00
−8
−73
−82
+0.3
+6
−9
−52
−76
−87
−0.0005
0
−14
83
−58
−48
−44
dB
dB
dB
mV
%
dB
dB
dB
2.00
2.04
2.89
2.89
2.145
2.13
2.97
2.99
2.19
2.21
3.04
3.11
3
V
V
V
V
mA
13
42
85
25
22
36
nV/√Hz
nV/√Hz
nV/√Hz
nV/√Hz
−0.3
−6
−17
100 mV rms, 1 kHz
100 mV rms, 20 kHz
MBIEN = 1
MBI = 1, MPERF = 0
MBI = 1, MPERF = 1
MBI = 0, MPERF = 0
MBI = 0, MPERF = 1
MBI = 0, MPERF = 1
1 kHz to 20 kHz
MBI = 0, MPERF = 0
MBI = 0, MPERF = 1
MBI = 1, MPERF = 0
MBI = 1, MPERF = 1
DAC performance excludes mixers and
headphone amplifier
All DACs
98
95
−0.15
24
0.375
95
Bits
dB
dB
0.92 (2.60)
V rms (V p-p)
101
98
dB
dB
20 Hz to 20 kHz, −60 dBFS input, line
output mode
With A-Weighted Filter (RMS)
No Filter (RMS)
95
93.5
Rev. 0 | Page 4 of 88
ADAU1461
Parameter
Total Harmonic Distortion + Noise
Line Output Mode
Headphone Output Mode
Signal-to-Noise Ratio
With A-Weighted Filter (RMS)
No Filter (RMS)
Mute Attenuation
Mixer 3 and Mixer 4 Muted
Mixer 5, Mixer 6, and Mixer 7 Muted
All Volume Controls Muted
Interchannel Gain Mismatch
Offset Error
Gain Error
Interchannel Isolation
Power Supply Rejection Ratio
DAC TO HEADPHONE/EARPIECE
OUTPUT
Full-Scale Output Voltage (0 dB)
Total Harmonic Distortion + Noise
Capless Headphone Mode
Headphone Output Mode
Interchannel Isolation
Power Supply Rejection Ratio
REFERENCE
Common-Mode Reference Output
Test Conditions/Comments
0 dBFS, 10 kΩ load
Min
Typ
Max
Unit
−92
−89
−77
−79
dB
dB
Line output mode
101
98
MX3RM, MX3LM, MX4RM, MX4LM = 0,
MX3AUXG[3:0], MX4AUXG[3:0] = 0000,
MX3G1[3:0], MX3G2[3:0] = 0000,
MX4G1[3:0], MX4G2[3:0] = 0000
MX5G3[1:0], MX5G4[1:0], MX6G3[1:0],
MX6G4[1:0], MX7[1:0] = 00
LOUTM, ROUTM = 0
MONOM, LHPM, RHPM = 0
−0.3
−22
−10
1 kHz, 0 dBFS input signal
CM capacitor = 20 μF, 100 mV p-p @ 1 kHz
LOUTx, ROUTx, LHP, RHP in headphone
output mode; PO = output power per
channel
Scales linearly with AVDD
−4 dBFS, 16 Ω load, PO = 21.1 mW
−4 dBFS, 32 Ω load, PO = 10.6 mW
−2 dBFS, 16 Ω load
−2 dBFS, 32 Ω load
0 dBFS, 10 kΩ load
1 kHz, 0 dBFS input signal, 32 Ω load
Referred to GND
Referred to CM (capless headphone
mode)
CM capacitor = 20 μF, 100 mV p-p @ 1 kHz
CM pin
dB
dB
−85
−78
dB
−89
−80
dB
−82
−74
−0.005
0
+3
100
70
−74
−69
+0.3
+22
+10
dB
dB
dB
mV
%
dB
dB
0.92 (2.60)
−82
−82
−78
−75
−86
−71
−65
−77
V rms (V p-p)
dB
dB
dB
dB
dB
73
50
dB
dB
67
dB
1.62
1.65
1.67
V
Min
Typ
Max
Unit
2.88
−67
3.09
−77
dB
dB
dB
dB
dB
−0.5
−5
−22
+0.5
+5
−6
dB
mV
%
ANALOG PERFORMANCE SPECIFICATIONS, −40°C < TA < +105°C
IOVDD = 3.3 V ± 10%.
Table 2.
Parameter
SINGLE-ENDED LINE INPUT
Dynamic Range
With A-Weighted Filter (RMS)
No Filter (RMS)
Total Harmonic Distortion + Noise
Input Mixer Gain per Step
Mute Attenuation
Test Conditions/Comments
20 Hz to 20 kHz, −60 dB input
74
71
−1 dBFS
−12 dB to +6 dB range
LINPG[2:0], LINNG[2:0] = 000,
RINPG[2:0], RINNG[2:0] = 000,
MX1AUXG[2:0], MX2AUXG[2:0] = 000
Interchannel Gain Mismatch
Offset Error
Gain Error
Rev. 0 | Page 5 of 88
ADAU1461
Parameter
PSEUDO-DIFFERENTIAL PGA INPUT
Dynamic Range
With A-Weighted Filter (RMS)
No Filter (RMS)
Total Harmonic Distortion + Noise
PGA Boost Gain Error
Mute Attenuation
Interchannel Gain Mismatch
Offset Error
Gain Error
Common-Mode Rejection Ratio
FULL DIFFERENTIAL PGA INPUT
Dynamic Range
With A-Weighted Filter (RMS)
No Filter (RMS)
Total Harmonic Distortion + Noise
PGA Boost Gain Error
Mute Attenuation
Interchannel Gain Mismatch
Offset Error
Gain Error
Common-Mode Rejection Ratio
MICROPHONE BIAS
Bias Voltage
0.65 × AVDD
0.90 × AVDD
Noise in the Signal Bandwidth
DAC TO LINE OUTPUT
Dynamic Range
With A-Weighted Filter (RMS)
No Filter (RMS)
Total Harmonic Distortion + Noise
Line Output Mode
Headphone Output Mode
Mute Attenuation
Mixer 3 and Mixer 4 Muted
Mixer 5, Mixer 6, and Mixer 7 Muted
All Volume Controls Muted
Test Conditions/Comments
Min
Typ
Max
Unit
−11
−75
−7
dB
dB
dB
dB
−0.6
−6
−24
−64
−53
−73
−82
+0.6
+6
−3
−38
−43
dB
dB
dB
mV
%
dB
dB
−11
−70
−7
dB
dB
dB
dB
−0.4
−6
−21
−64
−53
−73
−82
+0.4
+6
−7
−38
−43
dB
dB
dB
mV
%
dB
dB
1.85
1.87
2.65
2.65
11
2.45
2.45
3.40
3.40
36
V
V
V
V
nV/√Hz
20 Hz to 20 kHz, −60 dB input
94
91
−1 dBFS
20 dB gain setting (RDBOOST[1:0],
LDBOOST[1:0] = 10)
PGA muted
LDMUTE, RDMUTE = 0
RDBOOST[1:0], LDBOOST[1:0] = 00
100 mV rms, 1 kHz
100 mV rms, 20 kHz
Differential PGA inputs
20 Hz to 20 kHz, −60 dB input
89
86
−1 dBFS
20 dB gain setting (RDBOOST[1:0],
LDBOOST[1:0] = 10)
PGA muted
LDMUTE, RDMUTE = 0
RDBOOST[1:0], LDBOOST[1:0] = 00
100 mV rms, 1 kHz
100 mV rms, 20 kHz
MBIEN = 1
MBI = 1, MPERF = 0
MBI = 1, MPERF = 1
MBI = 0, MPERF = 0
MBI = 0, MPERF = 1
1 kHz to 20 kHz
20 Hz to 20 kHz, −60 dB input, line
output mode
85
78
dB
dB
0 dBFS, 10 kΩ load
MX3RM, MX3LM, MX4RM, MX4LM = 0,
MX3AUXG[3:0], MX4AUXG[3:0] = 0000,
MX3G1[3:0], MX3G2[3:0] = 0000,
MX4G1[3:0], MX4G2[3:0] = 0000
MX5G3[1:0], MX5G4[1:0], MX6G3[1:0],
MX6G4[1:0], MX7[1:0] = 00
LOUTM, ROUTM = 0
MONOM, LHPM, RHPM = 0
Rev. 0 | Page 6 of 88
−76
−78
dB
dB
−77
dB
−77
dB
−74
−69
dB
dB
ADAU1461
Parameter
Interchannel Gain Mismatch
Offset Error
Gain Error
DAC TO HEADPHONE/EARPIECE
OUTPUT
Total Harmonic Distortion + Noise
Capless Headphone Mode
Headphone Output Mode
REFERENCE
Common-Mode Reference Output
Test Conditions/Comments
Min
−0.3
−22
−10
Typ
Max
+0.3
+22
+10
Unit
dB
mV
%
−61
−63
−76
dB
dB
dB
1.83
V
LOUTx, ROUTx, LHP, RHP in headphone
output mode; PO = output power per
channel
−2 dBFS, 16 Ω load
−2 dBFS, 32 Ω load
0 dBFS, 10 kΩ load
CM pin
1.47
POWER SUPPLY SPECIFICATIONS
Master clock = 12.288 MHz, input sample rate = 48 kHz, input tone = 1 kHz, ADC input @ −1 dBFS, DAC input @ 0 dBFS,
−40°C < TA < +105°C, IOVDD = 3.3 V ± 10%. For total power consumption, add the IOVDD current listed in Table 3.
Table 3.
Parameter
SUPPLIES
Voltage
Digital I/O Current (IOVDD)
Slave Mode
Master Mode
Test Conditions/Comments
DVDDOUT
AVDD
IOVDD
20 pF capacitive load on all digital pins
fS = 48 kHz
fS = 96 kHz
fS = 8 kHz
fS = 48 kHz
fS = 96 kHz
fS = 8 kHz
Min
Typ
Max
Unit
2.97
2.97
1.56
3.3
3.3
3.65
3.65
V
V
V
0.48
0.9
0.13
1.51
3
0.27
mA
mA
mA
mA
mA
mA
5.24
6.57
mA
mA
5.55
6.90
mA
mA
30.9
32.25
mA
mA
56.75
58
mA
mA
Analog Current (AVDD)
Record Stereo Differential to ADC
DAC Stereo Playback to Line Output
DAC Stereo Playback to Headphone
DAC Stereo Playback to Capless Headphone
PLL bypass
Integer PLL
10 kΩ load
PLL bypass
Integer PLL
32 Ω load
PLL bypass
Integer PLL
32 Ω load
PLL bypass
Integer PLL
Rev. 0 | Page 7 of 88
ADAU1461
DIGITAL FILTERS
Table 4.
Parameter
ADC DECIMATION FILTER
Pass Band
Pass-Band Ripple
Transition Band
Stop Band
Stop-Band Attenuation
Group Delay
DAC INTERPOLATION FILTER
Pass Band
Pass-Band Ripple
Transition Band
Stop Band
Stop-Band Attenuation
Group Delay
Mode
All modes, typ @ 48 kHz
Factor
Min
0.4375 fS
Max
Unit
22.9844/fS
21
±0.015
24
27
67
479
kHz
dB
kHz
kHz
dB
μs
0.4535 fS
0.3646 fS
22
35
kHz
kHz
dB
dB
kHz
kHz
kHz
kHz
dB
dB
μs
μs
0.5 fS
0.5625 fS
48 kHz mode, typ @ 48 kHz
96 kHz mode, typ @ 96 kHz
48 kHz mode, typ @ 48 kHz
96 kHz mode, typ @ 96 kHz
48 kHz mode, typ @ 48 kHz
96 kHz mode, typ @ 96 kHz
48 kHz mode, typ @ 48 kHz
96 kHz mode, typ @ 96 kHz
48 kHz mode, typ @ 48 kHz
96 kHz mode, typ @ 96 kHz
48 kHz mode, typ @ 48 kHz
96 kHz mode, typ @ 96 kHz
Typ
±0.01
±0.05
0.5 fS
0.5 fS
0.5465 fS
0.6354 fS
24
48
26
61
69
68
521
115
25/fS
11/fS
DIGITAL INPUT/OUTPUT SPECIFICATIONS
−40°C < TA < +105°C, IOVDD = 3.3 V ± 10%.
Table 5.
Parameter
INPUT SPECIFICATIONS
Input Voltage High (VIH)
Input Voltage Low (VIL)
Input Leakage
Pull-Ups/Pull-Downs Disabled
Pull-Ups Enabled
Pull-Downs Enabled
Input Capacitance
OUTPUT SPECIFICATIONS
Output Voltage High (VOH)
Output Voltage Low (VOL)
Test Conditions/Comments
Min
Typ
Max
Unit
0.3 × IOVDD
V
V
+0.17
+0.17
−0.5
+0.7
−0.5
8.3
+0.18
5
μA
μA
μA
μA
μA
μA
μA
pF
0.1 × IOVDD
V
V
0.7 × IOVDD
IIH @ VIH = 3.3 V
IIL @ VIL = 0 V
IIL @ VIL = 0 V (MCLK pin)
IIH @ VIH = 3.3 V
IIL @ VIL = 0 V
IIH @ VIH = 3.3 V
IIL @ VIL = 0 V
−0.17
−0.17
−13.5
−0.7
−13.5
2.7
−0.18
IOH = 2 mA @ 3.3 V
IOL = 2 mA @ 3.3 V
0.8 × IOVDD
Rev. 0 | Page 8 of 88
ADAU1461
DIGITAL TIMING SPECIFICATIONS
−40°C < TA < +105°C, IOVDD = 3.3 V ± 10%.
Table 6. Digital Timing
Parameter
MASTER CLOCK
tMP
tMP
tMP
tMP
SERIAL PORT
tBIL
tBIH
tLIS
tLIH
tSIS
tSIH
tSODM
SPI PORT
fCCLK
tCCPL
tCCPH
tCLS
tCLH
tCLPH
tCDS
tCDH
tCOD
I2C PORT
fSCL
tSCLH
tSCLL
tSCS
tSCH
tDS
tSCR
tSCF
tSDR
tSDF
tBFT
DIGITAL MICROPHONE
tDCF
tDCR
tDDV
tDDH
tMIN
74
37
24.7
18.5
Limit
tMAX
Unit
Description
488
244
162.7
122
ns
ns
ns
ns
MCLK period, 256 × fS mode.
MCLK period, 512 × fS mode.
MCLK period, 768 × fS mode.
MCLK period, 1024 × fS mode.
50
ns
ns
ns
ns
ns
ns
ns
BCLK pulse width low.
BCLK pulse width high.
LRCLK setup. Time to BCLK rising.
LRCLK hold. Time from BCLK rising.
DAC_SDATA setup. Time to BCLK rising.
DAC_SDATA hold. Time from BCLK rising.
ADC_SDATA delay. Time from BCLK falling in master mode.
MHz
ns
ns
ns
ns
ns
ns
ns
ns
CCLK frequency.
CCLK pulse width low.
CCLK pulse width high.
CLATCH setup. Time to CCLK rising.
CLATCH hold. Time from CCLK rising.
CLATCH pulse width high.
CDATA setup. Time to CCLK rising.
CDATA hold. Time from CCLK rising.
COUT three-stated. Time from CLATCH rising.
kHz
μs
μs
μs
μs
ns
ns
ns
ns
ns
μs
SCL frequency.
SCL high.
SCL low.
Setup time; relevant for repeated start condition.
Hold time. After this period, the first clock is generated.
Data setup time.
SCL rise time.
SCL fall time.
SDA rise time.
SDA fall time.
Bus-free time. Time between stop and start.
RLOAD = 1 MΩ, CLOAD = 14 pF.
Digital microphone clock fall time.
Digital microphone clock rise time.
Digital microphone delay time for valid data.
Digital microphone delay time for data three-stated.
5
5
5
5
5
5
10
10
10
5
10
10
5
5
50
400
0.6
1.3
0.6
0.6
100
300
300
300
300
0.6
22
0
10
10
30
12
ns
ns
ns
ns
Rev. 0 | Page 9 of 88
ADAU1461
DIGITAL TIMING DIAGRAMS
tLIH
tBIH
BCLK
tBIL
tLIS
LRCLK
tSIS
DAC_SDATA
LEFT-JUSTIFIED
MODE
MSB
MSB – 1
tSIH
tSIS
DAC_SDATA
I2S MODE
MSB
tSIH
tSIS
tSIS
DAC_SDATA
RIGHT-JUSTIFIED
MODE
LSB
MSB
tSIH
tSIH
8-BIT CLOCKS
(24-BIT DATA)
12-BIT CLOCKS
(20-BIT DATA)
08914-002
14-BIT CLOCKS
(18-BIT DATA)
16-BIT CLOCKS
(16-BIT DATA)
Figure 2. Serial Input Port Timing
tBIH
BCLK
tBIL
LRCLK
ADC_SDATA
LEFT-JUSTIFIED
MODE
tSODM
MSB
MSB – 1
tSODM
ADC_SDATA
I2S MODE
MSB
tSODM
ADC_SDATA
RIGHT-JUSTIFIED
MODE
MSB
LSB
8-BIT CLOCKS
(24-BIT DATA)
12-BIT CLOCKS
(20-BIT DATA)
08914-003
14-BIT CLOCKS
(18-BIT DATA)
16-BIT CLOCKS
(16-BIT DATA)
Figure 3. Serial Output Port Timing
Rev. 0 | Page 10 of 88
ADAU1461
tCLS
tCLH
tCLPH
tCCPL
tCCPH
CLATCH
CCLK
CDATA
tCDH
tCDS
COUT
08914-004
tCOD
Figure 4. SPI Port Timing
tDS
tSCH
tSCH
SDA
tSCR
tSCLH
tSCLL
tSCS
tSCF
08914-005
SCL
tBFT
Figure 5. I2C Port Timing
tDCF
tDCR
CLK
tDDH
DATA2
tDDV
tDDV
DATA1
DATA2
08914-006
DATA1/
DATA2 DATA1
tDDH
Figure 6. Digital Microphone Timing
Rev. 0 | Page 11 of 88
ADAU1461
ABSOLUTE MAXIMUM RATINGS
THERMAL RESISTANCE
Table 7.
Parameter
Power Supply (AVDD)
Input Current (Except Supply Pins)
Analog Input Voltage (Signal Pins)
Digital Input Voltage (Signal Pins)
Operating Temperature Range
Storage Temperature Range
Rating
−0.3 V to +3.65 V
±20 mA
−0.3 V to AVDD + 0.3 V
−0.3 V to IOVDD + 0.3 V
−40°C to +105°C
−65°C to +150°C
Stresses above those listed under Absolute Maximum Ratings
may cause permanent damage to the device. This is a stress
rating only; functional operation of the device at these or any
other conditions above those indicated in the operational
section of this specification is not implied. Exposure to absolute
maximum rating conditions for extended periods may affect
device reliability.
θJA represents thermal resistance, junction-to-ambient; θJC represents thermal resistance, junction-to-case. All characteristics are
for a 4-layer board.
Table 8. Thermal Resistance
Package Type
32-Lead LFCSP
ESD CAUTION
Rev. 0 | Page 12 of 88
θJA
50.1
θJC
17
Unit
°C/W
ADAU1461
32
31
30
29
28
27
26
25
SCL/CCLK
SDA/COUT
ADDR1/CDATA
LRCLK/GPIO3
BCLK/GPIO2
DAC_SDATA/GPIO0
ADC_SDATA/GPIO1
DGND
PIN CONFIGURATION AND FUNCTION DESCRIPTIONS
1
2
3
4
5
6
7
8
PIN 1
INDICATOR
ADAU1461
TOP VIEW
(Not to Scale)
24
23
22
21
20
19
18
17
DVDDOUT
AVDD
AGND
MONOOUT
LHP
RHP
LOUTP
LOUTN
NOTES
1. THE EXPOSED PAD IS CONNECTED INTERNALLY TO THE
ADAU1461 GROUNDS. FOR INCREASED RELIABILITY OF THE
SOLDER JOINTS AND MAXIMUM THERMAL CAPABILITY, IT IS
RECOMMENDED THAT THE PAD BE SOLDERED TO THE
GROUND PLANE.
08914-007
AGND
LINP
LINN
RINP
RINN
RAUX
ROUTP
ROUTN
9
10
11
12
13
14
15
16
IOVDD
MCLK
ADDR0/CLATCH
JACKDET/MICIN
MICBIAS
LAUX
CM
AVDD
Figure 7. Pin Configuration
Table 9. Pin Function Descriptions
Pin No.
1
Mnemonic
IOVDD
Type 1
PWR
2
3
MCLK
ADDR0/CLATCH
D_IN
D_IN
4
JACKDET/MICIN
D_IN
5
6
7
MICBIAS
LAUX
CM
A_OUT
A_IN
A_OUT
8
AVDD
PWR
9
AGND
PWR
10
11
12
13
14
15
16
17
18
LINP
LINN
RINP
RINN
RAUX
ROUTP
ROUTN
LOUTN
LOUTP
A_IN
A_IN
A_IN
A_IN
A_IN
A_OUT
A_OUT
A_OUT
A_OUT
Description
Supply for Digital Input and Output Pins. The digital output pins are supplied from IOVDD,
which also sets the highest input voltage that should be seen on the digital input pins.
IOVDD should be set to 3.3 V. The current draw of this pin is variable because it is dependent
on the loads of the digital outputs. IOVDD should be decoupled to DGND with a 100 nF
capacitor and a 10 μF capacitor.
External Master Clock Input.
I2C Address Bit 0 (ADDR0).
SPI Latch Signal (CLATCH). Must go low at the beginning of an SPI transaction and high at the
end of a transaction. Each SPI transaction can take a different number of CCLKs to complete,
depending on the address and read/write bit that are sent at the beginning of the SPI
transaction.
Detect Insertion/Removal of Headphone Plug (JACKDET).
Digital Microphone Stereo Input (MICIN).
Bias Voltage for Electret Microphone.
Left Channel Single-Ended Auxiliary Input. Biased at AVDD/2.
AVDD/2 V Common-Mode Reference. A 10 μF to 47 μF standard decoupling capacitor should
be connected between this pin and AGND to reduce crosstalk between the ADCs and DACs.
This pin can be used to bias external analog circuits, as long as they are not drawing current
from CM (for example, the noninverting input of an op amp).
3.3 V Analog Supply for DAC and Microphone Bias. This pin should be decoupled locally to
AGND with a 100 nF capacitor.
Analog Ground. The AGND and DGND pins can be tied together on a common ground plane.
AGND should be decoupled locally to AVDD with a 100 nF capacitor.
Left Channel Noninverting Input or Single-Ended Input 0. Biased at AVDD/2.
Left Channel Inverting Input or Single-Ended Input 1. Biased at AVDD/2.
Right Channel Noninverting Input or Single-Ended Input 2. Biased at AVDD/2.
Right Channel Inverting Input or Single-Ended Input 3. Biased at AVDD/2.
Right Channel Single-Ended Auxiliary Input. Biased at AVDD/2.
Right Line Output, Positive. Biased at AVDD/2.
Right Line Output, Negative. Biased at AVDD/2.
Left Line Output, Negative. Biased at AVDD/2.
Left Line Output, Positive. Biased at AVDD/2.
Rev. 0 | Page 13 of 88
ADAU1461
Pin No.
19
20
21
Mnemonic
RHP
LHP
MONOOUT
Type 1
A_OUT
A_OUT
A_OUT
22
AGND
PWR
23
AVDD
PWR
24
DVDDOUT
PWR
25
DGND
PWR
26
ADC_SDATA/GPIO1
D_IO
27
DAC_SDATA/GPIO0
D_IO
28
BCLK/GPIO2
D_IO
29
LRCLK/GPIO3
D_IO
30
ADDR1/CDATA
D_IN
31
SDA/COUT
D_IO
32
SCL/CCLK
D_IN
EP
Exposed Pad
1
Description
Right Headphone Output. Biased at AVDD/2.
Left Headphone Output. Biased at AVDD/2.
Mono Output or Virtual Ground for Capless Headphone. Biased at AVDD/2 when set as mono
output.
Analog Ground. The AGND and DGND pins can be tied together on a common ground plane.
AGND should be decoupled locally to AVDD with a 100 nF capacitor.
3.3 V Analog Supply for ADC, Output Driver, and Input to Digital Supply Regulator. This pin
should be decoupled locally to AGND with a 100 nF capacitor.
Digital Core Supply Decoupling Point. The digital supply is generated from an on-board
regulator and does not require an external supply. DVDDOUT should be decoupled to DGND
with a 100 nF capacitor and a 10 μF capacitor.
Digital Ground. The AGND and DGND pins can be tied together on a common ground plane.
DGND should be decoupled to DVDDOUT and to IOVDD with 100 nF capacitors and 10 μF
capacitors.
ADC Serial Output Data (ADC_SDATA).
General-Purpose Input/Output 1 (GPIO1).
DAC Serial Input Data (DAC_SDATA).
General-Purpose Input/Output 0 (GPIO0).
Serial Data Port Bit Clock (BCLK).
General-Purpose Input/Output 2 (GPIO2).
Serial Data Port Frame Clock (LRCLK).
General-Purpose Input/Output 3 (GPIO3).
I2C Address Bit 1 (ADDR1).
