24-bit stereo audio ADC and DAC: >98 dB SNR
Sampling rates from 8 kHz to 96 kHz
Low power: 7 mW record, 7 mW playback, 48 kHz at 1.8 V
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: 1.8 V to 3.65 V
2
I
C and SPI control interfaces
Digital audio serial data I/O: stereo and time-division
multiplexing (TDM) modes
Software-controllable clickless mute
Software power-down
32-lead, 5 mm × 5 mm LFCSP
−40°C to +85°C operating temperature range
APPLICATIONS
Smartphones/multimedia phones
Digital still cameras/digital video cameras
Portable media players/portable audio players
Phone accessories products
FUNCTIONAL BLOCK DIAGRAM
Audio Codec with Integrated PLL
ADAU1361
GENERAL DESCRIPTION
The ADAU1361 is a low power, stereo audio codec that supports
stereo 48 kHz record and playback at 14 mW from a 1.8 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
ADAU1361 is ideal for battery-powered audio and telephony
applications.
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
ADAU1361 includes a stereo digital microphone input.
The ADAU1361 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.
The serial control bus supports the I
serial audio bus is programmable for I
and TDM modes. A programmable PLL supports flexible clock
generation for all standard integer rates and fractional master
clocks from 8 MHz to 27 MHz.
2
C and SPI protocols. The
2
S, left-/right-justified,
JACKDET/MICIN
LAUX
LINP
LINN
RINP
RINN
RAUX
MICBIAS
Rev. C
Information furnished by Analog Devices is believed to be accurate and reliable. However, no
responsibility is assumed by Anal og Devices for its use, nor for any infringements of patents or ot her
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.
Changes to Figure 60 and Figure 61 .............................................43
Changes to Table 26........................................................................45
Changes to R2: Digital Microphone/Jack Detection Control,
16,392 (0x4008) Section and Table 29..........................................47
Changes to Table 35........................................................................52
Changes to Table 36........................................................................53
Changes to R15: Serial Port Control 0, 16,405 (0x4015)
Section and Table 42 ....................................................................... 57
Change to Table 43..........................................................................58
Changes to Table 44, R18: Converter Control 1, 16,408
(0x4018) Section, and Table 45 .....................................................59
Changes to Table 53, R27: Playback L/R Mixer Right (Mixer 6)
Line Output Control, 16,417 (0x4021) Section, and Table 54...65
Changes to Table 55, R29: Playback Headphone Left Volume
Control, 16,419 (0x4023) Section, and Table 56.........................66
Changes to R42: Jack Detect Pin Control, 16,433 (0x4031)
Section and Table 69 ....................................................................... 73
1/09—Revision 0: Initial Version
2
C Port Section............38
Rev. C | Page 3 of 80
Page 4
ADAU1361
http://www.BDTIC.com/ADI
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, C
(digital output) = 20 pF, I
LOAD
unless otherwise noted. Performance of all channels is identical, exclusive of the interchannel gain mismatch and interchannel phase
deviation specifications.
ANALOG PERFORMANCE SPECIFICATIONS
Specifications guaranteed at 25°C (ambient).
Table 1.
Parameter Test Conditions/Comments Min Typ Max Unit
ANALOG-TO-DIGITAL CONVERTERS
ADC Resolution All ADCs 24 Bits
Digital Attenuation Step 0.375 dB
Digital Attenuation Range 95 dB
INPUT RESISTANCE
Single-Ended Line Input −12 dB gain 83 kΩ
0 dB gain 21 kΩ
6 dB gain 10.5 kΩ
PGA Inverting Inputs −12 dB gain 84.5 kΩ
0 dB gain 53 kΩ
35.25 dB gain 2 kΩ
PGA Noninverting Inputs All gains 105 kΩ
SINGLE-ENDED LINE INPUT
Full-Scale Input Voltage (0 dB) Scales linearly with AVDD AVDD/3.3 V rms
AVDD = 1.8 V 0.55 (1.56) V rms (V p-p)
AVDD = 3.3 V 1.0 (2.83) V rms (V p-p)
Dynamic Range 20 Hz to 20 kHz, −60 dB input
With A-Weighted Filter (RMS) AVDD = 1.8 V 94 dB
AVDD = 3.3 V 99 dB
No Filter (RMS) AVDD = 1.8 V 91 dB
AVDD = 3.3 V 96 dB
Total Harmonic Distortion + Noise −1 dBFS AVDD = 1.8 V −88 dB
AVDD = 3.3 V −90 dB
Signal-to-Noise Ratio
With A-Weighted Filter (RMS) AVDD = 1.8 V 94 dB
AVDD = 3.3 V 99 dB
No Filter (RMS) AVDD = 1.8 V 91 dB
AVDD = 3.3 V 96 dB
Gain per Step 3 dB
Total Gain Range −12 +6 dB
Mute Attenuation −87 dB
Interchannel Gain Mismatch 0.005 dB
Offset Error 0 mV
Gain Error −12 %
Interchannel Isolation 68 dB
Power Supply Rejection Ratio CM capacitor = 20 F
100 mV p-p @ 217 Hz 65 dB
100 mV p-p @ 1 kHz 67 dB
ADC performance excludes mixers
and PGA
(digital output) = 2 mA, VIH = 2 V, VIL = 0.8 V,
LOAD
Rev. C | Page 4 of 80
Page 5
ADAU1361
http://www.BDTIC.com/ADI
Parameter Test Conditions/Comments Min Typ Max Unit
PSEUDO-DIFFERENTIAL PGA INPUT
Full-Scale Input Voltage (0 dB) Scales linearly with AVDD AVDD/3.3 V rms
AVDD = 1.8 V 0.55 (1.56) V rms (V p-p)
AVDD = 3.3 V 1.0 (2.83) V rms (V p-p)
Dynamic Range 20 Hz to 20 kHz, −60 dB input
With A-Weighted Filter (RMS) AVDD = 1.8 V 92 dB
AVDD = 3.3 V 98 dB
No Filter (RMS) AVDD = 1.8 V 90 dB
AVDD = 3.3 V 95 dB
Total Harmonic Distortion + Noise −1 dBFS AVDD = 1.8 V −88 dB
AVDD = 3.3 V −89 dB
Signal-to-Noise Ratio
With A-Weighted Filter (RMS) AVDD = 1.8 V 92 dB
AVDD = 3.3 V 98 dB
No Filter (RMS) AVDD = 1.8 V 90 dB
AVDD = 3.3 V 95 dB
Volume Control Step PGA gain 0.75 dB
Volume Control Range PGA gain −12 +35.25 dB
PGA Boost 20 dB
Mute Attenuation −87 dB
Interchannel Gain Mismatch 0.005 dB
Offset Error 0 mV
Gain Error −14 %
Interchannel Isolation 83 dB
Common-Mode Rejection Ratio 100 mV rms, 1 kHz 65 dB
100 mV rms, 20 kHz 65 dB
FULL DIFFERENTIAL PGA INPUT Differential PGA inputs
Full-Scale Input Voltage (0 dB) Scales linearly with AVDD AVDD/3.3 V rms
AVDD = 1.8 V 0.55 (1.56) V rms (V p-p)
AVDD = 3.3 V 1.0 (2.83) V rms (V p-p)
Dynamic Range 20 Hz to 20 kHz, −60 dB input
With A-Weighted Filter (RMS) AVDD = 1.8 V 92 dB
AVDD = 3.3 V 98 dB
No Filter (RMS) AVDD = 1.8 V 90 dB
AVDD = 3.3 V 95 dB
Total Harmonic Distortion + Noise −1 dBFS
AVDD = 1.8 V −70 dB
AVDD = 3.3 V −78 dB
Signal-to-Noise Ratio
With A-Weighted Filter (RMS) AVDD = 1.8 V 92 dB
AVDD = 3.3 V 98 dB
No Filter (RMS) AVDD = 1.8 V 90 dB
AVDD = 3.3 V 95 dB
Volume Control Step PGA gain 0.75 dB
Volume Control Range PGA gain −12 +35.25 dB
PGA Boost 20 dB
Mute Attenuation −87 dB
Interchannel Gain Mismatch 0.005 dB
Offset Error 0 mV
Gain Error −14 %
Rev. C | Page 5 of 80
Page 6
ADAU1361
http://www.BDTIC.com/ADI
Parameter Test Conditions/Comments Min Typ Max Unit
Interchannel Isolation 83 dB
Common-Mode Rejection Ratio 100 mV rms, 1 kHz 65 dB
100 mV rms, 20 kHz 65 dB
MICROPHONE BIAS MBIEN = 1
Bias Voltage
0.65 × AVDD AVDD = 1.8 V, MBI = 1 1.17 V
AVDD = 3.3 V, MBI = 1 2.145 V
0.90 × AVDD AVDD = 1.8 V, MBI = 0 1.62 V
AVDD = 3.3 V, MBI = 0 2.97 V
Bias Current Source AVDD = 3.3 V, MBI = 0, MPERF = 1 3 mA
Noise in the Signal Bandwidth AVDD = 3.3 V, 1 kHz to 20 kHz MBI = 0, MPERF = 0 42 nV/√Hz
MBI = 0, MPERF = 1 85 nV/√Hz
MBI = 1, MPERF = 0 25 nV/√Hz
MBI = 1, MPERF = 1 37 nV/√Hz
DIGITAL-TO-ANALOG CONVERTERS
DAC Resolution All DACs 24 Bits
Digital Attenuation Step 0.375 dB
Digital Attenuation Range 95 dB
DAC TO LINE OUTPUT
Full-Scale Output Voltage (0 dB) Scales linearly with AVDD AVDD/3.3 V rms
AVDD = 1.8 V 0.50 (1.41) V rms (V p-p)
AVDD = 3.3 V 0.92 (2.60) V rms (V p-p)
Analog Volume Control Step Line output volume control 0.75 dB
Analog Volume Control Range Line output volume control −57 1 +6 dB
Mute Attenuation −87 dB
Dynamic Range
With A-Weighted Filter (RMS) AVDD = 1.8 V 96 dB
AVDD = 3.3 V 101 dB
No Filter (RMS) AVDD = 1.8 V 93.5 dB
AVDD = 3.3 V 98 dB
Total Harmonic Distortion + Noise −1 dBFS, line output mode dB
AVDD = 1.8 V −90 dB
AVDD = 3.3 V −92 dB
Signal-to-Noise Ratio Line output mode
With A-Weighted Filter (RMS) AVDD = 1.8 V 96 dB
AVDD = 3.3 V 101 dB
No Filter (RMS) AVDD = 1.8 V 93.5 dB
AVDD = 3.3 V 98 dB
Power Supply Rejection Ratio CM capacitor = 20 F 100 mV p-p @ 217 Hz 56 dB
100 mV p-p @ 1 kHz 70 dB
Gain Error 3 %
Interchannel Gain Mismatch 0.005 dB
Offset Error 0 mV
Interchannel Isolation 1 kHz, 0 dBFS input signal 100 dB
DAC performance excludes mixers and
headphone amplifier
20 Hz to 20 kHz, −60 dB input, line
output mode
Rev. C | Page 6 of 80
Page 7
ADAU1361
http://www.BDTIC.com/ADI
Parameter Test Conditions/Comments Min Typ Max Unit
= output power per channel
DAC TO HEADPHONE/EARPIECE
OUTPUT
Full-Scale Output Voltage (0 dB) Scales linearly with AVDD AVDD/3.3 V rms
AVDD = 1.8 V 0.50 (1.41) V rms (V p-p)
AVDD = 3.3 V 0.92 (2.60) V rms (V p-p)
Total Harmonic Distortion + Noise −4 dBFS
16 Ω load AVDD = 1.8 V, PO = 6.4 mW −76 dB
AVDD = 3.3 V, PO = 21.1 mW −82 dB
32 Ω load AVDD = 1.8 V, PO = 3.8 mW −82 dB
AVDD = 3.3 V, PO = 10.6 mW −82 dB
Power Supply Rejection Ratio CM capacitor = 20 F
100 mV p-p @ 217 Hz 56 dB
100 mV p-p @ 1 kHz 67 dB
Parameter Test Conditions/Comments Min Typ Max Unit
SUPPLIES
Voltage DVDDOUT 1.56 V
AVDD 1.8 3.3 3.65 V
IOVDD 1.63 3.3 3.65 V
Digital I/O Current (IOVDD = 1.8 V) 20 pF capacitive load on all digital pins
Slave Mode fS = 48 kHz 0.25 mA
f
f
Master Mode fS = 48 kHz 0.62 mA
f
f
Digital I/O Current (IOVDD = 3.3 V) 20 pF capacitive load on all digital pins
Slave Mode fS = 48 kHz 0.48 mA
f
f
Master Mode fS = 48 kHz 1.51 mA
f
f
Analog Current (AVDD) See Table 3
= 96 kHz 0.48 mA
S
= 8 kHz 0.07 mA
S
= 96 kHz 1.23 mA
S
= 8 kHz 0.11 mA
S
= 96 kHz 0.9 mA
S
= 8 kHz 0.13 mA
S
= 96 kHz 3 mA
S
= 8 kHz 0.27 mA
S
Rev. C | Page 7 of 80
Page 8
ADAU1361
http://www.BDTIC.com/ADI
TYPICAL CURRENT CONSUMPTION
Master clock = 12.288 MHz, input sample rate = 48 kHz, input tone = 1 kHz, normal power management settings, ADC input @ −1 dBFS,
DAC input @ 0 dBFS. For total power consumption, add the IOVDD current listed in Tab le 2 .
