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Электронный компонент: WM8971

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WM8971L
Stereo CODEC for Portable Audio Applications
WOLFSON MICROELECTRONICS plc
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Production Data, August 2005, Rev 4.1
Copyright
2005 Wolfson Microelectronics plc
DESCRIPTION
The WM8971L is a low power, high quality stereo codec
designed for portable digital audio applications.
The device integrates complete interfaces to stereo or mono
microphones and a stereo headphone. External component
requirements are drastically reduced as no separate
microphone or headphone amplifiers are required.
Advanced on-chip digital signal processing performs
graphic equaliser, and automatic level control for the
microphone or line input.
The WM8971L can operate as a master or a slave, with
various master clock frequencies including 12 or 24MHz for
USB devices, or standard 256f
s
rates like 12.288MHz and
24.576MHz. Different audio sample rates such as 96kHz,
48kHz, 44.1kHz are generated directly from the master
clock without the need for an external PLL.
The WM8971L operates at supply voltages down to 1.8V,
although the digital core can operate at voltages down to
1.42V to save power, and the maximum for all supplies is
3.6 Volts. Different sections of the chip can also be powered
down under software control.
The WM8971L is supplied in a very small and thin 5x5mm
QFN package, ideal for use in hand-held and portable
systems.
FEATURES
DAC SNR 98dB (`A' weighted), THD 84dB at 48kHz, 3.3V
ADC SNR 95dB (`A' weighted), THD -82dB at 48kHz, 3.3V
Complete Stereo / Mono Microphone Interface
- Programmable ALC / Noise Gate
On-chip 400mW BTL Speaker Driver (mono)
On-chip
Headphone
Driver
-
>40mW output power on 16
/ 3.3V
-
THD 80dB at 20mW, SNR 90dB with 16
load
-
No DC blocking capacitors required (capless mode)
Separately mixed mono output
Digital Graphic Equaliser
Low
Power
-
7mW stereo playback (1.8V / 1.5V supplies)
-
14mW record and playback (1.8V / 1.5V supplies)
Low Supply Voltages
-
Analogue 1.8V to 3.6V
-
Digital core: 1.42V to 3.6V
-
Digital I/O: 1.8V to 3.6V
256fs / 384fs or USB master clock rates: 12MHz, 24MHz
Audio sample rates: 8, 11.025, 16, 22.05, 24, 32, 44.1, 48,
88.2, 96kHz generated internally from master clock
5x5x0.9mm QFN package
APPLICATIONS
Digital
Still
Cameras
MP3 Player / Recorders
Minidisc Player / Recorders
Portable Digital Music Systems
BLOCK DIAGRAM
WM8971L
Production Data
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PD Rev 4.1 August 2005
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TABLE OF CONTENTS
DESCRIPTION .......................................................................................................1
FEATURES.............................................................................................................1
APPLICATIONS .....................................................................................................1
BLOCK DIAGRAM .................................................................................................1
TABLE OF CONTENTS .........................................................................................2
PIN CONFIGURATION...........................................................................................4
ORDERING INFORMATION ..................................................................................4
PIN DESCRIPTION ................................................................................................5
ABSOLUTE MAXIMUM RATINGS .........................................................................6
RECOMMENDED OPERATION CONDITIONS .....................................................6
ELECTRICAL CHARACTERISTICS ......................................................................7
OUTPUT PGA'S LINEARITY ......................................................................................... 9
HEADPHONE OUTPUT THD VERSUS POWER......................................................... 10
SPEAKER THD AND NOISE VERSUS POWER ......................................................... 11
POWER CONSUMPTION ....................................................................................12
SIGNAL TIMING REQUIREMENTS .....................................................................13
SYSTEM CLOCK TIMING............................................................................................ 13
AUDIO INTERFACE TIMING MASTER MODE ......................................................... 13
AUDIO INTERFACE TIMING SLAVE MODE ............................................................ 14
INTERNAL POWER ON RESET CIRCUIT ..........................................................17
DEVICE DESCRIPTION .......................................................................................18
INTRODUCTION.......................................................................................................... 18
INPUT SIGNAL PATH.................................................................................................. 18
AUTOMATIC LEVEL CONTROL (ALC) ....................................................................... 25
OUTPUT SIGNAL PATH.............................................................................................. 28
ANALOGUE OUTPUTS ............................................................................................... 33
ENABLING THE OUTPUTS ......................................................................................... 35
HEADPHONE SWITCH ............................................................................................... 35
THERMAL SHUTDOWN .............................................................................................. 37
HEADPHONE OUTPUT ............................................................................................... 37
DIGITAL AUDIO INTERFACE...................................................................................... 38
AUDIO INTERFACE CONTROL .................................................................................. 42
CLOCKING AND SAMPLE RATES .............................................................................. 44
CONTROL INTERFACE .............................................................................................. 46
POWER SUPPLIES ..................................................................................................... 47
POWER MANAGEMENT ............................................................................................. 47
REGISTER MAP...................................................................................................50
DIGITAL FILTER CHARACTERISTICS ...............................................................51
TERMINOLOGY........................................................................................................... 51
DAC FILTER RESPONSES ......................................................................................... 52
ADC FILTER RESPONSES ......................................................................................... 53
DE-EMPHASIS FILTER RESPONSES ........................................................................ 54
HIGHPASS FILTER ..................................................................................................... 55
APPLICATIONS INFORMATION .........................................................................56
RECOMMENDED EXTERNAL COMPONENTS........................................................... 56
LINE INPUT CONFIGURATION................................................................................... 57
MICROPHONE INPUT CONFIGURATION .................................................................. 57
MINIMISING POP NOISE AT THE ANALOGUE OUTPUTS ........................................ 57
POWER MANAGEMENT EXAMPLES ......................................................................... 58
Production Data
WM8971L
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PD Rev 4.1 August 2005
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IMPORTANT NOTICE ..........................................................................................60
ADDRESS:................................................................................................................... 60
WM8971L
Production Data
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PD Rev 4.1 August 2005
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PIN CONFIGURATION
ORDERING INFORMATION
ORDER CODE
TEMPERATURE RANGE
PACKAGE
MOISTURE
SENSITIVITY LEVEL
PEAK SOLDERING
TEMPERATURE
WM8971LGEFL
-25
C to +85C 32-lead
QFN
(5x5x0.9mm)
(Pb-free)
MSL1 260C
WM8971LGEFL/R
-25
C to +85C 32-lead
QFN
(5x5x0.9mm)
(Pb-free, tape and reel)
MSL1 260C
Note:
Reel quantity = 3500
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PD Rev 4.1 August 2005
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PIN DESCRIPTION
PIN NO
NAME
TYPE
DESCRIPTION
1 MCLK
Digital Input
Master Clock
2 DCVDD
Supply Digital
Core
Supply
3 DBVDD
Supply
Digital Buffer (I/O) Supply
4 DGND
Supply
Digital Ground (return path for both DCVDD and DBVDD)
5 BCLK
Digital Input / Output
Audio Interface Bit Clock
6 DACDAT
Digital Input
DAC Digital Audio Data
7 DACLRC
Digital Input / Output
Audio Interface Left / Right Clock/Clock Out
8 ADCDAT
Digital Output
ADC Digital Audio Data
9 ADCLRC
Digital Input / Output
Audio Interface Left / Right Clock
10 MONOOUT
Analogue Output
Mono Output
11 NC
No Connect
No Connect
12 ROUT1
Analogue Output
Right Output 1 (Line or Headphone)
13 LOUT1
Analogue Output
Left Output 1 (Line or Headphone)
14 HPGND
Supply
Supply for Analogue Output Drivers (LOUT1/2, ROUT1/2)
15 ROUT2
Analogue Output
Right Output 1 (Line or Headphone or Speaker)
16 LOUT2
Analogue Output
Left Output 1 (Line or Headphone or Speaker)
17 HPVDD
Supply
Supply for Analogue Output Drivers (LOUT1/2, ROUT1/2,
MONOUT)
18 AVDD
Supply Analogue
Supply
19 AGND
Supply
Analogue Ground (return path for AVDD)
20 VREF
Analogue Output
Reference Voltage Decoupling Capacitor
21 VMID
Analogue Output
Midrail Voltage Decoupling Capacitor
22 MICBIAS
Analogue Output
Microphone Bias
23 HPDETECT
Analogue Input
Headphone Plug-in Detection
24 NC
No Connect
No Connect
25 MIC
Analogue Input
Single Ended Microphone Input
26 NC
No Connect
No Connect
27 RINPUT1
Analogue Input
Right Channel Input 1
28 LINPUT1
Analogue Input
Left Channel Input 1
29 MODE
Digital Input
Control Interface Selection
30 CSB
Digital Input
Chip Select / Device Address Selection
31 SDIN
Digital Input/Output
Control Interface Data Input / 2-wire Acknowledge output
32
SCLK
Digital Input
Control Interface Clock Input
Note:
It is recommended that the QFN ground paddle should be connected to analogue ground on the application PCB.
WM8971L
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PD Rev 4.1 August 2005
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ABSOLUTE MAXIMUM RATINGS
Absolute Maximum Ratings are stress ratings only. Permanent damage to the device may be caused by continuously
operating at or beyond these limits. Device functional operating limits and guaranteed performance specifications are given
under Electrical Characteristics at the test conditions specified.
ESD Sensitive Device. This device is manufactured on a CMOS process. It is therefore generically susceptible
to damage from excessive static voltages. Proper ESD precautions must be taken during handling and storage
of this device.
Wolfson tests its package types according to IPC/JEDEC J-STD-020B for Moisture Sensitivity to determine acceptable storage
conditions prior to surface mount assembly. These levels are:
MSL1 = unlimited floor life at <30
C / 85% Relative Humidity. Not normally stored in moisture barrier bag.
MSL2 = out of bag storage for 1 year at <30
C / 60% Relative Humidity. Supplied in moisture barrier bag.
MSL3 = out of bag storage for 168 hours at <30
C / 60% Relative Humidity. Supplied in moisture barrier bag.
The Moisture Sensitivity Level for each package type is specified in Ordering Information.
CONDITION MIN
MAX
Supply voltages
-0.3V +3.63V
Voltage range digital inputs
DGND -0.3V
DBVDD +0.3V
Voltage range analogue inputs
AGND -0.3V
AVDD +0.3V
Operating temperature range, T
A
-25
C +85C
Storage temperature after soldering
-65
C +150C
Notes
1.