SPI Data Input (CDATA).
I2C Data (SDA). This pin is a bidirectional open-collector input/output. The line connected to
this pin should have a 2 kΩ pull-up resistor.
SPI Data Output (COUT). This pin is used for reading back registers and memory locations. It is
three-state when an SPI read is not active.
I2C Clock (SCL). This pin is always an open-collector input when in I2C control mode. The line
connected to this pin should have a 2 kΩ pull-up resistor.
SPI Clock (CCLK). This pin can run continuously or be gated off between SPI transactions.
Exposed Pad. The exposed pad is connected internally to the ADAU1461 grounds. For
increased reliability of the solder joints and maximum thermal capability, it is recommended
that the pad be soldered to the ground plane. See the Exposed Pad PCB Design section for
more information.
A_IN = analog input, A_OUT = analog output, D_IN = digital input, D_IO = digital input/output, PWR = power.
Rev. 0 | Page 14 of 88
ADAU1461
28
–30
26
–35
24
–40
22
–45
20
–50
–55
16
14
12
10
–65
–70
–75
–80
8
–85
6
–90
4
–95
2
–100
0
–60
–50
–40
–30
–20
–10
0
DIGITAL 1kHz INPUT SIGNAL (dBFS)
–105
–60
–50
–40
–30
–20
–10
0
DIGITAL 1kHz INPUT SIGNAL (dBFS)
Figure 8. Headphone Amplifier Power vs. Input Level, 16 Ω Load
Figure 11. Headphone Amplifier THD + N vs. Input Level, 16 Ω Load
18
0
16
–10
–20
14
–30
12
THD + N (dBV)
STEREO OUTPUT POWER (mW)
–60
08914-056
THD + N (dBV)
18
08914-055
STEREO OUTPUT POWER (mW)
TYPICAL PERFORMANCE CHARACTERISTICS
10
8
6
–40
–50
–60
–70
–80
4
–90
2
–40
–30
–20
–10
0
DIGITAL 1kHz INPUT SIGNAL (dBFS)
–60
–50
–40
–30
–20
–10
08914-058
–50
08914-057
–100
0
–60
0
DIGITAL 1kHz INPUT SIGNAL (dBFS)
Figure 9. Headphone Amplifier Power vs. Input Level, 32 Ω Load
Figure 12. Headphone Amplifier THD + N vs. Input Level, 32 Ω Load
0
0.04
−10
0.02
−30
MAGNITUDE (dBFS)
MAGNITUDE (dBFS)
−20
−40
−50
−60
−70
0
–0.02
–0.04
−80
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
FREQUENCY (NORMALIZED TO fS)
0.9
1.0
0
08914-008
−100
Figure 10. ADC Decimation Filter, 64× Oversampling, Normalized to fS
0.05
0.10
0.15
0.20
0.25
0.30
FREQUENCY (NORMALIZED TO fS)
0.35
0.40
08914-009
–0.06
−90
Figure 13. ADC Decimation Filter Pass-Band Ripple, 64× Oversampling,
Normalized to fS
Rev. 0 | Page 15 of 88
0.10
–10
0.08
–20
0.06
–30
0.04
–40
–50
–60
–70
0.02
0
–0.02
–0.04
–80
–0.06
–90
–0.08
–100
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1.0
FREQUENCY (NORMALIZED TO fS)
–0.10
0
Figure 14. ADC Decimation Filter, 128× Oversampling, Normalized to fS
0.05
0.10
0.15
0.20
0.25
0.30
0.35
0.40
0.45
0.50
FREQUENCY (NORMALIZED TO fS)
08914-011
MAGNITUDE (dBFS)
0
08914-010
MAGNITUDE (dBFS)
ADAU1461
Figure 17. ADC Decimation Filter Pass-Band Ripple, 128× Oversampling,
Normalized to fS
0
0.04
–10
0.02
–30
MAGNITUDE (dBFS)
MAGNITUDE (dBFS)
–20
–40
–50
–60
–70
0
−0.02
−0.04
–80
–90
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1.0
FREQUENCY (NORMALIZED TO fS)
0
Figure 15. ADC Decimation Filter, 128× Oversampling, Double-Rate Mode,
Normalized to fS
0.05
0.10
0.15
0.20
0.25
0.30
0.35
0.40
FREQUENCY (NORMALIZED TO fS)
08914-013
0
08914-012
–100
−0.06
Figure 18. ADC Decimation Filter Pass-Band Ripple, 128× Oversampling,
Double-Rate Mode, Normalized to fS
0
0.20
−10
0.15
0.10
−30
MAGNITUDE (dBFS)
−40
−50
−60
−70
0
–0.05
–0.10
−80
–0.15
−90
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
FREQUENCY (NORMALIZED TO fS)
0.9
1.0
–0.20
08914-014
−100
0.05
Figure 16. DAC Interpolation Filter, 64× Oversampling, Double-Rate Mode,
Normalized to fS
0
0.05
0.10
0.15
0.20
0.25
0.30
FREQUENCY (NORMALIZED TO fS)
0.35
0.40
08914-015
MAGNITUDE (dBFS)
−20
Figure 19. DAC Interpolation Filter Pass-Band Ripple, 64× Oversampling,
Double-Rate Mode, Normalized to fS
Rev. 0 | Page 16 of 88
0.05
–10
0.04
–20
0.03
–30
0.02
–40
–50
–60
–70
0.01
0
–0.01
–0.02
–80
–0.03
–90
–0.04
–100
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1.0
FREQUENCY (NORMALIZED TO fS)
–0.05
Figure 20. DAC Interpolation Filter, 128× Oversampling, Normalized to fS
0
0.05
0.10
0.15
0.20
0.25
0.30
0.35
0.40
0.45
0.50
08914-017
MAGNITUDE (dBFS)
0
08914-016
MAGNITUDE (dBFS)
ADAU1461
FREQUENCY (NORMALIZED TO fS)
Figure 23. DAC Interpolation Filter Pass-Band Ripple, 128× Oversampling,
Normalized to fS
0
0.20
−10
0.15
0.10
−30
MAGNITUDE (dBFS)
−40
−50
−60
−70
0
–0.05
–0.10
−80
–0.15
−90
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1.0
FREQUENCY (NORMALIZED TO fS)
–0.20
08914-018
−100
0.05
Figure 21. DAC Interpolation Filter, 128× Oversampling, Double-Rate Mode,
Normalized to fS
0
0.05
0.10
0.15
0.20
0.25
0.30
0.35
0.40
FREQUENCY (NORMALIZED TO fS)
Figure 24. DAC Interpolation Filter Pass-Band Ripple, 128× Oversampling,
Double-Rate Mode, Normalized to fS
12
90
11
80
10
70
IMPEDANCE (kΩ)
7
6
5
4
3
60
50
40
30
20
2
10
0
100
200
300
400
500
600
700
800
900 1000 1100
INSTRUCTIONS
0
GAIN (dB)
Figure 22. Typical DSP Current Draw
Figure 25. Input Impedance vs. Gain for Analog Inputs
Rev. 0 | Page 17 of 88
08914-125
0
35.00
32.75
30.50
28.25
26.00
23.75
21.50
19.25
17.00
14.75
12.50
10.25
8.00
5.75
3.50
1.25
–1.00
–3.25
–5.50
–7.75
–10.00
–12.25
1
08914-065
CURRENT (mA)
9
8
08914-019
MAGNITUDE (dBFS)
−20
ADAU1461
SYSTEM BLOCK DIAGRAMS
10µF
+
0.1µF
10µF
10µF
+
+
0.1µF
0.1µF
0.1µF
1.2nH
THE INPUT CAPACITOR VALUE DEPENDS ON THE
INPUT IMPEDANCE, WHICH VARIES WITH THE
VOLUME SETTING.
DVDDOUT
IOVDD
AVDD
9.1pF
AVDD
10µF
LOUTP
LINP
LEFT
MICROPHONE
EARPIECE
SPEAKER
LOUTN
LINN
10µF
RHP
CAPLESS
HEADPHONE
OUTPUT
MONOOUT
2kΩ
LHP
ROUTP
ADAU1461
MICBIAS
EARPIECE
SPEAKER
ROUTN
2kΩ
10µF
RINN
RIGHT
MICROPHONE
RINP
10µF
ADC_SDATA/GPIO1
JACK
DETECTION
SIGNAL
JACKDET/MICIN
DAC_SDATA/GPIO0
SERIAL DATA
LRCLK/GPIO3
AUX LEFT
1kΩ
BCLK/GPIO2
10µF
LAUX
ADDR1/CDATA
10µF
RAUX
SDA/COUT
SYSTEM
CONTROLLER
SCL/CCLK
1kΩ
AGND
CM
0.1µF
10µF
+
08914-045
CLOCK
SOURCE
ADDR0/CLATCH
MCLK
AGND
49.9Ω
DGND
AUX RIGHT
Figure 26. System Block Diagram
Rev. 0 | Page 18 of 88
ADAU1461
10µF
+
0.1µF
10µF
10µF
+
+
0.1µF
0.1µF
0.1µF
1.2nH
THE INPUT CAPACITOR VALUE DEPENDS ON THE
INPUT IMPEDANCE, WHICH VARIES WITH THE
VOLUME SETTING.
DVDDOUT
IOVDD
AVDD
9.1pF
AVDD
MICBIAS
LOUTP
VDD
10µF
SINGLE-ENDED
ANALOG
OUTPUT
MICROPHONE
EARPIECE
SPEAKER
LOUTN
LINN
RHP
CM
CAPLESS
HEADPHONE
OUTPUT
MONOOUT
LINP
GND
LHP
ADAU1461
ROUTP
EARPIECE
SPEAKER
ROUTN
VDD
10µF
SINGLE-ENDED
ANALOG
OUTPUT
MICROPHONE
RINN
CM
RINP
GND
ADC_SDATA/GPIO1
JACK
DETECTION
SIGNAL
JACKDET/MICIN
DAC_SDATA/GPIO0
SERIAL DATA
LRCLK/GPIO3
AUX LEFT
1kΩ
BCLK/GPIO2
10µF
LAUX
ADDR1/CDATA
10µF
RAUX
SDA/COUT
SYSTEM
CONTROLLER
SCL/CCLK
1kΩ
AGND
CM
0.1µF
10µF
+
08914-059
CLOCK
SOURCE
ADDR0/CLATCH
MCLK
AGND
49.9Ω
DGND
AUX RIGHT
Figure 27. System Block Diagram with Analog Microphones
Rev. 0 | Page 19 of 88
ADAU1461
10µF
+
0.1µF
10µF
10µF
+
+
0.1µF
0.1µF
0.1µF
1.2nH
DVDDOUT
IOVDD
AVDD
MICBIAS
RHP
CM
LHP
LINP
10µF
LINN
DIGITAL DATA
MICROPHONE
L/R SELECT
2.5V TO 5.0V
MONOOUT
CLK
0.1µF
CAPLESS
HEADPHONE
OUTPUT
AVDD
BCLK
VDD
9.1pF
RINN
GND
RINP
22nF
LOUTP
ADAU1461
22nF
LOUTN
BCLK
CLK
22nF
REXT
L/R SELECT
VDD
VDD
INL+
INL–
SSM2306
OUTL+
OUTL–
CLASS-D 2W
STEREO SPEAKER
DRIVER
INR+
OUTR+
OUTR–
INR–
SD
GND
LEFT
SPEAKER
RIGHT
SPEAKER
GND
GND
JACKDET/MICIN
AUX LEFT
1kΩ
DAC_SDATA/GPIO0
10µF
LAUX
SERIAL DATA
LRCLK/GPIO3
SHUTDOWN
ADC_SDATA/GPIO1
BCLK/GPIO2
10µF
RAUX
ADDR1/CDATA
SDA/COUT
SYSTEM
CONTROLLER
1kΩ
SCL/CCLK
CLOCK
SOURCE
MCLK
ADDR0/CLATCH
AGND
49.9Ω
AGND
CM
0.1µF
10µF
+
08914-060
AUX RIGHT
DGND
0.1µF
REXT
REXT
ROUTN
DIGITAL DATA
MICROPHONE
REXT
22nF
ROUTP
VDD
0.1µF
Figure 28. System Block Diagram with Digital Microphones and SSM2306 Class-D Speaker Driver
Rev. 0 | Page 20 of 88
ADAU1461
THEORY OF OPERATION
The ADAU1461 is a low power audio codec with an integrated
stream-oriented DSP core, making it an all-in-one package that
offers high quality audio, low power, small size, and many
advanced features. The stereo ADC and stereo DAC each have
an SNR of at least +98 dB and a THD + N of at least −90 dB.
The serial data port is compatible with I2S, left-justified, rightjustified, and TDM modes for interfacing to digital audio data.
The operating voltage is 3.3 V, with an on-board regulator
generating the internal digital supply voltage.
The record signal path includes very flexible input configurations
that can accept differential and single-ended analog microphone
inputs as well as a digital microphone input. A microphone bias
pin provides seamless interfacing to electret microphones. Input
configurations can accept up to six single-ended analog signals
or variations of stereo differential or stereo single-ended signals
with two additional auxiliary single-ended inputs. Each input
signal has its own programmable gain amplifier (PGA) for volume
adjustment and can be routed directly to the playback path output
mixers, bypassing the ADCs. An automatic level control (ALC)
can also be implemented to keep the recording volume constant.
The ADCs and DACs are high quality, 24-bit Σ-Δ converters
that operate at selectable 64× or 128× oversampling ratios. The
base sampling rate of the converters is set by the input clock rate
and can be further scaled with the converter control register
settings. The converters can operate at sampling frequencies
from 8 kHz to 96 kHz. The ADCs and DACs also include very
fine-step digital volume controls.
The playback path allows input signals and DAC outputs to be
mixed into various output configurations. Headphone drivers
are available for a stereo headphone output, and the other output
pins are capable of differentially driving an earpiece speaker.
Capless headphone outputs are possible with the use of the
mono output as a virtual ground connection. The stereo line
outputs can be used as either single-ended or differential
outputs and as an optional mix-down mono output.
The DSP core introduces many features that make this codec
unique and optimized for audio processing. The program and
parameter RAMs can be loaded with custom audio processing
signal flow built using the SigmaStudio graphical programming
software from Analog Devices, Inc. The values stored in the
parameter RAM control individual signal processing blocks,
such as equalization filters, dynamics processors, audio delays,
and mixer levels.
The SigmaStudio software is used to program and control the
SigmaDSP through the control port. Along with designing and
tuning a signal flow, the tools can be used to configure all of the
DSP registers. The SigmaStudio graphical interface allows anyone with digital or analog audio processing knowledge to easily
design DSP signal flow and port it to a target application. At the
same time, it provides enough flexibility and programmability
for an experienced DSP programmer to have in-depth control
of the design. In SigmaStudio, the user can connect graphical
blocks (such as biquad filters, dynamics processors, mixers, and
delays), compile the design, and load the program and parameter
files into the ADAU1461 memory through the control port.
Signal processing blocks available in the provided libraries
include the following:
•
•
•
•
•
•
•
•
•
•
•
•
•
Enhanced stereo capture
Single- and double-precision biquad filters
FIR filters
Dynamics processors with peak or rms detection for mono
and multichannel dynamics
Mixers and splitters
Tone and noise generators
Fixed and variable gain
Loudness
Delay
Stereo enhancement
Dynamic bass boost
Noise and tone sources
Level detectors
Additional processing blocks are always being developed.
Analog Devices also provides proprietary and third-party
algorithms for applications such as matrix decoding, bass
enhancement, and surround virtualizers. Contact Analog
Devices (www.analog.com) for information about licensing
these algorithms.
The ADAU1461 can generate its internal clocks from a wide
range of input clocks by using the on-board fractional PLL.
The PLL accepts inputs from 8 MHz to 27 MHz.
The ADAU1461 is provided in a small, 32-lead, 5 mm × 5 mm
LFCSP with an exposed bottom pad.
Rev. 0 | Page 21 of 88
ADAU1461
STARTUP, INITIALIZATION, AND POWER
This section describes the procedure for properly starting up
the ADAU1461. The following sequence provides a high level
approach to the proper initiation of the system.
1.
2.
3.
4.
POWER REDUCTION MODES
Sections of the ADAU1461 chip can be turned on and off as
needed to reduce power consumption. These include the ADCs,
the DACs, the PLL, and the DSP core.
Apply power to the ADAU1461.
Lock the PLL to the input clock (if using the PLL).
Enable the core clock.
Load the register settings.
The digital filters of the ADCs and DACs can each be set to oversampling ratios of 64× or 128× (default). Setting the oversampling
ratios to 64× for these filters lowers power consumption with a
minimal impact on performance. See the Digital Filters section
for specifications; see the Typical Performance Characteristics
section for graphs of these filters.
See the Startup section for more information about the proper
start-up sequence.
POWER-UP SEQUENCE
DIGITAL POWER SUPPLY
The ADAU1461 uses a power-on reset (POR) circuit to
reset the registers upon power-up. The POR monitors the
DVDDOUT pin and generates a reset signal whenever power
is applied to the chip. During the reset, the ADAU1461 is set
to the default values documented in the register map (see the
Control Registers section). Typically, with a 10 μF capacitor on
AVDD, the POR takes approximately 14 ms.
The digital power supply for the ADAU1461 is generated from
an internal regulator. This regulator generates a 1.5 V supply
internally. The only external connection to this regulator is the
DVDDOUT bypassing point. A 100 nF capacitor and a 10 μF
capacitor should be connected between this pin and DGND.
INPUT/OUTPUT POWER SUPPLY
1.5V
The power for the digital output pins is supplied from IOVDD,
and this pin also sets the highest input voltage that should be
seen on the digital input pins. IOVDD should be set to 3.3 V; no
digital input signal should be at a voltage level higher than the
one on IOVDD. The current draw of this pin is variable because
it depends on the loads of the digital outputs. IOVDD should be
decoupled to DGND with a 100 nF capacitor and a 10 μF
capacitor.
1.35V
DVDDOUT
0.95V
AVDD
PART READY
POR
POR
ACTIVE
POR
FINISHED
08914-061
CLOCK GENERATION AND MANAGEMENT
POR ACTIVE
Figure 29. Power-On Reset Sequence
The PLL lock time is dependent on the MCLK rate. Typical
lock times are provided in Table 10. The DSP can be enabled
immediately after the PLL is locked.
Table 10. PLL Lock Times
PLL Mode
Fractional
Fractional
Integer
Fractional
Fractional
Fractional
Fractional
Fractional
Fractional
Integer
Fractional
Fractional
MCLK Frequency
8 MHz
12 MHz
12.288 MHz
13 MHz
14.4 MHz
19.2 MHz
19.68 MHz
19.8 MHz
24 MHz
24.576 MHz
26 MHz
27 MHz
Lock Time (Typical)
3.5 ms
3.0 ms
2.96 ms
2.4 ms
2.4 ms
2.98 ms
2.98 ms
2.98 ms
2.95 ms
2.96 ms
2.4 ms
2.4 ms
The ADAU1461 uses a flexible clocking scheme that enables the
use of many different input clock rates. The PLL can be bypassed
or used, resulting in two different approaches to clock management. For more information about clocking schemes, PLL
configuration, and sampling rates, see the Clocking and
Sampling Rates section.
Case 1: PLL Is Bypassed
If the PLL is bypassed, the core clock is derived directly from
the MCLK input. The rate of this clock must be set properly in
Register R0 (clock control register, Address 0x4000) using the
INFREQ[1:0] bits. When the PLL is bypassed, supported external
clock rates are 256 × fS, 512 × fS, 768 × fS, and 1024 × fS, where fS
is the base sampling rate. The core clock of the chip is off until
the core clock enable bit (COREN) is asserted. If a clock slower
than 1024 × fS is directly input to the ADAU1461 (bypassing the
PLL), the number of available SigmaDSP processing cycles is
reduced, and the DSPSR bits in Register R57 (Address 0x40EB)
should be adjusted accordingly.
Rev. 0 | Page 22 of 88
ADAU1461
Case 2: PLL Is Used
The core clock to the entire chip is off during the PLL lock
acquisition period. The user can poll the lock bit to determine
when the PLL has locked. After lock is acquired, the ADAU1461
can be started by asserting the core clock enable bit (COREN)
in Register R0 (clock control register, Address 0x4000). This bit
enables the core clock to all the internal blocks of the ADAU1461.
To program the PLL during initialization or reconfiguration of
the clock setting, the following procedure must be followed:
1.
2.
3.
4.
5.
PLL Lock Acquisition
During the lock acquisition period, only Register R0 (Address
0x4000) and Register R1 (Address 0x4002) are accessible
through the control port. Because all other registers require a
valid master clock for reading and writing, do not attempt to
access any other register. Any read or write is prohibited until
the core clock enable bit (COREN) and the lock bit are both
asserted.
Power down the PLL.
Reset the PLL control register.
Start the PLL.
Poll the lock bit.
Assert the core clock enable bit after the PLL lock
is acquired.
The PLL control register (Register R1, Address 0x4002) is a
48-bit register where all bits must be written with a single
continuous write to the control port.
Rev. 0 | Page 23 of 88
ADAU1461
CLOCKING AND SAMPLING RATES
R57: DSP SAMPLING
RATE SETTING
DSPSR[3:0]
fS/0.5, 1, 1.5, 2, 3, 4, 6
R1: PLL CONTROL REGISTER
MCLK
÷X
R0: CLOCK
CONTROL REGISTER
× (R + N/M)
CLKSRC
INFREQ[1:0]
256 × fS, 512 × fS,
768 × fS, 1024 × fS
ADCs
R17: CONVERTER
SAMPLING RATE
CORE
CLOCK
DACs
CONVSR[2:0]
fS/0.5, 1, 1.5, 2, 3, 4, 6
R64: SERIAL PORT
SAMPLING RATE
ADC_SDATA/GPIO1
BCLK/GPIO2
LRCLK/GPIO3
DAC_SDATA/GPIO0
SERIAL
DATA INPUT/
OUTPUT PORT
08914-020
SPSR[2:0]
fS/0.5, 1, 1.5, 2, 3, 4, 6
Figure 30. Clock Tree Diagram
CORE CLOCK
Clocks for the converters, the serial ports, and the DSP are
derived from the core clock. The core clock can be derived
directly from MCLK or it can be generated by the PLL. The
CLKSRC bit (Bit 3 in Register R0, Address 0x4000) determines
the clock source.
To utilize the maximum amount of DSP instructions, the core
clock should run at a rate of 1024 × fS.
Table 11. Clock Control Register (Register R0, Address 0x4000)
Bits
3
Bit Name
CLKSRC
The INFREQ[1:0] bits should be set according to the expected
input clock rate selected by CLKSRC; this value also determines
the core clock rate and the base sampling frequency, fS.
[2:1]
INFREQ[1:0]
For example, if the input to CLKSRC = 49.152 MHz (from
PLL), then
0
COREN
INFREQ[1:0] = 1024 × fS
fS = 49.152 MHz/1024 = 48 kHz
The PLL output clock rate is always 1024 × fS, and the clock
control register automatically sets the INFREQ[1:0] bits to
1024 × fS when using the PLL. When using a direct clock, the
INFREQ[1:0] frequency should be set according to the MCLK
pin clock rate and the desired base sampling frequency.
Rev. 0 | Page 24 of 88
Settings
0: Direct from MCLK pin (default)
1: PLL clock
00: 256 × fS (default)
01: 512 × fS
10: 768 × fS
11: 1024 × fS
0: Core clock disabled (default)
1: Core clock enabled
ADAU1461
SAMPLING RATES
PLL
The ADCs, DACs, and serial port share a common sampling
rate that is set in Register R17 (Converter Control 0 register,
Address 0x4017). The CONVSR[2:0] bits set the sampling rate
as a ratio of the base sampling frequency. The DSP sampling
rate is set in Register R57 (DSP sampling rate setting register,
Address 0x40EB) using the DSPSR[3:0] bits, and the serial port
sampling rate is set in Register R64 (serial port sampling rate
register, Address 0x40F8) using the SPSR[2:0] bits.
The PLL uses the MCLK as a reference to generate the core
clock. PLL settings are set in Register R1 (PLL control register,
Address 0x4002). Depending on the MCLK frequency, the PLL
must be set for either integer or fractional mode. The PLL can
accept input frequencies in the range of 8 MHz to 27 MHz.
Table 12. 48 kHz Base Sampling Rate Divisions
Base Sampling
Frequency
fS = 48 kHz
Sampling Rate Scaling
fS/1
fS/6
fS/4
fS/3
fS/2
fS/1.5
fS/0.5
÷X
MCLK
× (R + N/M)
TO PLL
CLOCK DIVIDER
08914-021
It is recommended that the sampling rates for the converters,
serial ports, and DSP be set to the same value, unless appropriate
compensation filtering is done within the DSP. Table 12 and
Table 13 list the sampling rate divisions for common base
sampling rates.
All six bytes in the PLL control register must be written with a
single continuous write to the control port.
Figure 31. PLL Block Diagram
Integer Mode
Integer mode is used when the MCLK is an integer (R) multiple
of the PLL output (1024 × fS).