Table 3.
Typical AVDD Current
Operating Voltage Audio Path Clock Generation
AVDD = IOVDD = 3.3 V
DAC stereo playback to line output (10 kΩ)
AVDD = IOVDD = 1.8 V
DAC stereo playback to line output (10 kΩ)
Direct MCLK 5.24 Record stereo differential to ADC
Integer PLL 6.57
Direct MCLK 5.55
Integer PLL 6.90
Direct MCLK 55.5 DAC stereo playback to headphone (16 Ω)
Integer PLL 56.8
Direct MCLK 30.9 DAC stereo playback to headphone (32 Ω)
Integer PLL 32.25
Direct MCLK 56.75 DAC stereo playback to capless headphone (32 Ω)
Integer PLL 58
Direct MCLK 1.9 Record aux stereo bypass to line output (10 kΩ)
Integer PLL 3.3
Direct MCLK 4.25 Record stereo differential to ADC
Integer PLL 5.55
Direct MCLK 4.7
Integer PLL 5.7
Direct MCLK 30.81 DAC stereo playback to headphone (16 Ω)
Integer PLL 32
Direct MCLK 18.3 DAC stereo playback to headphone (32 Ω)
Integer PLL 19.5
Direct MCLK 32.6 DAC stereo playback to capless headphone (32 Ω)
Integer PLL 33.7
Direct MCLK 1.9 Record aux stereo bypass to line output (10 kΩ)
Integer PLL 3.07
Consumption (mA)
Rev. C | Page 8 of 80
Page 9
ADAU1361
http://www.BDTIC.com/ADI
TYPICAL POWER MANAGEMENT MEASUREMENTS
Master clock = 12.288 MHz, integer PLL, input sample rate = 48 kHz, input tone = 1 kHz. Pseudo-differential input to ADCs, DACs to
line output with 10 k load. ADC input @ −1 dBFS, DAC input @ 0 dBFS. In Tab l e 4 , the mixer boost and power management conditions
are set for MXBIAS[1:0], ADCBIAS[1:0], HPBIAS[1:0], and DACBIAS[1:0]. RBIAS[1:0] and PBIAS[1:0] do not have an extreme power
saving mode and are therefore set for power saving mode in the extreme power saving rows in Tab l e 4 .
50 ns ADC_SDATA delay. Time from BCLK falling in master mode.
SODM
SPI PORT
f
10 MHz CCLK frequency.
CCLK
t
10 ns CCLK pulse width low.
CCPL
t
10 ns CCLK pulse width high.
CCPH
t
5 ns
CLS
t
10 ns
CLH
t
10 ns
CLPH
t
5 ns CDATA setup. Time to CCLK rising.
CDS
t
5 ns CDATA hold. Time from CCLK rising.
CDH
t
50 ns
COD
I2C PORT
f
400 kHz SCL frequency.
SCL
t
0.6 µs SCL high.
SCLH
t
1.3 µs SCL low.
SCLL
t
0.6 µs Setup time; relevant for repeated start condition.
SCS
t
0.6 µs Hold time. After this period, the first clock is generated.
SCH
tDS 100 ns Data setup time.
t
300 ns SCL rise time.
SCR
t
300 ns SCL fall time.
SCF
t
300 ns SDA rise time.
SDR
t
300 ns SDA fall time.
SDF
t
0.6 µs Bus-free time. Time between stop and start.
BFT
DIGITAL MICROPHONE R
t
10 ns Digital microphone clock fall time.
DCF
t
10 ns Digital microphone clock rise time.
DCR
t
22 30 ns Digital microphone delay time for valid data.
DDV
t
0 12 ns Digital microphone delay time for data three-stated.
DDH
Unit Description
setup. Time to CCLK rising.
CLATCH
hold. Time from CCLK rising.
CLATCH
pulse width high.
CLATCH
COUT three-stated. Time from CLATCH
= 1 MΩ, C
LOAD
LOAD
rising.
= 14 pF.
Rev. C | Page 11 of 80
Page 12
ADAU1361
http://www.BDTIC.com/ADI
DIGITAL TIMING DIAGRAMS
t
LIH
t
SIS
LSB
t
SIH
07679-002
RIGHT-JUSTIFIED
BCLK
LRCLK
DAC_SDATA
LEFT-JUSTIFIED
MODE
DAC_SDATA
2
I
S MODE
DAC_SDATA
MODE
BCLK
t
BIH
t
BIL
t
LIS
t
SIS
MSB
t
SIH
8-BIT CLOCKS
(24-BIT DATA)
12-BIT CLO CKS
(20-BIT DATA)
14-BIT CLO CKS
(18-BIT DATA)
16-BIT CLO CKS
(16-BIT DATA)
t
SIS
MSB – 1
MSB
t
SIH
t
SIS
MSB
t
SIH
Figure 2. Serial Input Port Timing
t
BIH
t
BIL
LRCLK
ADC_SDATA
LEFT-JUSTIFIED
RIGHT-JUSTIFIED
MODE
ADC_SDATA
2
I
S MODE
ADC_SDATA
MODE
t
SODM
MSB
8-BIT CLO CKS
(24-BIT DAT A)
12-BIT CLOCKS
(20-BIT DAT A)
14-BIT CLOCKS
(18-BIT DAT A)
16-BIT CLOCKS
(16-BIT DAT A)
t
SODM
MSB – 1
MSB
Figure 3. Serial Output Port Timing
Rev. C | Page 12 of 80
t
SODM
MSB
LSB
07679-003
Page 13
ADAU1361
http://www.BDTIC.com/ADI
t
CLS
t
CCPL
CLATCH
CCLK
CDATA
COUT
t
CCPH
t
t
CDS
CDH
Figure 4. SPI Port Timing
t
t
SCLH
DS
t
SCH
SDA
t
SCH
t
SCR
t
CLH
t
COD
t
CLPH
07679-004
SCL
t
SCLL
t
SCF
Figure 5. I
t
2
C Port Timing
SCS
t
BFT
07679-005
t
CLK
DATA1/
DATA1DATA1DATA2DATA2
DATA2
t
DCF
DDH
t
DDV
t
DDH
t
DCR
t
DDV
07679-006
Figure 6. Digital Microphone Timing
Rev. C | Page 13 of 80
Page 14
ADAU1361
http://www.BDTIC.com/ADI
ABSOLUTE MAXIMUM RATINGS
Table 8.
Parameter Rating
Power Supply (AVDD) −0.3 V to +3.65 V
Input Current (Except Supply Pins) ±20 mA
Analog Input Voltage (Signal Pins) −0.3 V to AVDD + 0.3 V
Digital Input Voltage (Signal Pins) −0.3 V to IOVDD + 0.3 V
Operating Temperature Range −40°C to +85°C
Storage Temperature Range −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.
THERMAL RESISTANCE
θJA represents thermal resistance, junction-to-ambient; θJC represents thermal resistance, junction-to-case. All characteristics are
for a 4-layer board.
1. THE EXPOS E D PAD IS CONNECTE D INTERNALLY T O THE
ADAU1361 GROUNDS. F OR INCREASED REL IABILITY OF THE
SOLDER JOINTS AND MAXIMUM THERMAL CAP ABILITY, IT IS
RECOMMENDED THAT THE PAD BE SO LDERED TO T HE
GROUND PLANE.
Figure 7. Pin Configuration
Table 10. Pin Function Descriptions
Pin No. Mnemonic Type1 Description
1 IOVDD PWR
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 between 1.8 V and 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 F capacitor.
2 MCLK D_IN External Master Clock Input.
3
ADDR0/CLATCH
D_IN I
2
C 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.
4 JACKDET/MICIN D_IN Detect Insertion/Removal of Headphone Plug (JACKDET).
Digital Microphone Stereo Input (MICIN).
5 MICBIAS A_OUT Bias Voltage for Electret Microphone.
6 LAUX A_IN Left Channel Single-Ended Auxiliary Input. Biased at AVDD/2.
7 CM A_OUT
AVDD/2 V Common-Mode Reference. A 10 F to 47 F 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).
8 AVDD PWR
1.8 V to 3.65 V Analog Supply for DAC and Microphone Bias. This pin should be decoupled
locally to AGND with a 100 nF capacitor.
9 AGND PWR
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.