Analogue and digital grounds must always be within 0.3V of each other.
2.
All digital and analogue supplies are completely independent from each other.
3.
DCVDD must be less than or equal to AVDD and DBVDD.
RECOMMENDED OPERATION CONDITIONS
PARAMETER SYMBOL
MIN
TYP
MAX
UNIT
Digital supply range (Core)
DCVDD 1.42
2.0
3.6
V
Digital supply range (Buffer)
DBVDD 1.7
2.0
3.6
V
Analogue supplies range
AVDD, HPVDD
1.8
2.0
3.6
V
Ground
DGND,AGND, HPGND
0
V
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ELECTRICAL CHARACTERISTICS
Test Conditions
DCVDD = 1.5V, DBVDD = 3.3V, AVDD = HPVDD = 3.3V, T
A
= +25
o
C, 1kHz signal, fs = 48kHz, PGA gain = 0dB, 24-bit audio
data unless otherwise stated.
PARAMETER SYMBOL
TEST
CONDITIONS
MIN
TYP
MAX
UNIT
Analogue Inputs (LINPUT1, RINPUT1, MIC) to ADC
AVDD = 3.3V
1.0
Full Scale Input Signal Level
(for ADC 0dB Input at 0dB Gain)
V
INFS
AVDD = 1.8V
0.545
V rms
L/RINPUT1 to ADC,
PGA gain = 0dB
22
L/RINPUT1 to ADC,
PGA gain = +30dB
1.5
L/RINPUT1 DC
measurement
16
Input Resistance
L/RINPUT1 unused
17
k
Input Capacitance
10
pF
AVDD = 3.3V
80
95
Signal to Noise Ratio
(A-weighted)
SNR
AVDD = 1.8V
90
dB
Dynamic Range
-60dBFs
90
95
dB
-1dBFs input,
AVDD = 3.3V
-82
0.008
Total Harmonic Distortion
THD
-1dBFs input,
AVDD = 1.8V
-74
0.02
dB
%
ADC Channel Separation
1kHz signal
85
dB
Channel Matching
1kHz
signal 0.2 dB
Analogue Outputs (LOUT1/2, ROUT1/2, MONOOUT)
0dB Full scale output voltage
AVDD/3.3
Vrms
1kHz, full scale signal
90
Mute attenuation
MONOOUT pin
81
dB
Channel Separation
analogue in
to analogue out
85 dB
DAC to Line-Out (L/ROUT2 with 10k
/ 50pF load)
AVDD=3.3V 90
98
Signal to Noise Ratio
(A-weighted)
SNR
AVDD=1.8V
93
dB
AVDD=3.3V
-84
Total Harmonic Distortion
THD
AVDD=1.8V
-80
dB
Channel Separation
1kHz signal
100
dB
Headphone Output (LOUT1/ROUT1, using capacitors)
Output Power per channel
P
O
Output power is very closely correlated with THD; see below.
HPVDD=1.8V, R
L
=32
P
O
=5mW
0.016
-76
HPVDD=1.8V, R
L
=16
P
O
=5mW
0.022
-73
HPVDD=3.3V, R
L
=32
,
P
O
=20mW
0.013
-78
Total Harmonic Distortion
THD
HPVDD=3.3V, R
L
=16
,
P
O
=20mW
0.018
-75
%
dB
HPVDD = 3.3V
92
96
Signal to Noise Ratio
(A-weighted)
SNR
HPVDD = 1.8V
96
dB
WM8971L
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PD Rev 4.1 August 2005
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Test Conditions
DCVDD = 1.5V, DBVDD = 3.3V, AVDD = HPVDD = 3.3V, T
A
= +25
o
C, 1kHz signal, fs = 48kHz, PGA gain = 0dB, 24-bit audio
data unless otherwise stated.
PARAMETER SYMBOL
TEST
CONDITIONS
MIN
TYP
MAX
UNIT
Speaker Output (LOUT2/ROUT2 with 8
bridge tied load, ROUT2INV=1)
Output Power at 1% THD
P
O
THD = 1%
330
mW (rms)
Abs. Max Power Output
P
O
max
500
mW
(rms)
Total Harmonic Distortion
THD Po=200mW,
R
L
=8
,
HPVDD=3.3V
-63
0.07
dB
%
Signal to Noise Ratio
(A-weighted)
SNR HPVDD=3.3V,
R
L
=8
95
dB
Analogue Reference Levels
Midrail Reference Voltage
VMID
3%
AVDD/2
+3%
V
Buffered Reference Voltage
VREF
3%
AVDD/2
+3%
V
Microphone Bias
Bias Voltage
V
MICBIAS
3mA load current
5%
0.9
AVDD +
5%
V
Bias Current Source
I
MICBIAS
3
mA
Output Noise Voltage
Vn
1K to 20kHz
15
nV/
Hz
Digital Input / Output
Input HIGH Level
V
IH
0.7
DBVDD
V
Input LOW Level
V
IL
0.3
DBVDD V
Output HIGH Level
V
OH
I
OH
= +1mA
0.9
DBVDD
V
Output LOW Level
V
OL
I
OL
= -1mA
0.1
DBVDD V
HPDETECT (pin 23)
Input HIGH Level
V
IH
0.7
AVDD
V
Input LOW Level
V
IL
0.3
AVDD V
Production Data
WM8971L
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PD Rev 4.1 August 2005
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OUTPUT PGA'S LINEARITY
Output PGA Gains
-70.000
-60.000
-50.000
-40.000
-30.000
-20.000
-10.000
0.000
10.000
40
50
60
70
80
90
100
110
120
130
XXXVOL Register Setting (binary)
M
easu
r
e
d
Gai
n
[d
B
]
LOUT1
ROUT1
LOUT2
ROUT2
MONOOUT
Output PGA Gain Step Size
0.000
0.250
0.500
0.750
1.000
1.250
1.500
1.750
2.000
40
50
60
70
80
90
100
110
120
130
XXXVOL Register Setting (binary)
S
t
ep
S
i
z
e
[
d
B]
LOUT1
ROUT1
LOUT2
ROUT2
MONOOUT
WM8971L
Production Data
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PD Rev 4.1 August 2005
10
HEADPHONE OUTPUT THD VERSUS POWER
Headphone Pow er vs THD+N (32Ohm load)
-100
-80
-60
-40
-20
0
0
5
10
15
20
25
30
Pow er (m W)
TH
D
+
N
(
d
B
)
AVDD=1.8V
AVDD=1.8V, capless
AVDD=3.3V
AVDD=3.3V, capless
Headphone Pow er vs THD+N (16Ohm load)
-100
-80
-60
-40
-20
0
0
10
20
30
40
50
60
Pow er (m W)
TH
D
+
N
(
d
B
)
AVDD=1.8V
AVDD=1.8V, capless
AVDD=3.3V
AVDD=3.3V, capless
Production Data
WM8971L
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PD Rev 4.1 August 2005
11
SPEAKER THD AND NOISE VERSUS POWER
WM8971L
Production Data
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PD Rev 4.1 August 2005
12
POWER CONSUMPTION
The power consumption of the WM8971L depends on the following factors.
Supply voltages: Reducing the supply voltages also reduces supply currents, and therefore results in significant power
savings, especially in the digital sections of the WM8971L.
Operating mode: Significant power savings can be achieved by always disabling parts of the WM8971L that are not
used (e.g. mic pre-amps, unused outputs, DAC, ADC, etc.)