For example, if MCLK = 12.288 MHz and fS = 48 kHz, then
Sampling Rate
48 kHz
8 kHz
12 kHz
16 kHz
24 kHz
32 kHz
96 kHz
PLL required output = 1024 × 48 kHz = 49.152 MHz
R = 49.152 MHz/12.288 MHz = 4
In integer mode, the values set for N and M are ignored.
Fractional Mode
Fractional mode is used when the MCLK is a fractional
(R + (N/M)) multiple of the PLL output.
For example, if MCLK = 12 MHz and fS = 48 kHz, then
Table 13. 44.1 kHz Base Sampling Rate Divisions
PLL required output = 1024 × 48 kHz = 49.152 MHz
Base Sampling
Frequency
fS = 44.1 kHz
R + (N/M) = 49.152 MHz/12 MHz = 4 + (12/125)
Sampling Rate Scaling
fS/1
fS/6
fS/4
fS/3
fS/2
fS/1.5
fS/0.5
Sampling Rate
44.1 kHz
7.35 kHz
11.025 kHz
14.7 kHz
22.05 kHz
29.4 kHz
88.2 kHz
Common fractional PLL parameter settings for 44.1 kHz and
48 kHz sampling rates can be found in Table 15 and Table 16.
The PLL outputs a clock in the range of 41 MHz to 54 MHz,
which should be taken into account when calculating PLL
values and MCLK frequencies.
Table 14. PLL Control Register (Register R1, Address 0x4002)
Bits
[47:32]
Bit Name
M[15:0]
[31:16]
N[15:0]
[14:11]
R[3:0]
Description
Denominator of the fractional PLL: 16-bit binary number
0x00FD: M = 253 (default)
Numerator of the fractional PLL: 16-bit binary number
0x000C: N = 12 (default)
Integer part of PLL: four bits, only values 2 to 8 are valid
0010: R = 2 (default)
0011: R = 3
0100: R = 4
0101: R = 5
0110: R = 6
0111: R = 7
1000: R = 8
Rev. 0 | Page 25 of 88
ADAU1461
Bits
[10:9]
Bit Name
X[1:0]
8
Type
1
Lock
0
PLLEN
Description
PLL input clock divider
00: X = 1 (default)
01: X = 2
10: X = 3
11: X = 4
PLL operation mode
0: Integer (default)
1: Fractional
PLL lock (read-only bit)
0: PLL unlocked (default)
1: PLL locked
PLL enable
0: PLL disabled (default)
1: PLL enabled
Table 15. Fractional PLL Parameter Settings for fS = 44.1 kHz (PLL Output = 45.1584 MHz = 1024 × fS)
MCLK Input (MHz)
8
12
13
14.4
19.2
19.68
19.8
24
26
27
Input Divider (X)
1
1
1
2
2
2
2
2
2
2
Integer (R)
5
3
3
6
4
4
4
3
3
3
Denominator (M)
625
625
8125
125
125
1025
1375
625
8125
1875
Numerator (N)
403
477
3849
34
88
604
772
477
3849
647
R2: PLL Control Setting (Hex)
0x0271 0193 2901
0x0271 01DD 1901
0x1FBD 0F09 1901
0x007D 0022 3301
0x007D 0058 2301
0x0401 025C 2301
0x055F 0304 2301
0x0271 01DD 1B01
0x1FBD 0F09 1B01
0x0753 0287 1B01
Table 16. Fractional PLL Parameter Settings for fS = 48 kHz (PLL Output = 49.152 MHz = 1024 × fS)
MCLK Input (MHz)
8
12
13
14.4
19.2
19.68
19.8
24
26
27
Input Divider (X)
1
1
1
2
2
2
2
2
2
2
Integer (R)
6
4
3
6
5
4
4
4
3
3
Denominator (M)
125
125
1625
75
25
205
825
125
1625
1125
Numerator (N)
18
12
1269
62
3
204
796
12
1269
721
R2: PLL Control Setting (Hex)
0x007D 0012 3101
0x007D 000C 2101
0x0659 04F5 1901
0x004B 003E 3301
0x0019 0003 2B01
0x00CD 00CC 2301
0x0339 031C 2301
0x007D 000C 2301
0x0659 04F5 1B01
0x0465 02D1 1B01
Table 17. Integer PLL Parameter Settings for fS = 48 kHz (PLL Output = 49.152 MHz = 1024 × fS)
MCLK Input (MHz)
12.288
24.576
1
Input Divider (X)
1
1
Integer (R)
4
2
Denominator (M)
Don’t care
Don’t care
X = don’t care.
Rev. 0 | Page 26 of 88
Numerator (N)
Don’t care
Don’t care
R2: PLL Control Setting (Hex) 1
0xXXXX XXXX 2001
0xXXXX XXXX 1001
ADAU1461
RECORD SIGNAL PATH
MICIN LEFT
DIGITAL
MICROPHONE
INTERFACE
JACKDET/MICIN
MICIN RIGHT
LINNG[2:0]
MIXER 1
(LEFT RECORD
MIXER)
–12dB TO +6dB
PGA
LDBOOST[1:0]
LINN
–12dB TO
+35.25dB
LINP
LEFT
ADC
MUTE/0dB/20dB
LINPG[2:0]
–12dB TO +6dB
MIXER 1
OUTPUT
(TO PLAYBACK
MIXER)
ALCSEL[2:0]
LDVOL[5:0]
INSEL
ALC
CONTROL
DECIMATOR/
ALC/
DIGITAL
VOLUME
MX1AUXG[2:0]
LAUX
–12dB TO +6dB
AUXILIARY
BYPASS
MX2AUXG[2:0]
RAUX
–12dB TO +6dB
MIXER 2
OUTPUT
(TO PLAYBACK
MIXER)
RINPG[2:0]
–12dB TO +6dB
PGA
RDBOOST[1:0]
RINP
–12dB TO
+35.25dB
RINN
RIGHT
ADC
MUTE/0dB/20dB
MIXER 2
(RIGHT RECORD
MIXER)
RINNG[2:0]
INSEL
–12dB TO +6dB
RDVOL[5:0]
08914-022
ALCSEL[2:0]
ALC
CONTROL
Figure 32. Record Signal Path
INPUT SIGNAL PATHS
The ADAU1461 can accept both line level and microphone
inputs. The analog inputs can be configured in a single-ended
or differential configuration. There is also an input for a digital
microphone. The analog inputs are biased at AVDD/2. Unused
input pins should be connected to CM.
Each of the six analog inputs has individual gain controls (boost
or cut). The input signals are mixed and routed to an ADC. The
mixed input signals can also bypass the ADCs and be routed
directly to the playback mixers. Left channel inputs are mixed
before the left ADC; however, it is possible to route the mixed
analog signal around the ADC and output it into a left or right
output channel. The same capabilities apply to the right channel
and the right ADC.
Signals are inverted through the PGAs and the mixers. The
result of this inversion is that differential signals input through
the PGA are output from the ADCs at the same polarity as they
are input. Single-ended inputs that pass through the mixer but
not through the PGA are inverted. The ADCs are noninverting.
The input impedance of the analog inputs varies with the gain
of the PGA. This impedance ranges from 1.7 kΩ at the 35.25 dB
gain setting to 80.4 kΩ at the −12 dB setting. This range is shown
in Figure 25.
Rev. 0 | Page 27 of 88
ADAU1461
Analog Microphone Inputs
Analog Line Inputs
For microphone inputs, configure the part in either stereo
pseudo-differential mode or stereo full differential mode.
Line input signals can be accepted by any analog input. It is
possible to route signals on the RINN, RINP, LINN, and LINP
pins around the differential amplifier to their own amplifier and
to use these pins as single-ended line inputs by disabling the
LDEN and RDEN bits (Bit 0 in Register R8, Address 0x400E,
and Bit 0 in Register R9, Address 0x400F). Figure 35 depicts a
stereo single-ended line input using the RINN and LINN pins.
The LINN and LINP pins are the inverting and noninverting
inputs for the left channel, respectively. The RINN and RINP
pins are the inverting and noninverting inputs for the right
channel, respectively.
For a differential microphone input, connect the positive signal
to the noninverting input of the PGA and the negative signal to
the inverting input of the PGA, as shown in Figure 33. The PGA
settings are controlled with Register R8 (left differential input
volume control register, Address 0x400E) and Register R9 (right
differential input volume control register, Address 0x400F). The
PGA must first be enabled by setting the RDEN and LDEN bits.
The LAUX and RAUX pins are single-ended line inputs. They
can be used together as a stereo single-ended auxiliary input, as
shown in Figure 35. These inputs can bypass the input gain
control, mixers, and ADCs to directly connect to the output
playback mixers (see auxiliary bypass in Figure 32).
ADAU1461
LINNG[2:0]
ADAU1461
LINP
LEFT
PGA
LDBOOST[1:0]
LEFT LINE
INPUT
LINN
LEFT AUX
INPUT
LAUX
–12dB TO +6dB
LEFT
MICROPHONE
LINN
MUTE/
0dB/20dB
–12dB TO
+35.25dB
2kΩ
RIGHT AUX
INPUT
MICBIAS
RAUX
AUXILIARY
BYPASS
RINNG[2:0]
RINN
RDBOOST[1:0]
–12dB TO +6dB
RIGHT
MICROPHONE
RINP
MUTE/
0dB/20dB
Figure 35. Stereo Single-Ended Line Input with Stereo Auxiliary Bypass
08914-052
–12dB TO
+35.25dB
Figure 33. Stereo Differential Microphone Configuration
The PGA can also be used for single-ended microphone inputs.
Connect LINP and/or RINP to the CM pin. In this configuration, the signal connects to the inverting input of the PGA,
LINN and/or RINN, as shown in Figure 34.
ADAU1461
LINN
LEFT
MICROPHONE
2kΩ
LEFT
PGA
LDBOOST[1:0]
LINP
CM
–12dB TO
+35.25dB
MUTE/
0dB/20dB
MICBIAS
RIGHT
PGA
2kΩ
RINP
RDBOOST[1:0]
RINN
–12dB TO
+35.25dB
MUTE/
0dB/20dB
08914-053
RIGHT
MICROPHONE
RINN
08914-054
RIGHT
PGA
2kΩ
RIGHT LINE
INPUT
Figure 34. Stereo Single-Ended Microphone Configuration
Rev. 0 | Page 28 of 88
ADAU1461
Digital Microphone Input
Microphone Bias
When using a digital microphone connected to the JACKDET/
MICIN pin, the JDFUNC[1:0] bits in Register R2 (Address 0x4008)
must be set to 10 to enable the microphone input and disable the
jack detection function. The ADAU1461 must operate in master
mode and source BCLK to the input clock of the digital microphone. The DSPRUN bit must also be asserted in Register R62
(DSP run register, Address 0x40F6) for digital microphone
operation.
The MICBIAS pin provides a voltage reference for electret analog
microphones. The MICBIAS voltage is set in Register R10
(record microphone bias control register, Address 0x4010). In
this register, the MICBIAS output can be enabled or disabled.
Additional options include high performance operation and a
gain boost. The gain boost provides two different voltage biases:
0.65 × AVDD or 0.90 × AVDD. When enabled, the high performance bit increases supply current to the microphone bias
circuit to decrease rms input noise.
The digital microphone signal bypasses record path mixers and
ADCs and is routed directly into the decimation filters. The
digital microphone and ADCs share decimation filters and,
therefore, both cannot be used simultaneously. The digital
microphone input select bit, INSEL, can be set in Register R19
(ADC control register, Address 0x4019). Figure 36 depicts the
digital microphone interface and signal routing.
JACKDET/MICIN
R2: DIGITAL MICROPHONE/
JACK DETECTION
CONTROL
The ADAU1461 uses two 24-bit Σ-Δ analog-to-digital converters (ADCs) with selectable oversampling ratios of 64× or
128× (selected by Bit 3 in Register R17, Address 0x4017).
The full-scale input to the ADCs (0 dBFS) is 1.0 V rms with
AVDD = 3.3 V. This full-scale analog input will output a digital
signal at −1.38 dBFS. This gain offset is built into the ADAU1461
to prevent clipping. The full-scale input level scales linearly with
the level of AVDD.
TO JACK
DETECTION
CIRCUIT
DIGITAL MICROPHONE
INTERFACE
LEFT
CHANNEL
ANALOG-TO-DIGITAL CONVERTERS
ADC Full-Scale Level
JDFUNC[1:0]
RIGHT
ADC
The MICBIAS pin can also be used to cleanly supply voltage
to digital microphones or analog microphones with separate
power supply pins.
RIGHT
CHANNEL
For single-ended and pseudo-differential signals, the full-scale
value corresponds to the signal level at the pins, 0 dBFS.
LEFT
ADC
The full differential full-scale input level is measured after the
differential amplifier, which corresponds to −6 dBFS at each pin.
R19: ADC CONTROL
INSEL
DECIMATORS
Figure 36. Digital Microphone Interface Block Diagram
08914-023
Signal levels above the full-scale value cause the ADCs to clip.
Digital ADC Volume Control
The digital ADC volume can be attenuated before DSP processing using Register R20 (left input digital volume register,
Address 0x401A) and Register R21 (right input digital volume
register, Address 0x401B).
High-Pass Filter
By default, a high-pass filter is used in the ADC path to remove
dc offsets; this filter can be enabled or disabled in Register R19
(ADC control register, Address 0x4019). At fS = 48 kHz, the
corner frequency of this high-pass filter is 2 Hz.
Rev. 0 | Page 29 of 88
ADAU1461
AUTOMATIC LEVEL CONTROL (ALC)
•
The ADAU1461 contains a hardware automatic level control
(ALC). The ALC is designed to continuously adjust the PGA
gain to keep the recording volume constant as the input level
varies.
For optimal noise performance, the ALC uses the analog PGA
to adjust the gain instead of using a digital method. This ensures
that the ADC noise is not amplified at low signal levels.
Extremely small gain step sizes are used to ensure high audio
quality during gain changes.
To use the ALC function, the inputs must be applied either
differentially or pseudo-differentially to input pins LINN and
LINP, for the left channel, and RINN and RINP, for the right
channel. The ALC function is not available for the auxiliary line
input pins, LAUX and RAUX.
•
A block diagram of the ALC block is shown in Figure 37. The
ALC logic receives the ADC output signals and analyzes these
digital signals to set the PGA gain. The ALC control registers
are used to control the time constants and output levels, as
described in this section.
ANALOG
INPUT
RIGHT
I2 C
CONTROL
PGA
–12dB TO +35.25dB
0.75dB STEP SIZE
LEFT
ADC
MUTE
SERIAL
PORTS
RIGHT
ADC
ALC
DIGITAL
08914-024
ANALOG
INPUT
LEFT
•
•
Figure 37. ALC Architecture
ALC PARAMETERS
The ALC function is controlled with the ALC control registers
(Address 0x4011 through Address 0x4014) using the following
parameters:
•
•
ALCSEL[2:0]: The ALC select bits are used to enable the
ALC and set the mode to left only, right only, stereo, or
DSP. In stereo mode, the greater of the left or right inputs
is used to calculate the gain, and the same gain is then
applied to both the left and right channels. In DSP mode,
the PGA gain is controlled by the SigmaDSP core.
ALCTARG[3:0]: The ALC target is the desired input
recording level that the ALC attempts to achieve.
ALCATCK[3:0]: The ALC attack time sets how fast the
ALC starts attenuating after a sudden increase in input
level above the ALC target. Although it may seem that
the attack time should be set as fast as possible to avoid
clipping on transients, using a moderate value results in
better overall sound quality. If the value is too fast, the
ALC overreacts to very short transients, causing audible
gain-pumping effects, which sounds worse than using a
moderate value that allows brief periods of clipping on
transients. A typical setting for music recording is 384 ms.
A typical setting for voice recording is 24 ms.
ALCHOLD[3:0]: These bits set the ALC hold time. When
the output signal falls below the target output level, the
gain is not increased unless the output remains below the
target level for the period of time set by the hold time bits.
The hold time is used to prevent the gain from modulating
on a steady low frequency sine wave signal, which would
cause distortion.
ALCDEC[3:0]: The ALC decay time sets how fast the ALC
increases the PGA gain after a sudden decrease in input level
below the ALC target. A very slow setting can be used if the
main function of the ALC is to set a music recording level.
A faster setting can be used if the function of the ALC is to
compress the dynamic range of a voice recording. Using a
very fast decay time can cause audible artifacts such as noise
pumping or distortion. A typical setting for music recording
is 24.58 sec. A typical setting for voice recording is 1.54 sec.
ALCMAX[2:0]: The maximum ALC gain bits are used to
limit the maximum gain that can be programmed into the
ALC. This can be used to prevent excessive noise in the
recording for small input signals. Note that setting this
register to a low value may prevent the ALC from reaching
its target output level, but this behavior is often desirable to
achieve the best overall sound.
Figure 38 shows the dynamic behavior of the PGA gain for a
tone-burst input. The target output is achieved for three different input levels, with the effect of attack, hold, and decay shown
in the figure. Note that for very small signals, the maximum PGA
gain may prevent the ALC from achieving its target level; in the
same way, for very large inputs, the minimum PGA gain may
prevent the ALC from achieving its target level (assuming that
the target output level is set to a very low value). The effects of
the PGA gain limit are shown in the input/output graph of
Figure 39.
Rev. 0 | Page 30 of 88
ADAU1461
the threshold for 250 ms before the noise gate operates.
Hysteresis is used so that the threshold for coming out of the
mute state is 6 dB higher than the threshold for going into the
mute state. There are four operating modes for the noise gate.
INPUT
Noise Gate Mode 0 (see Figure 40) is selected by setting the
NGTYP[1:0] bits to 00. In this mode, the current state of the
PGA gain is held at its current state when the noise gate logic is
activated. This prevents a large increase in background noise
during periods of silence. When using this mode, it is advisable
to use a relatively slow decay time. This is because the noise gate
takes at least 250 ms to activate, and if the PGA gain has already
increased to a large value during this time, the value at which
the gain is held will be large.
GAIN
OUTPUT
THRESHOLD
HOLD DECAY
TIME
TIME
08914-025
INPUT
ATTACK
TIME
Figure 38. Basic ALC Operation
ANALOG
GAIN
MAX GAIN = 30dB
250ms
MIN PGA
GAIN POINT
DIGITAL
MUTE
TARGET
08914-026
INPUT LEVEL (dB)
OUTPUT
08914-027
OUTPUT LEVEL (dB)
MAX GAIN = 18dB
GAIN HELD
INTERNAL
NOISE GATE
ENABLE SIGNAL
MAX GAIN = 24dB
Figure 39. Effect of Varying the Maximum Gain Parameter
NOISE GATE FUNCTION
Figure 40. Noise Gate Mode 0 (PGA Gain Hold)
•
•
•
NGTYP[1:0]: The noise gate type is set to one of four
modes by writing to the NGTYP[1:0] bits.
NGEN: The noise gate function is enabled by writing to the
NGEN bit.
NGTHR[4:0]: The threshold for muting the output is set by
writing to the NGTHR[4:0] bits.
Noise Gate Mode 1 (see Figure 41) is selected by setting the
NGTYP[1:0] bits to 01. In this mode, the ADAU1461 does a
simple digital mute of the ADC output. Although this mode
completely eliminates any background noise, the effect of an
abrupt mute may not be pleasant to the ear.
THRESHOLD
INPUT
ANALOG
GAIN
250ms
INTERNAL
NOISE GATE
ENABLE SIGNAL
One common problem with noise gate functions is chatter,
where a small signal that is close to the noise gate threshold
varies in amplitude, causing the noise gate function to open and
close rapidly. This causes an unpleasant sound.
To reduce this effect, the noise gate in the ADAU1461 uses a
combination of a timeout period and hysteresis. The timeout
period is set to 250 ms, so the signal must consistently be below
Rev. 0 | Page 31 of 88
DIGITAL
MUTE
OUTPUT
08914-028
When using the ALC, one potential problem is that for small
input signals, the PGA gain can become very large. A side effect
of this is that the noise is amplified along with the signal of
interest. To avoid this situation, the ADAU1461 noise gate can
be used. The noise gate cuts off the ADC output when its signal
level is below a set threshold. The noise gate is controlled using
the following parameters in the ALC Control 3 register
(Address 0x4014):
Figure 41. Noise Gate Mode 1 (Digital Mute)
ADAU1461
Noise Gate Mode 3 (see Figure 43) is selected by setting the
NGTYP[1:0] bits to 11. This mode is the same as Mode 2 except
that at the end of the PGA fade gain interval, a digital mute is
performed. In general, this mode is the best-sounding mode,
because the audible effect of the digital hard mute is reduced by
the fact that the gain has already faded to a low level before the
mute occurs.
Noise Gate Mode 2 (see Figure 42) is selected by setting the
NGTYP[1:0] bits to 10. In this mode, the ADAU1461 improves
the sound of the noise gate operation by first fading the PGA
gain over a period of about 100 ms to the minimum PGA gain
value. The ADAU1461 does not do a hard mute after the fade is
complete, so some small background noise will still exist.
THRESHOLD
THRESHOLD
INPUT
ANALOG
GAIN
INPUT
ANALOG
GAIN
250ms
MIN GAIN
250ms
MIN GAIN
100ms
100ms
INTERNAL
NOISE GATE
ENABLE SIGNAL
INTERNAL
NOISE GATE
ENABLE SIGNAL
DIGITAL
MUTE
DIGITAL
MUTE
OUTPUT
08914-029
08914-030
OUTPUT
Figure 43. Noise Gate Mode 3 (Analog Fade/Digital Mute)
Figure 42. Noise Gate Mode 2 (Analog Fade)
Rev. 0 | Page 32 of 88
ADAU1461
PLAYBACK SIGNAL PATH
MX3G1[3:0]
LEFT INPUT MIXER
–15dB TO +6dB
MX3G2[3:0]
RIGHT INPUT MIXER
MIXER 3
(LEFT
PLAYBACK
MIXER)
–15dB TO +6dB
MX3AUXG[3:0]
LAUX
LHPVOL[5:0]
–15dB TO +6dB
–57dB TO +6dB
MIXER 5
(LEFT L/R
PLAYBACK
MIXER)
LEFT DAC
LHP
LOUTVOL[5:0]
MX3LM
–57dB TO +6dB
LOUTP
MX5G3[1:0]
RIGHT DAC
MX3RM
–1
MX6G3[1:0]
LOUTN
MONOVOL[5:0]
MX7[1:0]
MIXER 7
(MONO MIXER)
MONOOUT
–57dB TO +6dB
–1
MX4G1[3:0]
LEFT INPUT MIXER
–15dB TO +6dB
MX5G4[1:0]
MX4G2[3:0]
RIGHT INPUT MIXER
MIXER 6
(RIGHT L/R
PLAYBACK
MIXER)
ROUTVOL[5:0]
–57dB TO +6dB
ROUTP
MX6G4[1:0]
–15dB TO +6dB
RHPVOL[5:0]
MX4AUXG[3:0]
RAUX
ROUTN
–15dB TO +6dB
–57dB TO +6dB
MIXER 4
(RIGHT
PLAYBACK
MIXER)
LEFT DAC
RHP
08914-031
MX4LM
RIGHT DAC
MX4RM
Figure 44. Playback Signal Path
OUTPUT SIGNAL PATHS
Routing Flexibility
The outputs of the ADAU1461 can be configured as a variety of
differential or single-ended outputs. All analog output pins are
capable of driving headphone or earpiece speakers. There are
selectable output paths for stereo signals or a downmixed mono
output. The line outputs can drive a load of at least 10 kΩ or can
be put into HP mode to drive headphones or earpiece speakers.
The analog output pins are biased at AVDD/2.
The playback path contains five mixers (Mixer 3 to Mixer 7)
that perform the following functions:
With a 0 dBFS digital input and AVDD = 3.3 V, the full-scale
output level is 920 mV rms.
Signals are inverted through the mixers and volume controls.
The result of this inversion is that the polarity of the differential
outputs and the headphone outputs is preserved. The singleended mono output is inverted. The DACs are noninverting.
•
•
•
Mix signals from the record path and the DACs.
Mix or swap the left and right channels.
Mix a mono signal or generate a common-mode output.
Mixer 3 and Mixer 4 are dedicated to mixing signals from the
record path and the DACs. Each of these two mixers can accept
signals from the left and right DACs, the left and right input
mixers, and the dedicated channel auxiliary input. Signals
coming from the record path can be boosted or cut before the
playback mixer.
For example, the MX4G2[3:0] bits set the gain from the output
of Mixer 2 (right record channel) to the input of Mixer 4, hence
the naming convention.
Signals coming from the DACs have digital volume attenuation controls set in Register R20 (left input digital volume
register, Address 0x401A) and Register R21 (right input digital
volume register, Address 0x401B).
Rev. 0 | Page 33 of 88
ADAU1461
HEADPHONE OUTPUT
Headphone Output Power-Up/Power-Down Sequencing
The LHP and RHP pins can be driven by either a line output
driver or a headphone driver by setting the HPMODE bit in
Register R30 (playback headphone right volume control register,
Address 0x4024). The headphone outputs can drive a load of at
least 16 Ω.
To prevent pops when turning on the headphone outputs, the
user must wait at least 4 ms to unmute these outputs after
enabling the headphone output with the HPMODE bit. This is
because of an internal capacitor that must charge before these
outputs can be used. Figure 46 and Figure 47 illustrate the
headphone power-up/power-down sequencing.