10 LINP A_IN Left Channel Noninverting Input or Single-Ended Input 0. Biased at AVDD/2.
11 LINN A_IN Left Channel Inverting Input or Single-Ended Input 1. Biased at AVDD/2.
12 RINP A_IN Right Channel Noninverting Input or Single-Ended Input 2. Biased at AVDD/2.
13 RINN A_IN Right Channel Inverting Input or Single-Ended Input 3. Biased at AVDD/2.
14 RAUX A_IN Right Channel Single-Ended Auxiliary Input. Biased at AVDD/2.
15 ROUTP A_OUT Right Line Output, Positive. Biased at AVDD/2.
16 ROUTN A_OUT Right Line Output, Negative. Biased at AVDD/2.
17 LOUTN A_OUT Left Line Output, Negative. Biased at AVDD/2.
18 LOUTP A_OUT Left Line Output, Positive. Biased at AVDD/2.
Rev. C | Page 15 of 80
Page 16
ADAU1361
http://www.BDTIC.com/ADI
Pin No. Mnemonic Type1 Description
19 RHP A_OUT Right Headphone Output. Biased at AVDD/2.
20 LHP A_OUT Left Headphone Output. Biased at AVDD/2.
21 MONOOUT A_OUT
22 AGND PWR
23 AVDD PWR
24 DVDDOUT PWR
25 DGND PWR
26 ADC_SDATA D_OUT ADC Serial Output Data.
27 DAC_SDATA D_IN DAC Serial Input Data.
28 BCLK D_IO Serial Data Port Bit Clock.
29 LRCLK D_IO Serial Data Port Frame Clock.
30 ADDR1/CDATA D_IN I2C Address Bit 1 (ADDR1).
31 SDA/COUT D_IO
32 SCL/CCLK D_IN
EP Exposed Pad
1
A_IN = analog input, A_OUT = analog output, D_IN = digital input, D_IO = digital input/output, D_OUT = digital output, PWR = power.
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.
1.8 V to 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 F 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 F
capacitors.
SPI Data Input (CDATA).
2
C Data (SDA). This pin is a bidirectional open-collector input/output. The line connected to
I
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.
2
C Clock (SCL). This pin is always an open-collector input when in I2C control mode. The line
I
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 ADAU1361 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.
Rev. C | Page 16 of 80
Page 17
ADAU1361
–
http://www.BDTIC.com/ADI
TYPICAL PERFORMANCE CHARACTERISTICS
28
26
24
22
20
18
16
14
12
10
8
6
STEREO OUTPUT POWER (mW)
4
2
0
–600–10–20–30–40–50
DIGITAL 1kHz INPUT S IGNAL (dBFS)
Figure 8. Headphone Amplifier Power vs. Input Level, 16 Ω Load
07679-068
30
–35
–40
–45
–50
–55
–60
–65
–70
–75
THD + N (dBV)
–80
–85
–90
–95
–100
–105
–600–10–20–30–40–50
DIGITAL 1kHz INPUT S IGNAL (dBFS)
Figure 11. Headphone Amplifier THD + N vs. Input Level, 16 Ω Load
07679-069
18
16
14
12
10
8
6
4
STEREO OUTPUT POWER (mW)
2
0
–600–10–20–30–40–50
DIGITAL 1kHz INPUT S IGNAL (dBFS)
Figure 9. Headphone Amplifier Power vs. Input Level, 32 Ω Load
0
10
20
30
40
50
60
MAGNITUDE (dBFS)
70
80
90
100
00.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1.0
FREQUENCY (NORMALIZED TO
f
)
S
Figure 10. ADC Decimation Filter, 64× Oversampling, Normalized to f
0
–10
–20
–30
–40
–50
–60
THD + N (dBV)
–70
–80
–90
–100
–600–10–20–30–40–50
07679-070
DIGITAL 1kHz INPUT SIGNAL (dBF S )
07679-071
Figure 12. Headphone Amplifier THD + N vs. Input Level, 32 Ω Load
Figure 22. Input Impedance vs. Gain for Analog Inputs
3.50
1.25
–1.00
–3.25
–5.50
–7.75
–10.00
–12.25
7679-125
Rev. C | Page 19 of 80
Page 20
ADAU1361
http://www.BDTIC.com/ADI
SYSTEM BLOCK DIAGRAMS
FROM VOLTAGE
REGULATOR
THE INPUT CAPACITOR VALUE DEPENDS ON THE
INPUT IMPE DANCE , WHICH VARIE S W ITH THE
VOLUME SETTING.
10µF
LEFT
MICROPHONE
10µF
2k
2k
10µF
LINP
LINN
MICBIAS
10µF
0.1µF
(1.8V TO 3.3V)
10µF
++
0.1µF
AVDDIOVDDAVDDDVDDOUT
MONOOUT
ADAU1361
1.2nH
LOUTP
LOUTN
RHP
LHP
ROUTP
ROUTN
10µF
+
0.1µF
0.1µF
9.1pF
EARPIECE
SPEAKER
CAPLESS
HEADPHONE
OUTPUT
EARPIECE
SPEAKER
RIGHT
MICROPHONE
AUX LEFT
AUX RIGHT
1k
1k
10µF
JACK
DETECTION
SIGNAL
CLOCK
SOURCE
10µF
10µF
49.9
RINN
RINP
JACKDET/MICIN
LAUX
RAUX
MCLK
DGND
AGND
Figure 25. System Block Diagram
ADC_SDATA
DAC_SDATA
LRCLK
BCLK
ADDR1/CDATA
SDA/COUT
SCL/CCLK
ADDR0/CLATCH
CM
AGND
SERIAL DATA
SYSTEM
CONTROLLER
0.1µF10µF
+
07679-045
Rev. C | Page 20 of 80
Page 21
ADAU1361
http://www.BDTIC.com/ADI
THE INPUT CAPACITOR VALUE DEPENDS ON THE
INPUT IMPE DANCE , WHICH VARIE S W ITH THE
VOLUME SETTING.
V
DD
SINGLE-ENDED
ANALOG
MICROPHONE
GND
V
DD
SINGLE-ENDED
ANALOG
MICROPHONE
GND
OUTPUT
OUTPUT
10µF
CM
10µF
CM
10µF
0.1µF
MICBIAS
LINN
LINP
RINN
RINP
10µF
0.1µF
FROM VOLTAGE
++
ADAU1361
REGULATOR
(1.8V TO 3.3V)
AVDDIOVDDAVDDDVDDOUT
1.2nH
LOUTP
LOUTN
RHP
MONOOUT
LHP
ROUTP
ROUTN
10µF
+
0.1µF
0.1µF
9.1pF
EARPIECE
SPEAKER
CAPLESS
HEADPHONE
OUTPUT
EARPIECE
SPEAKER
AUX LEFT
AUX RIGHT
1k
1k
JACK
DETECTION
SIGNAL
CLOCK
SOURCE
JACKDET/MICIN
10µF
10µF
49.9
LAUX
RAUX
MCLK
DGND
AGND
Figure 26. System Block Diagram with Analog Microphones
ADC_SDATA
DAC_SDATA
LRCLK
BCLK
ADDR1/CDATA
SDA/COUT
SCL/CCLK
ADDR0/CLATCH
CM
AGND
SERIAL DATA
SYSTEM
CONTROLLER
0.1µF10µF
+
07679-072
Rev. C | Page 21 of 80
Page 22
ADAU1361
http://www.BDTIC.com/ADI
10µF
0.1µF
MICBIAS
LINP
LINN
RINN
RINP
JACKDET/MICIN
LAUX
RAUX
MCLK
0.1µF
0.1µF
MICROPHONE
V
DD
MICROPHONE
V
DD
AUX LEFT
AUX RIG HT
CLK
DIGITAL
CLK
DIGITAL
1k
1k
BCLK
CM
DATA
GNDL/R SELECT
BCLK
DATA
GNDL/R SELECT
10µF
10µF
49.9
FROM VOLTAGE
10µF
++
0.1µF
ADAU1361
REGULATOR
(1.8V TO 3.3V)
AVDDIOVDDAVDDDVDDOUT
ADC_SDATA
DAC_SDATA
ADDR1/CDATA
ADDR0/CLATCH
RHP
MONOOUT
LHP
LOUTP
LOUTN
ROUTP
ROUTN
LRCLK
BCLK
SDA/COUT
SCL/CCLK
1.2nH
10µF
+
0.1µF
0.1µF
22nF
22nF
22nF
22nF
9.1pF
CAPLESS
HEADPHONE
OUTPUT
R
EXT
R
EXT
R
EXT
R
EXT
SERIAL DATA
SYSTEM
CONTROLLER
10µF
INL+
INL–
CLASS-D 2W
STEREO SPEAKER
INR+
INR–
2.5V T O 5.0V
0.1µF
VDDVDD
SSM2306
DRIVER
GNDSDGND
SHUTDOWN
OUTL+
OUTL–
OUTR+
OUTR–
LEFT
SPEAKER
RIGHT
SPEAKER
CLOCK
SOURCE
Figure 27. System Block Diagram with Digital Microphones and SSM2306 Class-D Speaker Driver
DGND
Rev. C | Page 22 of 80
AGND
AGND
CM
0.1µF10µF
+
07679-073
Page 23
ADAU1361
http://www.BDTIC.com/ADI
THEORY OF OPERATION
The ADAU1361 is an audio codec that offers high quality audio,
low power, and small package size. 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 I
justified, right-justified, and TDM modes for interfacing to digital
audio data. The operating voltage range is 1.8 V to 3.65 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.
2
S, left-
The ADCs and DACs are high quality, 24-bit Σ- 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 ADAU1361 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 ADAU1361 is provided in a small, 32-lead, 5 mm × 5 mm
LFCSP with an exposed bottom pad.
Rev. C | Page 23 of 80
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ADAU1361
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STARTUP, INITIALIZATION, AND POWER
This section describes the procedure for properly starting up
the ADAU1361. The following sequence provides a high level
approach to the proper initiation of the system.
1. Apply power to the ADAU1361.
2. Lock the PLL to the input clock (if using the PLL).
3. Enable the core clock.
4. Load the register settings.
POWER-UP SEQUENCE
The ADAU1361 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 ADAU1361 is set
to the default values documented in the register map (see the
Control Registers section). Typically, with a 10 F capacitor on
AVDD, the POR takes approximately 14 ms.
1.5V
DVDDOUT
AVDD
POR
POR
ACTIVE
The PLL lock time is dependent on the MCLK rate. Typical
lock times are provided in Tab l e 1 1 .
Table 11. PLL Lock Times
PLL Mode MCLK Frequency Lock Time (Typical)
Fractional 8 MHz 3.5 ms
Fractional 12 MHz 3.0 ms
Integer 12.288 MHz 2.96 ms
Fractional 13 MHz 2.4 ms
Fractional 14.4 MHz 2.4 ms
Fractional 19.2 MHz 2.98 ms
Fractional 19.68 MHz 2.98 ms
Fractional 19.8 MHz 2.98 ms
Fractional 24 MHz 2.95 ms
Integer 24.576 MHz 2.96 ms
Fractional 26 MHz 2.4 ms
Fractional 27 MHz 2.4 ms
1.35V
PART READY
POR
FINISHED
Figure 28. Power-On Reset Sequence
0.95V
POR ACTIVE
POWER REDUCTION MODES
Sections of the ADAU1361 chip can be turned on and off as
needed to reduce power consumption. These include the ADCs,
the DACs, and the PLL.