Control Register
R23
Other settings
Tot. Power
Bit
VMI
D
SEL
VR
EF
AIN
L
AIN
R
A
DCL
A
DCR
MI
C
B
DA
CL
DA
CR
LOUT
1
RO
UT
1
LOUT
2
RO
UT
2
MO
N
O
A
DCO
S
R
DA
CO
S
R
VSEL
V
I (mA)
V
I (mA)
V
I (mA)
V
I (mA)
mW
OFF
00 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 11
Clocks stopped
3.3 0.0016 3.3 0.0190 3.3 0.0080 3.3 0.0002
0.0950
01
2.5 0.0008 2.5 0.0170 2.5 0.0050 2.5 0.0000
0.0570
00
1.8 0.0005 1.5 0.0120 1.5 0.0029 1.8 0.0000
0.0233
Standby
10 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 11
Interface Stopped
3.3 0.3900 3.3 0.0390 3.3 0.0080 3.3 0.0000
1.4421
(500 KOhm VMID string)
01
2.5 0.2880 2.5 0.0170 2.5 0.0050 2.5 0.0000
0.7750
00
1.8 0.1970 1.5 0.0120 1.5 0.0030 1.8 0.0000
0.3771
Playback to Line-out
01 1 0 0 0 0 0 1 1 0 0 1 1 0 0 0 11
3.3 3.7310 3.3 5.6600 3.3 0.3000 3.3 0.2370
32.7624
01
2.5 2.6940 2.5 3.8600 2.5 0.2200 2.5 0.2100
17.4600
00
1.8 1.8820 1.5 2.1400 1.5 0.1240 1.8 0.1500
7.0536
Playback to Line-out
01 1 0 0 0 0 0 1 1 1 1 0 0 0 0 1 11
3.3 3.5170 3.3 4.6470 3.3 0.3000 3.3 0.9500
31.0662
(64x oversampling mode)
01
2.5 2.5760 2.5 3.2030 2.5 0.2200 2.5 0.6480
16.6175
00
1.8 1.7760 1.5 1.7590 1.5 0.1240 1.8 0.4130
6.7647
Playback to 16 Ohm Headphone 01 1 0 0 0 0 0 1 1 1 1 0 0 0 0 0 11
3.3 3.7260 3.3 5.6700 3.3 0.3000 3.3 0.9530
35.1417
01
2.5 2.7530 2.5 3.9250 2.5 0.2200 2.5 0.6570
18.8875
00
1.8 1.8900 1.5 2.1410 1.5 0.1240 1.8 0.4150
7.5465
Playback to 8 Ohm BTL Speaker 01 1 0 0 0 0 0 1 1 0 0 1 1 0 0 0 11
R24, ROUT2INV=1
3.3 3.8820 3.3 5.6470 3.3 0.3000 3.3 0.2830
33.3696
01
2.5 2.8780 2.5 3.9390 2.5 0.2200 2.5 0.2100
18.1175
00
1.8 1.9800 1.5 2.1630 1.5 0.1240 1.8 0.1510
7.2663
Headphone Amp
01 1 0 0 0 0 0 0 0 1 1 0 0 0 0 0 11
Clocks Stopped
3.3 1.8400 3.3 0.0200 3.3 0.0080 3.3 0.9540
9.3126
(line-in to 16 Ohm headphone)
01
2.5 1.3300 2.5 0.0190 2.5 0.0050 2.5 0.6400
4.9850
00
1.8 0.9300 1.5 0.0130 1.5 0.0030 1.8 0.4100
2.4360
Speaker Amp
01 1 0 0 0 0 0 0 0 0 0 1 1 0 0 0 11
Clocks Stopped
3.3 1.9780 3.3 0.0200 3.3 0.0080 3.3 0.3310
7.7121
(line-in to 8 Ohm speaker)
01
R24, ROUT2INV=1
2.5 1.4300 2.5 0.0190 2.5 0.0050 2.5 0.2430
4.2425
00
1.8 0.9860 1.5 0.0130 1.5 0.0030 1.8 0.1760
2.1156
Phone Call
01 1 0 0 0 0 1 0 0 0 0 1 1 1 0 0 11
Clocks Stopped
3.3 2.5230 3.3 0.0370 3.3 0.0080 3.3 0.4420
9.9330
(mono line-in to headphone,
01
2.5 1.8520 2.5 0.0190 2.5 0.0050 2.5 0.3200
5.4900
mic to MONOOUT)
00
1.8 1.2900 1.5 0.0130 1.5 0.0030 1.8 0.2240
2.7492
Record from Line-in
01 1 1 1 1 1 0 0 0 0 0 0 0 0 0 0 11
3.3 8.6600 3.3 6.5700 3.3 0.3330 3.3 0.0000
51.3579
01
2.5 7.7100 2.5 4.2800 2.5 0.2320 2.5 0.0000
30.5550
00
1.8 6.8000 1.5 2.2100 1.5 0.1350 1.8 0.0000
15.7575
Record from Line-in
01 1 1 1 1 1 0 0 0 0 0 0 0 0 1 0 11
3.3 5.0720 3.3 5.9100 3.3 0.3390 3.3 0.0000
37.3593
(64x oversampling mode)
01
2.5 4.2550 2.5 3.7500 2.5 0.2320 2.5 0.0000
20.5925
00
1.8 3.5900 1.5 1.9100 1.5 0.1350 1.8 0.0000
9.5295
Record from mono microphone
01 1 1 0 1 0 1 0 0 0 0 0 0 0 0 0 11
3.3 4.9330 3.3 6.5400 3.3 0.3390 3.3 0.0000
38.9796
01
R23, DATSEL=01
2.5 4.2970 2.5 4.2500 2.5 0.2400 2.5 0.0000
21.9675
00
1.8 3.7210 1.5 2.2200 1.5 0.1370 1.8 0.0000
10.2333
Stereo Record & Playback
01 1 1 1 1 1 1 1 1 0 0 1 1 0 0 0 11
3.3 11.927 3.3 10.870 3.3 0.3320 3.3 0.2820
77.2563
01
2.5 10.112 2.5 7.3600 2.5 0.2340 2.5 0.2060
44.7800
00
1.8 7.3910 1.5 4.0610 1.5 0.1320 1.8 0.1480
19.8597
Stereo Record & Playback
01 1 1 1 1 1 1 1 1 0 0 1 1 0 1 1 11
3.3 8.1090 3.3 9.3300 3.3 0.3330 3.3 0.2820
59.5782
(64x oversampling mode)
01
2.5 6.5500 2.5 6.3020 2.5 0.2340 2.5 0.2070
33.2325
00
1.8 4.7000 1.5 3.3800 1.5 0.1320 1.8 0.1490
13.9962
DBVDD
HPVDD
R24
R25 (19h)
R26 (1Ah)
AVDD
DCVDD
Table 1 Supply Current Consumption
Notes:
1.
All figures are at T
A
= +25
o
C, Slave Mode, fs = 48kHz, MCLK = 12.288 MHz (256fs), with zero signal (quiescent)
2.
The power dissipated in the headphone or speaker is not included in the above table.
Production Data
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PD Rev 4.1 August 2005
13
SIGNAL TIMING REQUIREMENTS
SYSTEM CLOCK TIMING
MCLK
t
MCLKL
t
MCLKH
t
MCLKY
Figure 1 System Clock Timing Requirements
Test Conditions
CLKDIV2=0, DCVDD = 1.42V, DBVDD = 3.3V, DGND = 0V, T
A
= +25
o
C, Slave Mode fs = 48kHz, MCLK = 384fs, 24-bit data,
unless otherwise stated.
PARAMETER SYMBOL
MIN
TYP
MAX
UNIT
System Clock Timing Information
MCLK System clock pulse width high
T
MCLKL
21
ns
MCLK System clock pulse width low
T
MCLKH
21
ns
MCLK System clock cycle time
T
MCLKY
54
ns
MCLK duty cycle
T
MCLKDS
60:40
40:60
Test Conditions
CLKDIV2=1, DCVDD = 1.42V, DBVDD = 3.3V, DGND = 0V, T
A
= +25
o
C, Slave Mode fs = 48kHz, MCLK = 384fs, 24-bit data,
unless otherwise stated.
PARAMETER SYMBOL
MIN
TYP
MAX
UNIT
System Clock Timing Information
MCLK System clock pulse width high
T
MCLKL
10
ns
MCLK System clock pulse width low
T
MCLKH
10
ns
MCLK System clock cycle time
T
MCLKY
27
ns
AUDIO INTERFACE TIMING MASTER MODE
BCLK
(Output)
ADCDAT
ADCLRC/
DACLRC
(Outputs)
t
DL
DACDAT
t
DDA
t
DHT
t
DST
Figure 2 Digital Audio Data Timing Master Mode (see Control Interface)
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Figure 3 Bit Clock Mode
Test Conditions
DCVDD = 1.42V, DBVDD = 3.3V, DGND = 0V, T
A
= +25
o
C, Slave Mode, fs = 48kHz, MCLK = 256fs, 24-bit data, unless
otherwise stated.
PARAMETER SYMBOL
MIN
TYP
MAX
UNIT
Bit Clock Timing Information
BCLK rise time (10pF load)
t
BCLKR
3
ns
BCLK fall time (10pF load)
t
BCLKF
3
ns
BCLK duty cycle (normal mode, BCLK = MCLK/n)
t
BCLKDS
50:50
BCLK duty cycle (USB mode, BCLK = MCLK)
t
BCLKDS
T
MCLKDS
Audio Data Input Timing Information
ADCLRC/DACLRC propagation delay from BCLK falling edge
t
DL
10
ns
ADCDAT propagation delay from BCLK falling edge
t
DDA
10
ns
DACDAT setup time to BCLK rising edge
t
DST
10
ns
DACDAT hold time from BCLK rising edge
t
DHT
10
ns
AUDIO INTERFACE TIMING SLAVE MODE
BCLK
DACLRC/
ADCLRC
t
BCH
t
BCL
t
BCY
DACDAT
ADCDAT
t
LRSU
t
DS
t
LRH
t
DH
t
DD
Figure 4 Digital Audio Data Timing Slave Mode
Test Conditions
DCVDD = 1.42V, DBVDD = 3.3V, DGND = 0V, T
A
= +25
o
C, Slave Mode, fs = 48kHz, MCLK = 256fs, 24-bit data, unless
otherwise stated.
PARAMETER SYMBOL
MIN
TYP
MAX
UNIT
Audio Data Input Timing Information
BCLK cycle time
t
BCY
50
ns
BCLK pulse width high
t
BCH
20
ns
BCLK pulse width low
t
BCL
20
ns
ADCLRC/DACLRC set-up time to BCLK rising edge
t
LRSU
10
ns
ADCLRC/DACLRC hold time from BCLK rising edge
t
LRH
10
ns
DACDAT hold time from BCLK rising edge
t
DH
10
ns
ADCDAT propagation delay from BCLK falling edge
t
DD
10
ns
Note:
BCLK period should always be greater than or equal to MCLK period.
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CONTROL INTERFACE TIMING 3-WIRE MODE
CSB
SCLK
SDIN
t
CSL
t
DHO
t
DSU
t
CSH
t
SCY
t
SCH
t
SCL
t
SCS
LSB
t
CSS
Figure 5 Control Interface Timing 3-Wire Serial Control Mode
Test Conditions
DCVDD = 1.42V, DBVDD = 3.3V, DGND = 0V, T
A
= +25
o
C, Slave Mode, fs = 48kHz, MCLK = 256fs, 24-bit data, unless
otherwise stated.
PARAMETER SYMBOL
MIN
TYP
MAX
UNIT
Program Register Input Information
SCLK rising edge to CSB rising edge
t
SCS
80
ns
SCLK pulse cycle time
t
SCY
200
ns
SCLK pulse width low
t
SCL
80
ns
SCLK pulse width high
t
SCH
80
ns
SDIN to SCLK set-up time
t
DSU
40
ns
SCLK to SDIN hold time
t
DHO
40
ns
CSB pulse width low
t
CSL
40
ns
CSB pulse width high
t
CSH
40
ns
CSB rising to SCLK rising
t
CSS
40
ns
Pulse width of spikes that will be suppressed
t
PS
0
5
ns
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CONTROL INTERFACE TIMING 2-WIRE MODE
SDIN
SCLK
t
3
t
1
t
6
t
2
t
7
t
5
t
4
t
3
t
8
t
9
Figure 6 Control Interface Timing 2-Wire Serial Control Mode
Test Conditions
DCVDD = 1.42V, DBVDD = 3.3V, DGND = 0V, T
A
= +25
o
C, Slave Mode, fs = 48kHz, MCLK = 256fs, 24-bit data, unless
otherwise stated.