Separate volume controls for the left and right channels range
from −57 dB to +6 dB. Slew can be applied to all the playback
volume controls using the ASLEW[1:0] bits in Register R34
(playback pop/click suppression register, Address 0x4028).
For capless headphones, configure the MONOOUT pin before
unmuting the headphone outputs.
USER
DEFINED
Capless Headphone Configuration
4ms
HPMODE
1 = HEADPHONE
The headphone outputs can be configured in a capless output
configuration with the MONOOUT pin used as a dc virtual
ground reference. Figure 45 depicts a typical playback path in
a capless headphone configuration. Table 18 lists the register
settings for this configuration. As shown in this table, the
MONOOUT pin outputs common mode (AVDD/2), which
is used as the virtual headphone reference.
LEFT
DAC
MX3LM
MIXER 3
08914-046
RHPM AND LHPM
1 = UNMUTE
INTERNAL
PRECHARGE
LHPVOL[5:0]
Figure 46. Headphone Output Power-Up Timing
LHP
MX3EN
MIXER 7
HPMODE
0 = LINE OUTPUT
RHPVOL[5:0]
RHP
MX4EN
Figure 47. Headphone Output Power-Down Timing
Ground-Centered Headphone Configuration
Figure 45. Capless Headphone Configuration Diagram
Table 18. Capless Headphone Register Settings
Register
R36
R22
R24
R28
R33
R29
R30
Bit Name
DACEN[1:0]
MX3EN
MX3LM
MX4EN
MX4RM
MX7EN
MX7[1:0]
MONOM
MOMODE
LHPVOL[5:0]
LHPM
HPMODE
RHPVOL[5:0]
RHPM
08914-047
MIXER 4
MOMODE
08914-062
MX7EN
MX4RM
USER DEFINED
MONOOUT
MX7[1:0]
RIGHT
DAC
RHPM AND LHPM
0 = MUTE
MONOM
Setting
11 = both DACs on
1 = enable Mixer 3
1 = unmute left DAC input
1 = enable Mixer 4
1 = unmute right DAC input
1 = enable Mixer 7
00 = common-mode output
1 = unmute mono output
1 = headphone output
Desired volume for LHP output
1 = unmute left headphone output
1 = headphone output
Desired volume for RHP output
1 = unmute right headphone output
The headphone outputs can also be configured as groundcentered outputs by placing coupling capacitors on the LHP
and RHP pins. Ground-centered headphones should use the
AGND pin as the ground reference.
When the headphone outputs are configured in this manner,
the capacitors create a high-pass filter on the outputs. The
corner frequency of this filter, at which point its attenuation
is 3 dB, is calculated by the following formula:
f3dB = 1/(2π × R × C)
where:
C is the capacitor value.
R is the impedance of the headphones.
For a typical headphone impedance of 16 Ω and a 47 μF
capacitor, the corner frequency is 211 Hz.
Rev. 0 | Page 34 of 88
ADAU1461
Jack Detection
LINE OUTPUTS
When the JACKDET/MICIN pin is set to the jack detect function, a flag on this pin can be used to mute the line outputs
when headphones are plugged into the jack. This pin can be
configured in Register R2 (digital microphone/jack detection
control register, Address 0x4008). The JDFUNC[1:0] bits set the
functionality of the JACKDET/MICIN pin.
The line output pins (LOUTP, LOUTN, ROUTP, and ROUTN)
can be used to drive both differential and single-ended loads. In
their default settings, these pins can drive typical line loads of
10 kΩ or greater, but they can also be put into headphone mode
by setting the LOMODE bit in Register R31 (playback line output
left volume control register, Address 0x4025) and the ROMODE
bit in Register R32 (playback line output right volume control
register, Address 0x4026). In headphone mode, the line output
pins are capable of driving headphone and earpiece speakers of
16 Ω or greater. The output impedance of the line outputs is
approximately 1 kΩ.
POP-AND-CLICK SUPPRESSION
Upon power-up, precharge circuitry is enabled to suppress pops
and clicks. After power-up, the precharge circuitry can be put
into a low power mode using the POPMODE bit in Register R34
(playback pop/click suppression register, Address 0x4028).
The precharge time depends on the capacitor value on the CM
pin and the RC time constant of the load. For a typical line output
load, the precharge time is between 2 ms and 3 ms. After this
precharge time, the POPMODE bit can be set to low power mode.
Changing any register settings that affect the signal path can
cause pops and clicks on the analog outputs. To avoid these pops
and clicks, mute the appropriate outputs using Register R29 to
Register R32 (Address 0x4023 to Address 0x4026). Unmute the
analog outputs after the changes are made.
When the line output pins are used in single-ended mode,
LOUTP and ROUTP should be used to output the signals, and
LOUTN and ROUTN should be left unconnected.
The volume controls for these outputs range from −57 dB to
+6 dB. Slew can be applied to all the playback volume controls
using the ASLEW[1:0] bits in Register R34 (playback pop/click
suppression register, Address 0x4028).
The MX5G4[1:0], MX5G3[1:0], MX6G3[1:0], and MX6G4[1:0]
bits can all provide a 6 dB gain boost to the line outputs. This
gain boost allows single-ended output signals to achieve 0 dBV
(1.0 V rms) and differential output signals to achieve up to
6 dBV (2.0 V rms). For more information, see Register R26
(playback L/R mixer left (Mixer 5) line output control register,
Address 0x4020) and Register R27 (playback L/R mixer right
(Mixer 6) line output control register, Address 0x4021).
LEFT DAC
MIXER 3
MX5G3[1:0]
MIXER 5
LOUTVOL[5:0]
LOUTP
–1
–1
RIGHT DAC
MIXER 4
MX6G4[1:0]
MIXER 6
ROUTN
ROUTVOL[5:0]
ROUTP
Figure 48. Differential Line Output Configuration
Rev. 0 | Page 35 of 88
LOUTN
08914-063
Additional settings for jack detection include debounce time
(JDDB[1:0] bits) and detection polarity (JDPOL bit). Because
the jack detection and digital microphone share a pin, both
functions cannot be used simultaneously.
ADAU1461
CONTROL PORTS
The ADAU1461 can operate in one of two control modes:
•
•
2
I C control
SPI control
The ADAU1461 has both a 4-wire SPI control port and a
2-wire I2C bus control port. Both ports can be used to set the
registers. The part defaults to I2C mode, but it can be put into
SPI control mode by pulling the CLATCH pin low three times.
The control port is capable of full read/write operation for all
addressable registers. The ADAU1461 must have a valid master
clock in order to write to all registers except for Register R0
(Address 0x4000) and Register R1 (Address 0x4002).
All addresses can be accessed in both a single-address mode
or a burst mode. The first byte (Byte 0) of a control port write
contains the 7-bit chip address plus the R/W bit. The next two
bytes (Byte 1 and Byte 2) together form the subaddress of the
register location within the ADAU1461. This subaddress must
be two bytes long because the memory locations within the
ADAU1461 are directly addressable and their sizes exceed the
range of single-byte addressing. All subsequent bytes (starting
with Byte 3) contain the data, such as control port data, program
data, or parameter data. The number of bytes per word depends
on the type of data that is being written.
The ADAU1461 has several mechanisms for updating signal processing parameters in real time without causing pops or clicks. If
large blocks of data need to be downloaded, the output of the DSP
core can be halted (using the DSPRUN bit in the DSP run register,
Address 0x40F6), new data can be loaded, and the device can be
restarted. This is typically done during the booting sequence at
start-up or when loading a new program into RAM.
The control port pins are multifunctional, depending on the
mode in which the part is operating. Table 19 describes these
multiple functions.
Table 19. Control Port Pin Functions
Pin Name
SCL/CCLK
SDA/COUT
ADDR1/CDATA
ADDR0/CLATCH
I2C Mode
SCL: input clock
SDA: open-collector
input/output
I2C Address Bit 1: input
I2C Address Bit 0: input
SPI Mode
CCLK: input clock
COUT: output
CDATA: input
CLATCH: input
BURST MODE WRITING AND READING
Burst mode addressing, where the subaddresses are automatically
incremented at word boundaries, can be used for writing large
amounts of data to contiguous registers. This increment happens
automatically after a single-word write or read unless a stop condition is encountered (I2C) or CLATCH is brought high (SPI). A
burst write starts like a single-word write, but following the first
data-word, the data-word for the next immediate address can be
written immediately without sending its two-byte address.
The registers in the ADAU1461 are one byte wide with the
exception of the PLL control register, which is six bytes wide.
The autoincrement feature knows the word length at each
subaddress, so the subaddress does not need to be specified
manually for each address in a burst write.
The subaddresses are autoincremented by 1 following each read
or write of a data-word, regardless of whether there is a valid register or RAM word at that address. Address holes in the register
map can be written to or read from without consequence. In the
ADAU1461, these address holes exist at Address 0x4001, Address
0x4003 to Address 0x4007, Address 0x402E, Address 0x4032 to
Address 0x4035, Address 0x4037 to Address 0x40BF, Address
0x40C5, Address 0x40CA to Address 0x40CF, Address 0x40D5
to Address 0x40EA, and Address 0x40EC to Address 0x40F1. A
single-byte write to these registers is ignored by the ADAU1461,
and a read returns a single byte 0x00.
I2C PORT
The ADAU1461 supports a 2-wire serial (I2C-compatible)
microprocessor bus driving multiple peripherals. Two pins,
serial data (SDA) and serial clock (SCL), carry information
between the ADAU1461 and the system I2C master controller.
In I2C mode, the ADAU1461 is always a slave on the bus,
meaning that it cannot initiate a data transfer. Each slave device
is recognized by a unique address. The address and R/W byte
format is shown in Table 20. The address resides in the first
seven bits of the I2C write. Bits[5:6] of the I2C address for the
ADAU1461 are set by the levels on the ADDR1 and ADDR0
pins. The LSB of the address—the R/W bit—specifies either a
read or write operation. Logic Level 1 corresponds to a read
operation, and Logic Level 0 corresponds to a write operation.
Table 20. ADAU1461 I2C Address and Read/Write Byte Format
Bit 0
0
Bit 1
1
Bit 2
1
Bit 3
1
Bit 4
0
Bit 5
ADDR1
Bit 6
ADDR0
Bit 7
R/W
The SDA and SCL pins should each have a 2 kΩ pull-up resistor
on the line connected to it. The voltage on these signal lines
should not be higher than IOVDD (3.3 V).
Addressing
Initially, each device on the I2C bus is in an idle state and
monitors the SDA and SCL lines for a start condition and
the proper address. The I2C master initiates a data transfer by
establishing a start condition, defined by a high-to-low transition
on SDA while SCL remains high. This indicates that an address/
data stream follows. All devices on the bus respond to the start
condition and shift the next eight bits (the 7-bit address plus the
R/W bit) MSB first. The device that recognizes the transmitted
address responds by pulling the data line low during the ninth
clock pulse. This ninth bit is known as an acknowledge bit. All
other devices withdraw from the bus at this point and return to
the idle condition.
Rev. 0 | Page 36 of 88
ADAU1461
the user should only issue one start condition, one stop condition,
or a single stop condition followed by a single start condition. If
an invalid subaddress is issued by the user, the ADAU1461 does
not issue an acknowledge and returns to the idle condition.
The R/W bit determines the direction of the data. A Logic 0 on
the LSB of the first byte means that the master will write information to the peripheral, whereas a Logic 1 means that the
master will read information from the peripheral after writing
the subaddress and repeating the start address. A data transfer
takes place until a stop condition is encountered. A stop
condition occurs when SDA transitions from low to high while
SCL is held high. Figure 49 shows the timing of an I2C write,
and Figure 50 shows an I2C read.
If the user exceeds the highest subaddress while in autoincrement
mode, one of two actions is taken. In read mode, the ADAU1461
outputs the highest subaddress register contents until the master
device issues a no acknowledge, indicating the end of a read. A
no acknowledge condition is where the SDA line is not pulled
low on the ninth clock pulse on SCL. If the highest subaddress
location is reached while in write mode, the data for the invalid
byte is not loaded into any subaddress register, a no acknowledge
is issued by the ADAU1461, and the part returns to the idle
condition.
Stop and start conditions can be detected at any stage during the
data transfer. If these conditions are asserted out of sequence with
normal read and write operations, the ADAU1461 immediately
jumps to the idle condition. During a given SCL high period,
SCL
0
SDA
1
1
1
R/W
0
ADDR1 ADDR0
START BY
MASTER
FRAME 1
CHIP ADDRESS BYTE
ACK BY
ADAU1461
ACK BY
ADAU1461
FRAME 2
SUBADDRESS BYTE 1
SCL
(CONTINUED)
ACK BY
ADAU1461
FRAME 3
SUBADDRESS BYTE 2
ACK BY
ADAU1461
FRAME 4
DATA BYTE 1
STOP BY
MASTER
08914-032
SDA
(CONTINUED)
Figure 49. I2C Write to ADAU1461 Clocking
SCL
SDA
START BY
MASTER
0
1
1
1
0
R/W
ADDR1 ADDR0
ACK BY
ADAU1461
ACK BY
ADAU1461
FRAME 1
CHIP ADDRESS BYTE
FRAME 2
SUBADDRESS BYTE 1
SCL
(CONTINUED)
SDA
(CONTINUED)
0
FRAME 3
SUBADDRESS BYTE 2
ACK BY
ADAU1461
1
REPEATED
START BY MASTER
1
1
0
R/W
ADDR1 ADDR0
ACK BY
ADAU1461
FRAME 4
CHIP ADDRESS BYTE
SCL
(CONTINUED)
ACK BY
MASTER
STOP BY
MASTER
FRAME 5
READ DATA BYTE 1
Figure 50. I2C Read from ADAU1461 Clocking
Rev. 0 | Page 37 of 88
08914-033
SDA
(CONTINUED)
ADAU1461
I2C Read and Write Operations
This causes the ADAU1461 SDA to reverse and begin driving
data back to the master. The master then responds every ninth
pulse with an acknowledge pulse to the ADAU1461.
Figure 51 shows the format of a single-word write operation.
Every ninth clock pulse, the ADAU1461 issues an acknowledge
by pulling SDA low.
Figure 54 shows the format of a burst mode read sequence. This
figure shows an example of a read from sequential single-byte
registers. The ADAU1461 increments its subaddress register
after every byte because the requested subaddress corresponds
to a register or memory area with a 1-byte word length. The
ADAU1461 always decodes the subaddress and sets the autoincrement circuit so that the address increments after the
appropriate number of bytes.
Figure 52 shows the format of a burst mode write sequence. This
figure shows an example of a write to sequential single-byte
registers. The ADAU1461 increments its subaddress register
after every byte because the requested subaddress corresponds
to a register or memory area with a 1-byte word length.
Figure 53 shows the format of a single-word read operation. Note
that the first R/W bit is 0, indicating a write operation. This is
because the subaddress still needs to be written to set up the
internal address. After the ADAU1461 acknowledges the receipt
of the subaddress, the master must issue a repeated start command
followed by the chip address byte with the R/W bit set to 1 (read).
S
AS
Chip address,
R/W = 0
Subaddress high byte
Figure 51 to Figure 54 use the following abbreviations:
S = start bit
P = stop bit
AM = acknowledge by master
AS = acknowledge by slave
AS
Subaddress low byte
AS
Data Byte 1
P
Figure 51. Single-Word I2C Write Format
S
Chip address,
R/W = 0
AS
Subaddress
high byte
AS
Subaddress
low byte
AS
AS
Data
Byte 1
AS
Data
Byte 2
Data
Byte 3
AS
Data
Byte 4
AS
…
P
Figure 52. Burst Mode I2C Write Format
S
Chip address,
R/W = 0
AS
Subaddress high
byte
AS
Subaddress low
byte
AS
S
Chip address,
R/W = 1
AS
P
Data
Byte 1
Figure 53. Single-Word I2C Read Format
S
Chip address,
R/W = 0
AS
Subaddress
high byte
AS
Subaddress
low byte
AS
S
Chip address,
R/W = 1
Figure 54. Burst Mode I2C Read Format
Rev. 0 | Page 38 of 88
AS
Data
Byte 1
AM
Data
Byte 2
AM
…
P
ADAU1461
SPI PORT
Chip Address R/W
2
By default, the ADAU1461 is in I C mode, but it can be put into
SPI control mode by pulling CLATCH low three times. This is
done by performing three dummy writes to the SPI port (the
ADAU1461 does not acknowledge these three writes). Beginning
with the fourth SPI write, data can be written to or read from
the IC. The ADAU1461 can be taken out of SPI mode only by
a full reset initiated by power-cycling the IC.
The LSB of the first byte of an SPI transaction is a R/W bit. This bit
determines whether the communication is a read (Logic Level 1)
or a write (Logic Level 0). This format is shown in Table 21.
The SPI port uses a 4-wire interface, consisting of the CLATCH,
CCLK, CDATA, and COUT signals, and it is always a slave port.
The CLATCH signal should go low at the beginning of a transaction and high at the end of a transaction. The CCLK signal
latches CDATA on a low-to-high transition. COUT data is shifted
out of the ADAU1461 on the falling edge of CCLK and should
be clocked into a receiving device, such as a microcontroller, on
the CCLK rising edge. The CDATA signal carries the serial input
data, and the COUT signal carries the serial output data. The
COUT signal remains three-state until a read operation is requested.
This allows other SPI-compatible peripherals to share the same
readback line. All SPI transactions have the same basic format
shown in Table 22. A timing diagram is shown in Figure 4. All
data should be written MSB first.
Subaddress
Table 21. ADAU1461 SPI Address and Read/Write Byte Format
Bit 0
0
Bit 1
0
Bit 2
0
Bit 3
0
Bit 4
0
Bit 5
0
Bit 6
0
The 16-bit subaddress word is decoded into a location in one of
the registers. This subaddress is the location of the appropriate
register. The MSBs of the subaddress are zero-padded to bring
the word to a full 2-byte length.
Data Bytes
The number of data bytes varies according to the register being
accessed. During a burst mode write, an initial subaddress is
written followed by a continuous sequence of data for consecutive register locations.
A sample timing diagram for a single-word SPI write operation
to a register is shown in Figure 55. A sample timing diagram of
a single-word SPI read operation is shown in Figure 56. The
COUT pin goes from being three-state to being driven at the
beginning of Byte 3. In this example, Byte 0 to Byte 2 contain
the addresses and R/W bit, and subsequent bytes carry the data.
Table 22. Generic Control Word Format
Byte 0
chip_adr[6:0], R/W
Byte 2
subaddr[7:0]
Byte 3
data
Byte 4 1
data
Continues to end of data.
CLATCH
CDATA
BYTE 0
BYTE 1
BYTE 2
08914-038
CCLK
BYTE 3
Figure 55. SPI Write to ADAU1461 Clocking (Single-Word Write Mode)
CLATCH
CCLK
CDATA
COUT
BYTE 1
BYTE 0
BYTE 2
HIGH-Z
DATA
Figure 56. SPI Read from ADAU1461 Clocking (Single-Word Read Mode)
Rev. 0 | Page 39 of 88
HIGH-Z
08914-039
1
Byte 1
subaddr[15:8]
Bit 7
R/W
ADAU1461
SERIAL DATA INPUT/OUTPUT PORTS
If the PLL of the ADAU1461 is not used, the serial data clocks
must be synchronous with the ADAU1461 master clock input.
The LRCLK and BCLK pins are used to clock both the serial
input and output ports. The ADAU1461 can be set as the master
or the slave in a system. Because there is only one set of serial
data clocks, the input and output ports must always be both
master or both slave.
Register R15 and Register R16 (serial port control registers,
Address 0x4015 and Address 0x4016) allow control of clock
polarity and data input modes. The valid data formats are I2S,
left-justified, right-justified (24-/20-/18-/16-bit), and TDM. In
all modes except for the right-justified modes, the serial port
inputs an arbitrary number of bits up to a limit of 24. Extra bits
do not cause an error, but they are truncated internally.
The serial port can operate with an arbitrary number of BCLK
transitions in each LRCLK frame. The LRCLK in TDM mode
can be input to the ADAU1461 either as a 50% duty cycle clock
or as a bit-wide pulse.
When the LRCLK is set as a pulse, a 47 pF capacitor should be
connected between the LRCLK pin and ground (see Figure 57).
This capacitor is necessary in both master and slave modes to
properly align the LRCLK signal to the serial data stream.
ADAU1461
LRCLK
47pF
08914-071
The flexible serial data input and output ports of the ADAU1461
can be set to accept or transmit data in 2-channel format or in
a 4-channel or 8-channel TDM stream to interface to external
ADCs or DACs. Data is processed in twos complement, MSB
first format. The left channel data field always precedes the right
channel data field in 2-channel streams. In TDM mode, Slot 0
to Slot 3 are in the first half of the audio frame, and Slot 4 to
Slot 7 are in the second half of the frame. The serial modes and
the position of the data in the frame are set in Register R15 to
Register R18 (serial port and converter control registers,
Address 0x4015 to Address 0x4018).
BCLK
Figure 57. LRCLK Capacitor Alignment, TDM Pulse Mode
In TDM 8 mode, the ADAU1461 can be a master for fS up to
48 kHz. Table 23 lists the modes in which the serial output port
can function.
Table 23. Serial Output Port Master/Slave Mode Capabilities
fS
48 kHz
96 kHz
2-Channel Modes (I2S, LeftJustified, Right-Justified)
Master and slave
Master and slave
8-Channel TDM
Master and slave
Slave
Table 24 describes the proper configurations for standard audio
data formats.
Table 24. Data Format Configurations
Format
I2S
(see Figure 58)
Left-Justified (see
Figure 59)
Right-Justified
(see Figure 60)
TDM with Clock
(see Figure 61)
TDM with Pulse
(see Figure 62)
LRCLK Polarity (LRPOL)
Frame begins on falling edge
LRCLK Mode
(LRMOD)
50% duty cycle
Frame begins on rising edge
50% duty cycle
Frame begins on rising edge
50% duty cycle
Frame begins on falling edge
50% duty cycle
Frame begins on rising edge
Pulse
BCLK Polarity
(BPOL)
Data changes
on falling edge
Data changes
on falling edge
Data changes
on falling edge
Data changes
on falling edge
Data changes
on falling edge
Rev. 0 | Page 40 of 88
BCLK Cycles/Audio
Frame (BPF[2:0])
32 to 64
32 to 64
32 to 64
64 to 256
64 to 256
Data Delay from LRCLK
Edge (LRDEL[1:0])
Delayed from LRCLK edge
by 1 BCLK
Aligned with LRCLK edge
Delayed from LRCLK edge
by 8 or 16 BCLKs
Delayed from start of word
clock by 1 BCLK
Delayed from start of word
clock by 1 BCLK
ADAU1461
LEFT CHANNEL
LRCLK
RIGHT CHANNEL
BCLK
LSB
MSB
LSB
MSB
08914-040
SDATA
1/fS
2
Figure 58. I S Mode—16 Bits to 24 Bits per Channel
MSB
LSB
MSB
LSB
08914-041
SDATA
RIGHT CHANNEL
LEFT CHANNEL
LRCLK
BCLK
1/fS
Figure 59. Left-Justified Mode—16 Bits to 24 Bits per Channel
RIGHT CHANNEL
SDATA
MSB
LSB
MSB
LSB
08914-042
LEFT CHANNEL
LRCLK
BCLK
1/fS
Figure 60. Right-Justified Mode—16 Bits to 24 Bits per Channel
LRCLK
256 BCLKs
BCLK
SDATA
32 BCLKs
SLOT 0
SLOT 1
SLOT 2
SLOT 3
SLOT 4
SLOT 5
SLOT 6
SLOT 7
LRCLK
MSB – 1
MSB – 2
08914-043
BCLK
MSB
SDATA
Figure 61. TDM 8 Mode
LRCLK
BCLK
MSB TDM
MSB TDM
CH
8
CH
0
SLOT 0
SLOT 1
SLOT 2
SLOT 3
SLOT 4
SLOT 5
SLOT 6
SLOT 7
08914-044
SDATA
32
BCLKs
Figure 62. TDM 8 Mode with Pulse Word Clock
Rev. 0 | Page 41 of 88
ADAU1461
APPLICATIONS INFORMATION
POWER SUPPLY BYPASS CAPACITORS
GROUNDING
Each analog and digital power supply pin should be bypassed to
its nearest appropriate ground pin with a single 100 nF capacitor. The connections to each side of the capacitor should be as
short as possible, and the trace should stay on a single layer with
no vias. For maximum effectiveness, locate the capacitor equidistant from the power and ground pins or, when equidistant
placement is not possible, slightly closer to the power pin.
Thermal connections to the ground planes should be made
on the far side of the capacitor.
A single ground plane should be used in the application layout.
Components in an analog signal path should be placed away
from digital signals.
Each supply signal on the board should also be bypassed with a
single bulk capacitor (10 μF to 47 μF).
VDD
EXPOSED PAD PCB DESIGN
The ADAU1461 has an exposed pad on the underside of the
LFCSP. This pad is used to couple the package to the PCB for
heat dissipation when using the outputs to drive earpiece or
headphone loads. When designing a board for the ADAU1461,
special consideration should be given to the following:
•
GND
•
CAPACITOR
A copper layer equal in size to the exposed pad should be
on all layers of the board, from top to bottom, and should
connect somewhere to a dedicated copper board layer (see
Figure 65).