In addition, the control registers can be used to configure some
functions for power saving, normal, or enhanced performance
operation. See the Control Registers section for more
information.
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.
DIGITAL POWER SUPPLY
The digital power supply for the ADAU1361 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 F
capacitor should be connected between this pin and DGND.
INPUT/OUTPUT POWER SUPPLY
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 between
1.8 V and 3.3 V; no digital input signal should be at a voltage
level higher than the one on IOVDD. The current draw of this
07679-074
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 F capacitor.
CLOCK GENERATION AND MANAGEMENT
The ADAU1361 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 × f
is the base sampling rate. The core clock of the chip is off until
the core clock enable bit (COREN) is asserted.
, 512 × fS, 768 × fS, and 1024 × fS, where fS
S
Rev. C | Page 24 of 80
Page 25
ADAU1361
http://www.BDTIC.com/ADI
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 ADAU1361
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 ADAU1361.
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.
To program the PLL during initialization or reconfiguration of
the clock setting, the following procedure must be followed:
1. Power down the PLL.
2. Reset the PLL control register.
3. Start the PLL.
4. Poll the lock bit.
5. 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. C | Page 25 of 80
Page 26
ADAU1361
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CLOCKING AND SAMPLING RATES
ADCsDACs
MCLK
R1: PLL CONTROL REG ISTER
÷ X
× (R + N/M)
R0: CLOCK
CONTROL RE GISTER
CLKSRC
INFREQ[1:0]
256 ×
f
, 512 ×
S
f
, 1024 ×
768 ×
S
Figure 29. Clock Tree Diagram
CORE
CLOCK
f
,
S
f
S
R17: CONVERTE R
CONTROL 0 RE GISTER
CONVSR[2:0]
f
/0.5, 1, 1.5, 2, 3, 4, 6
S
SERIAL DATA
INPUT/OUTPUT
PORT
BCLK
LRCLK
ADC_SDATA
DAC_SDATA
07679-020
CORE CLOCK
Clocks for the converters and serial ports 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.
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, f
For example, if the input to CLKSRC = 49.152 MHz (from
PLL), then
INFREQ[1:0] = 1024 × f
f
= 49.152 MHz/1024 = 48 kHz
S
The PLL output clock rate is always 1024 × f
S
, and the clock
S
control register automatically sets the INFREQ[1:0] bits to
1024 × f
when using the PLL. When using a direct clock, the
S
INFREQ[1:0] frequency should be set according to the MCLK
pin clock rate and the desired base sampling frequency.
Table 12. Clock Control Register (Register R0, Address 0x4000)
Bits Bit Name Settings
3 CLKSRC
0: Direct from MCLK pin (default)
1: PLL clock
[2:1] INFREQ[1:0]
0 COREN
00: 256 × f
01: 512 × f
10: 768 × f
11: 1024 × f
0: Core clock disabled (default)
(default)
S
S
S
S
1: Core clock enabled
.
S
SAMPLING RATES
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.
Tabl e 1 3 and Tabl e 14 list the sampling rate divisions for
common base sampling rates.
Table 13. 48 kHz Base Sampling Rate Divisions
Base Sampling
Frequency
= 48 kHz
f
S
Table 14. 44.1 kHz Base Sampling Rate Divisions
Base Sampling
Frequency Sampling Rate Scaling Sampling Rate
= 44.1 kHz
f
S
Sampling Rate Scaling Sampling Rate
fS/1 48 kHz
fS/6 8 kHz
f
/4 12 kHz
S
f
/3 16 kHz
S
f
/2 24 kHz
S
f
/1.5 32 kHz
S
/0.5 96 kHz
f
S
fS/1 44.1 kHz
fS/6 7.35 kHz
f
/4 11.025 kHz
S
f
/3 14.7 kHz
S
f
/2 22.05 kHz
S
f
/1.5 29.4 kHz
S
/0.5 88.2 kHz
f
S
Rev. C | Page 26 of 80
Page 27
ADAU1361
http://www.BDTIC.com/ADI
PLL
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.
All six bytes in the PLL control register must be written with a
single continuous write to the control port.
TO PLL
MCLK
÷ X
× (R + N/M)
Figure 30. PLL Block Diagram
Integer Mode
Integer mode is used when the MCLK is an integer (R) multiple
of the PLL output (1024 × f
).
S
For example, if MCLK = 12.288 MHz and f
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.
CLOCK DIVIDER
= 48 kHz, then
S
07679-021
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 f
= 48 kHz, then
S
PLL required output = 1024 × 48 kHz = 49.152 MHz
R + (N/M) = 49.152 MHz/12 MHz = 4 + (12/125)
Common fractional PLL parameter settings for 44.1 kHz and
48 kHz sampling rates can be found in Table 16 and Ta b le 1 7.
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 15. PLL Control Register (Register R1, Address 0x4002)
Bits Bit Name Description
[47:32] M[15:0] Denominator of the fractional PLL: 16-bit binary number
0x00FD: M = 253 (default)
[31:16] N[15:0] Numerator of the fractional PLL: 16-bit binary number
0x000C: N = 12 (default)
[14:11] R[3:0] 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
The ADAU1361 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.
MIXER 1
OUTPUT
(TO P LAYBACK
MIXER)
AUXILIARY
BYPASS
MIXER 2
OUTPUT
(TO PLAYBACK
MIXER)
MIXER 2
(RIGHT RECORD
MIXER)
RIGHT
ADC
INSEL
INSEL
DECIMATOR/
ALC/
DIGITAL
VOLUME
07679-022
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 k at the 35.25 dB
gain setting to 80.4 k at the −12 dB setting. This range is shown
in Figure 22.
Rev. C | Page 29 of 80
Page 30
ADAU1361
http://www.BDTIC.com/ADI
Analog Microphone Inputs
For microphone inputs, configure the part in either stereo
pseudo-differential mode or stereo full differential mode.
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 32. 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 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 33.
Analog Line Inputs
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 34 depicts a
stereo single-ended line input using the RINN and LINN pins.
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 34. These inputs can bypass the input gain
control, mixers, and ADCs to directly connect to the output
playback mixers (see auxiliary bypass in Figure 31).
ADAU1361
LINNG[2:0]
LEFT LINE
INPUT
LEFT AUX
INPUT
RIGHT AUX
INPUT
RIGHT LINE
INPUT
Figure 34. Stereo Single-Ended Line Input with Stereo Auxiliary Bypass
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 ADAU1361 must operate in
master mode and source BCLK to the input clock of the digital
microphone.
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 35 depicts the
digital microphone interface and signal routing.
JACKDET/MICIN
R2: DIGIT AL MICROPHONE/
JACK DETECTION
CONTROL
JDFUNC[1:0]
TO JACK
DETECTION
CIRCUIT
DIGITAL MICROPHONE
RIGHT
ADC
LEFT
ADC
INTERFACE
LEFT
CHANNEL
RIGHT
CHANNEL
R19: ADC CONTROL
INSEL
ANALOG-TO-DIGITAL CONVERTERS
The ADAU1361 uses two 24-bit Σ- analog-to-digital converters (ADCs) with selectable oversampling ratios of 64× or
128× (selected by Bit 3 in Register R17, Address 0x4017).
ADC Full-Scale Level
The full-scale input to the ADCs (0 dBFS) depends on AVDD.
At AVDD = 3.3 V, the full-scale input level is 1.0 V rms. This
full-scale analog input will output a digital signal at −1.38 dBFS.
This gain offset is built into the ADAU1361 to prevent clipping.
The full-scale input level scales linearly with the level of AVDD.
For single-ended and pseudo-differential signals, the full-scale
value corresponds to the signal level at the pins, 0 dBFS.
The full differential full-scale input level is measured after the
differential amplifier, which corresponds to −6 dBFS at each pin.
Signal levels above the full-scale value cause the ADCs to clip.
Digital ADC Volume Control
The digital ADC volume can be attenuated 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 f
corner frequency of this high-pass filter is 2 Hz.
= 48 kHz, the
S
DECIMATORS
Figure 35. Digital Microphone Interface Block Diagram
07679-023
Microphone Bias
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 MICBIAS pin can also be used to cleanly supply voltage
to digital microphones or analog microphones with separate
power supply pins.
Rev. C | Page 31 of 80
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ADAU1361
A
A
http://www.BDTIC.com/ADI
AUTOMATIC LEVEL CONTROL (ALC)
The ADAU1361 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 36. 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.
NALOG
INPUT
LEFT
NALOG
INPUT
RIGHT
I2C
CONTROL
PGA
–12dB TO +3 5.25dB
0.75dB STEP SIZE
ALC
DIGITAL
Figure 36. ALC Architecture
LEFT
ADC
RIGHT
ADC
MUTE
SERIAL
PORTS
ALC PARAMETERS
The ALC function is controlled with the ALC control registers
(Address 0x4011 through Address 0x4014) using the following
parameters:
07679-024
•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.
•ALCSEL[2:0]: The ALC select bits are used to enable the
ALC and set the mode to left only, right only, or stereo. 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.
•ALCTARG[3:0]: The ALC target is the desired input
recording level that the ALC attempts to achieve.
Figure 37 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 38.
Rev. C | Page 32 of 80
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the threshold for 250 ms before the noise gate operates.
Hysteresis is used so that the threshold for coming out of the
INPUT
GAIN
OUTPUT
TARGET
DECAY
HOLD
TIME
TIME
Figure 37. Basic ALC Operation
MAX GAIN = 30dB
MAX GAIN = 24dB
MAX GAIN = 18dB
ATTACK
TIME
MIN PGA
GAIN POINT
07679-025
mute state is 6 dB higher than the threshold for going into the
mute state. There are four operating modes for the noise gate.
Noise Gate Mode 0 (see Figure 39) 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.
THRESHOLD
INPUT
ANALOG
GAIN
INTERNAL
NOISE GAT E
ENABLE SIGNAL
DIGITAL
MUTE
250ms
GAIN HELD
OUTPUT L EV E L (dB)
INPUT LEVEL (dB)
Figure 38. Effect of Varying the Maximum Gain Parameter
NOISE GATE FUNCTION
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 ADAU1361 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):
•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.
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 ADAU1361 uses a
combination of a timeout period and hysteresis. The timeout
period is set to 250 ms, so the signal must consistently be below
7679-026
OUTPUT
07679-027
Figure 39. Noise Gate Mode 0 (PGA Gain Hold)
Noise Gate Mode 1 (see Figure 40) is selected by setting the
NGTYP[1:0] bits to 01. In this mode, the ADAU1361 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
INTERNAL
NOISE GATE
ENABLE SIGNAL
DIGITAL
MUTE
OUTPUT
Figure 40. Noise Gate Mode 1 (Digital Mute)
250ms
07679-028
Rev. C | Page 33 of 80
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Noise Gate Mode 2 (see Figure 41) is selected by setting the
NGTYP[1:0] bits to 10. In this mode, the ADAU1361 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 ADAU1361 does not do a hard mute after the fade is
complete, so some small background noise will still exist.