PARAMETER SYMBOL
MIN
TYP
MAX
UNIT
Program Register Input Information
SCLK Frequency
0
400
kHz
SCLK Low Pulse-Width
t
1
1.3
us
SCLK High Pulse-Width
t
2
600
ns
Hold Time (Start Condition)
t
3
600
ns
Setup Time (Start Condition)
t
4
600
ns
Data Setup Time
t
5
100
ns
SDIN, SCLK Rise Time
t
6
300
ns
SDIN, SCLK Fall Time
t
7
300
ns
Setup Time (Stop Condition)
t
8
600
ns
Data Hold Time
t
9
900
ns
Pulse width of spikes that will be suppressed
t
ps
0
5
ns
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INTERNAL POWER ON RESET CIRCUIT
VDD
T1
GND
AVDD
DCVDD
DGND
Internal PORB
Power on Reset
Circuit
Figure 7 Internal Power on Reset Circuit Schematic
The WM8971 includes an internal Power-On-Reset Circuit, as shown in Figure 7, which is used to
reset the digital logic into a default state after power up. The power on reset circuit is powered from
DCVDD and monitors DCVDD and AVDD. It asserts PORB low if DCVDD or AVDD are below a
minimum threshold.
Figure 8 Typical Power-Up Sequence
Figure 8 shows a typical power-up sequence. When DCVDD and AVDD go above the minimum
thresholds, Vpord_dcvdd and Vpord_avdd, there is enough voltage for the circuit to guarantee the
Power on Reset is asserted low and the chip is held in reset. In this condition, all writes to the control
interface are ignored. When DCVDD rises to Vpor_dcvdd_on and AVDD rises to Vpor_avdd_on,
PORB is released high and all registers are in their default state and writes to the control interface
may take place. If DCVDD and AVDD rise at different rates then PORB will only be released when
DCVDD and AVDD have both exceeded the Vpor_dcvdd_on and Vpor_avdd_on thresholds.
On power down, PORB is asserted low whenever DCVDD drops below the minimum threshold
Vpor_dcvdd_off or AVDD drops below the minimum threshold Vpor_avdd_off.
SYMBOL MIN TYP MAX UNIT
V
pord_dcvdd
0.4 0.6 0.8 V
V
por_dcvdd_on
0.9 1.26 1.6 V
V
por_avdd_on
0.5 0.7 0.9 V
V
por_avdd_off
0.4
0.6
0.8
V
Table 2 Typical POR Operation (typical values, not tested)
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DEVICE DESCRIPTION
INTRODUCTION
The WM8971L is a low power audio codec offering a combination of high quality audio, advanced
features, low power and small size. These characteristics make it ideal for portable digital audio
applications such as Digital Still Cameras, MP3 and minidisk player / recorders. Stereo 24-bit multi-
bit delta sigma ADCs and DACs are used with oversampling digital interpolation and decimation
filters.
The device includes a stereo analogue input which can be switched internally. This input can be used
as either a line level input, as a microphone input or be selected as a mono differential input. A mono
input is also included which can be configured as a microphone or line input.
When recording a programmable gain amplifier with automatic level control (ALC) keeps the
recording volume constant. The on-chip stereo ADC and DAC are of a high quality using a multi-bit,
low-order oversampling architecture to deliver optimum performance with low power consumption.
The DAC output signal first enters an analogue mixer where an analogue input and/or the post-ALC
signal can be added to it. This mix is available on line and headphone outputs.
The WM8971L has a configurable digital audio interface where ADC data can be read and digital
audio playback data fed to the DAC. It supports a number of audio data formats including I
2
S, DSP
Mode (a burst mode in which frame sync plus 2 data packed words are transmitted), MSB-First, left
justified and MSB-First, right justified, and can operate in master or slave modes.
The WM8971L uses a unique clocking scheme that can generate many commonly used audio
sample rates from either a 12.00MHz USB clock or an industry standard 256/384 f
s
clock. This
feature eliminates the common requirement for an external phase-locked loop (PLL) in applications
where the master clock is not an integer multiple of the sample rate. Sample rates of 8kHz,
11.025kHz, 12kHz, 16kHz, 22.05kHz, 24kHz, 32kHz, 44.1kHz, 48kHz, 88.2kHz and 96kHz can be
generated. The digital filters used for recording and playback are optimised for each sampling rate
used.
To allow full software control over all its features, the WM8971L offers a choice of 2 or 3 wire MPU
control interface. It is fully compatible and an ideal partner for a wide range of industry standard
microprocessors, controllers and DSPs.
The design of the WM8971L has given much attention to power consumption without compromising
performance. It operates at very low voltages, and includes the ability to power off parts of the
circuitry under software control, including standby and power off modes.
INPUT SIGNAL PATH
The input signal path for each channel consists of a switch to select between three analogue inputs,
followed by a PGA (programmable gain amplifier) and an optional microphone gain boost. A
differential input of (LINPUT1 RINPUT1) may also be selected. The gain of the PGA can be
controlled either by the user or by the on-chip ALC function (see Automatic Level Control).
The signal then enters an ADC where it is digitised. Alternatively, the two channels can also be
mixed in the analogue domain and digitised in one ADC while the other ADC is switched off. The
mono-mix signal appears on both digital output channels.
SIGNAL INPUTS
The WM8971L has three high impedance, low capacitance AC coupled analogue inputs, LINPUT1,
RINPUT1, and MIC. The LINSEL and RINSEL control bits select between them. These inputs can be
configured as microphone or line inputs by enabling or disabling the microphone gain boost. A
differential input, LINPUT1-RINPUT1 may also be selected using L/RINSEL.
The signal inputs are biased internally to the reference voltage VREF. Whenever the line inputs are
muted or the device placed into standby mode, the inputs are kept biased to VREF using special
anti-thump circuitry. This reduces any audible clicks that may otherwise be heard when changing
inputs.
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REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
7:6
LINSEL
00
Left Channel Input Select
00 = LINPUT1
11 = Differential
R32 (20h)
ADC Signal
Path Control
(Left)
5:4
LMICBOOST
00
Left Channel Microphone Gain Boost
00 = Boost off (bypassed)
01 = 13dB boost
10 = 20dB boost
11 = 29dB boost
7:6
RINSEL
00
Right Channel Input Select
00 = RINPUT1
01 = MIC
11 = Differential
R33 (21h)
ADC Signal
Path Control
(Right)
5:4
RMICBOOST
00
Right Channel Microphone Gain Boost
00 = Boost off (bypassed)
01 = 13dB boost
10 = 20dB boost
11 = 29dB boost
Table 3 Input Software Control
MONO MIXING
The stereo ADC can operate as a stereo or mono device, or the two channels can be mixed to mono,
either in the analogue domain (i.e. before the ADC) or in the digital domain (after the ADC).
MONOMIX selects the mode of operation. The user also has the flexibility to select the data output
from the audio interface using DATSEL. The default is for left and right channel ADC data to be
output, but the interface may also be configured so that e.g. left channel ADC data is output as both
left and right data for when an analogue mono mix is selected.
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
R31 (1Fh)
ADC input
Mode
7:6
MONOMIX
[1:0]
00 00:
Stereo
01: Analogue Mono Mix (using left ADC)
10: Analogue Mono Mix (using right ADC)
11: Digital Mono Mix
Table 4 Mono Mixing
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
R23 (17h)
Additional
Control (1)
3:2
DATSEL
[1:0]
00
00: left data=left ADC; right data =right ADC
01: left data =left ADC; right data = left ADC
10: left data = right ADC; right data =right ADC
11: left data = right ADC; right data = left ADC
Table 5 ADC Data Output Configuration
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The MICBIAS output provides a low noise reference voltage suitable for biasing electret type
microphones and the associated external resistor biasing network. Refer to the Applications
Information section for recommended external components. The output can be enabled or disables
using the MICB control bit (see also the "Power Management" section).
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
R25 (19h)
Power
Management (1)
1
MICB
0
Microphone Bias Enable
0 = OFF (high impedance output)
1 = ON
Table 6 Microphone Bias Control
The internal MICBIAS circuitry is shown below. Note that the is a maximum source current capability
for MICBIAS is 3mA. The external biasing resistors therefore must be large enough to limit the
MICBIAS current to 3mA.
AGND
MICBIAS
= 1.8 x VMID
= 0.9 X AVDD
VMID
internal
resistor
internal
resistor
MICB
Figure 9 Microphone Bias Schematic
PGA CONTROL
The PGA matches the input signal level to the ADC input range. The PGA gain is logarithmically
adjustable from +30dB to 17.25dB in 0.75dB steps. Each PGA can be controlled either by the user
or by the ALC function (see Automatic Level Control). When ALC is enabled for one or both
channels, then writing to the corresponding PGA control register has no effect.
The gain is independently adjustable on both Right and Left Line Inputs. However, by setting the
LIVU or RIVU bits whilst programming the PGA gain, both channels are simultaneously updated.
This reduces the required number of software writes required. Setting the LZCEN and RZCEN bits
enables a zero-cross detector which ensures that PGA gain changes only occur when the signal is at
zero, eliminating any zipper noise. If zero cross is enabled a timeout is also available to update the
gain if a zero cross does not occur. This function may be enabled by setting TOEN in register R23
(17h).
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The inputs can also be muted in the analogue domain under software control. The software control
registers are shown in Table 7. If zero crossing is enabled, it is necessary to enable zero cross
timeout to un-mute the input PGAs. This is because their outputs will not cross zero when muted.
Alternatively, zero cross can be disabled before sending the un-mute command.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
8
LIVU
0
Left Volume Update
0 = Store LINVOL in intermediate
latch (no gain change)
1 = Update left and right channel
gains (left = LINVOL, right =
intermediate latch)
7
LINMUTE
1
Left Channel Input Analogue Mute
1 = Enable Mute
0 = Disable Mute
Note: LIVU must be set to un-mute.
6
LZCEN
0
Left Channel Zero Cross Detector
1 = Change gain on zero cross only
0 = Change gain immediately
R0 (00h)
Left Channel
PGA
5:0
LINVOL
[5:0]
010111
( 0dB )
Left Channel Input Volume Control
111111 = +30dB
111110 = +29.25dB
. . 0.75dB steps down to
000000 = -17.25dB
8 RIVU
0
Right
Volume
Update
0 = Store RINVOL in intermediate
latch (no gain change)
1 = Update left and right channel
gains (right = RINVOL, left =
intermediate latch)
7
RINMUTE
1
Right Channel Input Analogue Mute
1 = Enable Mute
0 = Disable Mute
Note: RIVU must be set to un-mute.