Vias should be placed to connect all layers of copper,
allowing for efficient heat and energy conductivity. For an
example, see Figure 66, which has nine vias arranged in a
3 inch × 3 inch grid in the pad area.
08914-048
TO GND
TOP
GROUND
POWER
BOTTOM
Figure 63. Recommended Power Supply Bypass Capacitor Layout
VIAS
COPPER SQUARES
Figure 65. Exposed Pad Layout Example, Side View
GSM NOISE FILTER
In mobile phone applications, excessive 217 Hz GSM noise on
the analog supply pins can degrade the audio quality. To avoid
this problem, it is recommended that an L-C filter be used in
series with the bypass capacitors for the AVDD pins. This filter
should consist of a 1.2 nH inductor and a 9.1 pF capacitor in
series between AVDD and ground, as shown in Figure 64.
10µF
08914-051
+
0.1µF
0.1µF
1.2nH 9.1pF
AVDD
08914-049
AVDD
Figure 66. Exposed Pad Layout Example, Top View
Figure 64. GSM Filter on the Analog Supply Pins
Rev. 0 | Page 42 of 88
08914-050
TO VDD
ADAU1461
DSP CORE
SIGNAL PROCESSING
PROGRAM COUNTER
The ADAU1461 is designed to provide all audio signal processing
functions commonly used in stereo or mono low power record
and playback systems. The signal processing flow is designed
using the SigmaStudio software, which allows graphical entry
and real-time control of all signal processing functions.
The execution of instructions in the core is governed by a program
counter, which sequentially steps through the addresses of the
program RAM. The program counter starts every time that a
new audio frame is clocked into the core. SigmaStudio inserts
a jump-to-start command at the end of every program. The
program counter increments sequentially until it reaches this
command and then jumps to the program start address and
waits for the next audio frame to clock into the core.
Many of the signal processing functions are coded using full,
56-bit, double-precision arithmetic data. The input and output
word lengths of the DSP core are 24 bits. Four extra headroom
bits are used in the processor to allow internal gains of up to
24 dB without clipping. Additional gains can be achieved by
initially scaling down the input signal in the DSP signal flow.
ARCHITECTURE
The DSP core consists of a simple 28-/56-bit multiply-accumulate
(MAC) unit with two sources: a data source and a coefficient
source. The data source can come from the data RAM, a ROM
table of commonly used constant values, or the audio inputs to
the core. The coefficient source can come from the parameter
RAM or from a ROM table of commonly used constant values.
The two sources are multiplied in a 28-bit fixed-point multiplier
and then the signal is input to the 56-bit adder; the result is usually
stored in one of three 56-bit accumulator registers. The accumulators can be output from the core (in 28-bit format) or can
optionally be written back into the data or parameter RAMs.
DATA SOURCE
(DATA RAM,
ROM CONSTANTS,
AUDIO INPUTS)
28
56
The SigmaDSP core was designed specifically for audio processing
and therefore includes several features intended for maximizing
efficiency. These include hardware decibel conversion and audiospecific ROM constants.
STARTUP
Before the DSPRUN bit is set or any settings are written to the
parameter RAM, the DSP core must be enabled by setting the
DSPEN bit in Register R61 (Address 0x40F5).
The following steps should be performed every time that a new
program is loaded to the SigmaDSP core, or any time that the
DSPRUN bit is disabled and reenabled.
1.
2.
3.
4.
5.
COEFFICIENT SOURCE
(PARAMETER RAM,
ROM CONSTANTS)
28
FEATURES
Changing any register setting or RAM can cause pops and
clicks on the analog outputs. To avoid these pops and clicks,
mute the appropriate outputs using Register R29 to Register R32
(Address 0x4023 to Address 0x4026). Unmute the analog outputs after the startup procedure is completed.
28
TRUNCATOR
56
56
DATA OPERATIONS
(ACCUMULATORS (3), dB CONVERSION,
BIT OPERATORS, BIT SHIFTER, ...)
56
TRUNCATOR
08914-067
28
OUTPUTS
Set the DSPSR[3:0] bits in Register R57 (Address 0x40EB)
to 1111 (none).
Set the DSPRUN bit in Register R62 (Address 0x40F6) to 0.
Download the rest of the registers, the program RAM, and
the parameter RAM.
Set the DSPRUN bit in Register R62 to 1.
Set the DSPSR[3:0] bits in Register R57 to the operational
setting (default value is 0001).
Figure 67. Simplified DSP Core Architecture
Rev. 0 | Page 43 of 88
ADAU1461
NUMERIC FORMATS
DSP systems commonly use a standard numeric format.
Fractional numeric systems are specified by an A.B format,
where A is the number of bits to the left of the decimal point
and B is the number of bits to the right of the decimal point.
The ADAU1461 uses numeric format 5.23 for both the
parameter and data values.
Numeric Format 5.23
The serial port accepts up to 24 bits on the input and is signextended to the full 28 bits of the DSP core. This allows internal
gains of up to 24 dB without internal clipping.
A digital clipper circuit is used between the output of the DSP
core and the DACs or serial port outputs (see Figure 68). This
circuit clips the top four bits of the signal to produce a 24-bit
output with a range of 1.0 (minus 1 LSB) to −1.0. Figure 68
shows the maximum signal levels at each point in the data flow
in both binary and decibel levels.
Linear range: −16.0 to (+16.0 − 1 LSB)
DATA IN
1.23
(0dB)
SERIAL
PORT
1.23
(0dB)
SIGNAL
PROCESSING
(5.23 FORMAT)
5.23
(24dB)
DIGITAL
CLIPPER
5.23
(24dB)
1.23
(0dB)
Figure 68. Numeric Precision and Clipping Structure
PROGRAMMING
On power-up, the ADAU1461 must be configured with a clocking scheme and then loaded with register settings. After the codec
signal path is set up, the DSP core can be programmed. There
are 1024 instruction cycles per audio sample, resulting in an
internal clock rate of 49.152 MHz when fS = 48 kHz.
The part can be programmed easily using SigmaStudio, a graphical
tool provided by Analog Devices (see Figure 69). No knowledge
of writing line-level DSP code is required. More information
about SigmaStudio can be found at www.analog.com.
08914-069
Examples:
1000 0000 0000 0000 0000 0000 0000 = −16.0
1110 0000 0000 0000 0000 0000 0000 = −4.0
1111 1000 0000 0000 0000 0000 0000 = −1.0
1111 1110 0000 0000 0000 0000 0000 = −0.25
1111 1111 0011 0011 0011 0011 0011 = −0.1
1111 1111 1111 1111 1111 1111 1111 = (1 LSB below 0)
0000 0000 0000 0000 0000 0000 0000 = 0
0000 0000 1100 1100 1100 1100 1101 = 0.1
0000 0010 0000 0000 0000 0000 0000 = 0.25
0000 1000 0000 0000 0000 0000 0000 = 1.0
0010 0000 0000 0000 0000 0000 0000 = 4.0
0111 1111 1111 1111 1111 1111 1111 = (16.0 − 1 LSB)
08914-068
4-BIT SIGN EXTENSION
Figure 69. SigmaStudio Screen Shot
Rev. 0 | Page 44 of 88
ADAU1461
PROGRAM RAM, PARAMETER RAM, AND DATA RAM
Table 25. RAM Map and Read/Write Modes
Memory
Parameter RAM
Program RAM
Size
1024 × 32
1024 × 40
Address Range
0 to 1023 (0x0000 to 0x03FF)
2048 to 3071 (0x0800 to 0x0BFF)
Table 25 shows the RAM map (the ADAU1461 register map is
provided in the Control Registers section). The address space
encompasses a set of registers and three RAMs: program,
parameter, and data. The program RAM and parameter RAM
are not initialized on power-up and are in an unknown state
until written to.
PROGRAM RAM
The program RAM contains the 40-bit operation codes that
are executed by the core. The SigmaStudio compiler calculates
maximum instructions per frame for a project and generates an
error when the value exceeds the maximum allowable instructions
per frame based on the sample rate of the signals in the core.
Because the end of a program contains a jump-to-start command,
the unused program RAM space does not need to be filled with
no-operation (NOP) commands.
PARAMETER RAM
The parameter RAM is 32 bits wide and occupies Address 0
to Address 1023. Each parameter is padded with four 0s before
the MSB to extend the 28-bit word to a full 4-byte width. The
data format of the parameter RAM is twos complement, 5.23.
This means that the coefficients can range from +16.0 (minus
1 LSB) to −16.0, with 1.0 represented by the binary word
0000 1000 0000 0000 0000 0000 0000 or by the hexadecimal
word 0x00 0x80 0x00 0x00.
The parameter RAM can be written to directly or with a safeload write. The direct write mode of operation is typically used
during a complete new loading of the RAM using burst mode
addressing to avoid any clicks or pops in the outputs. Note that
this mode can be used during live program execution, but because
there is no handshaking between the core and the control port,
the parameter RAM is unavailable to the DSP core during control
writes, resulting in pops and clicks in the audio stream.
SigmaStudio automatically assigns the first eight positions to
safeload parameters; therefore, project-specific parameters start
at Address 0x0008.
The parameter RAM should not be written to until the DSPEN
bit has been set in Register R61 (Address 0x40F5).
DATA RAM
Read
Yes
Yes
Write
Yes
Yes
Write Modes
Direct, safeload
Direct
When implementing blocks, such as delays, that require large
amounts of data RAM space, data RAM utilization should be
taken into account. The SigmaDSP core processes delay times
in one-sample increments; therefore, the total pool of delay
available to the user equals 4096 multiplied by the sample
period. For a fS,DSP of 48 kHz, the pool of available delay is a
maximum of about 86 ms, where fS,DSP is the DSP core sampling
rate. In practice, this much data memory is not available to the
user because every block in a design uses a few data memory
locations for its processing. In most DSP programs, this does
not significantly affect the total delay time. The SigmaStudio
compiler manages the data RAM and indicates whether the
number of addresses needed in the design exceeds the maximum number available.
READ/WRITE DATA FORMATS
The read/write formats of the control port are designed to be
byte oriented to allow for easy programming of common microcontroller chips. To fit into a byte-oriented format, 0s are added
to the data fields before the MSB to extend the data-word to
eight bits. For example, 28-bit words written to the parameter
RAM are preceded by four leading 0s to equal 32 bits (four bytes);
40-bit words written to the program RAM are not preceded by
0s because they are already a full five bytes. These zero-padded
data fields are appended to a 3-byte field consisting of a 7-bit
chip address, a read/write bit, and a 16-bit RAM/register address.
The control port knows how many data bytes to expect based
on the address given in the first three bytes.
The total number of bytes for a single-location write command
can vary from one byte (for a control register write) to five bytes
(for a program RAM write). Burst mode can be used to fill
contiguous register or RAM locations. A burst mode write begins
by writing the address and data of the first RAM or register location
to be written. Rather than ending the control port transaction
(by issuing a stop command in I2C mode or by bringing the
CLATCH signal high in SPI mode after the data-word), as
would be done in a single-address write, the next data-word
can be written immediately without specifying its address. The
ADAU1461 control port autoincrements the address of each write
even across the boundaries of the different RAMs and registers.
Table 27 and Table 29 show examples of burst mode writes.
The ADAU1461 data RAM is used to store audio data-words for
processing, as well as certain run-time parameters. SigmaStudio
provides the data and address information for writing to and
reading from the data RAM.
Rev. 0 | Page 45 of 88
ADAU1461
Table 26. Parameter RAM Read/Write Format (Single Address)
Byte 0
chip_adr[6:0], R/W
Byte 1
param_adr[15:8]
Byte 2
param_adr[7:0]
Byte 3
0000, param[27:24]
Bytes[4:6]
param[23:0]
Table 27. Parameter RAM Block Read/Write Format (Burst Mode)
Byte 0
chip_adr[6:0], R/W
Byte 1
param_adr[15:8]
Byte 2
param_adr[7:0]
Byte 3
0000, param[27:24]
Bytes[4:6]
param[23:0]
<—param_adr—>
Bytes[7:10]
Bytes[11:14]
param_adr + 1
param_adr + 2
Table 28. Program RAM Read/Write Format (Single Address)
Byte 0
chip_adr[6:0], R/W
Byte 1
prog_adr[15:8]
Byte 2
prog_adr[7:0]
Bytes[3:7]
prog[39:0]
Table 29. Program RAM Block Read/Write Format (Burst Mode)
Byte 0
chip_adr[6:0], R/W
Byte 1
prog_adr[15:8]
Byte 2
prog_adr[7:0]
SOFTWARE SAFELOAD
To update parameters in real time while avoiding pop and click
noises on the output, the ADAU1461 uses a software safeload
mechanism. The software safeload mechanism enables the
SigmaDSP core to load new parameters into RAM while guaranteeing that the parameters are not in use. This prevents an
undesirable condition where an instruction could execute with
a mix of old and new parameters.
SigmaStudio sets up the necessary code and parameters automatically for new projects. The safeload code, along with other
initialization code, fills the first 39 locations in program RAM.
The first eight parameter RAM locations (Address 0x0000 to
Address 0x0007) are configured by default in SigmaStudio as
described in Table 30.
Table 30. Software Safeload Parameter RAM Defaults
Address (Hex)
0x0000
0x0001
0x0002
0x0003
0x0004
0x0005
0x0006
0x0007
Function
Modulo RAM size
Safeload Data 1
Safeload Data 2
Safeload Data 3
Safeload Data 4
Safeload Data 5
Safeload target address (offset of −1)
Number of words to write/safeload trigger
Bytes[3:7]
prog[39:0]
Bytes[8:12]
Bytes[13:17]
<—prog_adr—>
prog_adr + 1
prog_adr + 2
Parameter RAM Address 0x0001 to Address 0x0005 are the five
data slots for storing the data to be safeloaded. The safeload
parameter space contains five data slots by default because most
standard signal processing algorithms have five parameters or less.
Address 0x0006 is the target address in parameter RAM (with
an offset of −1). This designates the first address to be written.
If more than one word is written, the address increments automatically for each data-word. Up to five sequential parameter
RAM locations can be updated with safeload during each audio
frame. The target address offset of −1 is used because the write
address is calculated relative to the address of the data, which
starts at Address 0x0001. Therefore, to update a parameter at
Address 0x000A, the target address is 0x0009.
Address 0x0007 designates the number of words to be written
into the parameter RAM during the safeload. A biquad filter
uses all five safeload data addresses. A simple mono gain cell
uses only one safeload data address. Writing to Address 0x0007
also triggers the safeload write to occur in the next audio frame.
The safeload mechanism is software based and executes once
per audio frame. Therefore, system designers must take care
when designing the communication protocol. A delay equal to
or greater than the sampling period (the inverse of sampling
frequency) is required between each safeload write. A sample
rate of 48 kHz equates to a delay of at least 21 μs. If this delay
is not observed, the downloaded data is corrupted.
Address 0x0000, which controls the modulo RAM size, is set
by SigmaStudio and is based on the dynamic address generator
mode of the project.
Rev. 0 | Page 46 of 88
ADAU1461
SOFTWARE SLEW
Because algorithms that use software slew generally require more
RAM than their nonslew equivalents, they should be used only
in situations where a parameter will change during operation of
the device.
Figure 70 shows an example of volume slew applied to a sine wave.
The target value takes an additional space in parameter RAM,
and the current value of the parameter is updated in the nonmodulo section of data RAM. Assignment of parameters and
nonmodulo data RAM is handled by the SigmaStudio compiler
and does not need to be programmed manually.
Slew parameters can follow several different curves, including
an RC-type curve and a linear curve. These curve types are
coded into each algorithm and cannot be modified by the user.
Rev. 0 | Page 47 of 88
NEW TARGET
VALUE
SLEW
CURVE
INITIAL
VALUE
08914-070
When the values of signal processing parameters are changed
abruptly in real time, they sometimes cause pop and click
sounds to appear on the audio outputs. To avoid pops and
clicks, some algorithms in SigmaStudio implement a software
slew functionality. Algorithms using software slew set a target
value for a parameter and continuously update the value of that
parameter until it reaches the target.
Figure 70. Example of Volume Slew
ADAU1461
GENERAL-PURPOSE INPUT/OUTPUT
The serial data input/output pins (Pin 26 to Pin 29) are shared
with the general-purpose input/output function. Each of these
four pins can be set to only one of these functions. The function
of these pins is set in the serial data/GPIO pin configuration
register (Register R60, Address 0x40F4).
The GPIOx pins can be used as inputs or outputs. These pins
are readable and can be set through the control port or directly
by the SigmaDSP core. When configured as inputs, the GPIOx
pins can be used with push-button switches or rotary encoders
to control DSP program settings. These pins can also be used
with digital outputs to drive LEDs or external logic to indicate
the status of internal signals and control other devices. Examples
of this use include indicating signal overload, signal present,
and button press confirmation.
When configured as an output, each GPIO pin can typically
drive 2 mA, which is enough current to directly drive some
high efficiency LEDs. Standard LEDs require about 20 mA of
current and can be driven from a GPIO output with an external
transistor or buffer. Because of problems that can arise from
simultaneously driving or sinking a large amount of current on
many pins, avoid connecting high efficiency LEDs directly to
many or all of the GPIO pins when designing the application.
If many LEDs are required, use an external driver. When the
GPIO pins are configured as open-collector outputs, they
should be pulled up to a maximum voltage equal to the voltage
set on IOVDD.
The configuration of the GPIO functions is set up in the
GPIO pin control registers (Register R48 to Register R51,
Address 0x40C6 to Address 0x40C9).
GPIO PINS SET FROM THE CONTROL PORT
The GPIO pins can also be configured to be directly controlled
from the I2C/SPI control port. When the pins are set to this
mode, four memory locations are enabled for the GPIO pin
settings. The physical settings on the GPIO pins mirror the
settings of the LSB of these 4-byte-wide memory locations.
Table 31. GPIOx Pin Memory Settings (Set from Control Port)
Memory Location
Decimal
Hex
1568
0x0620
1569
0x0621
1570
0x0622
1571
0x0623
Rev. 0 | Page 48 of 88
Bits[31:1]
Reserved
Reserved
Reserved
Reserved
Bit 0
GPIO0SET
GPIO1SET
GPIO2SET
GPIO3SET
ADAU1461
CONTROL REGISTERS
Table 32. Register Map
Reg
R0
R1
Address
0x4000
0x4002
Name
Clock control
PLL control
Bit 7
Bit 6
Bit 5
Reserved
Reserved
R2
R3
R4
R5
R6
R7
R8
R9
R10
R11
R12
R13
R14
R15
R16
R17
R18
R19
R20
R21
R22
R23
R24
R25
R26
R27
R28
R29
R30
R31
R32
R33
R34
R35
R36
R37
R38
R39
R40
R41
R42
R67
R43
R44
R45
R46
0x4008
0x4009
0x400A
0x400B
0x400C
0x400D
0x400E
0x400F
0x4010
0x4011
0x4012
0x4013
0x4014
0x4015
0x4016
0x4017
0x4018
0x4019
0x401A
0x401B
0x401C
0x401D
0x401E
0x401F
0x4020
0x4021
0x4022
0x4023
0x4024
0x4025
0x4026
0x4027
0x4028
0x4029
0x402A
0x402B
0x402C
0x402D
0x402F
0x4030
0x4031
0x4036
0x40C0
0x40C1
0x40C2
0x40C3
Dig mic/jack detect
Reserved
Rec Mixer Left 0
Rec Mixer Left 1
Rec Mixer Right 0
Rec Mixer Right 1
Left diff input vol
Right diff input vol
Record mic bias
ALC 0
ALC 1
ALC 2
ALC 3
Serial Port 0
Serial Port 1
Converter 0
Converter 1
ADC control
Left digital vol
Right digital vol
Play Mixer Left 0
Play Mixer Left 1
Play Mixer Right 0
Play Mixer Right 1
Play L/R mixer left
Play L/R mixer right
Play L/R mixer mono
Play HP left vol
Play HP right vol
Line output left vol
Line output right vol
Play mono output
Pop/click suppress
Play power mgmt
DAC Control 0
DAC Control 1
DAC Control 2
Serial port pad
Control Port Pad 0
Control Port Pad 1
Jack detect pin
Dejitter control
Cyclic redundancy
check
JDDB[1:0]
Bit 4
Bit 3
CLKSRC
M[15:8]
M[7:0]
N[15:8]
N[7:0]
R[3:0]
Reserved
JDFUNC[1:0]
Bit 2
Bit 1
INFREQ[1:0]
Bit 0
COREN
X[1:0]
Lock
Reserved
Type
PLLEN
JDPOL
Reserved
Reserved
LINPG[2:0]
Reserved
Reserved
LINNG[2:0]
LDBOOST[1:0]
RINPG[2:0]
Reserved
MX1EN
MX1AUXG[2:0]
RINNG[2:0]
RDBOOST[1:0]
LDVOL[5:0]
RDVOL[5:0]
MX2EN
MX2AUXG[2:0]
LDMUTE
LDEN
RDMUTE
RDEN
MBI
Reserved
MBIEN
ALCSEL[2:0]
ALCTARG[3:0]
ALCDEC[3:0]
NGTHR[4:0]
CHPF[1:0]
MS
MSBP
LRDEL[1:0]
CONVSR[2:0]
ADPAIR[1:0]
INSEL
ADCEN[1:0]
Reserved
MPERF
PGASLEW[1:0]
ALCMAX[2:0]
ALCHOLD[3:0]
ALCATCK[3:0]
NGTYP[1:0]
NGEN
Reserved
SPSRS
LRMOD
BPOL
LRPOL
BPF[2:0]
ADTDM
DATDM
Reserved
DAPAIR[1:0]
DAOSR
ADOSR
Reserved
Reserved
ADCPOL
HPF
DMPOL
DMSW
LADVOL[7:0]
RADVOL[7:0]
Reserved
MX3RM
MX3LM
MX3AUXG[3:0]
MX3EN
MX3G2[3:0]
MX3G1[3:0]
Reserved
MX4RM
MX4LM
MX4AUXG[3:0]
MX4EN
MX4G2[3:0]
MX4G1[3:0]
Reserved
MX5G4[1:0]
MX5G3[1:0]
MX5EN
Reserved
MX6G4[1:0]
MX6G3[1:0]
MX6EN
Reserved
MX7[1:0]
MX7EN
LHPVOL[5:0]
LHPM
HPEN
RHPVOL[5:0]
RHPM
HPMODE
LOUTVOL[5:0]
LOUTM
LOMODE
ROUTVOL[5:0]
ROUTM
ROMODE
MONOVOL[5:0]
MONOM
MOMODE
Reserved
POPMODE POPLESS
ASLEW[1:0]
Reserved
Reserved
PREN
PLEN
DACMONO[1:0]
DACPOL
Reserved
DEMPH
DACEN[1:0]
LDAVOL[7:0]
RDAVOL[7:0]
ADCSDP[1:0]
DACSDP[1:0]
LRCLKP[1:0]
BCLKP[1:0]
CDATP[1:0]
CLCHP[1:0]
SCLP[1:0]
SDAP[1:0]
Reserved
SDASTR
Reserved
JDSTR
Reserved
JDP[1:0]
Reserved
DEJIT[7:0]
CRC[31:24]
CRC[23:16]
CRC[15:8]
CRC[7:0]
Rev. 0 | Page 49 of 88
Default
00000000
00000000
11111101
00000000
00001100
00010000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00010000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000010
00000010
00000010
00000010
00000010
00000000
00000000
00000000
00000000
00000000
10101010
10101010
00000000
00001000
00000011
00000000
00000000
00000000
00000000
ADAU1461
Reg
R47
R48
R49
R50
R51
R52
R53
R54
R55
R56
R57
Address
0x40C4
0x40C6
0x40C7
0x40C8
0x40C9
0x40D0
0x40D1
0x40D2
0x40D3
0x40D4
0x40EB
R58
0x40F2
R59
0x40F3
R60
0x40F4
R61
R62
R63
R64
0x40F5
0x40F6
0x40F7
0x40F8
R65
R66
0x40F9
0x40FA
Name
CRC enable
GPIO0 pin control
GPIO1 pin control
GPIO2 pin control
GPIO3 pin control
Watchdog enable
Watchdog value
Watchdog error
DSP sampling rate
setting
Serial input route
control
Serial output route
control
Serial data/GPIO
pin configuration
DSP enable
DSP run
DSP slew modes
Serial port
sampling rate
Clock Enable 0
Clock Enable 1
Bit 7
Bit 6
Bit 5
Bit 4
Reserved
Bit 3
Bit 2
Reserved
DSPSR[3:0]
Default
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000000
00000001
Reserved
SINRT[3:0]
00000000
Reserved
SOUTRT[3:0]
00000000
Reserved
Reserved
Reserved
Reserved
Bit 1
GPIO0[3:0]
GPIO1[3:0]
GPIO2[3:0]
GPIO3[3:0]
Reserved
DOG[23:16]
DOG[15:8]
DOG[7:0]
Reserved
Reserved
DOGEN
DOGER
LRGP3
Reserved
Reserved
MOSLW
Reserved
BGP2
SLEWPD
ALCPD
DECPD
Reserved
SDOGP1
ROSLW
LOSLW
RHPSLW
SPSR[2:0]
SOUTPD
INTPD
SINPD
CLK1
Reserved
Reserved
Bit 0
CRCEN
SDIGP0
00000000
DSPEN
DSPRUN
LHPSLW
00000000
00000000
00000000
00000000
SPPD
CLK0
00000000
00000000
CONTROL REGISTER DETAILS
All registers except for the PLL control register are 1-byte write and read registers.