THRESHOLD
INPUT
Noise Gate Mode 3 (see Figure 42) 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.
The outputs of the ADAU1361 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 k or can
be put into HP mode to drive headphones or earpiece speakers.
The analog output pins are biased at AVDD/2.
With a 0 dBFS digital input and AVDD = 1.8 V, the full-scale
output level is 500 mV rms; when 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.
07679-031
Routing Flexibility
The playback path contains five mixers (Mixer 3 to Mixer 7)
that perform the following functions:
• 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. C | Page 35 of 80
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HEADPHONE OUTPUT
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 .
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).
Capless Headphone Configuration
The headphone outputs can be configured in a capless output
configuration with the MONOOUT pin used as a dc virtual
ground reference. Figure 44 depicts a typical playback path in
a capless headphone configuration. Tab l e 1 9 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.
HPMODE 1 = headphone output
RHPVOL[5:0] Desired volume for RHP output
RHPM 1 = unmute right headphone output
Headphone Output Power-Up/Power-Down Sequencing
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 45 and Figure 46 illustrate the
headphone power-up/power-down sequencing.
For capless headphones, configure the MONOOUT pin before
unmuting the headphone outputs.
USER
DEFINED
4ms
HPMODE
1 = HEADPHONE
RHPM AND LHPM
1 = UNMUTE
INTERNAL
PRECHARGE
Figure 45. Headphone Output Power-Up Timing
07679-046
RHPM AND LHPM
0 = MUTE
HPMODE
0 = LINE OUTPUT
Figure 46. Headphone Output Power-Down Timing
USER DEFINED
07679-047
Ground-Centered Headphone Configuration
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:
f
= 1/(2π × R × C)
3dB
where:
C is the capacitor value.
R is the impedance of the headphones.
For a typical headphone impedance of 16 and a 47 F
capacitor, the corner frequency is 211 Hz.
Rev. C | Page 36 of 80
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ADAU1361
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Jack Detection
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.
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.
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.
LINE OUTPUTS
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 k 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 or greater. The output impedance of the line outputs is
approximately 1 k.
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
RIGHT DAC
–1
–1
MIXER 4
Figure 47. Differential Line Output Configuration
MX6G4[1:0]
MIXER 6
ROUTVOL[5:0]
LOUTN
ROUTN
ROUTP
07679-076
Rev. C | Page 37 of 80
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ADAU1361
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CONTROL PORTS
The ADAU1361 can operate in one of two control modes:
2
C control
• I
• SPI control
The ADAU1361 has both a 4-wire SPI control port and a
2
2-wire I
registers. The part defaults to I
SPI control mode by pulling the
C bus control port. Both ports can be used to set the
2
C mode, but it can be put into
CLATCH
pin low three times.
The control port is capable of full read/write operation for all
addressable registers. The ADAU1361 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 ADAU1361. This subaddress must
be two bytes long because the memory locations within the
ADAU1361 are directly addressable and their sizes exceed the
range of single-byte addressing. All subsequent bytes (starting
with Byte 3) contain the data. The number of bytes per word
depends on the type of data that is being written.
The control port pins are multifunctional, depending on the
mode in which the part is operating. Ta b le 2 0 describes these
multiple functions.
ADDR1/CDATA I2C Address Bit 1: input CDATA: input
ADDR0/CLATCH
SDA: open-collector
input/output
2
I
C Address Bit 0: input
COUT: output
: input
CLATCH
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 (I
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 ADAU1361 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.
2
C) or
CLATCH
is brought high (SPI). A
The subaddresses are autoincremented by 1 following each
read or write of a data-word, regardless of whether there is a
valid register word at that address. Address holes in the register
map can be written to or read from without consequence. In
the ADAU1361, these address holes exist at Address 0x4001,
Address 0x4003 to Address 0x4007, Address 0x402E, and
Address 0x4032 to Address 0x4035. A single-byte write to these
registers is ignored by the ADAU1361, and a read returns a
single byte 0x00.
I2C PORT
The ADAU1361 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 ADAU1361 and the system I
2
In I
C mode, the ADAU1361 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/
format is shown in . The address resides in the first
seven bits of the I
Tabl e 21
2
C write. Bits[5:6] of the I2C address for the
ADAU1361 are set by the levels on the ADDR1 and ADDR0
pins. The LSB of the address—the R/
read or write operation. Logic Level 1 corresponds to a read
operation, and Logic Level 0 corresponds to a write operation.
Table 21. ADAU1361 I
Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7
0 1 1 1 0 ADDR1 ADDR0
2
C Address and Read/
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 (1.8 V to 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 I
2
C 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
W
R/
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.
2
C master controller.
W
W
bit—specifies either a
Write
Byte Format
byte
R/W
Rev. C | Page 38 of 80
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ADAU1361
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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. shows the timing of an I
and shows an I
Figure 49
Figure 48
2
C read.
2
C write,
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 ADAU1361 immediately
jumps to the idle condition. During a given SCL high period,
the user should only issue one start condition, one stop condition,
SCL
111
SDA
START BY
MASTER
0
CHIP ADDRESS BYT E
0
FRAME 1
R/W
ADDR0ADDR1
ACK BY
ADAU1361
or a single stop condition followed by a single start condition. If
an invalid subaddress is issued by the user, the ADAU1361 does
not issue an acknowledge and returns to the idle condition.
If the user exceeds the highest subaddress while in autoincrement
mode, one of two actions is taken. In read mode, the ADAU1361
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 ADAU1361, and the part returns to the idle
condition.
ACK BY
FRAME 2
SUBADDRESS BYTE 1
ADAU1361
(CONTINUED)
(CONTINUED)
SCL
(CONTINUED)
SDA
(CONTINUED)
SCL
SDA
START BY
MASTER
SCL
SDA
SCL
(CONTINUED)
SUBADDRESS BYTE 2
0111
FRAME 1
CHIP ADDRESS BYT E
FRAME 3
SUBADDRESS BYTE 2
FRAME 3
0
Figure 48. I
ADDR0ADDR1
ACK BY
ADAU1361
2
C Write to ADAU1361 Clocking
R/W
ACK BY
ADAU1361
ACK BY
ADAU1361
REPEATED
START BY MASTER
FRAME 4
DATA BYTE 1
FRAME 2
SUBADDRESS BYTE 1
0111
FRAME 4
CHIP ADDRESS BYTE
0
ADDR1
ADAU1361
ACK BY
ADAU1361
ADDR0
ACK BY
R/W
STOP BY
MASTER
ACK BY
ADAU1361
07679-032
SDA
(CONTINUED)
FRAME 5
READ DATA BYTE 1
Figure 49. I
ACK BY
MASTER
2
C Read from ADAU1361 Clocking
STOP BY
MASTER
Rev. C | Page 39 of 80
7679-033
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ADAU1361
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I2C Read and Write Operations
Figure 50 shows the format of a single-word write operation.
Every ninth clock pulse, the ADAU1361 issues an acknowledge
by pulling SDA low.
Figure 51 shows the format of a burst mode write sequence. This
figure shows an example of a write to sequential single-byte
registers. The ADAU1361 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 52 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 ADAU1361 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
Chip address,
R/W
= 0
AS Subaddress high byte AS Subaddress low byte AS Data Byte 1 P
Figure 50. Single-Word I2C Write Format
S
Chip address,
R/W = 0
AS
Subaddress
high byte
AS
Subaddress
low byte
AS
Figure 51. Burst Mode I2C Write Format
S
Chip address,
R/W
= 0
AS
Subaddress high
byte
AS
Subaddress low
byte
Figure 52. Single-Word I2C Read Format
S
Chip address,
R/W = 0
AS
Subaddress
high byte
AS
Subaddress
low byte
Figure 53. Burst Mode I2C Read Format
AS S
This causes the ADAU1361 SDA to reverse and begin driving
data back to the master. The master then responds every ninth
pulse with an acknowledge pulse to the ADAU1361.
Figure 53 shows the format of a burst mode read sequence. This
figure shows an example of a read from sequential single-byte
registers. The ADAU1361 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
ADAU1361 always decodes the subaddress and sets the autoincrement circuit so that the address increments after the
appropriate number of bytes.
Figure 50 to Figure 53 use the following abbreviations:
S = start bit
P = stop bit
AM = acknowledge by master
AS = acknowledge by slave
Data
Byte 1
AS
Data
Byte 2
AS S
Chip address,
R/W = 1
AS
Chip address,
R/W
AS
Data
Byte 1
Data
Byte 3
= 1
AM
AS
Data
Byte 4
AS
Data
Byte 2
AS … P
Data
Byte 1
AM … P
P
Rev. C | Page 40 of 80
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ADAU1361
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SPI PORT
By default, the ADAU1361 is in I2C mode, but it can be put into
SPI control mode by pulling
CLATCH
done by performing three dummy writes to the SPI port (the
ADAU1361 does not acknowledge these three writes). Beginning
with the fourth SPI write, data can be written to or read from
the IC. The ADAU1361 can be taken out of SPI mode only by
a full reset initiated by power-cycling the IC.
The SPI port uses a 4-wire interface, consisting of the
CCLK, CDATA, and COUT signals, and it is always a slave port.
CLATCH
The
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 ADAU1361 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 . A timing diagram is shown in . All
Tabl e 2 3Figure 4
data should be written MSB first.
low three times. This is
CLATCH
,
Chip Address R/W
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 . Tabl e 22
Table 22. ADAU1361 SPI Address and Read/
Write
Byte Format
Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7
0 0 0 0 0 0 0
R/W
Subaddress
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 54. A sample timing diagram of
a single-word SPI read operation is shown in Figure 55. 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 23. Generic Control Word Format
Byte 0 Byte 1 Byte 2 Byte 3 Byte 41
chip_adr[6:0], R/W
1
Continues to end of data.
subaddr[15:8] subaddr[7:0] data data
CLATCH
CCLK
CDATA
BYTE 0BYTE 1BYTE 2BYTE 3
Figure 54. SPI Write to ADAU1361 Clocking (Single-Word Write Mode)
CLATCH
CCLK
CDATA
COUT
BYTE 0
HIGH-Z
Figure 55. SPI Read from ADAU1361 Clocking (Single-Word Read Mode)
BYTE 1
BYTE 2
DATA
HIGH-Z
07679-038
07679-039
Rev. C | Page 41 of 80
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ADAU1361
47pF
http://www.BDTIC.com/ADI
SERIAL DATA INPUT/OUTPUT PORTS
The flexible serial data input and output ports of the ADAU1361
can be set to accept or transmit data in 2-channel format or in a
4-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 and Slot 1 are
in the first half of the audio frame, and Slot 2 and Slot 3 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).
If the PLL of the ADAU1361 is not used, the serial data clocks
must be synchronous with the ADAU1361 master clock input.
The LRCLK and BCLK pins are used to clock both the serial
input and output ports. The ADAU1361 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 I
2
S,
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 ADAU1361 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 56).