6
RZCEN
0
Right Channel Zero Cross Detector
1 = Change gain on zero cross only
0 = Change gain immediately
R1 (01h)
Right Channel
PGA
5:0 RINVOL
[5:0]
010111
( 0dB )
Right Channel Input Volume Control
111111 = +30dB
111110 = +29.25dB
. . 0.75dB steps down to
000000 = -17.25dB
R23 (17h)
Additional
Control (1)
0 TOEN
0
Timeout
Enable
0 : Timeout Disabled
1 : Timeout Enabled
Table 7 Input PGA Software Control
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ANALOGUE TO DIGITAL CONVERTER (ADC)
The WM8971L uses a multi-bit, oversampled sigma-delta ADC for each channel. The use of multi-bit
feedback and high oversampling rates reduces the effects of jitter and high frequency noise. The
ADC Full Scale input level is proportional to AVDD. With a 3.3V supply voltage, the full scale level is
1.0 Volts r.m.s. Any voltage greater than full scale may overload the ADC and cause distortion.
ADC DIGITAL FILTER
The ADC filters perform true 24 bit signal processing to convert the raw multi-bit oversampled data
from the ADC to the correct sampling frequency to be output on the digital audio interface. The digital
filter path is illustrated in Figure 10.
FROM ADC
DIGITAL
HPF
DIGITAL
FILTER
TO DIGITAL
AUDIO
INTERFACE
DIGITAL
DECIMATOR
ADCHPD
Figure 10 ADC Digital Filter
The ADC digital filters contain a digital high pass filter, selectable via software control. The high-pass
filter response is detailed in the Digital Filter Characteristics section. When the high-pass filter is
enabled the dc offset is continuously calculated and subtracted from the input signal. By setting
HPOR, the last calculated dc offset value is stored when the high-pass filter is disabled and will
continue to be subtracted from the input signal. If the DC offset is changed, the stored and
subtracted value will not change unless the high-pass filter is enabled. This feature can be used for
calibration purposes. In addition the high-pass filter may be enabled separately on the left and right
channels (see Table 9).
The output data format can be programmed by the user to accommodate stereo or monophonic
recording on both inputs. The polarity of the output signal can also be changed under software
control. The software control is shown in Table 8.
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REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
6:5 ADCPOL
[1:0]
00
00 = Polarity not inverted
01 = L polarity invert
10 = R polarity invert
11 = L and R polarity invert
4
HPOR
0
Store dc offset when high-pass
Filter disabled
1 = store offset
0 = clear offset
ADC high-pass filter enable
(Digital)
HPFLREN = 0
1 = Disable High-pass filter on left
and right channels
0 = Enable High-pass filter on left
and right channels
R5 (05h)
ADC and DAC
Control
0 ADCHPD 0
HPFLREN = 1
0 = High-pass enabled on left,
disabled on right
1 = High-pass enabled on right,
disabled on left
R27 (1Bh)
5
HPFLREN
0
ADC high-pass filter left or right
enable
0 = High-pass filter enable/disable
on left and right channels
controlled by ADCHPD
1 = High-pass filter enabled on left
or right channel, as selected by
ADCHPD
Table 8 ADC Signal Path Control
HPFLREN ADCHPD
HIGH
PASS
MODE
0 0
High-pass filter enabled on left and right
channels
0 1
High-pass filter disabled on left and right
channels
1 0
High-pass filter enabled on left channel,
disabled on right channel
1 1
High-pass filter disabled on left channel,
enabled on right channel
Table 9 ADC High Pass Filter Enable Modes
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DIGITAL ADC VOLUME CONTROL
The output of the ADCs can be digitally amplified or attenuated over a range from 97dB to +30dB in
0.5dB steps. The volume of each channel can be controlled separately. The gain for a given eight-bit
code X is given by:
0.5
(X-195) dB for 1 X 255;
MUTE for X = 0
The LAVU and RAVU control bits control the loading of digital volume control data. When LAVU or
RAVU are set to 0, the LADCVOL or RADCVOL control data will be loaded into the respective control
register, but will not actually change the digital gain setting. Both left and right gain settings are
updated when either LAVU or RAVU are set to 1. This makes it possible to update the gain of both
channels simultaneously.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
7:0 LADCVOL
[7:0]
11000011
( 0dB )
Left ADC Digital Volume Control
0000 0000 = Digital Mute
0000 0001 = -97dB
0000 0010 = -96.5dB
... 0.5dB steps up to
1111 1111 = +30dB
R21 (15h)
Left ADC
Digital Volume
8
LAVU
0
Left ADC Volume Update
0 = Store LADCVOL in intermediate
latch (no gain change)
1 = Update left and right channel
gains (left = LADCVOL, right =
intermediate latch)
7:0 RADCVOL
[7:0]
11000011
( 0dB )
Right ADC Digital Volume Control
0000 0000 = Digital Mute
0000 0001 = -97dB
0000 0010 = -96.5dB
... 0.5dB steps up to
1111 1111 = +30dB
R22 (16h)
Right ADC
Digital Volume
8
RAVU
0
Right ADC Volume Update
0 = Store RADCVOL in intermediate
latch (no gain change)
1 = Update left and right channel
gains (left = intermediate latch, right
= RADCVOL)
Table 10 ADC Digital Volume Control
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AUTOMATIC LEVEL CONTROL (ALC)
The WM8971L has an automatic level control that aims to keep a constant recording volume
irrespective of the input signal level. This is achieved by continuously adjusting the PGA gain so that
the signal level at the ADC input remains constant. A digital peak detector monitors the ADC output
and changes the PGA gain if necessary. Note that when the ALC function is enabled, the settings of
registers 0 and 1 (LINVOL, LIVU, LIZC, LINMUTE, RINVOL, RIVU, RIZC and RINMUTE) are
ignored.
hold
time
decay
time
attack
time
input
signal
signal
after
ALC
PGA
gain
ALC
target
level
Figure 11 ALC Operation
The ALC function is enabled using the ALCSEL control bits. When enabled, the recording volume
can be programmed between 6dB and 28.5dB (relative to ADC full scale) using the ALCL register
bits. An upper limit for the PGA gain can be imposed by setting the MAXGAIN control bits.
HLD, DCY and ATK control the hold, decay and attack times, respectively:
Hold time is the time delay between the peak level detected being below target and the PGA gain
beginning to ramp up. It can be programmed in power-of-two (2
n
) steps, e.g. 2.67ms, 5.33ms,
10.67ms etc. up to 43.7s. Alternatively, the hold time can also be set to zero. The hold time only
applies to gain ramp-up, there is no delay before ramping the gain down when the signal level is
above target.
Decay (Gain Ramp-Up) Time is the time that it takes for the PGA gain to ramp up across 90% of its
range (e.g. from 15B up to 27.75dB). The time it takes for the recording level to return to its target
value therefore depends on both the decay time and on the gain adjustment required. If the gain
adjustment is small, it will be shorter than the decay time. The decay time can be programmed in
power-of-two (2
n
) steps, from 24ms, 48ms, 96ms, etc. to 24.58s.
Attack (Gain Ramp-Down) Time is the time that it takes for the PGA gain to ramp down across 90%
of its range (e.g. from 27.75dB down to -15B gain). The time it takes for the recording level to return
to its target value therefore depends on both the attack time and on the gain adjustment required. If
the gain adjustment is small, it will be shorter than the attack time. The attack time can be
programmed in power-of-two (2
n
) steps, from 6ms, 12ms, 24ms, etc. to 6.14s.
When operating in stereo, the peak detector takes the maximum of left and right channel peak
values, and any new gain setting is applied to both left and right PGAs, so that the stereo image is
preserved. However, the ALC function can also be enabled on one channel only. In this case, only
one PGA is controlled by the ALC mechanism, while the other channel runs independently with its
PGA gain set through the control register.
When one ADC channel is unused, the peak detector disregards that channel. The ALC function can
also operate when the two ADC outputs are mixed to mono in the digital domain, but not if they are
mixed to mono in the analogue domain, before entering the ADCs.
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REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
8:7 ALCSEL
[1:0]
00
(OFF)
ALC function select
00 = ALC off (PGA gain set by register)
01 = Right channel only
10 = Left channel only
11 = Stereo (PGA registers unused)
Note: ensure that LINVOL and
RINVOL settings (reg. 0 and 1) are
the same before entering this mode.
6:4 MAXGAIN
[2:0]
111
(+30dB)
Set Maximum Gain of PGA
111 : +30dB
110 : +24dB
....(-6dB steps)
001 : -6dB
000 : -12dB
R17 (11h)
ALC Control 1
3:0 ALCL
[3:0]
1011
(-12dB)
ALC target sets signal level at ADC
input
0000 = -28.5dB FS
0001 = -27.0dB FS
... (1.5dB steps)
1110 = -7.5dB FS
1111 = -6dB FS
7 ALCZC
0
(zero
cross
off)
ALC uses zero cross detection circuit.
R18 (12h)
ALC Control 2
3:0 HLD
[3:0]
0000
(0ms)
ALC hold time before gain is increased.
0000 = 0ms
0001 = 2.67ms
0010 = 5.33ms
... (time doubles with every step)
1111 = 43.691s
7:4 DCY
[3:0]
0011
(192ms)
ALC decay (gain ramp-up) time
0000 = 24ms
0001 = 48ms
0010 = 96ms
... (time doubles with every step)
1010 or higher = 24.58s
R19 (13h)
ALC Control 3
3:0 ATK
[3:0]
0010
(24ms)
ALC attack (gain ramp-down) time
0000 = 6ms
0001 = 12ms
0010 = 24ms
... (time doubles with every step)
1010 or higher = 6.14s
Table 11 ALC Control
PEAK LIMITER
To prevent clipping when a large signal occurs just after a period of quiet, the ALC circuit includes a
limiter function. If the ADC input signal exceeds 87.5% of full scale (1.16dB), the PGA gain is
ramped down at the maximum attack rate (as when ATK = 0000), until the signal level falls below
87.5% of full scale. This function is automatically enabled whenever the ALC is enabled.
Note:
If ATK = 0000, then the limiter makes no difference to the operation of the ALC. It is designed to
prevent clipping when long attack times are used.
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NOISE GATE
When the signal is very quiet and consists mainly of noise, the ALC function may cause "noise
pumping", i.e. loud hissing noise during silence periods. The WM8971L has a noise gate function
that prevents noise pumping by comparing the signal level at the LINPUT1/2/3 and/or RINPUT1/2/3
pins against a noise gate threshold, NGTH. The noise gate cuts in when:
Signal level at ADC [dB] < NGTH [dB] + PGA gain [dB] + Mic Boost gain [dB]
This is equivalent to:
Signal level at input pin [dB] < NGTH [dB]
The ADC output can then either be muted or digitally attenuated by 18dB. Alternatively, the PGA gain
can be held constant (preventing it from ramping up as it normally would when the signal is quiet).