R0: Clock Control, 16,384 (0x4000)
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Bit 3
CLKSRC
Bit 2
Bit 1
INFREQ[1:0]
Bit 0
COREN
Table 33. Clock Control Register
Bits
3
Bit Name
CLKSRC
[2:1]
INFREQ[1:0]
0
COREN
Description
Clock source select.
0 = direct from MCLK pin (default).
1 = PLL clock.
Input clock frequency. Sets the core clock rate that generates the core clock. If the PLL is used, this value is
automatically set to 1024 × fS.
Setting
Input Clock Frequency
00
256 × fS (default)
01
512 × fS
10
768 × fS
11
1024 × fS
Core clock enable. Only the R0 and R1 registers can be accessed when this bit is set to 0 (core clock disabled).
0 = core clock disabled (default).
1 = core clock enabled.
Rev. 0 | Page 50 of 88
ADAU1461
R1: PLL Control, 16,386 (0x4002)
Byte
0
1
2
3
4
5
Bit 7
Bit 6
Reserved
Bit 5
Bit 4
Bit 3
M[15:8]
M[7:0]
N[15:8]
N[7:0]
R[3:0]
Reserved
Bit 2
Bit 1
X[1:0]
Lock
Bit 0
Type
PLLEN
Table 34. PLL Control Register
Byte
0
1
Bits
[7:0]
[7:0]
Bit Name
M[15:8]
M[7:0]
2
3
[7:0]
[7:0]
N[15:8]
N[7:0]
4
[6:3]
R[3:0]
4
[2:1]
X[1:0]
4
0
Type
5
1
Lock
5
0
PLLEN
Description
PLL denominator MSB. This value is concatenated with M[7:0] to make up a 16-bit number.
PLL denominator LSB. This value is concatenated with M[15:8] to make up a 16-bit number.
M[15:8] (MSB)
M[7:0] (LSB)
Value of M
00000000
00000000
0
…
…
…
00000000
11111101
253 (default)
…
…
…
11111111
11111111
65,535
PLL numerator MSB. This value is concatenated with N[7:0] to make up a 16-bit number.
PLL numerator LSB. This value is concatenated with N[15:8] to make up a 16-bit number.
N[15:8] (MSB)
N[7:0] (LSB)
Value of N
00000000
00000000
0
…
…
…
00000000
00001100
12 (default)
…
…
…
11111111
11111111
65,535
PLL integer setting.
Setting
Value of R
0010
2 (default)
0011
3
0100
4
0101
5
0110
6
0111
7
1000
8
PLL input clock divider.
Setting
Value of X
00
1 (default)
01
2
10
3
11
4
Type of PLL. When set to integer mode, the values of M and N are ignored.
0 = integer (default).
1 = fractional.
PLL lock. This read-only bit is flagged when the PLL has finished locking.
0 = PLL unlocked (default).
1 = PLL locked.
PLL enable.
0 = PLL disabled (default).
1 = PLL enabled.
Rev. 0 | Page 51 of 88
ADAU1461
R2: Digital Microphone/Jack Detection Control, 16,392 (0x4008)
Bit 7
Bit 6
JDDB[1:0]
Bit 5
Bit 4
JDFUNC[1:0]
Bit 3
Bit 2
Reserved
Bit 1
Bit 0
JDPOL
Table 35. Digital Microphone/Jack Detection Control Register
Bits
[7:6]
Bit Name
JDDB[1:0]
[5:4]
JDFUNC[1:0]
0
JDPOL
Description
Jack detect debounce time.
Setting
Debounce Time
00
5 ms (default)
01
10 ms
10
20 ms
11
40 ms
JACKDET/MICIN pin function. Enables or disables the jack detect function or configures the pin for a digital
microphone input.
Setting
Pin Function
00
Jack detect off (default)
01
Jack detect on
10
Digital microphone input
11
Reserved
Jack detect polarity. Detects high or low signal.
0 = detect high signal (default).
1 = detect low signal.
R4: Record Mixer Left (Mixer 1) Control 0, 16,394 (0x400A)
This register controls the gain of single-ended inputs for the left channel record path. The left channel record mixer is referred to as Mixer 1.
Bit 7
Reserved
Bit 6
Bit 5
LINPG[2:0]
Bit 4
Bit 3
Bit 2
LINNG[2:0]
Table 36. Record Mixer Left (Mixer 1) Control 0 Register
Bits
[6:4]
Bit Name
LINPG[2:0]
[3:1]
LINNG[2:0]
0
MX1EN
Description
Gain for a left channel single-ended input from the LINP pin, input to Mixer 1.
Setting
Gain
000
Mute (default)
001
−12 dB
010
−9 dB
011
−6 dB
100
−3 dB
101
0 dB
110
3 dB
111
6 dB
Gain for a left channel single-ended input from the LINN pin, input to Mixer 1.
Setting
Gain
000
Mute (default)
001
−12 dB
010
−9 dB
011
−6 dB
100
−3 dB
101
0 dB
110
3 dB
111
6 dB
Left channel mixer enable in the record path. Referred to as Mixer 1.
0 = mixer disabled (default).
1 = mixer enabled.
Rev. 0 | Page 52 of 88
Bit 1
Bit 0
MX1EN
ADAU1461
R5: Record Mixer Left (Mixer 1) Control 1, 16,395 (0x400B)
This register controls the gain boost of the left channel differential PGA input and the gain for the left channel auxiliary input in the
record path. The left channel record mixer is referred to as Mixer 1.
Bit 7
Bit 6
Reserved
Bit 5
Bit 4
Bit 3
LDBOOST[1:0]
Bit 2
Bit 1
MX1AUXG[2:0]
Bit 0
Table 37. Record Mixer Left (Mixer 1) Control 1 Register
Bits
[4:3]
Bit Name
LDBOOST[1:0]
[2:0]
MX1AUXG[2:0]
Description
Left channel differential PGA input gain boost, input to Mixer 1. The left differential input uses the LINP (positive
signal) and LINN (negative signal) pins.
Setting
Gain Boost
00
Mute (default)
01
0 dB
10
20 dB
11
Reserved
Left single-ended auxiliary input gain from the LAUX pin in the record path, input to Mixer 1.
Setting
Auxiliary Input Gain
000
Mute (default)
001
−12 dB
010
−9 dB
011
−6 dB
100
−3 dB
101
0 dB
110
3 dB
111
6 dB
Rev. 0 | Page 53 of 88
ADAU1461
R6: Record Mixer Right (Mixer 2) Control 0, 16,396 (0x400C)
This register controls the gain of single-ended inputs for the right channel record path. The right channel record mixer is referred to as
Mixer 2.
Bit 7
Reserved
Bit 6
Bit 5
RINPG[2:0]
Bit 4
Bit 3
Bit 2
RINNG[2:0]
Bit 1
Table 38. Record Mixer Right (Mixer 2) Control 0 Register
Bits
[6:4]
Bit Name
RINPG[2:0]
[3:1]
RINNG[2:0]
0
MX2EN
Description
Gain for a right channel single-ended input from the RINP pin, input to Mixer 2.
Setting
Gain
000
Mute (default)
001
−12 dB
010
−9 dB
011
−6 dB
100
−3 dB
101
0 dB
110
3 dB
111
6 dB
Gain for a right channel single-ended input from the RINN pin, input to Mixer 2.
Setting
Gain
000
Mute (default)
001
−12 dB
010
−9 dB
011
−6 dB
100
−3 dB
101
0 dB
110
3 dB
111
6 dB
Right channel mixer enable in the record path. Referred to as Mixer 2.
0 = mixer disabled (default).
1 = mixer enabled.
Rev. 0 | Page 54 of 88
Bit 0
MX2EN
ADAU1461
R7: Record Mixer Right (Mixer 2) Control 1, 16,397 (0x400D)
This register controls the gain boost of the right channel differential PGA input and the gain for the right channel auxiliary input in the
record path. The right channel record mixer is referred to as Mixer 2.
Bit 7
Bit 6
Reserved
Bit 5
Bit 4
Bit 3
RDBOOST[1:0]
Bit 2
Bit 1
MX2AUXG[2:0]
Bit 0
Table 39. Record Mixer Right (Mixer 2) Control 1 Register
Bits
[4:3]
Bit Name
RDBOOST[1:0]
[2:0]
MX2AUXG[2:0]
Description
Right channel differential PGA input gain boost, input to Mixer 2. The right differential input uses the RINP
(positive signal) and RINN (negative signal) pins.
Setting
Gain Boost
00
Mute (default)
01
0 dB
10
20 dB
11
Reserved
Right single-ended auxiliary input gain from the RAUX pin in the record path, input to Mixer 2.
Setting
Auxiliary Input Gain
000
Mute (default)
001
−12 dB
010
−9 dB
011
−6 dB
100
−3 dB
101
0 dB
110
3 dB
111
6 dB
R8: Left Differential Input Volume Control, 16,398 (0x400E)
This register enables the differential path and sets the volume control for the left differential PGA input.
Bit 7
Bit 6
Bit 5
Bit 4
LDVOL[5:0]
Bit 3
Bit 2
Bit 1
LDMUTE
Bit 0
LDEN
Table 40. Left Differential Input Volume Control Register
Bits
[7:2]
Bit Name
LDVOL[5:0]
1
LDMUTE
0
LDEN
Description
Left channel differential PGA input volume control. The left differential input uses the LINP (positive signal) and
LINN (negative signal) pins. Each step corresponds to a 0.75 dB increase in gain. See Table 90 for a complete list
of the volume settings.
Setting
Volume
000000
−12 dB (default)
000001
−11.25 dB
…
…
010000
0 dB
…
…
111110
34.5 dB
111111
35.25 dB
Left differential input mute control.
0 = mute (default).
1 = unmute.
Left differential PGA enable. When enabled, the LINP and LINN pins are used as a full differential pair. When
disabled, these two pins are configured as two single-ended inputs with the signals routed around the PGA.
0 = disabled (default).
1 = enabled.
Rev. 0 | Page 55 of 88
ADAU1461
R9: Right Differential Input Volume Control, 16,399 (0x400F)
This register enables the differential path and sets the volume control for the right differential PGA input.
Bit 7
Bit 6
Bit 5
Bit 4
RDVOL[5:0]
Bit 3
Bit 2
Bit 1
RDMUTE
Bit 0
RDEN
Table 41. Right Differential Input Volume Control Register
Bits
[7:2]
Bit Name
RDVOL[5:0]
1
RDMUTE
0
RDEN
Description
Right channel differential PGA input volume control. The right differential input uses the RINP (positive signal)
and RINN (negative signal) pins. Each step corresponds to a 0.75 dB increase in gain. See Table 90 for a complete
list of the volume settings.
Setting
Volume
000000
−12 dB (default)
000001
−11.25 dB
…
…
010000
0 dB
…
…
111110
34.5 dB
111111
35.25 dB
Right differential input mute control.
0 = mute (default).
1 = unmute.
Right differential PGA enable. When enabled, the RINP and RINN pins are used as a full differential pair. When
disabled, these two pins are configured as two single-ended inputs with the signals routed around the PGA.
0 = disabled (default).
1 = enabled.
R10: Record Microphone Bias Control, 16,400 (0x4010)
This register controls the MICBIAS pin settings for biasing electret type analog microphones.
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Bit 3
MPERF
Bit 2
MBI
Bit 1
Reserved
Bit 0
MBIEN
Table 42. Record Microphone Bias Control Register
Bits
3
Bit Name
MPERF
2
MBI
0
MBIEN
Description
Microphone bias is enabled for high performance or normal operation. High performance operation sources
more current to the microphone.
0 = normal operation (default).
1 = high performance.
Microphone voltage bias as a fraction of AVDD.
0 = 0.90 × AVDD (default).
1 = 0.65 × AVDD.
Enables the MICBIAS output.
0 = disabled (default).
1 = enabled.
Rev. 0 | Page 56 of 88
ADAU1461
R11: ALC Control 0, 16,401 (0x4011)
Bit 7
Bit 6
PGASLEW[1:0]
Bit 5
Bit 4
ALCMAX[2:0]
Bit 3
Bit 2
Bit 1
ALCSEL[2:0]
Bit 0
Table 43. ALC Control 0 Register
Bits
[7:6]
Bit Name
PGASLEW[1:0]
[5:3]
ALCMAX[2:0]
[2:0]
ALCSEL[2:0]
Description
PGA volume slew time when the ALC is off. The slew time is the period of time that a volume increase or decrease
takes to ramp up or ramp down to the target volume set in Register R8 (left differential input volume control)
and Register R9 (right differential input volume control).
Setting
Slew Time
00
24 ms (default)
01
48 ms
10
96 ms
11
Off
The maximum ALC gain sets a limit to the amount of gain that the ALC can provide to the input signal. This
protects small signals from excessive amplification.
Setting
Maximum ALC Gain
000
−12 dB (default)
001
−6 dB
010
0 dB
011
6 dB
100
12 dB
101
18 dB
110
24 dB
111
30 dB
ALC select. These bits set the channels that are controlled by the ALC. When set to right only, the ALC responds
only to the right channel input and controls the gain of the right PGA amplifier only. When set to left only, the
ALC responds only to the left channel input and controls the gain of the left PGA amplifier only. When set to
stereo, the ALC responds to the greater of the left or right channel and controls the gain of both the left and
right PGA amplifiers. DSP control allows the PGA gain to be set within the DSP or from external GPIO inputs.
These bits must be off if manual control of the volume is desired.
Setting
Channels
000
Off (default)
001
Right only
010
Left only
011
Stereo
100
DSP control
101
Reserved
110
Reserved
111
Reserved
Rev. 0 | Page 57 of 88
ADAU1461
R12: ALC Control 1, 16,402 (0x4012)
Bit 7
Bit 6
Bit 5
ALCHOLD[3:0]
Bit 4
Bit 3
Bit 2
Bit 1
ALCTARG[3:0]
Bit 0
Table 44. ALC Control 1 Register
Bits
[7:4]
Bit Name
ALCHOLD[3:0]
[3:0]
ALCTARG[3:0]
Description
ALC hold time. The ALC hold time is the amount of time that the ALC waits after a decrease in input level before
increasing the gain to achieve the target level. The recommended minimum setting is 21 ms (0011) to prevent
distortion of low frequency signals. The hold time doubles with every 1-bit increase.
Setting
Hold Time
0000
2.67 ms (default)
0001
5.34 ms
0010
10.68 ms
0011
21.36 ms
0100
42.72 ms
0101
85.44 ms
0110
170.88 ms
0111
341.76 ms
1000
683.52 ms
1001
1.367 sec
1010
2.7341 sec
1011
5.4682 sec
1100
10.936 sec
1101
21.873 sec
1110
43.745 sec
1111
87.491 sec
ALC target. The ALC target sets the desired ADC input level. The PGA gain is adjusted by the ALC to reach this
target level. The recommended target level is between −16 dB and −10 dB to accommodate transients without
clipping the ADC.
Setting
ALC Target
0000
−28.5 dB (default)
0001
−27 dB
0010
−25.5 dB
0011
−24 dB
0100
−22.5 dB
0101
−21 dB
−19.5 dB
0110
0111
−18 dB
1000
−16.5 dB
1001
−15 dB
1010
−13.5 dB
1011
−12 dB
1100
−10.5 dB
1101
−9 dB
1110
−7.5 dB
1111
−6 dB
Rev. 0 | Page 58 of 88
ADAU1461
R13: ALC Control 2, 16,403 (0x4013)
Bit 7
Bit 6
Bit 5
ALCATCK[3:0]
Bit 4
Bit 3
Bit 2
Bit 1
ALCDEC[3:0]
Bit 0
Table 45. ALC Control 2 Register
Bits
[7:4]
Bit Name
ALCATCK[3:0]
[3:0]
ALCDEC[3:0]
Description
ALC attack time. The attack time sets how fast the ALC starts attenuating after an increase in input level above
the target. A typical setting for music recording is 384 ms, and a typical setting for voice recording is 24 ms.
Setting
Attack Time
0000
6 ms (default)
0001
12 ms
0010
24 ms
0011
48 ms
0100
96 ms
0101
192 ms
0110
384 ms
0111
768 ms
1000
1.54 sec
1001
3.07 sec
1010
6.14 sec
1011
12.29 sec
1100
24.58 sec
1101
49.15 sec
1110
98.30 sec
1111
196.61 sec
ALC decay time. The decay time sets how fast the ALC increases the PGA gain after a decrease in input level
below the target. A typical setting for music recording is 24.58 seconds, and a typical setting for voice recording
is 1.54 seconds.
Setting
Decay Time
0000
24 ms
0001
48 ms
0010
96 ms
0011
192 ms
0100
384 ms
0101
768 ms
0110
1.54 sec
0111
3.07 sec
1000
6.14 sec
1001
12.29 sec
1010
24.58 sec
1011
49.15 sec
1100
98.30 sec
1101
196.61 sec
1110
393.22 sec
1111
786.43 sec
Rev. 0 | Page 59 of 88
ADAU1461
R14: ALC Control 3, 16,404 (0x4014)
Bit 7
Bit 6
NGTYP[1:0]
Bit 5
NGEN
Bit 4
Bit 3
Bit 2
NGTHR[4:0]
Bit 1
Bit 0
Table 46. ALC Control 3 Register
Bits
[7:6]
Bit Name
NGTYP[1:0]
5
NGEN
[4:0]
NGTHR[4:0]
Description
Noise gate type. When the input signal falls below the threshold for 250 ms, the noise gate can hold a constant
PGA gain, mute the ADC output, fade the PGA gain to the minimum gain value, or fade then mute.
Setting
Noise Gate
00
Hold PGA constant (default)
01
Mute ADC output (digital mute)
10
Fade to PGA minimum value (analog fade)
11
Fade then mute (analog fade/digital mute)
Noise gate enable.
0 = disabled (default).
1 = enabled.
Noise gate threshold. When the input signal falls below the threshold for 250 ms, the noise gate is activated.
A 1 LSB increase corresponds to a −1.5 dB change. See Table 91 for a complete list of the threshold settings.
Setting
Threshold
00000
−76.5 dB (default)
00001
−75 dB
…
…
11110
−31.5 dB
11111
−30 dB
R15: Serial Port Control 0, 16,405 (0x4015)
Bit 7
Reserved
Bit 6
SPSRS
Bit 5
LRMOD
Bit 4
BPOL
Bit 3
LRPOL
Bit 2
Bit 1
CHPF[1:0]
Bit 0
MS
Table 47. Serial Port Control 0 Register
Bits
6
Bit Name
SPSRS
5
LRMOD
4
BPOL
3
LRPOL
[2:1]
CHPF[1:0]
0
MS
Description
Serial port sampling rate source.
0 = converter rate set in Register R17 (default).
1 = DSP rate set in Register R57.
LRCLK mode sets the LRCLK for either a 50% duty cycle or a pulse. The pulse mode should be at least 1 BCLK wide.
0 = 50% duty cycle (default).
1 = pulse mode.
BCLK polarity sets the BCLK edge that triggers a change in audio data. This can be set for the falling or rising
edge of the BCLK.
0 = falling edge (default).
1 = rising edge.
LRCLK polarity sets the LRCLK edge that triggers the beginning of the left channel audio frame. This can be set
for the falling or rising edge of the LRCLK.
0 = falling edge (default).
1 = rising edge.
Channels per frame sets the number of channels per LRCLK frame.
Setting
Channels per LRCLK Frame
00
Stereo (default)
01
TDM 4
10
TDM 8
11
Reserved
Serial data port bus mode. Both LRCLK and BCLK are master of the serial port when set in master mode and are
serial port slave in slave mode.
0 = slave mode (default).
1 = master mode.
Rev. 0 | Page 60 of 88
ADAU1461
R16: Serial Port Control 1, 16,406 (0x4016)
Bit 7
Bit 6
BPF[2:0]
Bit 5
Bit 4
ADTDM
Bit 3
DATDM
Table 48. Serial Port Control 1 Register
Bits
[7:5]
Bit Name
BPF[2:0]
4
ADTDM
3
DATDM
2
MSBP
[1:0]
LRDEL[1:0]
Description
Number of bit clock cycles per LRCLK audio frame.
Setting
Bit Clock Cycles
000
64 (default)
001
Reserved
010
48
011
128
100
256
Reserved
101
110
Reserved
111
Reserved
ADC serial audio data channel position in TDM mode.
0 = left first (default).
1 = right first.
DAC serial audio data channel position in TDM mode.
0 = left first (default).
1 = right first.
MSB position in the LRCLK frame.
0 = MSB first (default).
1 = LSB first.
Data delay from LRCLK edge (in BCLK units).
Setting
Delay (Bit Clock Cycles)
00
1 (default)
01
0
10
8
11
16
Rev. 0 | Page 61 of 88
Bit 2
MSBP
Bit 1
Bit 0
LRDEL[1:0]
ADAU1461
R17: Converter Control 0, 16,407 (0x4017)
Bit 7
Reserved
Bit 6
Bit 5
DAPAIR[1:0]
Bit 4
DAOSR
Bit 3
ADOSR
Bit 2
Bit 1
CONVSR[2:0]
Bit 0
Table 49. Converter Control 0 Register
Bits
[6:5]
Bit Name
DAPAIR[1:0]
4
DAOSR
3
ADOSR
[2:0]
CONVSR[2:0]
Description
On-chip DAC serial data selection in TDM 4 or TDM 8 mode.
Setting
Pair
00
First pair (default)
01
Second pair
10
Third pair
11
Fourth pair
DAC oversampling ratio. This bit cannot be set for 64× when CONVSR[2:0] is set to 96 kHz.
0 = 128× (default).
1 = 64×.
ADC oversampling ratio. This bit cannot be set for 64× when CONVSR[2:0] is set to 96 kHz.
0 = 128× (default).
1 = 64×.
Converter sampling rate. The ADCs and DACs operate at the sampling rate set in this register. The converter rate
selected is a ratio of the base sampling rate, fS. The base sampling rate is determined by the operating frequency
of the core clock.
Setting
Sampling Rate
Base Sampling Rate (fS = 48 kHz)
000
fS
48 kHz, base (default)
001
fS/6
8 kHz
010
fS/4
12 kHz
011
fS/3
16 kHz
100
fS/2
24 kHz
101
fS/1.5
32 kHz
110
fS/0.5
96 kHz
111
Reserved
R18: Converter Control 1, 16,408 (0x4018)
Bit 7
Bit 6
Bit 5
Bit 4
Reserved
Bit 3
Table 50. Converter Control 1 Register
Bits
[1:0]
Bit Name
ADPAIR[1:0]
Description
On-chip ADC serial data selection in TDM 4 or TDM 8 mode.
Setting
Pair
00
First pair (default)
01
Second pair
10
Third pair
11
Fourth pair
Rev. 0 | Page 62 of 88
Bit 2
Bit 1
Bit 0
ADPAIR[1:0]
ADAU1461
R19: ADC Control, 16,409 (0x4019)
Bit 7
Reserved
Bit 6
ADCPOL
Bit 5
HPF
Bit 4
DMPOL
Bit 3
DMSW
Bit 2
INSEL
Bit 1
Bit 0
ADCEN[1:0]
Table 51. ADC Control Register
Bits
6
Bit Name
ADCPOL
5
HPF
4
DMPOL
3
DMSW
2
INSEL
[1:0]
ADCEN[1:0]
Description
Invert input polarity.
0 = normal (default).
1 = inverted.
ADC high-pass filter select. At 48 kHz, f3dB = 2 Hz.
0 = off (default).
1 = on.
Digital microphone data polarity swap.
0 = invert polarity.
1 = normal (default).
Digital microphone channel swap. Normal operation sends the left channel on the rising edge of the clock and
the right channel on the falling edge of the clock.
0 = normal (default).
1 = swap left and right channels.
Digital microphone input select. When asserted, the on-chip ADCs are off, BCLK is master at 128 × fS, and
ADC_SDATA is expected to have left and right channels interleaved.
0 = digital microphone inputs off, ADCs enabled (default).
1 = digital microphone inputs enabled, ADCs off.
ADC enable.
Setting
ADCs Enabled
00
Both off (default)
01
Left on
10
Right on
11
Both on
R20: Left Input Digital Volume, 16,410 (0x401A)
Bit 7
Bit 6
Bit 5
Bit 4
Bit 3
LADVOL[7:0]
Bit 2
Bit 1
Bit 0
Table 52. Left Input Digital Volume Register
Bits
[7:0]
Bit Name
LADVOL[7:0]
Description
Controls the digital volume attenuation for left channel inputs from either the left ADC or the left digital
microphone input. Each bit corresponds to a 0.375 dB step with slewing between settings. See Table 92 for a
complete list of the volume settings.