This capacitor is necessary in both master and slave modes to
properly align the LRCLK signal to the serial data stream.
In TDM mode, the ADAU1361 can be a master for fS up to
48 kHz. Tabl e 2 4 lists the modes in which the serial output
port can function.
Table 24. Serial Output Port Master/Slave Mode Capabilities
2-Channel Modes (I
fS
48 kHz Master and slave Master and slave
96 kHz Master and slave Slave
Justified, Right-Justified) 4-Channel TDM
2
S, Left-
Tabl e 2 5 describes the proper configurations for standard audio
data formats.
Table 25. Data Format Configurations
LRCLK Mode
Format LRCLK Polarity (LRPOL)
I2S
(see Figure 57)
Left-Justified (see
Figure 58)
Right-Justified
(see Figure 59)
TDM with Clock
(see Figure 60)
TDM with Pulse
(see Figure 61)
Frame begins on falling edge 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
(LRMOD)
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
BCLK Cycles/Audio
Frame (BPF[2:0])
32 to 64
32 to 64
32 to 64
64 to 128
64 to 128
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
Rev. C | Page 42 of 80
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LRCL
BCLK
SDATAMSB
LRCLK
BCLK
SDATA
MSBLSBMSB
LRCLK
BCLK
SDATA
LEFT CHANNEL
MSB
S
LEFT CHANNEL
Figure 57. I
LSB
1/f
2
S Mode—16 Bits to 24 Bits per Channel
S
1/
f
Figure 58. Left-Justified Mode—16 Bits to 24 Bits per Channel
LEFT CHANNEL
MSBLSBMSB
1/
f
S
Figure 59. Right-Justified Mode—16 Bits to 24 Bits per Channel
LRCLK
BCLK
DAT
32 BCLKs
SLOT 0SLOT 2
128 BCLKs
SLOT 1SLOT 3
RIGHT CHANNE L
RIGHT CHANNEL
RIGHT CHANNE L
LSB
LSB
LSB
07679-040
07679-041
07679-042
LRCLK
BCLK
MSBMSB – 1 MSB – 2
SDATA
07679-043
Figure 60. TDM 4 Mode
LRCLK
BCLK
MSB TDM
SDATA
CH
0
SLOT 0SLOT 1SLOT 2SLOT 3
32 BCLKs
07679-044
Figure 61. TDM 4 Mode with Pulse Word Clock
Rev. C | Page 43 of 80
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APPLICATIONS INFORMATION
POWER SUPPLY BYPASS CAPACITORS
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.
Each supply signal on the board should also be bypassed with a
single bulk capacitor (10 F to 47 F).
DD GND
CAPACITOR
TO VDD
TO GND
Figure 62. Recommended Power Supply Bypass Capacitor Layout
07679-048
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 63.
GROUNDING
A single ground plane should be used in the application layout.
Components in an analog signal path should be placed away
from digital signals.
EXPOSED PAD PCB DESIGN
The ADAU1361 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 ADAU1361,
special consideration should be given to the following:
•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 64).
•Vias should be placed to connect all layers of copper,
allowing for efficient heat and energy conductivity. For an
example, see Figure 65, which has nine vias arranged in a
3 inch × 3 inch grid in the pad area.
TOP
GROUND
POWER
BOTTOM
VIAS
Figure 64. Exposed Pad Layout Example, Side View
COPPER SQUARES
07679-050
10µF
+
0.1µF
0.1µF
9.1pF1.2nH
VDD AVDD
Figure 63. GSM Filter on the Analog Supply Pins
7679-049
Rev. C | Page 44 of 80
07679-051
Figure 65. Exposed Pad Layout Example, Top View
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CONTROL REGISTERS
Table 26. Register Map
Reg Address Name Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0 Default
R0 0x4000 Clock control Reserved CLKSRC INFREQ[1:0] COREN 00000000
R1 0x4002 PLL control
Reserved R[3:0] X[1:0] Type 00010000
Reserved Lock PLLEN 00000000
R2 0x4008 Dig mic/jack detect JDDB[1:0] JDFUNC[1:0] Reserved JDPOL 00000000
R3 0x4009 Rec power mgmt Reserved MXBIAS[1:0] ADCBIAS[1:0] RBIAS[1:0] Reserved 00000000
R4 0x400A Rec Mixer Left 0 Reserved LINPG[2:0] LINNG[2:0] MX1EN 00000000
R5 0x400B Rec Mixer Left 1 Reserved LDBOOST[1:0] MX1AUXG[2:0] 00000000
R6 0x400C Rec Mixer Right 0 Reserved RINPG[2:0] RINNG[2:0] MX2EN 00000000
R7 0x400D Rec Mixer Right 1 Reserved RDBOOST[1:0] MX2AUXG[2:0] 00000000
R8 0x400E Left diff input vol LDVOL[5:0] LDMUTE LDEN 00000000
R9 0x400F Right diff input vol RDVOL[5:0] RDMUTE RDEN 00000000
R10 0x4010 Record mic bias Reserved MPERF MBI Reserved MBIEN 00000000
R11 0x4011 ALC 0 PGASLEW[1:0] ALCMAX[2:0] ALCSEL[2:0] 00000000
R12 0x4012 ALC 1 ALCHOLD[3:0] ALCTARG[3:0] 00000000
R13 0x4013 ALC 2 ALCATCK[3:0] ALCDEC[3:0] 00000000
R14 0x4014 ALC 3 NGTYP[1:0] NGEN NGTHR[4:0] 00000000
R15 0x4015 Serial Port 0 DITHEN Reserved LRMOD BPOL LRPOL CHPF[1:0] MS 00000000
R16 0x4016 Serial Port 1 BPF[2:0] ADTDM DATDM MSBP LRDEL[1:0] 00000000
R17 0x4017 Converter 0 Reserved DAPAIR[1:0] DAOSR ADOSR CONVSR[2:0] 00000000
R18 0x4018 Converter 1 Reserved ADPAIR[1:0] 00000000
R19 0x4019 ADC control Reserved ADCPOL HPF DMPOL DMSW INSEL ADCEN[1:0] 00010000
R20 0x401A Left digital vol LADVOL[7:0] 00000000
R21 0x401B Right digital vol RADVOL[7:0] 00000000
R22 0x401C Play Mixer Left 0 Reserved MX3RM MX3LM MX3AUXG[3:0] MX3EN 00000000
R23 0x401D Play Mixer Left 1 MX3G2[3:0] MX3G1[3:0] 00000000
R24 0x401E Play Mixer Right 0 Reserved MX4RM MX4LM MX4AUXG[3:0] MX4EN 00000000
R25 0x401F Play Mixer Right 1 MX4G2[3:0] MX4G1[3:0] 00000000
R26 0x4020 Play L/R mixer left Reserved MX5G4[1:0] MX5G3[1:0] MX5EN 00000000
R27 0x4021 Play L/R mixer right Reserved MX6G4[1:0] MX6G3[1:0] MX6EN 00000000
R28 0x4022 Play L/R mixer mono Reserved MX7[1:0] MX7EN 00000000
R29 0x4023 Play HP left vol LHPVOL[5:0] LHPM HPEN 00000010
R30 0x4024 Play HP right vol RHPVOL[5:0] RHPM HPMODE 00000010
R31 0x4025 Line output left vol LOUTVOL[5:0] LOUTM LOMODE 00000010
R32 0x4026 Line output right vol ROUTVOL[5:0] ROUTM ROMODE 00000010
R33 0x4027 Play mono output MONOVOL[5:0] MONOM MOMODE 00000010
R34 0x4028 Pop/click suppress Reserved POPMODE POPLESS ASLEW[1:0] Reserved 00000000
R35 0x4029 Play power mgmt HPBIAS[1:0] DACBIAS[1:0] PBIAS[1:0] PREN PLEN 00000000
R36 0x402A DAC Control 0 DACMONO[1:0] DACPOL Reserved DEMPH DACEN[1:0] 00000000
R37 0x402B DAC Control 1 LDAVOL[7:0] 00000000
R38 0x402C DAC Control 2 RDAVOL[7:0] 00000000
R39 0x402D Serial port pad ADCSDP[1:0] DACSDP[1:0] LRCLKP[1:0] BCLKP[1:0] 10101010
R40 0x402F Control Port Pad 0 CDATP[1:0] CLCHP[1:0] SCLP[1:0] SDAP[1:0] 10101010
R41 0x4030 Control Port Pad 1 Reserved SDASTR 00000000
R42 0x4031 Jack detect pin Reserved JDSTR Reserved JDP[1:0] Reserved 00001000
R67 0x4036 Dejitter control DEJIT[7:0] 00000011
M[15:8] 00000000
M[7:0] 11111101
N[15:8] 00000000
N[7:0] 00001100
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CONTROL REGISTER DETAILS
All registers except for the PLL control register are 1-byte write and read registers.
ByteBit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
0 M[15:8]
1 M[7:0]
2 N[15:8]
3 N[7:0]
4 Reserved R[3:0] X[1:0] Type
5 Reserved Lock PLLEN
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 × f
Setting Input Clock Frequency
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.
.
S
Table 28. PLL Control Register
Byte Bits Bit Name Description
0 [7:0] M[15:8] PLL denominator MSB. This value is concatenated with M[7:0] to make up a 16-bit number.
1 [7:0] M[7:0] PLL denominator LSB. This value is concatenated with M[15:8] to make up a 16-bit number.
00000000 00000000 0
… … …
00000000 11111101 253 (default)
… … …
11111111 11111111 65,535
2 [7:0] N[15:8] PLL numerator MSB. This value is concatenated with N[7:0] to make up a 16-bit number.
3 [7:0] N[7:0] PLL numerator LSB. This value is concatenated with N[15:8] to make up a 16-bit number.
Table 29. Digital Microphone/Jack Detection Control Register
Bits Bit Name Description
[7:6] JDDB[1:0] Jack detect debounce time.
00 5 ms (default)
01 10 ms
10
11
[5:4] JDFUNC[1:0]
00 Jack detect off (default)
01 Jack detect on
10
11
0 JDPOL Jack detect polarity. Detects high or low signal.
Setting Debounce Time
20 ms
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
Digital microphone input
Reserved
0 = detect high signal (default).
1 = detect low signal.
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R3: Record Power Management, 16,393 (0x4009)
This register manages the power consumption for the record path. In particular, the current distribution for the mixer boosts, ADCs,
record path mixers, and PGAs can be set to one of four modes. These settings are normal operation, power saving mode, enhanced
performance mode, and extreme power saving mode. Each of these modes draws current from a central bias. Enhanced performance
mode offers the highest performance with the trade-off of higher power consumption.
00 Normal operation (default)
01 Boost Level 1
10 Boost Level 2
11 Boost Level 3
[4:3] ADCBIAS[1:0] ADC bias control. Sets the bias current for the ADCs based on the mode of operation selected.
00 Normal operation (default)
01 Extreme power saving
10 Enhanced performance
11 Power saving
[2:1] RBIAS[1:0] Record path bias control. Sets the bias current for the PGAs and mixers in the record path.