The table below summarises the noise gate control register. The NGTH control bits set the noise
gate threshold with respect to the ADC full-scale range. The threshold is adjusted in 1.5dB steps.
Levels at the extremes of the range may cause inappropriate operation, so care should be taken with
setup of the function. Note that the noise gate only works in conjunction with the ALC function, and
always operates on the same channel(s) as the ALC (left, right, both, or none).
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
7:3
NGTH
[4:0]
00000
Noise gate threshold
00000 -76.5dBfs
00001 -75dBfs
... 1.5 dB steps
11110 -31.5dBfs
11111 -30dBfs
2:1
NGG
[1:0]
00
Noise gate type
X0 = PGA gain held constant
01 = mute ADC output
10 = reserved (do not use this setting)
11 = reserved (do not use this setting)
R20 (14h)
Noise Gate
Control
0
NGAT
0
Noise gate function enable
1 = enable
0 = disable
Table 12 Noise Gate Control
Note:
The performance of the ADC may degrade at high input signal levels if the monitor bypass mux is
selected with MIC boost and ALC enabled.
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
8
ADCDIV2
0
ADC 6dB attenuate enable
0 = disabled (0dB)
1 = -6dB enabled
R5 (05h)
ADC and DAC
control
7
DACDIV2
0
DAC 6dB attenuate enable
0 = disabled (0dB)
1 = -6dB enabled
Table 13 ADC and DAC 6dB Attenuation Select
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OUTPUT SIGNAL PATH
The WM8971L output signal paths consist of digital filters, DACs, analogue mixers and output
drivers. The digital filters and DACs are enabled when the WM8971L is in `playback only' or `record
and playback' mode. The mixers and output drivers can be separately enabled by individual control
bits (see Analogue Outputs). Thus it is possible to utilise the analogue mixing and amplification
provided by the WM8971L, irrespective of whether the DACs are running or not.
The WM8971L receives digital input data on the DACDAT pin. The digital filter block processes the
data to provide the following functions:
Digital volume control
Graphic equaliser and Dynamic Bass Boost
Sigma-Delta
Modulation
Two high performance sigma-delta audio DACs convert the digital data into two analogue signals (left
and right). These can then be mixed with analogue signals from the LINPUT1/2/3 and RINPUT1/2/3
pins, and the mix is fed to the output drivers, LOUT1/ROUT1, LOUT2/ROUT2 and MONOOUT.
LOUT1/ROUT1: can drive a 16
or 32 stereo headphone or stereo line output.
LOUT2/ROUT2: can drive a 16
or 32 stereo headphone or stereo line output, or an
8
mono speaker.
MONOOUT: can drive a mono line output or other load down to 10k
DIGITAL DAC VOLUME CONTROL
The signal volume from each DAC can be controlled digitally, in the same way as the ADC volume
(see Digital ADC Volume Control). The gain and attenuation range is 127dB to 0dB in 0.5dB steps.
The level of attenuation for an eight-bit code X is given by:
0.5
(X-255) dB for 1 X 255;
MUTE for X = 0
The LDVU and RDVU control bits control the loading of digital volume control data. When LDVU or
RDVU are set to 0, the LDACVOL or RDACVOL control data is loaded into an intermediate register,
but the actual gain does not change. Both left and right gain settings are updated simultaneously
when either LDVU or RDVU are set to 1.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
8
LDVU
0
Left DAC Volume Update
0 = Store LDACVOL in intermediate
latch (no gain change)
1 = Update left and right channel
gains (left = LDACVOL, right =
intermediate latch)
R10 (0Ah)
Left Channel
Digital Volume
7:0 LDACVOL
[7:0]
11111111
( 0dB )
Left DAC Digital Volume Control
0000 0000 = Digital Mute
0000 0001 = -127dB
0000 0010 = -126.5dB
... 0.5dB steps up to
1111 1111 = 0dB
8
RDVU
0
Right DAC Volume Update
0 = Store RDACVOL in intermediate
latch (no gain change)
1 = Update left and right channel
gains (left = intermediate latch, right
= RDACVOL)
R11 (0Bh)
Right Channel
Digital Volume
7:0 RDACVOL
[7:0]
11111111
( 0dB )
Right DAC Digital Volume Control
similar to LDACVOL
Table 14 Digital Volume Control
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GRAPHIC EQUALISER
The WM8971L has a digital graphic equaliser and adaptive bass boost function. This function
operates on digital audio data before it is passed to the audio DACs. Bass enhancement can take
two different forms:
Linear bass control: bass signals are amplified or attenuated by a user programmable
gain. This is independent of signal volume, and very high bass gains on loud signals
may lead to signal clipping.
Adaptive bass boost: The bass volume is amplified by a variable gain. When the bass
volume is low, it is boosted more than when the bass volume is high. This method is
recommended because it prevents clipping, and usually sounds more pleasant to the
human ear.
Treble control applies a user programmable gain, without any adaptive boost function. Bass and
treble control are completely independent with separately programmable gains and filter
characteristics.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
7 BB
0
Bass
Boost
0 = Linear bass control
1 = Adaptive bass boost
6
BC
0
Bass Filter Characteristic
0 = Low Cutoff (130Hz at 48kHz sampling)
1 = High Cutoff (200Hz at 48kHz sampling)
Bass Intensity
Code BB=0 BB=1
0000 +9dB 15
(max)
0001 +9dB 14
0010 +7.5dB
13
0011 +6dB 12
0100 +4.5dB
11
0101 +3dB 10
0110 +1.5dB
9
0111 0dB 8
1000 -1.5dB
7
1001 -3dB 6
1010 -4.5dB
5
1011 -6dB 4
1100 -6dB 3
1101 -6dB 2
1110 -6dB 1
R12 (0Ch)
Bass Control
3:0 BASS
[3:0]
1111
(Disabled)
1111 Bypass
(OFF)
6 TC
0
Treble
Filter
Characteristic
0 = High Cutoff (8kHz at 48kHz sampling)
1 = Low Cutoff (4kHz at 48kHz sampling)
R13 (0Dh)
Treble Control
3:0 TRBL
[3:0]
1111
(Disabled)
Treble Intensity
0000 or 0001 = +9dB
0010 = +7.5dB
... (1.5dB steps)
1011 to 1110 = -6dB
1111 = Disable
Table 15 Graphic Equaliser
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DIGITAL TO ANALOGUE CONVERTER (DAC)
After passing through the graphic equaliser filters, digital `de-emphasis' can be applied to the audio
data if necessary (e.g. when the data comes from a CD with pre-emphasis used in the recording).
De-emphasis filtering is available for sample rates of 48kHz, 44.1kHz and 32kHz.
The WM8971L also has a Soft Mute function, which gradually attenuates the volume of the digital
signal to zero. When removed, the gain will ramp back up to the digital gain setting. This function is
enabled by default. To play back an audio signal, it must first be disabled by setting the DACMU bit
to zero.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
2:1
DEEMP
[1:0]
00 De-emphasis
Control
11 = 48kHz sample rate
10 = 44.1kHz sample rate
01 = 32kHz sample rate
00 = No De-emphasis
R5 (05h)
ADC and DAC
Control
3
DACMU
1
Digital Soft Mute
1 = mute
0 = no mute (signal active)
Table 16 DAC Control
The digital audio data is converted to oversampled bit streams in the on-chip, true 24-bit digital
interpolation filters. The bitstream data enters two multi-bit, sigma-delta DACs, which convert them to
high quality analogue audio signals. The multi-bit DAC architecture reduces high frequency noise and
sensitivity to clock jitter. It also uses a Dynamic Element Matching technique for high linearity and
low distortion.
In normal operation, the left and right channel digital audio data is converted to analogue in two
separate DACs. However, it is also possible to disable one channel, so that the same signal (left or
right) appears on both analogue output channels. Additionally, there is a mono-mix mode where the
two audio channels are mixed together digitally and then converted to analogue using only one DAC,
while the other DAC is switched off. The mono-mix signal can be selected to appear on both
analogue output channels.
The DAC output defaults to non-inverted. Setting DACINV will invert the DAC output phase on both
left and right channels.
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
5:4
DMONOMIX
[1:0]
00
DAC mono mix
00: stereo
01: mono ((L+R)/2) into DACL, `0' into
DACR
10: mono ((L+R)/2) into DACR, `0' into
DACL
11: mono ((L+R)/2) into DACL and
DACR
R23 (17h)
Additional
Control (1)
1
DACINV 0
DAC
phase
invert
0 : non-inverted
1 : inverted
Table 17 DAC Mono Mix and Phase Invert Select
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OUTPUT MIXERS
The WM8971L provides the option to mix the DAC output signal with analogue line-in signals from
the LINPUT1, RINPUT1, MIC pins or a mono differential input (LINPUT1 RINPUT1). The level of
the mixed-in signals can be controlled with PGAs (Programmable Gain Amplifiers).
The mono mixer is designed to allow a number of signal combinations to be mixed, including the
possibility of mixing both the right and left channels together to produce a mono output. To prevent
overloading of the mixer when full-scale DAC left and right signals are input, the mixer inputs from
the DAC outputs each have a fixed gain of -6dB. The bypass path inputs to the mono mixer have
variable gain as determined by R38/R39 bits [6:4].