Setting
Volume Attenuation
00000000
0 dB (default)
00000001
−0.375 dB
00000010
−0.75 dB
…
…
11111110
−95.25 dB
11111111
−95.625 dB
Rev. 0 | Page 63 of 88
ADAU1461
R21: Right Input Digital Volume, 16,411 (0x401B)
Bit 7
Bit 6
Bit 5
Bit 4
Bit 3
RADVOL[7:0]
Bit 2
Bit 1
Bit 0
Table 53. Right Input Digital Volume Register
Bits
[7:0]
Bit Name
RADVOL[7:0]
Description
Controls the digital volume attenuation for right channel inputs from either the right ADC or the right digital
microphone input. Each bit corresponds to a 0.375 dB step with slewing between settings. See Table 92 for a
complete list of the volume settings.
Setting
Volume Attenuation
00000000
0 dB (default)
00000001
−0.375 dB
00000010
−0.75 dB
…
…
11111110
−95.25 dB
11111111
−95.625 dB
R22: Playback Mixer Left (Mixer 3) Control 0, 16,412 (0x401C)
Bit 7
Reserved
Bit 6
MX3RM
Bit 5
MX3LM
Bit 4
Bit 3
Bit 2
MX3AUXG[3:0]
Bit 1
Bit 0
MX3EN
Table 54. Playback Mixer Left (Mixer 3) Control 0 Register
Bits
6
Bit Name
MX3RM
5
MX3LM
[4:1]
MX3AUXG[3:0]
0
MX3EN
Description
Mixer input mute. Mutes the right DAC input to the left channel playback mixer (Mixer 3).
0 = muted (default).
1 = unmuted.
Mixer input mute. Mutes the left DAC input to the left channel playback mixer (Mixer 3).
0 = muted (default).
1 = unmuted.
Mixer input gain. Controls the left channel auxiliary input gain to the left channel playback mixer (Mixer 3).
Setting
Gain
0000
Mute (default)
0001
−15 dB
0010
−12 dB
0011
−9 dB
0100
−6 dB
0101
−3 dB
0110
0 dB
0111
3 dB
1000
6 dB
Mixer 3 enable.
0 = disabled (default).
1 = enabled.
Rev. 0 | Page 64 of 88
ADAU1461
R23: Playback Mixer Left (Mixer 3) Control 1, 16,413 (0x401D)
Bit 7
Bit 6
Bit 5
MX3G2[3:0]
Bit 4
Bit 3
Bit 2
Bit 1
MX3G1[3:0]
Bit 0
Table 55. Playback Mixer Left (Mixer 3) Control 1 Register
Bits
[7:4]
Bit Name
MX3G2[3:0]
[3:0]
MX3G1[3:0]
Description
Bypass gain control. The signal from the right channel record mixer (Mixer 2) bypasses the converters and gain
can be applied before the left playback mixer (Mixer 3).
Setting
Gain
0000
Mute (default)
0001
−15 dB
0010
−12 dB
0011
−9 dB
0100
−6 dB
0101
−3 dB
0110
0 dB
0111
3 dB
1000
6 dB
Bypass gain control. The signal from the left channel record mixer (Mixer 1) bypasses the converters and gain
can be applied before the left playback mixer (Mixer 3).
Setting
Gain
0000
Mute (default)
0001
−15 dB
0010
−12 dB
0011
−9 dB
0100
−6 dB
0101
−3 dB
0110
0 dB
0111
3 dB
1000
6 dB
Rev. 0 | Page 65 of 88
ADAU1461
R24: Playback Mixer Right (Mixer 4) Control 0, 16,414 (0x401E)
Bit 7
Reserved
Bit 6
MX4RM
Bit 5
MX4LM
Bit 4
Bit 3
Bit 2
MX4AUXG[3:0]
Bit 1
Bit 0
MX4EN
Table 56. Playback Mixer Right (Mixer 4) Control 0 Register
Bits
6
Bit Name
MX4RM
5
MX4LM
[4:1]
MX4AUXG[3:0]
0
MX4EN
Description
Mixer input mute. Mutes the right DAC input to the right channel playback mixer (Mixer 4).
0 = muted (default).
1 = unmuted.
Mixer input mute. Mutes the left DAC input to the right channel playback mixer (Mixer 4).
0 = muted (default).
1 = unmuted.
Mixer input gain. Controls the right channel auxiliary input gain to the right channel playback mixer (Mixer 4).
Setting
Gain
0000
Mute (default)
0001
−15 dB
0010
−12 dB
0011
−9 dB
0100
−6 dB
0101
−3 dB
0110
0 dB
0111
3 dB
1000
6 dB
Mixer 4 enable.
0 = disabled (default).
1 = enabled.
Rev. 0 | Page 66 of 88
ADAU1461
R25: Playback Mixer Right (Mixer 4) Control 1, 16,415 (0x401F)
Bit 7
Bit 6
Bit 5
MX4G2[3:0]
Bit 4
Bit 3
Bit 2
Bit 1
MX4G1[3:0]
Bit 0
Table 57. Playback Mixer Right (Mixer 4) Control 1 Register
Bits
[7:4]
Bit Name
MX4G2[3:0]
[3:0]
MX4G1[3:0]
Description
Bypass gain control. The signal from the right channel record mixer (Mixer 2) bypasses the converters and gain
can be applied before the right playback mixer (Mixer 4).
Setting
Gain
0000
Mute (default)
0001
−15 dB
0010
−12 dB
0011
−9 dB
0100
−6 dB
0101
−3 dB
0110
0 dB
0111
3 dB
1000
6 dB
Bypass gain control. The signal from the left channel record mixer (Mixer 1) bypasses the converters and gain
can be applied before the right playback mixer (Mixer 4).
Setting
Gain
0000
Mute (default)
0001
−15 dB
0010
−12 dB
0011
−9 dB
0100
−6 dB
0101
−3 dB
0110
0 dB
0111
3 dB
1000
6 dB
R26: Playback L/R Mixer Left (Mixer 5) Line Output Control, 16,416 (0x4020)
Bit 7
Bit 6
Reserved
Bit 5
Bit 4
Bit 3
MX5G4[1:0]
Bit 2
Bit 1
MX5G3[1:0]
Bit 0
MX5EN
Table 58. Playback L/R Mixer Left (Mixer 5) Line Output Control Register
Bits
[4:3]
Bit Name
MX5G4[1:0]
[2:1]
MX5G3[1:0]
0
MX5EN
Description
Mixer input gain boost. The signal from the right channel playback mixer (Mixer 4) can be enabled and boosted
in the playback L/R mixer left (Mixer 5).
Setting
Gain Boost
00
Mute (default)
01
0 dB output (−6 dB gain on each of the two inputs)
10
6 dB output (0 dB gain on each of the two inputs)
11
Reserved
Mixer input gain boost. The signal from the left channel playback mixer (Mixer 3) can be enabled and boosted in
the playback L/R mixer left (Mixer 5).
Setting
Gain Boost
00
Mute (default)
01
0 dB output (−6 dB gain on each of the two inputs)
10
6 dB output (0 dB gain on each of the two inputs)
11
Reserved
Mixer 5 enable.
0 = disabled (default).
1 = enabled.
Rev. 0 | Page 67 of 88
ADAU1461
R27: Playback L/R Mixer Right (Mixer 6) Line Output Control, 16,417 (0x4021)
Bit 7
Bit 6
Reserved
Bit 5
Bit 4
Bit 3
MX6G4[1:0]
Bit 2
Bit 1
MX6G3[1:0]
Bit 0
MX6EN
Table 59. Playback L/R Mixer Right (Mixer 6) Line Output Control Register
Bits
[4:3]
Bit Name
MX6G4[1:0]
[2:1]
MX6G3[1:0]
0
MX6EN
Description
Mixer input gain boost. The signal from the right channel playback mixer (Mixer 4) can be enabled and boosted
in the playback L/R mixer right (Mixer 6).
Setting
Gain Boost
00
Mute (default)
01
0 dB output (−6 dB gain on each of the two inputs)
10
6 dB output (0 dB gain on each of the two inputs)
11
Reserved
Mixer input gain boost. The signal from the left channel playback mixer (Mixer 3) can be enabled and boosted in
the playback L/R mixer right (Mixer 6).
Setting
Gain Boost
00
Mute (default)
01
0 dB output (−6 dB gain on each of the two inputs)
10
6 dB output (0 dB gain on each of the two inputs)
11
Reserved
Mixer 6 enable.
0 = disabled (default).
1 = enabled.
R28: Playback L/R Mixer Mono Output (Mixer 7) Control, 16,418 (0x4022)
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Bit 3
Bit 2
Bit 1
MX7[1:0]
Bit 0
MX7EN
Table 60. Playback L/R Mixer Mono Output (Mixer 7) Control Register
Bits
[2:1]
Bit Name
MX7[1:0]
0
MX7EN
Description
L/R mono playback mixer (Mixer 7). Mixes the left and right playback mixers (Mixer 3 and Mixer 4) with either a
0 dB or 6 dB gain boost. Additionally, this mixer can operate as a common-mode output, which is used as the
virtual ground in a capless headphone configuration.
Setting
Gain Boost
00
Common-mode output (default)
01
0 dB output (−6 dB gain on each of the two inputs)
10
6 dB output (0 dB gain on each of the two inputs)
11
Reserved
Mixer 7 enable.
0 = disabled (default).
1 = enabled.
Rev. 0 | Page 68 of 88
ADAU1461
R29: Playback Headphone Left Volume Control, 16,419 (0x4023)
Bit 7
Bit 6
Bit 5
Bit 4
LHPVOL[5:0]
Bit 3
Bit 2
Bit 1
LHPM
Bit 0
HPEN
Table 61. Playback Headphone Left Volume Control Register
Bits
[7:2]
Bit Name
LHPVOL[5:0]
1
LHPM
0
HPEN
Description
Headphone volume control for left channel, LHP output. Each 1-bit step corresponds to a 1 dB increase in volume.
See Table 93 for a complete list of the volume settings.
Setting
Volume
000000
−57 dB (default)
…
…
111001
0 dB
…
…
111111
6 dB
Headphone mute for left channel, LHP output (active low).
0 = mute.
1 = unmute (default).
Headphone volume control enable. Logical OR with the HPMODE bit in Register R30. If either the HPEN bit or
the HPMODE bit is set to 1, the headphone output is enabled.
0 = disabled (default).
1 = enabled.
R30: Playback Headphone Right Volume Control, 16,420 (0x4024)
Bit 7
Bit 6
Bit 5
Bit 4
RHPVOL[5:0]
Bit 3
Bit 2
Bit 1
RHPM
Bit 0
HPMODE
Table 62. Playback Headphone Right Volume Control Register
Bits
[7:2]
Bit Name
RHPVOL[5:0]
1
RHPM
0
HPMODE
Description
Headphone volume control for right channel, RHP output. Each 1-bit step corresponds to a 1 dB increase in
volume. See Table 93 for a complete list of the volume settings.
Setting
Volume
000000
−57 dB (default)
…
…
111001
0 dB
…
…
111111
6 dB
Headphone mute for right channel, RHP output (active low).
0 = mute.
1 = unmute (default).
RHP and LHP output mode. These pins can be configured for either line outputs or headphone outputs. Logical
OR with the HPEN bit in Register R29. If either the HPMODE bit or the HPEN bit is set to 1, the headphone output
is enabled.
0 = enable line output (default).
1 = enable headphone output.
Rev. 0 | Page 69 of 88
ADAU1461
R31: Playback Line Output Left Volume Control, 16,421 (0x4025)
Bit 7
Bit 6
Bit 5
Bit 4
LOUTVOL[5:0]
Bit 3
Bit 2
Bit 1
LOUTM
Bit 0
LOMODE
Table 63. Playback Line Output Left Volume Control Register
Bits
[7:2]
Bit Name
LOUTVOL[5:0]
1
LOUTM
0
LOMODE
Description
Line output volume control for left channel, LOUTN and LOUTP outputs. Each 1-bit step corresponds to a 1 dB
increase in volume. See Table 93 for a complete list of the volume settings.
Setting
Volume
000000
−57 dB (default)
…
…
111001
0 dB
…
…
111111
6 dB
Line output mute for left channel, LOUTN and LOUTP outputs (active low).
0 = mute.
1 = unmute (default).
Line output mode for left channel, LOUTN and LOUTP outputs. These pins can be configured for either line
outputs or headphone outputs. To drive earpiece speakers, set this bit to 1 (headphone output).
0 = line output (default).
1 = headphone output.
R32: Playback Line Output Right Volume Control, 16,422 (0x4026)
Bit 7
Bit 6
Bit 5
Bit 4
ROUTVOL[5:0]
Bit 3
Bit 2
Bit 1
ROUTM
Bit 0
ROMODE
Table 64. Playback Line Output Right Volume Control Register
Bits
[7:2]
Bit Name
ROUTVOL[5:0]
1
ROUTM
0
ROMODE
Description
Line output volume control for right channel, ROUTN and ROUTP outputs. Each 1-bit step corresponds to a 1 dB
increase in volume. See Table 93 for a complete list of the volume settings.
Setting
Volume
000000
−57 dB (default)
…
…
111001
0 dB
…
…
111111
6 dB
Line output mute for right channel, ROUTN and ROUTP outputs (active low).
0 = mute.
1 = unmute (default).
Line output mode for right channel, ROUTN and ROUTP outputs. These pins can be configured for either line
outputs or headphone outputs. To drive earpiece speakers, set this bit to 1 (headphone output).
0 = line output (default).
1 = headphone output.
Rev. 0 | Page 70 of 88
ADAU1461
R33: Playback Mono Output Control, 16,423 (0x4027)
Bit 7
Bit 6
Bit 5
Bit 4
MONOVOL[5:0]
Bit 3
Bit 2
Bit 1
MONOM
Bit 0
MOMODE
Table 65. Playback Mono Output Control Register
Bits
[7:2]
Bit Name
MONOVOL[5:0]
1
MONOM
0
MOMODE
Description
Mono output volume control. Each 1-bit step corresponds to a 1 dB increase in volume. If MX7[1:0] in Register R28
is set for common-mode output, volume control is disabled. See Table 93 for a complete list of the volume settings.
Setting
Volume
000000
−57 dB (default)
…
…
111001
0 dB
…
…
111111
6 dB
Mono output mute (active low).
0 = mute.
1 = unmute (default).
Headphone mode enable. If MX7[1:0] in Register R28 is set for common-mode output for a capless headphone
configuration, this bit should be set to 1 ( headphone output).
0 = line output (default).
1 = headphone output.
R34: Playback Pop/Click Suppression, 16,424 (0x4028)
Bit 7
Bit 6
Reserved
Bit 5
Bit 4
POPMODE
Bit 3
POPLESS
Bit 2
Bit 1
ASLEW[1:0]
Bit 0
Reserved
Table 66. Playback Pop/Click Suppression Register
Bits
4
Bit Name
POPMODE
3
POPLESS
[2:1]
ASLEW[1:0]
Description
Pop suppression circuit power saving mode. The pop suppression circuits charge faster in normal operation;
however, after they are charged, they can be put into low power operation.
0 = normal (default).
1 = low power.
Pop suppression disable. The pop suppression circuits are enabled by default. They can be disabled to save
power; however, disabling the circuits increases the risk of pops and clicks.
0 = enabled (default).
1 = disabled.
Analog volume slew rate for playback volume controls.
Setting
Slew Rate
00
21.25 ms (default)
01
42.5 ms
10
85 ms
11
Off
R35: Playback Power Management, 16,425 (0x4029)
Bit 7
Bit 6
Bit 5
Bit 4
Reserved
Bit 3
Table 67. Playback Power Management Register
Bits
1
Bit Name
PREN
0
PLEN
Description
Playback right channel enable.
0 = disabled (default).
1 = enabled.
Playback left channel enable.
0 = disabled (default).
1 = enabled.
Rev. 0 | Page 71 of 88
Bit 2
Bit 1
PREN
Bit 0
PLEN
ADAU1461
R36: DAC Control 0, 16,426 (0x402A)
Bit 7
Bit 6
DACMONO[1:0]
Bit 5
DACPOL
Bit 4
Bit 3
Reserved
Bit 2
DEMPH
Bit 1
Bit 0
DACEN[1:0]
Table 68. DAC Control 0 Register
Bits
[7:6]
Bit Name
DACMONO[1:0]
5
DACPOL
2
DEMPH
[1:0]
DACEN[1:0]
Description
DAC mono mode. The DAC channels can be set to mono mode within the DAC and output on the left
channel, the right channel, or both channels.
Setting
Mono Mode
00
Stereo (default)
01
Left channel in mono mode
10
Right channel in mono mode
11
Both channels in mono mode
Invert input polarity of the DACs.
0 = normal (default).
1 = inverted.
DAC de-emphasis filter enable. The de-emphasis filter is designed for use with a sampling rate of 44.1 kHz only.
0 = disabled (default).
1 = enabled.
DAC enable.
Setting
DACs Enabled
00
Both off (default)
01
Left on
10
Right on
11
Both on
R37: DAC Control 1, 16,427 (0x402B)
Bit 7
Bit 6
Bit 5
Bit 4
Bit 3
LDAVOL[7:0]
Bit 2
Bit 1
Bit 0
Table 69. DAC Control 1 Register
Bits
[7:0]
Bit Name
LDAVOL[7:0]
Description
Controls the digital volume attenuation for left channel inputs from the left DAC. Each bit corresponds to a
0.375 dB step with slewing between settings. See Table 92 for a complete list of the volume settings.
Setting
Volume Attenuation
00000000
0 dB (default)
00000001
−0.375 dB
00000010
−0.75 dB
…
…
11111110
−95.25 dB
11111111
−95.625 dB
Rev. 0 | Page 72 of 88
ADAU1461
R38: DAC Control 2, 16,428 (0x402C)
Bit 7
Bit 6
Bit 5
Bit 4
Bit 3
RDAVOL[7:0]
Bit 2
Bit 1
Bit 0
Table 70. DAC Control 2 Register
Bits
[7:0]
Bit Name
RDAVOL[7:0]
Description
Controls the digital volume attenuation for right channel inputs from the right DAC. Each bit corresponds to a
0.375 dB step with slewing between settings. See Table 92 for a complete list of the volume settings.
Setting
Volume Attenuation
00000000
0 dB (default)
00000001
−0.375 dB
00000010
−0.75 dB
…
…
11111110
−95.25 dB
11111111
−95.625 dB
R39: Serial Port Pad Control, 16,429 (0x402D)
The optional pull-up/pull-down resistors are nominally 250 kΩ. When enabled, these pull-up/pull-down resistors set the serial port
signals to a defined state when the signal source becomes three-state.
Bit 7
Bit 6
ADCSDP[1:0]
Bit 5
Bit 4
DACSDP[1:0]
Bit 3
Table 71. Serial Port Pad Control Register
Bits
[7:6]
Bit Name
ADCSDP[1:0]
[5:4]
DACSDP[1:0]
[3:2]
LRCLKP[1:0]
[1:0]
BCLKP[1:0]
Description
ADC_SDATA pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
DAC_SDATA pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
LRCLK pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
BCLK pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
Rev. 0 | Page 73 of 88
Bit 2
LRCLKP[1:0]
Bit 1
Bit 0
BCLKP[1:0]
ADAU1461
R40: Control Port Pad Control 0, 16,431 (0x402F)
The optional pull-up/pull-down resistors are nominally 250 kΩ. When enabled, these pull-up/pull-down resistors set the control port
signals to a defined state when the signal source becomes three-state.
Bit 7
Bit 6
CDATP[1:0]
Bit 5
Bit 4
CLCHP[1:0]
Bit 3
Bit 2
SCLP[1:0]
Bit 1
Bit 0
SDAP[1:0]
Table 72. Control Port Pad Control 0 Register
Bits
[7:6]
Bit Name
CDATP[1:0]
[5:4]
CLCHP[1:0]
[3:2]
SCLP[1:0]
[1:0]
SDAP[1:0]
Description
CDATA pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
CLATCH pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
Pull-down
11
SCL/CCLK pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
SDA/COUT pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
R41: Control Port Pad Control 1, 16,432 (0x4030)
With IOVDD set to 3.3 V, the low and high drive strengths of the SDA/COUT pin are approximately 2.0 mA and 4.0 mA, respectively.
The high drive strength mode may be useful for generating a stronger ACK pulse in I2C mode, if needed.
Bit 7
Bit 6
Bit 5
Bit 4
Reserved
Bit 3
Table 73. Control Port Pad Control 1 Register
Bits
0
Bit Name
SDASTR
Description
SDA/COUT pin drive strength.
0 = low (default).
1 = high.
Rev. 0 | Page 74 of 88
Bit 2
Bit 1
Bit 0
SDASTR
ADAU1461
R42: Jack Detect Pin Control, 16,433 (0x4031)
With IOVDD set to 3.3 V, the low and high drive strengths of the JACKDET/MICIN pin are approximately 2.0 mA and 4.0 mA, respectively.
The optional pull-up/pull-down resistors are nominally 250 kΩ. When enabled, these pull-up/pull-down resistors set the input signals to
a defined state when the signal source becomes three-state.
Bit 7
Bit 6
Reserved
Bit 5
JDSTR
Bit 4
Reserved
Bit 3
Bit 2
JDP[1:0]
Bit 1
Bit 0
Reserved
Table 74. Jack Detect Pin Control Register
Bits
5
Bit Name
JDSTR
[3:2]
JDP[1:0]
Description
JACKDET/MICIN pin drive strength.
0 = low (default).
1 = high.
JACKDET/MICIN pad pull-up/pull-down configuration.
Setting
Configuration
00
Pull-up
01
Reserved
10
None (default)
11
Pull-down
R67: Dejitter Control, 16,438 (0x4036)
The dejitter control register allows the size of the dejitter window to be set, and also allows all dejitter circuits in the device to be activated or
bypassed. Dejitter circuits protect against duplicate samples or skipped samples due to jitter from the serial ports in slave mode. Disabling
and reenabling certain subsystems in the device—that is, the ADCs, serial ports, SigmaDSP core, and DACs—during operation can cause
the associated dejitter circuits to fail. As a result, audio data fails to be output to the next subsystem in the device.
When the serial ports are in master mode, the dejitter circuit can be bypassed by setting the dejitter window to 0. When the serial ports
are in slave mode, the dejitter circuit can be reinitialized prior to outputting audio from the device, guaranteeing that audio is output to
the next subsystem in the device. Any time that audio must pass through the ADCs, serial port, sound engine/DSP core, or DACs, the
dejitter circuit can be bypassed and reset by setting the dejitter window size to 0. In this way, the dejitter circuit can be immediately
reactivated, without a wait period, by setting the dejitter window size to the default value of 3.
Bit 7
Bit 6
Bit 5
Bit 4
Bit 3
DEJIT[7:0]
Table 75. Dejitter Control Register
Bits
[7:0]
Bit Name
DEJIT[7:0]
Description
Dejitter window size.
Window Size
00000000
…
00000011
…
00000101
Core Clock Cycles
0
…
3 (default)
…
5
Rev. 0 | Page 75 of 88
Bit 2
Bit 1
Bit 0
ADAU1461
R43 to R47: Cyclic Redundancy Check Registers, 16,576 to 16,580 (0x40C0 to 0x40C4)
The cyclic redundancy check (CRC) constantly checks the validity of the program RAM contents. SigmaStudio generates a 32-bit hash
sum, which must be written to four consecutive read-only 8-bit register locations. CRC must then be enabled. Every 1024 frames (21 ms
at 48 kHz), the IC generates its own 32-bit code and compares it to the one stored in these registers. If the codes do not match, a GPIO pin
is set high (CRC flag). This output flag must be enabled using the output CRC error sticky setting in the GPIO pin control register (see
Table 77). The 1-bit CRC error flag is reset when the CRCEN bit goes low. For example, a GPIO pin can be connected to an interrupt pin
on an external microcontroller, which triggers a rewrite of the corrupted memory.
By default, CRC is disabled (the CRCEN bit is set to 0). To enable continuous CRC checking, the user can set the CRCEN bit to 1 after
loading a program and sending the correct CRC, which is calculated by SigmaStudio. If an error occurs, it can be cleared by setting the
CRCEN bit low, fixing the error (presumably by reloading the program), and then setting the CRCEN bit high again.
Address
0x40C0
0x40C1
0x40C2
0x40C3
0x40C4
Bit 7
Bit 6
Bit 5
Bit 4
Bit 3
CRC[31:24]
CRC[23:16]
CRC[15:8]
CRC[7:0]
Reserved
Bit 2
Table 76. Cyclic Redundancy Check Registers
Register
R43
R44
R45
R46
R47
Address
Decimal
Hex
16,576
0x40C0
16,577
0x40C1
16,578
0x40C2
16,579
0x40C3
16,580
0x40C4
Bit Name
CRC[31:24]
CRC[23:16]
CRC[15:8]
CRC[7:0]
CRCEN
Description
CRC hash sum, Bits[31:24] (read-only register)
CRC hash sum, Bits[23:16] (read-only register)
CRC hash sum, Bits[15:8] (read-only register)
CRC hash sum, Bits[7:0] (read-only register)
CRC enable
0 = disabled (default)
1 = enabled
Rev. 0 | Page 76 of 88
Bit 1
Bit 0
CRCEN
ADAU1461
R48 to R51: GPIO Pin Control, 16,582 to 16,585 (0x40C6 to 0x40C9)
The GPIO pin control register sets the functionality of each GPIO pin as shown in Table 77. The GPIO functions use the same pins as the
serial port and must be enabled in the serial data/GPIO pin configuration register (Address 0x40F4). When the GPIO pins are set to
I2C/SPI port control mode, the pins are set through writes to memory locations described in Table 31. The value of the optional internal
pull-up is nominally 250 kΩ.