00 Normal operation (default)
01 Reserved
10 Enhanced performance
11 Power saving
Mixer amplifier bias boost. Sets the boost level for the bias current of the record path mixers. In some cases,
the boost level enhances the THD + N performance.
Setting Boost Level
Setting ADC Bias Control
Setting Record Path Bias Control
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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 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved LINPG[2:0] LINNG[2:0] MX1EN
Table 31. Record Mixer Left (Mixer 1) Control 0 Register
Bits Bit Name Description
[6:4] LINPG[2:0] Gain for a left channel single-ended input from the LINP pin, input to Mixer 1.
000 Mute (default)
001 −12 dB
010
011
100
101
110
111
[3:1] LINNG[2:0] Gain for a left channel single-ended input from the LINN pin, input to Mixer 1.
000 Mute (default)
001 −12 dB
010
011
100
101
110
111
0 MX1EN Left channel mixer enable in the record path. Referred to as Mixer 1.
Setting Gain
−9 dB
−6 dB
−3 dB
0 dB
3 dB
6 dB
Setting Gain
−9 dB
−6 dB
−3 dB
0 dB
3 dB
6 dB
0 = mixer disabled (default).
1 = mixer enabled.
Rev. C | Page 49 of 80
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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 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved
Table 32. Record Mixer Left (Mixer 1) Control 1 Register
Bits Bit Name Description
[4:3] LDBOOST[1:0]
00 Mute (default)
01 0 dB
10
11
[2:0] MX1AUXG[2:0] Left single-ended auxiliary input gain from the LAUX pin in the record path, input to Mixer 1.
000 Mute (default)
001 −12 dB
010
011
100
101
110
111 6 dB
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
Setting Auxiliary Input Gain
LDBOOST[1:0] MX1AUXG[2:0]
20 dB
Reserved
−9 dB
−6 dB
−3 dB
0 dB
3 dB
Rev. C | Page 50 of 80
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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 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved
RINPG[2:0] RINNG[2:0]
Table 33. Record Mixer Right (Mixer 2) Control 0 Register
Bits Bit Name Description
[6:4] RINPG[2:0] Gain for a right channel single-ended input from the RINP pin, input to Mixer 2.
000 Mute (default)
001 −12 dB
010
011
100
101
110
111
[3:1] RINNG[2:0] Gain for a right channel single-ended input from the RINN pin, input to Mixer 2.
000 Mute (default)
001 −12 dB
010
011
100
101
110
111
0 MX2EN Right channel mixer enable in the record path. Referred to as Mixer 2.
Setting Gain
−9 dB
−6 dB
−3 dB
0 dB
3 dB
6 dB
Setting Gain
−9 dB
−6 dB
−3 dB
0 dB
3 dB
6 dB
0 = mixer disabled (default).
1 = mixer enabled.
MX2EN
Rev. C | Page 51 of 80
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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 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved
Table 34. Record Mixer Right (Mixer 2) Control 1 Register
Bits Bit Name Description
[4:3] RDBOOST[1:0]
00 Mute (default)
01 0 dB
10 20 dB
11 Reserved
[2:0] MX2AUXG[2:0] Right single-ended auxiliary input gain from the RAUX pin in the record path, input to Mixer 2.
000 Mute (default)
001 −12 dB
010 −9 dB
011 −6 dB
101 0 dB
110 3 dB
111 6 dB
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
Setting Auxiliary Input Gain
100 −3 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 Bit 3 Bit 2 Bit 1 Bit 0
LDVOL[5:0]
RDBOOST[1:0] MX2AUXG[2:0]
LDMUTE LDEN
Table 35. Left Differential Input Volume Control Register
Bits Bit Name Description
[7:2] LDVOL[5:0]
000000 −12 dB (default)
000001 −11.25 dB
…
010000
…
111110
111111
1 LDMUTE Left differential input mute control.
0 LDEN
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 7 1 for a complete list
of the volume settings.
Setting Volume
…
0 dB
…
34.5 dB
35.25 dB
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. C | Page 52 of 80
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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 Bit 3 Bit 2 Bit 1 Bit 0
RDVOL[5:0]
Table 36. Right Differential Input Volume Control Register
Bits Bit Name Description
[7:2] RDVOL[5:0]
000000 −12 dB (default)
000001 −11.25 dB
…
010000
…
111110
111111
1 RDMUTE Right differential input mute control.
0 RDEN
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 71 for a complete
list of the volume settings.
Setting Volume
…
0 dB
…
34.5 dB
35.25 dB
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 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved MPERF MBI Reserved MBIEN
RDMUTE RDEN
Table 37. Record Microphone Bias Control Register
Bits Bit Name Description
3 MPERF
2 MBI Microphone voltage bias as a fraction of AVDD.
0 MBIEN Enables the MICBIAS output.
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.
0 = 0.90 × AVDD (default).
1 = 0.65 × AVDD.
0 = disabled (default).
1 = enabled.
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R11: ALC Control 0, 16,401 (0x4011)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
PGASLEW[1:0] ALCMAX[2:0] ALCSEL[2:0]
Table 38. ALC Control 0 Register
Bits Bit Name Description
[7:6] PGASLEW[1:0]
00 24 ms (default)
01 48 ms
10
11
[5:3] ALCMAX[2:0]
000 −12 dB (default)
001 −6 dB
010
011
100
101
110
111
[2:0] ALCSEL[2:0]
000 Off (default)
001 Right only
010
011
100
101
110
111 Reserved
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
96 ms
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
0 dB
6 dB
12 dB
18 dB
24 dB
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. These bits must be off if manual control of the volume is desired.
0000 −28.5 dB (default)
0001 −27 dB
0010
0011
0100
0101
0110
0111
1000
1001
1010
1011
1100
1101
1110
1111 −6 dB
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
10.68 ms
21.36 ms
42.72 ms
85.44 ms
170.88 ms
341.76 ms
683.52 ms
1.367 sec
2.7341 sec
5.4682 sec
10.936 sec
21.873 sec
43.745 sec
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.
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
24 ms
48 ms
96 ms
192 ms
384 ms
768 ms
1.54 sec
3.07 sec
6.14 sec
12.29 sec
24.58 sec
49.15 sec
98.30 sec
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.
00000 −76.5 dB (default)
00001 −75 dB
…
11110
11111 −30 dB
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
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 72 for a complete list of the threshold settings.
Setting Threshold
R15: Serial Port Control 0, 16,405 (0x4015)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
DITHEN Reserved LRMOD BPOL LRPOL
NGEN
NGTHR[4:0]
Fade to PGA minimum value (analog fade)
Fade then mute (analog fade/digital mute)
…
−31.5 dB
CHPF[1:0]
MS
Table 42. Serial Port Control 0 Register
Bits Bit Name Description
7 DITHEN Dither enable is applicable only for 16-bit data width modes.
0 = disabled (default).
1 = enabled.
5 LRMOD 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.
4 BPOL
3 LRPOL
[2:1] CHPF[1:0] Channels per frame sets the number of channels per LRCLK frame.
00 Stereo (default)
01 TDM 4
10
11
0 MS
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.
Setting Channels per LRCLK Frame
Reserved
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.
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R16: Serial Port Control 1, 16,406 (0x4016)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
BPF[2:0]
Table 43. Serial Port Control 1 Register
Bits Bit Name Description
[7:5] BPF[2:0] Number of bit clock cycles per LRCLK audio frame.
000 64 (default)
001 32
010
011
100
101
110
111
4 ADTDM ADC serial audio data channel position in TDM mode.
3 DATDM DAC serial audio data channel position in TDM mode.
2 MSBP MSB position in the LRCLK frame.
[1:0] LRDEL[1:0] Data delay from LRCLK edge (in BCLK units).
00 1 (default)
01 0
10
11 16
Setting Bit Clock Cycles
0 = left first (default).
1 = right first.
0 = left first (default).
1 = right first.
0 = MSB first (default).
1 = LSB first.
Setting Delay (Bit Clock Cycles)
ADTDM DATDM MSBP
48
128
Reserved
Reserved
Reserved
Reserved
8
LRDEL[1:0]
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R17: Converter Control 0, 16,407 (0x4017)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved
DAPAIR[1:0]
Table 44. Converter Control 0 Register
Bits Bit Name Description
[6:5] DAPAIR[1:0] On-chip DAC serial data selection in TDM mode.
00 First pair (default)
01 Second pair
10
11
4 DAOSR DAC oversampling ratio. This bit cannot be set for 64× when CONVSR[2:0] is set to 96 kHz.
3 ADOSR ADC oversampling ratio. This bit cannot be set for 64× when CONVSR[2:0] is set to 96 kHz.
[2:0] CONVSR[2:0]
000 fS 48 kHz, base (default)
001 fS/6 8 kHz
010 f
011 f
100 f
101 f
110 f
111 Reserved
Setting Pair
0 = 128× (default).
1 = 64×.
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. The serial port mirrors the converter sampling rates set in this register.
[1:0] ADPAIR[1:0] On-chip ADC serial data selection in TDM mode.
00 First pair (default)
01 Second pair
10
11 Fourth pair
Setting Pair
Third pair
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R19: ADC Control, 16,409 (0x4019)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved ADCPOL HPF DMPOL DMSW INSEL
Table 46. ADC Control Register
Bits Bit Name Description
6 ADCPOL Invert input polarity.
0 = normal (default).
1 = inverted.
5 HPF ADC high-pass filter select. At 48 kHz, f
0 = off (default).
1 = on.
4 DMPOL Digital microphone data polarity swap.
0 = invert polarity.
1 = normal (default).
3 DMSW
2 INSEL
[1:0] ADCEN[1:0] ADC enable.
00 Both off (default)
01 Left on
10
11 Both on
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 × f
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.
Setting ADCs Enabled
= 2 Hz.
3dB
Right on
R20: Left Input Digital Volume, 16,410 (0x401A)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
LADVOL[7:0]
ADCEN[1:0]
, and
S
Table 47. Left Input Digital Volume Register
Bits Bit Name Description
[7:0] LADVOL[7:0]
00000000 0 dB (default)
00000001 −0.375 dB
00000010
…
11111110
11111111 −95.625 dB
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 7 3 for a complete
list of the volume settings.
Setting Volume Attenuation
−0.75 dB
…
−95.25 dB
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R21: Right Input Digital Volume, 16,411 (0x401B)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
RADVOL[7:0]
Table 48. Right Input Digital Volume Register
Bits Bit Name Description
[7:0] RADVOL[7:0]
00000000 0 dB (default)
00000001 −0.375 dB
00000010
…
11111110
11111111 −95.625 dB
R22: Playback Mixer Left (Mixer 3) Control 0, 16,412 (0x401C)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved MX3RM MX3LM
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 73 for a
complete list of the volume settings.
Setting Volume Attenuation
−0.75 dB
…
−95.25 dB
MX3AUXG[3:0]
MX3EN
Table 49. Playback Mixer Left (Mixer 3) Control 0 Register
Bits Bit Name Description
6 MX3RM Mixer input mute. Mutes the right DAC input to the left channel playback mixer (Mixer 3).
0 = muted (default).