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
R34 (22h)
Left Mixer (1)
2:0
LMIXSEL
000
Left Input Selection for Output Mix
000 = LINPUT1
011 = Left ADC Input (after PGA /
MICBOOST)
100 = Differential input
R36 (24h)
Right Mixer
(1)
2:0
RMIXSEL
000
Right Input Selection for Output Mix
000 = RINPUT1
001 = MIC
010 = Reserved (do not use)
011 = Right ADC Input (after PGA /
MICBOOST)
100 = Differential input
Table 18 Output Mixer Signal Selection
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
8
LD2LO
0
Left DAC to Left Mixer
0 = Disable (Mute)
1 = Enable Path
7
LI2LO
0
LMIXSEL Signal to Left Mixer
0 = Disable (Mute)
1 = Enable Path
R34 (22h)
Left Mixer
Control (1)
6:4 LI2LOVOL
[2:0]
101
(-9dB)
LMIXSEL Signal to Left Mixer Volume
000 = +6dB
... (3dB steps)
111 = -15dB
8
RD2LO
0
Right DAC to Left Mixer
0 = Disable (Mute)
1 = Enable Path
7
RI2LO
0
RMIXSEL Signal to Left Mixer
0 = Disable (Mute)
1 = Enable Path
R35 (23h)
Left Mixer
Control (2)
6:4 RI2LOVOL
[2:0]
101
(-9dB)
RMIXSEL Signal to Left Mixer Volume
000 = +6dB
... (3dB steps)
111 = -15dB
Table 19 Left Output Mixer Control
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REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
8
LD2RO
0
Left DAC to Right Mixer
0 = Disable (Mute)
1 = Enable Path
7
LI2RO
0
LMIXSEL Signal to Right Mixer
0 = Disable (Mute)
1 = Enable Path
R36 (24h)
Right Mixer
Control (1)
6:4 LI2ROVOL
[2:0]
101
(-9dB)
LMIXSEL Signal to Right Mixer Volume
000 = +6dB
... (3dB steps)
111 = -15dB
8
RD2RO
0
Right DAC to Right Mixer
0 = Disable (Mute)
1 = Enable Path
7
RI2RO
0
RMIXSEL Signal to Right Mixer
0 = Disable (Mute)
1 = Enable Path
R37 (25h)
Right Mixer
Control (2)
6:4 RI2ROVOL
[2:0]
101
(-9dB)
RMIXSEL Signal to Right Mixer Volume
000 = +6dB
... (3dB steps)
111 = -15dB
Table 20 Right Output Mixer Control
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
8
LD2MO
0
Left DAC to Mono Mixer
0 = Disable (Mute)
1 = Enable Path
7
LI2MO
0
LMIXSEL Signal to Mono Mixer
0 = Disable (Mute)
1 = Enable Path
R38 (26h)
Mono Mixer
Control (1)
6:4 LI2MOVOL
[2:0]
101
(-9dB)
LMIXSEL Signal to Mono Mixer
Volume
000 = +6dB
... (3dB steps)
111 = -15dB
8
RD2MO
0
Right DAC to Mono Mixer
0 = Disable (Mute)
1 = Enable Path
7
RI2MO
0
RMIXSEL Signal to Mono Mixer
0 = Disable (Mute)
1 = Enable Path
R39 (27h)
Mono Mixer
Control (2)
6:4 RI2MOVOL
[2:0]
101
(-9dB)
RMIXSEL Signal to Mono Mixer
Volume
000 = +6dB
... (3dB steps)
111 = -15dB
Table 21 Mono Output Mixer Control
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ANALOGUE OUTPUTS
LOUT1/ROUT1 OUTPUTS
The LOUT1 and ROUT1 pins can drive a 16
or 32 headphone or a line output (see Headphone
Output and Line Output sections, respectively). The signal volume on LOUT1 and ROUT1 can be
independently adjusted under software control by writing to LOUT1VOL and ROUT1VOL,
respectively. Note that gains over 0dB may cause clipping if the signal is large. Any gain setting
below 0101111 (minimum) mutes the output driver. The corresponding output pin remains at the
same DC level (the reference voltage on the VREF pin), so that no click noise is produced when
muting or un-muting.
A zero cross detect on the analogue output may also be enabled when changing the gain setting to
minimize audible clicks and zipper noise as the gain updates. If zero cross is enabled a timeout is
also available to update the gain if a zero cross does not occur. This function may be enabled by
setting TOEN in register R23 (17h).
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
8 LO1VU
0
Left
Volume
Update
0 = Store LOUT1VOL in intermediate
latch (no gain change)
1 = Update left and right channel gains
(left = LOUT1VOL, right =
intermediate latch)
7
LO1ZC
0
Left zero cross enable
1 = Change gain on zero cross only
0 = Change gain immediately
R2 (02h)
LOUT1
Volume
6:0 LOUT1VOL
[6:0]
1111001
(0dB)
LOUT1 Volume
1111111 = +6dB
... (80 steps)
0110000 = -67dB
0101111 to 0000000 = Analogue
MUTE
8 RO1VU
0
Right
Volume
Update
0 = Store ROUT1VOL in intermediate
latch (no gain change)
1 = Update left and right channel gains
(left = intermediate latch, right =
ROUT1VOL)
7
RO1ZC
0
Right zero cross enable
1 = Change gain on zero cross only
0 = Change gain immediately
R3 (03h)
ROUT1
Volume
6:0 ROUT1VOL
[6:0]
1111001 ROUT1
Volume
Similar to LOUT1VOL
Table 22 LOUT1/ROUT1 Volume Control
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LOUT2/ROUT2 OUTPUTS
The LOUT2 and ROUT2 output pins are essentially similar to LOUT1 and ROUT1, but they are
independently controlled and can also drive an 8
mono speaker (see Speaker Output section). For
speaker drive, the ROUT2 signal must be inverted (ROUT2INV = 1), so that the left and right channel
are mixed to mono in the speaker [L(-R) = L+R].
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
6:0 LOUT2VOL
[6:0]
1111001
(0dB)
Similar to LOUT1VOL
7
LO2ZC
0
Left zero cross enable
1 = Change gain on zero cross only
0 = Change gain immediately
R40 (28h)
LOUT2
Volume
8 LO2VU 0
Same
as
LO1VU
6:0 ROUT2VOL
[6:0]
1111001
(0dB)
Similar ROUT1VOL
7
RO2ZC
0
Right zero cross enable
1 = Change gain on zero cross only
0 = Change gain immediately
R41 (29h)
ROUT2
Volume
8 RO2VU 0
Same
as
RO1VU
R24 (18h)
Additional
Control (2)
4 ROUT2INV
0
ROUT2 Invert
0 = No Inversion (0
phase shift)
1 = Signal inverted (180
phase shift)
Table 23 LOUT2/ROUT2 Volume Control
MONO OUTPUT
The MONOOUT pin can drive a mono line output. The signal volume on MONOOUT can be adjusted
under software control by writing to MONOOUTVOL.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
6:0 MONOOUT
VOL [6:0]
1111001
(0dB)
MONOOUT Volume
1111111 = +6dB
... (80 steps)
0110000 = -67dB
0101111 to 0000000 = Analogue MUTE
R42 (2Ah)
MONOOUT
Volume
7
MOZC
0
MONOOUT zero cross enable
1 = Change gain on zero cross only
0 = Change gain immediately
Table 24 MONOOUT Volume Control
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ENABLING THE OUTPUTS
Each analogue output of the WM8971L can be separately enabled or disabled. The analogue mixer
associated with each output is powered on or off along with the output pin. All outputs are disabled by
default. To save power, unused outputs should remain disabled.
Outputs can be enabled at any time, except when VREF is disabled (VR=0), as this may cause pop
noise (see "Power Management" and "Applications Information" sections)
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
6 LOUT1 0
LOUT1
Enable
5 ROUT1 0
ROUT1
Enable
4 LOUT2 0
LOUT2
Enable
3 ROUT2 0
ROUT2
Enable
R26 (1Ah)
Power
Management
(2)
2 MONO
0
MONOOUT
Enable
Note: All "Enable" bits are 1 = ON, 0 = OFF
Table 25 Analogue Output Control
Whenever an analogue output is disabled, it remains connected to VREF (pin 20) through a resistor.
This helps to prevent pop noise when the output is re-enabled. The resistance between VREF and
each output can be controlled using the VROI bit in register 27. The default is low (1.5k
), so that
any capacitors on the outputs can charge up quickly at start-up. If a high impedance is desired for
disabled outputs, VROI can then be set to 1, increasing the resistance to about 40k
.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
R27 (1Bh)
Additional (1)
6
VROI
0
VREF to analogue output resistance
0: 1.5 k
1: 40 k
Table 26 Disabled Outputs to VREF Resistance
HEADPHONE SWITCH
The HPDETECT pin can be used as a headphone switch control input to automatically disable the
speaker output and enable the headphone output e.g. when a headphone is plugged into a jack
socket. In this mode, enabled by setting HPSWEN, HPDETECT switches between headphone and
speaker outputs (e.g. when the pin is connected to a mechanical switch in the headphone socket to
detect plug-in). The HPSWPOL bit reverses the pin's polarity. Note that the LOUT1, ROUT1, LOUT2
and ROUT2 bits in register 26 must also be set for headphone and speaker output (see Table 27 and
Table 28).
HPSWEN HPSWPOL HPDETECT
(PIN23)
L/ROUT1
(REG. 26)
L/ROUT2
(REG. 26)
HEADPHONE
ENABLED
SPEAKER
ENABLED
0 X X 0 0 no no
0 X X 0 1 no yes
0 X X 1 0 yes no
0 X X 1 1 yes yes
1 0 0 X 0 no no
1 0 0 X 1 no yes
1 0 1 0 X no no
1 0 1 1 X yes no
1 1 0 0 X no no
1 1 0 1 X yes no
1 1 1 X 0 no no
1 1 1 X 1 no yes

Table 27 Headphone Switch Operation
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REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
6 HPSWEN 0
Headphone Switch Enable
0 : Headphone switch disabled
1 : Headphone switch enabled
R24 (18h)
Additional
Control (2)
5
HPSWPOL
0
Headphone Switch Polarity
0 : HPDETECT high = headphone
1 : HPDETECT high = speaker
Table 28 Headphone Switch
Figure 12 Example Headset Detection Circuit Using Normally-Open Switch
Figure 13 Example Headset Detection Circuit Using Normally-Closed Switch
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THERMAL SHUTDOWN
The speaker and headphone outputs can drive very large currents. To protect the WM8971L from
overheating a thermal shutdown circuit is included. If the device temperature reaches approximately
150
0
C and the thermal shutdown circuit is enabled (TSDEN = 1) then the analogue outputs
(OUT1L/R, OUT2L/R) will be disabled.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
R23 (17h)
Additional
Control (1)
8 TSDEN 0
Thermal Shutdown Enable
0 : thermal shutdown disabled
1 : thermal shutdown enabled
Table 29 Thermal Shutdown
HEADPHONE OUTPUT
Analogue outputs LOUT1/ROUT1 and LOUT2/ROUT2, can drive a 16
or 32 headphone load,
through DC blocking capacitors.