The output CRC error and output watchdog error settings are sticky, that is, once set, they remain set until the ADAU1461 is reset.
Address
0x40C6
0x40C7
0x40C8
0x40C9
Bit 7
Bit 6
Bit 5
Reserved
Reserved
Reserved
Reserved
Bit 4
Bit 3
Table 77. GPIO Pin Functionality Bit Settings
GPIOx[3:0] Bits
0000
0001
0010
0011
0100
0101
0110
0111
1000
1001
1010
1011
1100
1101
1110
1111
GPIO Pin Function
Input without debounce (default)
Input with debounce (0.3 ms)
Input with debounce (0.6 ms)
Input with debounce (0.9 ms)
Input with debounce (5 ms)
Input with debounce (10 ms)
Input with debounce (20 ms)
Input with debounce (40 ms)
Input controlled by I2C/SPI port
Output set by I2C/SPI port, with pull-up
Output set by I2C/SPI port, no pull-up
Output set by DSP core, with pull-up
Output set by DSP core, no pull-up
Reserved
Output CRC error (sticky)
Output watchdog error (sticky)
Table 78. GPIO Pin Control Registers
Register
R48
R49
R50
R51
Address
Decimal
Hex
16,582
0x40C6
16,583
0x40C7
16,584
0x40C8
16,585
0x40C9
Bit Name
GPIO0[3:0]
GPIO1[3:0]
GPIO2[3:0]
GPIO3[3:0]
Description
GPIO 0 pin function (see Table 77)
GPIO 1 pin function (see Table 77)
GPIO 2 pin function (see Table 77)
GPIO 3 pin function (see Table 77)
Rev. 0 | Page 77 of 88
Bit 2
Bit 1
GPIO0[3:0]
GPIO1[3:0]
GPIO2[3:0]
GPIO3[3:0]
Bit 0
ADAU1461
R52 to R56: Watchdog Registers, 16,592 to 16,596 (0x40D0 to 0x40D4)
A program counter watchdog is used when the core does block processing (which can span several samples). The watchdog flags an error
if the program counter reaches a specific 24-bit value (ranging from 0x000000 to 0xFFFFFF) that is set in the register map. This value
consists of three consecutive 8-bit register locations. The error flag sends a high signal to one of the GPIO pins. The watchdog function
must be enabled by setting the DOGEN bit high in Register R52 (Address 0x40D0).
The watchdog error bit (DOGER) is the 1-bit watchdog error flag that can be sent to a GPIO pin, as described in Table 77. This error flag
can connect, for example, to an interrupt pin on a microcontroller in the system. The flag is reset when the DOGEN bit goes low. This
flag can also be read back over the control port from Register R56 (Address 0x40D4).
Address
0x40D0
0x40D1
0x40D2
0x40D3
0x40D4
Bit 7
Bit 6
Bit 5
Bit 4
Bit 3
Reserved
DOG[23:16]
DOG[15:8]
DOG[7:0]
Reserved
Bit 2
Bit 1
Bit 0
DOGEN
DOGER
Table 79. Watchdog Registers
Register
R52
Address
Decimal
Hex
16,592
0x40D0
Bit Name
DOGEN
R53
R54
R55
16,593
16,594
16,595
DOG[23:16]
DOG[15:8]
DOG[7:0]
R56
16,596
0x40D1
0x40D2
0x40D3
0x40D4
DOGER
Description
Watchdog enable bit.
0 = disabled (default).
1 = enabled.
Watchdog value, Bits[23:16] (MSB).
Watchdog value, Bits[15:8].
Watchdog value, Bits[7:0].
DOG[23:16]
DOG[15:8]
00000000
00000000
…
…
11111111
11111111
Watchdog error (read-only bit).
0 = no error (default).
1 = error.
DOG[7:0]
00000000
…
11111111
Hex Value
0x000000 (default)
…
0xFFFFFF
R57: DSP Sampling Rate Setting, 16,619 (0x40EB)
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Bit 3
Bit 2
Bit 1
DSPSR[3:0]
Bit 0
Table 80. DSP Sampling Rate Setting Register
Bits
[3:0]
Bit Name
DSPSR[3:0]
Description
SigmaDSP core sampling rate. The DSP sampling rate is a ratio of the base sampling rate, fS. The base sampling rate
is determined by the operating frequency of the core clock. For most applications, the SigmaDSP core sampling
rate should equal the converter sampling rate (set using the CONVSR[2:0] bits in Register R17) and the serial
port sampling rate (set using the SPSR[2:0] bits in Register R64).
Setting
Sampling Rate
Base Sampling Rate (fS = 48 kHz)
0000
fS/0.5
96 kHz, base
0001
fS
48 kHz (default)
0010
fS/1.5
32 kHz
0011
fS/2
24 kHz
0100
fS/3
16 kHz
0101
fS/4
12 kHz
0110
fS/6
8 kHz
0111
Serial input data rate
1000
Serial output data rate
1111
None
Rev. 0 | Page 78 of 88
ADAU1461
R58: Serial Input Route Control, 16,626 (0x40F2)
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Bit 3
Bit 2
Bit 1
SINRT[3:0]
Bit 0
Table 81. Serial Input Route Control Register
Bits
[3:0]
Bit Name
SINRT[3:0]
Description
Serial data input routing. This register sets the input where the DACs receive serial data. This location can be
from the DSP or from any TDM slot on the serial port.
Setting
Routing
0000
DSP to DACs [L, R] (default)
0001
Serial input [L0, R0] to DACs [L, R]
0010
Reserved
0011
Serial input [L1, R1] to DACs [L, R]
0100
Reserved
0101
Serial input [L2, R2] to DACs [L, R]
0110
Reserved
0111
Serial input [L3, R3] to DACs [L, R]
1000
Reserved
1001
Serial input [R0, L0] to DACs [L, R]
1010
Reserved
1011
Serial input [R1, L1] to DACs [L, R]
1100
Reserved
1101
Serial input [R2, L2] to DACs [L, R]
1110
Reserved
1111
Serial input [R3, L3] to DACs [L, R]
R59: Serial Output Route Control, 16,627 (0x40F3)
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Bit 3
Bit 2
Bit 1
SOUTRT[3:0]
Bit 0
Table 82. Serial Output Route Control Register
Bits
[3:0]
Bit Name
SOUTRT[3:0]
Description
Serial data output routing. This register sets the output where the ADCs send serial data. This location can be to
the DSP or to any TDM slot on the serial port.
Setting
Routing
0000
ADCs [L, R] to DSP (default)
0001
ADCs [L, R] to serial output [L0, R0]
0010
Reserved
0011
ADCs [L, R] to serial output [L1, R1]
0100
Reserved
0101
ADCs [L, R] to serial output [L2, R2]
0110
Reserved
0111
ADCs [L, R] to serial output [L3, R3]
1000
Reserved
1001
ADCs [L, R] to serial output [R0, L0]
1010
Reserved
1011
ADCs [L, R] to serial output [R1, L1]
1100
Reserved
1101
ADCs [L, R] to serial output [R2, L2]
1110
Reserved
1111
ADCs [L, R] to serial output [R3, L3]
Rev. 0 | Page 79 of 88
ADAU1461
R60: Serial Data/GPIO Pin Configuration, 16,628 (0x40F4)
The serial data/GPIO pin configuration register controls the functionality of the serial data port pins. If the bits in this register are set to 1,
these pins are configured as GPIO interfaces to the SigmaDSP. If these bits are set to 0, they are configured as serial data I/O port pins.
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Bit 3
LRGP3
Bit 2
BGP2
Bit 1
SDOGP1
Bit 0
SDIGP0
Bit 2
Bit 1
Bit 0
DSPEN
Table 83. Serial Data/GPIO Pin Configuration Register
Bits
3
Bit Name
LRGP3
2
BGP2
1
SDOGP1
0
SDIGP0
Description
LRCLK or GPIO3 pin configuration select.
0 = LRCLK enabled (default).
1 = GPIO3 enabled.
BCLK or GPIO2 pin configuration select.
0 = BCLK enabled (default).
1 = GPIO2 enabled.
ADC_SDATA or GPIO1 pin configuration select.
0 = ADC_SDATA enabled (default).
1 = GPIO1 enabled.
DAC_SDATA or GPIO0 pin configuration select.
0 = DAC_SDATA enabled (default).
1 = GPIO0 enabled.
R61: DSP Enable, 16,629 (0x40F5)
Bit 7
Bit 6
Bit 5
Bit 4
Reserved
Bit 3
Table 84. DSP Enable Register
Bits
0
Bit Name
DSPEN
Description
Enables the DSP. Set this bit before writing to the parameter RAM and before setting the DSPRUN bit in
Register R62 (Address 0x40F6).
0 = DSP disabled (default).
1 = DSP enabled.
R62: DSP Run, 16,630 (0x40F6)
Bit 7
Bit 6
Bit 5
Bit 4
Reserved
Bit 3
Bit 2
Bit 1
Table 85. DSP Run Register
Bits
0
Bit Name
DSPRUN
Description
Run the DSP. Set the DSPEN bit in Register R61 (Address 0x40F5) before setting this bit.
0 = DSP off (default).
1 = run the DSP.
Rev. 0 | Page 80 of 88
Bit 0
DSPRUN
ADAU1461
R63: DSP Slew Modes, 16,631 (0x40F7)
The DSP slew modes register sets the slew source for each output. The slew source can be either the DSP (digital slew) or the codec (analog
slew). When these bits are set to Logic 0, the codec provides volume slew according to the ASLEW[1:0] bits in Register R34 (playback
pop/click suppression register, Address 0x4028). When these bits are set to Logic 1, the slew is provided and defined by the DSP program,
disabling the codec volume slew.
Bit 7
Bit 6
Reserved
Bit 5
Bit 4
MOSLW
Bit 3
ROSLW
Bit 2
LOSLW
Bit 1
RHPSLW
Bit 0
LHPSLW
Bit 3
Bit 2
Bit 1
SPSR[2:0]
Bit 0
Table 86. DSP Slew Modes Register
Bits
4
Bit Name
MOSLW
3
ROSLW
2
LOSLW
1
RHPSLW
0
LHPSLW
Description
Mono output slew generation.
0 = codec (default).
1 = DSP.
Line output right slew generation.
0 = codec (default).
1 = DSP.
Line output left slew generation.
0 = codec (default).
1 = DSP.
Headphone right slew generation.
0 = codec (default).
1 = DSP.
Headphone left slew generation.
0 = codec (default).
1 = DSP.
R64: Serial Port Sampling Rate, 16,632 (0x40F8)
Bit 7
Bit 6
Bit 5
Reserved
Bit 4
Table 87. Serial Port Sampling Rate Register
Bits
[2:0]
Bit Name
SPSR[2:0]
Description
Serial port sampling rate. The serial port sampling rate is a ratio of the base sampling rate, fS. The base sampling
rate is determined by the operating frequency of the core clock. For most applications, the serial port sampling
rate should equal the converter sampling rate (set using the CONVSR[2:0] bits in Register R17) and the DSP sampling
rate (set using the DSPSR[3:0] bits in Register R57).
Setting
Sampling Rate
Base Sampling Rate (fS = 48 kHz)
000
fS
48 kHz, base (default)
001
fS/6
8 kHz
010
fS/4
12 kHz
011
fS/3
16 kHz
100
fS/2
24 kHz
101
fS/1.5
32 kHz
110
fS/0.5
96 kHz
111
Reserved
Rev. 0 | Page 81 of 88
ADAU1461
R65: Clock Enable 0, 16,633 (0x40F9)
This register disables or enables the digital clock engine for different blocks within the ADAU1461. For maximum power saving, use this
register to disable blocks that are not being used.
Bit 7
Reserved
Bit 6
SLEWPD
Bit 5
ALCPD
Bit 4
DECPD
Bit 3
SOUTPD
Bit 2
INTPD
Bit 1
SINPD
Bit 0
SPPD
Table 88. Clock Enable 0 Register
Bits
6
Bit Name
SLEWPD
5
ALCPD
4
DECPD
3
SOUTPD
2
INTPD
1
SINPD
0
SPPD
Description
Codec slew digital clock engine enable. When powered down, the analog playback path volume controls are
disabled and stay set to their current state.
0 = powered down (default).
1 = enabled.
ALC digital clock engine enable.
0 = powered down (default).
1 = enabled.
Decimator resync (dejitter) digital clock engine enable.
0 = powered down (default).
1 = enabled.
Serial routing outputs digital clock engine enable.
0 = powered down (default).
1 = enabled.
Interpolator resync (dejitter) digital clock engine enable.
0 = powered down (default).
1 = enabled.
Serial routing inputs digital clock engine enable.
0 = powered down (default).
1 = enabled.
Serial port digital clock engine enable.
0 = powered down (default).
1 = enabled.
R66: Clock Enable 1, 16,634 (0x40FA)
This register enables Digital Clock Generator 0 and Digital Clock Generator 1. Digital Clock Generator 0 generates sample rates for the
ADCs, DACs, and DSP. Digital Clock Generator 1 generates BCLK and LRCLK for the serial port when the part is in master mode. For
maximum power saving, use this register to disable clocks that are not being used.
Bit 7
Bit 6
Bit 5
Bit 4
Reserved
Bit 3
Table 89. Clock Enable 1 Register
Bits
1
Bit Name
CLK1
0
CLK0
Description
Digital Clock Generator 1.
0 = off (default).
1 = on.
Digital Clock Generator 0.
0 = off (default).
1 = on.
Rev. 0 | Page 82 of 88
Bit 2
Bit 1
CLK1
Bit 0
CLK0
ADAU1461
Table 90. R8 and R9 Volume Settings
Binary Value
000000
000001
000010
000011
000100
000101
000110
000111
001000
001001
001010
001011
001100
001101
001110
001111
010000
010001
010010
010011
010100
010101
010110
010111
011000
011001
011010
011011
011100
011101
011110
011111
100000
100001
100010
100011
100100
100101
100110
100111
101000
101001
101010
101011
101100
101101
101110
101111
110000
110001
110010
Volume Setting (dB)
−12
−11.25
−10.5
−9.75
−9
−8.25
−7.5
−6.75
−6
−5.25
−4.5
−3.75
−3
−2.25
−1.5
−0.75
0
0.75
1.5
2.25
3
3.75
4.5
5.25
6
6.75
7.5
8.25
9
9.75
10.5
11.25
12
12.75
13.5
14.25
15
15.75
16.5
17.25
18
18.75
19.5
20.25
21
21.75
22.5
23.25
24
24.75
25.5
Binary Value
110011
110100
110101
110110
110111
111000
111001
111010
111011
111100
111101
111110
111111
Volume Setting (dB)
26.25
27
27.75
28.5
29.25
30
30.75
31.5
32.25
33
33.75
34.5
35.25
Table 91. R14 Noise Gate Threshold
Binary Value
00000
00001
00010
00011
00100
00101
00110
00111
01000
01001
01010
01011
01100
01101
01110
01111
10000
10001
10010
10011
10100
10101
10110
10111
11000
11001
11010
11011
11100
11101
11110
11111
Rev. 0 | Page 83 of 88
Noise Gate Threshold (dB)
−76.5
−75
−73.5
−72
−70.5
−69
−67.5
−66
−64.5
−63
−61.5
−60
−58.5
−57
−55.5
−54
−52.5
−51
−49.5
−48
−46.5
−45
−43.5
−42
−40.5
−39
−37.5
−36
−34.5
−33
−31.5
−30
ADAU1461
Table 92. R20, R21, R37, and R38 Volume Settings
Binary Value
00000000
00000001
00000010
00000011
00000100
00000101
00000110
00000111
00001000
00001001
00001010
00001011
00001100
00001101
00001110
00001111
00010000
00010001
00010010
00010011
00010100
00010101
00010110
00010111
00011000
00011001
00011010
00011011
00011100
00011101
00011110
00011111
00100000
00100001
00100010
00100011
00100100
00100101
00100110
00100111
00101000
00101001
00101010
00101011
00101100
00101101
00101110
00101111
Volume Attenuation (dB)
0
−0.375
−0.75
−1.125
−1.5
−1.875
−2.25
−2.625
−3
−3.375
−3.75
−4.125
−4.5
−4.875
−5.25
−5.625
−6
−6.375
−6.75
−7.125
−7.5
−7.875
−8.25
−8.625
−9
−9.375
−9.75
−10.125
−10.5
−10.875
−11.25
−11.625
−12
−12.375
−12.75
−13.125
−13.5
−13.875
−14.25
−14.625
−15
−15.375
−15.75
−16.125
−16.5
−16.875
−17.25
−17.625
Binary Value
00110000
00110001
00110010
00110011
00110100
00110101
00110110
00110111
00111000
00111001
00111010
00111011
00111100
00111101
00111110
00111111
01000000
01000001
01000010
01000011
01000100
01000101
01000110
01000111
01001000
01001001
01001010
01001011
01001100
01001101
01001110
01001111
01010000
01010001
01010010
01010011
01010100
01010101
01010110
01010111
01011000
01011001
01011010
01011011
01011100
01011101
01011110
01011111
Rev. 0 | Page 84 of 88
Volume Attenuation (dB)
−18
−18.375
−18.75
−19.125
−19.5
−19.875
−20.25
−20.625
−21
−21.375
−21.75
−22.125
−22.5
−22.875
−23.25
−23.625
−24
−24.375
−24.75
−25.125
−25.5
−25.875
−26.25
−26.625
−27
−27.375
−27.75
−28.125
−28.5
−28.875
−29.25
−29.625
−30
−30.375
−30.75
−31.125
−31.5
−31.875
−32.25
−32.625
−33
−33.375
−33.75
−34.125
−34.5
−34.875
−35.25
−35.625
ADAU1461
Binary Value
01100000
01100001
01100010
01100011
01100100
01100101
01100110
01100111
01101000
01101001
01101010
01101011
01101100
01101101
01101110
01101111
01110000
01110001
01110010
01110011
01110100
01110101
01110110
01110111
01111000
01111001
01111010
01111011
01111100
01111101
01111110
01111111
10000000
10000001
10000010
10000011
10000100
10000101
10000110
10000111
10001000
10001001
10001010
10001011
10001100
10001101
10001110
10001111
10010000
Volume Attenuation (dB)
−36
−36.375
−36.75
−37.125
−37.5
−37.875
−38.25
−38.625
−39
−39.375
−39.75
−40.125
−40.5
−40.875
−41.25
−41.625
−42
−42.375
−42.75
−43.125
−43.5
−43.875
−44.25
−44.625
−45
−45.375
−45.75
−46.125
−46.5
−46.875
−47.25
−47.625
−48
−48.375
−48.75
−49.125
−49.5
−49.875
−50.25
−50.625
−51
−51.375
−51.75
−52.125
−52.5
−52.875
−53.25
−53.625
−54
Binary Value
10010001
10010010
10010011
10010100
10010101
10010110
10010111
10011000
10011001
10011010
10011011
10011100
10011101
10011110
10011111
10100000
10100001
10100010
10100011
10100100
10100101
10100110
10100111
10101000
10101001
10101010
10101011
10101100
10101101
10101110
10101111
10110000
10110001
10110010
10110011
10110100
10110101
10110110
10110111
10111000
10111001
10111010
10111011
10111100
10111101
10111110
10111111
11000000
11000001
Rev. 0 | Page 85 of 88
Volume Attenuation (dB)
−54.375
−54.75
−55.125
−55.5
−55.875
−56.25
−56.625
−57
−57.375
−57.75
−58.125
−58.5
−58.875
−59.25
−59.625
−60
−60.375
−60.75
−61.125
−61.5
−61.875
−62.25
−62.625
−63
−63.375
−63.75
−64.125
−64.5
−64.875
−65.25
−65.625
−66
−66.375
−66.75
−67.125
−67.5
−67.875
−68.25
−68.625
−69
−69.375
−69.75
−70.125
−70.5
−70.875
−71.25
−71.625
−72
−72.375
ADAU1461
Binary Value
11000010
11000011
11000100
11000101
11000110
11000111
11001000
11001001
11001010
11001011
11001100
11001101
11001110
11001111
11010000
11010001
11010010
11010011
11010100
11010101
11010110
11010111
11011000
11011001
11011010
11011011
11011100
11011101
11011110
11011111
11100000
11100001
11100010
11100011
11100100
11100101
11100110
11100111
11101000
11101001
11101010
11101011
11101100
11101101
11101110
11101111
11110000
11110001
11110010
Volume Attenuation (dB)
−72.75
−73.125
−73.5
−73.875
−74.25
−74.625
−75
−75.375
−75.75
−76.125
−76.5
−76.875
−77.25
−77.625
−78
−78.375
−78.75
−79.125
−79.5
−79.875
−80.25
−80.625
−81
−81.375
−81.75
−82.125
−82.5
−82.875
−83.25
−83.625
−84
−84.375
−84.75
−85.125
−85.5
−85.875
−86.25
−86.625
−87
−87.375
−87.75
−88.125
−88.5
−88.875
−89.25
−89.625
−90
−90.375
−90.75
Binary Value
11110011
11110100
11110101
11110110
11110111
11111000
11111001
11111010
11111011
11111100
11111101
11111110
11111111
Volume Attenuation (dB)
−91.125
−91.5
−91.875
−92.25
−92.625
−93
−93.375
−93.75
−94.125
−94.5
−94.875
−95.25
−95.625
Table 93. R29 through R33 Volume Settings
Binary Value
000000
000001
000010
000011
000100
000101
000110
000111
001000
001001
001010
001011
001100
001101
001110
001111
010000
010001
010010
010011
010100
010101
010110
010111
011000
011001
011010
011011
011100
011101
011110
011111
100000
Rev. 0 | Page 86 of 88
Volume Setting (dB)
−57
−56
−55
−54
−53
−52
−51
−50
−49
−48
−47
−46
−45
−44
−43
−42
−41
−40
−39
−38
−37
−36
−35
−34
−33
−32
−31
−30
−29
−28
−27
−26
−25
ADAU1461
Binary Value
100001
100010
100011
100100
100101
100110
100111
101000
101001
101010
101011
101100
101101
101110
101111
110000
110001
110010
110011
110100
110101
110110
110111
111000
111001
111010
111011
111100
111101
111110
111111
Volume Setting (dB)
−24
−23
−22
−21
−20
−19
−18
−17
−16
−15
−14
−13
−12
−11
−10
−9
−8
−7
−6
−5
−4
−3
−2
−1
0
1
2
3
4
5
6
Rev. 0 | Page 87 of 88
ADAU1461
OUTLINE DIMENSIONS
0.60 MAX
5.00
BSC SQ
0.60 MAX
PIN 1
INDICATOR
25
24
PIN 1
INDICATOR
4.75
BSC SQ
0.50
0.40
0.30
1.00
0.85
0.80
EXPOSED
PAD
(BOTTOM VIEW)
17
16
0.80 MAX
0.65 TYP
12° MAX
0.30
0.23
0.18
3.65
3.50 SQ
3.35
9
8
0.25 MIN
3.50 REF
0.05 MAX
0.02 NOM
SEATING
PLANE
1
0.20 REF
COPLANARITY
0.08
FOR PROPER CONNECTION OF
THE EXPOSED PAD, REFER TO
THE PIN CONFIGURATION AND
FUNCTION DESCRIPTIONS
SECTION OF THIS DATA SHEET.
COMPLIANT TO JEDEC STANDARDS MO-220-VHHD-2
100608-A
TOP
VIEW
0.50
BSC
32
Figure 71. 32-Lead Lead Frame Chip Scale Package [LFCSP_VQ]
5 mm × 5 mm Body, Very Thin Quad
(CP-32-4)
Dimensions shown in millimeters
ORDERING GUIDE
Model 1, 2
ADAU1461WBCPZ
ADAU1461WBCPZ-R7
ADAU1461WBCPZ-RL
1
2
Temperature Range
−40°C to +105°C
−40°C to +105°C
−40°C to +105°C
Package Description
32-Lead Lead Frame Chip Scale Package [LFCSP_VQ]
32-Lead Lead Frame Chip Scale Package [LFCSP_VQ], 7” Tape and Reel
32-Lead Lead Frame Chip Scale Package [LFCSP_VQ], 13” Tape and Reel
Package Option
CP-32-4
CP-32-4
CP-32-4
Z = RoHS Compliant Part.
W = Qualified for Automotive Applications.
AUTOMOTIVE PRODUCTS
The ADAU1461 models are available with controlled manufacturing to support the quality and reliability requirements of automotive
applications. Note that these automotive models may have specifications that differ from the commercial models; therefore, designers
should review the Specifications section of this data sheet carefully. Only the automotive grade products shown are available for use in
automotive applications. Contact your local Analog Devices account representative for specific product ordering information and to
obtain the specific Automotive Reliability reports for these models.
©2010 Analog Devices, Inc. All rights reserved. Trademarks and
registered trademarks are the property of their respective owners.
D08914-0-6/10(0)
Rev. 0 | Page 88 of 88
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