1 = unmuted.
5 MX3LM Mixer input mute. Mutes the left DAC input to the left channel playback mixer (Mixer 3).
0 = muted (default).
1 = unmuted.
[4:1] MX3AUXG[3:0] Mixer input gain. Controls the left channel auxiliary input gain to the left channel playback mixer (Mixer 3).
0000 Mute (default)
0001 −15 dB
0010
0011
0100
0101
0110
0111
1000 6 dB
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
−12 dB
−9 dB
−6 dB
−3 dB
0 dB
3 dB
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
−12 dB
−9 dB
−6 dB
−3 dB
0 dB
3 dB
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R24: Playback Mixer Right (Mixer 4) Control 0, 16,414 (0x401E)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved MX4RM MX4LM
MX4AUXG[3:0]
Table 51. Playback Mixer Right (Mixer 4) Control 0 Register
Bits Bit Name Description
6 MX4RM Mixer input mute. Mutes the right DAC input to the right channel playback mixer (Mixer 4).
0 = muted (default).
1 = unmuted.
5 MX4LM Mixer input mute. Mutes the left DAC input to the right channel playback mixer (Mixer 4).
0 = muted (default).
1 = unmuted.
[4:1] MX4AUXG[3:0] Mixer input gain. Controls the right channel auxiliary input gain to the right channel playback mixer (Mixer 4).
0000 Mute (default)
0001 −15 dB
0010
0011
0100
0101
0110
0111
1000 6 dB
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
−12 dB
−9 dB
−6 dB
−3 dB
0 dB
3 dB
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
−12 dB
−9 dB
−6 dB
−3 dB
0 dB
3 dB
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R26: Playback L/R Mixer Left (Mixer 5) Line Output Control, 16,416 (0x4020)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved
MX5G4[1:0] MX5G3[1:0]
Table 53. Playback L/R Mixer Left (Mixer 5) Line Output Control Register
Bits Bit Name Description
[4:3] MX5G4[1:0]
00 Mute (default)
01 0 dB output (−6 dB gain on each of the two inputs)
10
11
[2:1] MX5G3[1:0]
00 Mute (default)
01 0 dB output (−6 dB gain on each of the two inputs)
10
11
0 MX5EN Mixer 5 enable.
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
6 dB output (0 dB gain on each of the two inputs)
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
6 dB output (0 dB gain on each of the two inputs)
Reserved
0 = disabled (default).
1 = enabled.
R27: Playback L/R Mixer Right (Mixer 6) Line Output Control, 16,417 (0x4021)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved
MX6G4[1:0] MX6G3[1:0]
MX5EN
MX6EN
Table 54. Playback L/R Mixer Right (Mixer 6) Line Output Control Register
Bits Bit Name Description
[4:3] MX6G4[1:0]
00 Mute (default)
01 0 dB output (−6 dB gain on each of the two inputs)
10
11
[2:1] MX6G3[1:0]
00 Mute (default)
01 0 dB output (−6 dB gain on each of the two inputs)
10
11
0 MX6EN Mixer 6 enable.
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
6 dB output (0 dB gain on each of the two inputs)
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
6 dB output (0 dB gain on each of the two inputs)
Reserved
00 Common-mode output (default)
01 0 dB output (−6 dB gain on each of the two inputs)
10
11
0 MX7EN Mixer 7 enable.
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
6 dB output (0 dB gain on each of the two inputs)
Reserved
0 = disabled (default).
1 = enabled.
R29: Playback Headphone Left Volume Control, 16,419 (0x4023)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
LHPVOL[5:0] LHPM HPEN
MX7EN
Table 56. Playback Headphone Left Volume Control Register
Bits Bit Name Description
[7:2] LHPVOL[5:0]
000000 −57 dB (default)
… …
111001
…
111111
1 LHPM Headphone mute for left channel, LHP output (active low).
0 HPEN Headphone output enable.
Headphone volume control for left channel, LHP output. Each 1-bit step corresponds to a 1 dB increase in volume.
See Table 74 for a complete list of the volume settings.
Setting Volume
0 dB
…
6 dB
0 = mute.
1 = unmute (default).
0 = disabled (default).
1 = enabled.
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R30: Playback Headphone Right Volume Control, 16,420 (0x4024)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
RHPVOL[5:0] RHPM
Table 57. Playback Headphone Right Volume Control Register
Bits Bit Name Description
[7:2] RHPVOL[5:0]
000000 −57 dB (default)
… …
111001
…
111111
1 RHPM Headphone mute for right channel, RHP output (active low).
0 HPMODE RHP and LHP output mode. These pins can be configured for either line outputs or headphone outputs.
Headphone volume control for right channel, RHP output. Each 1-bit step corresponds to a 1 dB increase in
volume. See Table 74 for a complete list of the volume settings.
Setting Volume
0 dB
…
6 dB
0 = mute.
1 = unmute (default).
0 = line output (default).
1 = headphone output.
R31: Playback Line Output Left Volume Control, 16,421 (0x4025)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
LOUTVOL[5:0]
LOUTM LOMODE
HPMODE
Table 58. Playback Line Output Left Volume Control Register
Bits Bit Name Description
[7:2] LOUTVOL[5:0]
000000 −57 dB (default)
… …
111001
…
111111
1 LOUTM Line output mute for left channel, LOUTN and LOUTP outputs (active low).
0 LOMODE
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 74 for a complete list of the volume settings.
Setting Volume
0 dB
…
6 dB
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.
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R32: Playback Line Output Right Volume Control, 16,422 (0x4026)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
ROUTVOL[5:0]
Table 59. Playback Line Output Right Volume Control Register
Bits Bit Name Description
[7:2] ROUTVOL[5:0]
000000 −57 dB (default)
… …
111001
…
111111
1 ROUTM Line output mute for right channel, ROUTN and ROUTP outputs (active low).
0 ROMODE
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 74 for a complete list of the volume settings.
Setting Volume
0 dB
…
6 dB
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).
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 74 for a complete list of the volume
settings.
Setting Volume
0 dB
…
6 dB
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).
[2:1] ASLEW[1:0] Analog volume slew rate for playback volume controls.
00 21.25 ms (default)
01 42.5 ms
10 85 ms
11 Off
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.
Setting Slew Rate
R35: Playback Power Management, 16,425 (0x4029)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
HPBIAS[1:0]
DACBIAS[1:0] PBIAS[1:0]
PREN PLEN
Reserved
Table 62. Playback Power Management Register
Bits Bit Name Description
[7:6] HPBIAS[1:0] Headphone bias control.
00 Normal operation (default)
01 Extreme power saving
10 Enhanced performance
11 Power saving
[5:4] DACBIAS[1:0] DAC bias control.
00 Normal operation (default)
01 Extreme power saving
10 Enhanced performance
11 Power saving
[3:2] PBIAS[1:0] Playback path channel bias control.
00 Normal operation (default)
01 Reserved
10 Enhanced performance
11 Power saving
1 PREN Playback right channel enable.
0 PLEN Playback left channel enable.
Setting Headphone Bias Control
Setting DAC Bias Control
Setting Playback Path Bias Control
0 = disabled (default).
1 = enabled.
0 = disabled (default).
1 = enabled.
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R36: DAC Control 0, 16,426 (0x402A)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
DACMONO[1:0]
DACPOL
Reserved
Table 63. DAC Control 0 Register
Bits Bit Name Description
[7:6] DACMONO[1:0]
00 Stereo (default)
01 Left channel in mono mode
10
11
5 DACPOL Invert input polarity of the DACs.
2 DEMPH DAC de-emphasis filter enable. The de-emphasis filter is designed for use with a sampling rate of 44.1 kHz only.
[1:0] DACEN[1:0] DAC enable.
00 Both off (default)
01 Left on
10
11 Both on
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
Right channel in mono mode
Both channels in mono mode
0 = normal (default).
1 = inverted.
0 = disabled (default).
1 = enabled.
Setting DACs Enabled
Right on
R37: DAC Control 1, 16,427 (0x402B)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
LDAVOL[7:0]
DEMPH
DACEN[1:0]
Table 64. DAC Control 1 Register
Bits Bit Name Description
[7:0] LDAVOL[7:0]
00000000 0 dB (default)
00000001 −0.375 dB
00000010
…
11111110
11111111 −95.625 dB
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 73 for a complete list of the volume settings.
Setting Volume Attenuation
−0.75 dB
…
−95.25 dB
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R38: DAC Control 2, 16,428 (0x402C)
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
RDAVOL[7:0]
Table 65. DAC Control 2 Register
Bits Bit Name Description
[7:0] RDAVOL[7:0]
00000000 0 dB (default)
00000001 −0.375 dB
00000010
…
11111110
11111111 −95.625 dB
R39: Serial Port Pad Control, 16,429 (0x402D)
The optional pull-up/pull-down resistors are nominally 250 k. 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 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
ADCSDP[1:0] DACSDP[1:0] LRCLKP[1:0] BCLKP[1:0]
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 73 for a complete list of the volume settings.
Setting Volume Attenuation
−0.75 dB
…
−95.25 dB
Table 66. Serial Port Pad Control Register
Bits Bit Name Description
[7:6] ADCSDP[1:0] ADC_SDATA pad pull-up/pull-down configuration.
The optional pull-up/pull-down resistors are nominally 250 k. 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 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
CDATP[1:0] CLCHP[1:0] SCLP[1:0] SDAP[1:0]
Table 67. Control Port Pad Control 0 Register
Bits Bit Name Description
[7:6] CDATP[1:0] CDATA pad pull-up/pull-down configuration.
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.
With IOVDD set to 1.8 V, the low and high drive strengths are approximately 0.8 mA and 1.7 mA, respectively. The high drive strength
mode may be useful for generating a stronger ACK pulse in I
Bit 7 Bit 6 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Setting Configuration
None (default)
Pull-down
pad pull-up/pull-down configuration.
CLATCH
Setting Configuration
None (default)
Pull-down
Setting Configuration
None (default)
Pull-down
Setting Configuration
None (default)
2
C mode, if needed.
Reserved SDASTR
Table 68. Control Port Pad Control 1 Register
Bits Bit Name Description
0 SDASTR SDA/COUT pin drive strength.
0 = low (default).
1 = high.
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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.
With IOVDD set to 1.8 V, the low and high drive strengths are approximately 0.8 mA and 1.7 mA, respectively. The optional pull-up/
pull-down resistors are nominally 250 k. 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 Bit 5 Bit 4 Bit 3 Bit 2 Bit 1 Bit 0
Reserved JDSTR Reserved
JDP[1:0]
Table 69. Jack Detect Pin Control Register
Bits Bit Name Description
5 JDSTR JACKDET/MICIN pin drive strength.
0 = low (default).
1 = high.
[3:2] JDP[1:0] JACKDET/MICIN pad pull-up/pull-down configuration.
00 Pull-up
01 Reserved
10
11 Pull-down
Setting Configuration
None (default)
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, 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, 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.