Figure 14 Recommended Headphone Output Configuration
The DC blocking capacitors and the load resistance together determine the lower cut-off frequency,
f
c
. Increasing the capacitance lowers f
c
, improving the bass response. Smaller capacitance values
will diminish the bass response. Assuming a 16 Ohm load and C1, C2 = 220
F:
f
c
= 1 / 2
R
L
C
1
= 1 / (2
x 16 x 220F) = 45 Hz
SPEAKER OUTPUT
LOUT2 and ROUT2 can differentially drive a mono 8
speaker as shown below.
Figure 15 Speaker Output Connection
The right channel is inverted by setting the ROUT2INV bit, so that the signal across the loudspeaker
is the sum of left and right channels.
LINE OUTPUT
The analogue outputs, LOUT1/ROUT1 and LOUT2/ROUT2, can be used as line outputs.
Recommended external components are shown below.
Figure 16 Recommended Circuit for Line Output
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The DC blocking capacitors and the load resistance together determine the lower cut-off frequency,
f
c
. Assuming a 10 kOhm load and C1, C2 = 1
F:
f
c
= 1 / 2
(R
L
+R
1
) C
1
= 1 / (2
x 10.1k x 1F) = 16 Hz
Increasing the capacitance lowers f
c
, improving the bass response. Smaller values of C1 and C2 will
diminish the bass response. The function of R1 and R2 is to protect the line outputs from damage
when used improperly.
DIGITAL AUDIO INTERFACE
The digital audio interface is used for inputting DAC data into the WM8971L and outputting ADC data
from it. It uses five pins:
ADCDAT: ADC data output
ADCLRC: ADC data alignment clock
DACDAT: DAC data input
DACLRC: DAC data alignment clock
BCLK: Bit clock, for synchronisation
The clock signals BCLK, ADCLRC and DACLRC can be outputs when the WM8971L operates as a
master, or inputs when it is a slave (see Master and Slave Mode Operation, below).
Four different audio data formats are supported:
Left
justified
Right
justified
I
2
S
DSP
mode
All four of these modes are MSB first. They are described in Audio Data Formats, below. Refer to the
Electrical Characteristic section for timing information.
MASTER AND SLAVE MODE OPERATION
The WM8971L can be configured as either a master or slave mode device. As a master device the
WM8971L generates BCLK, ADCLRC and DACLRC and thus controls sequencing of the data
transfer on ADCDAT and DACDAT. In slave mode, the WM8971L responds with data to clocks it
receives over the digital audio interface. The mode can be selected by writing to the MS bit (see
Table 23). Master and slave modes are illustrated below.
Figure 17 Master Mode
Figure 18 Slave Mode
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AUDIO DATA FORMATS
In Left Justified mode, the MSB is available on the first rising edge of BCLK following a LRCLK
transition. The other bits up to the LSB are then transmitted in order. Depending on word length,
BCLK frequency and sample rate, there may be unused BCLK cycles before each LRCLK transition.
Figure 19 Left Justified Audio Interface (assuming n-bit word length)
In Right Justified mode, the LSB is available on the last rising edge of BCLK before a LRCLK
transition. All other bits are transmitted before (MSB first). Depending on word length, BCLK
frequency and sample rate, there may be unused BCLK cycles after each LRCLK transition.
Figure 20 Right Justified Audio Interface (assuming n-bit word length)
In I
2
S mode, the MSB is available on the second rising edge of BCLK following a LRCLK transition.
The other bits up to the LSB are then transmitted in order. Depending on word length, BCLK
frequency and sample rate, there may be unused BCLK cycles between the LSB of one sample and
the MSB of the next.
Figure 21 I
2
S Justified Audio Interface (assuming n-bit word length)
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In DSP/PCM mode, the left channel MSB is available on either the 1
st
(mode B) or 2
nd
(mode A)
rising edge of BCLK (selectable by LRP) following a rising edge of LRC. Right channel data
immediately follows left channel data. Depending on word length, BCLK frequency and sample rate,
there may be unused BCLK cycles between the LSB of the right channel data and the next sample.
In device master mode, the LRC output will resemble the frame pulse shown in Figure 22 and Figure
23. In device slave mode, Figure 24 and Figure 25, it is possible to use any length of frame pulse
less than 1/fs, providing the falling edge of the frame pulse occurs greater than one BCLK period
before the rising edge of the next frame pulse.
Figure 22 DSP/PCM Mode Audio Interface (mode A, LRP=0, Master)
Figure 23 DSP/PCM Mode Audio Interface (mode B, LRP=1, Master)
Figure 24 DSP/PCM Mode Audio Interface (mode A, LRP=0, Slave)
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Figure 25 DSP/PCM Mode Audio Interface (mode B, LRP=0, Slave)
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AUDIO INTERFACE CONTROL
The register bits controlling audio format, word length and master / slave mode are summarised in
Table 30. MS selects audio interface operation in master or slave mode. In Master mode BCLK,
ADCLRC and DACLRC are outputs. The frequency of ADCLRC and DACLRC is set by the sample
rate control bits SR[4:0] and USB. In Slave mode BCLK, ADCLRC and DACLRC are inputs.
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
7
BCLKINV
0
BCLK invert bit (for master and slave
modes)
0 = BCLK not inverted
1 = BCLK inverted
6
MS
0
Master / Slave Mode Control
1 = Enable Master Mode
0 = Enable Slave Mode
5
LRSWAP
0
Left/Right channel swap
1 = swap left and right DAC data in
audio interface
0 = output left and right data as normal
right, left and i2s modes LRCLK
polarity
1 = invert LRCLK polarity
0 = normal LRCLK polarity
4 LRP
0
DSP Mode mode A/B select
1 = MSB is available on 1st BCLK rising
edge after LRC rising edge (mode B)
0 = MSB is available on 2nd BCLK rising
edge after LRC rising edge (mode A)
3:2
WL[1:0]
10
Audio Data Word Length
11 = 32 bits (see Note)
10 = 24 bits
01 = 20 bits
00 = 16 bits
R7 (07h)
Digital Audio
Interface
Format
1:0
FORMAT[1:0]
10
Audio Data Format Select
11 = DSP Mode
10 = I
2
S Format
01 = Left justified
00 = Right justified
Table 30 Audio Data Format Control
Note: Right Justified mode does not support 32-bit data.
AUDIO INTERFACE OUTPUT TRISTATE
Register bit TRI, register 24(18h) bit[3] can be used to tristate the ADCDAT pin and switch ADCLRC,
DACLRC and BCLK to inputs. In Slave mode (MASTER=0) ADCLRC, DACLRC and BCLK are by
default configured as inputs and only ADCDAT will be tri-stated, (see Table 31).
REGISTER
ADDRESS
BIT LABEL
DEFAULT DESCRIPTION
R24(18h)
Additional
Control (2)
3 TRI
0 Tristates ADCDAT and switches ADCLRC,
DACLRC and BCLK to inputs.
0 = ADCDAT is an output, ADCLRC, DACLRC
and BCLK are inputs (slave mode) or outputs
(master mode)
1 = ADCDAT is tristated, ADCLRC, DACLRC
and BCLK are inputs
Table 31 Tri-stating the Audio Interface
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MASTER MODE ADCLRC AND DACLRC ENABLE
In Master mode, by default ADCLRC is disabled when the ADC is disabled and DACLRC is disabled
when the DAC is disabled. Register bit LRCM, register 24(18h) bit[2] changes the control so that the
ADCLRC and DACLRC are disabled only when ADC and DAC are disabled. This enables the user to
use e.g. ADCLRC for both ADC and DAC LRCLK and disable the ADC when DAC only operation is
required, (see Table 32).
REGISTER
ADDRESS
BIT LABEL
DEFAULT DESCRIPTION
R24(18h)
Additional
Control (2)
2 LRCM
0 Selects disable mode for ADCLRC and
DACLRC
0 = ADCLRC disabled when ADC (Left and
Right) disabled, DACLRC disabled when
DAC (Left and Right) disabled.
1 = ADCLRC and DACLRC disabled only when
ADC (Left and Right) and DAC (Left and
Right) are disabled.
Table 32 ADCLRC/DACLRC Enable
BIT CLOCK MODE
The default master mode bit clock generator produces a bit clock frequency based on the sample
rate and input MCLK frequency as shown in Table 36. When enabled by setting the appropriate
BCM[1:0] bits, the bit clock mode (BCM) function overrides the default master mode bit clock
generator to produce the bit clock frequency shown in the table below:
REGISTER
ADDRESS
BIT LABEL
DEFAULT DESCRIPTION
R8 (08h)
Clocking and
Sample Rate
Control
8:7 BCM[1:0]
00
BCLK Frequency
00 = BCM function disabled
01 = MCLK/4
10 = MCLK/8
11 = MCLK/16
Table 33 Master Mode BCLK Frequency Control
The BCM mode bit clock generator produces 16 or 24 bit clock cycles per sample. The number of bit
clock cycles per sample in this mode is determined by the word length bits (WL[1:0]) in the Digital
Audio Interface Format register (R7). When these bits are set to 00, there will be 16 bit clock cycles
per sample. When these bits are set to 01, 10 or 11, there will be 24 bit clock cycles per sample.
Please refer to Figure 26.
The BCM generator uses the ADCLRC signal, hence the ADCLRC signal must be enabled when
using bit clock mode. To enable the ADCLRC signal, either the ADC must be powered up or, if the
ADC is not in use, the LRCM bit must be set to enable both the ADCLRC and DACLRC signals when
either the ADC or the DAC is enabled.
When the BCM function is enabled, the following restrictions apply:
1. The bit clock invert (BCLKINV) function is not available.
2. The DAC and ADC must be operated at the same sample rate.
3. DSP late digital audio interface mode is not available and must not be enabled.
Figure 26 Bit Clock Mode
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Note: The shaded bit clock cycles are present only when 24-bit mode is selected. Please refer to the
"Bit Clock Mode" description for details.
CLOCK OUTPUT
By default ADCLRC (pin 9) is the ADC word clock input/output. Under the control of ADCLRM[1:0],
register 27(1Bh) bits [8:7] the ADCLRC pin may be configured as a clock output. If ADCLRM is 01,
10 or 11 then ADCLRC pin is always an output even in slave mode or when TRI = `1', (see Table 34).
The ADC then uses the DACLRC pin as its LRCLK in both master and slave modes.
REGISTER
ADDRESS
BIT LABEL
DEFAULT DESCRIPTION
R27(1Bh)
Additional
Control (3)
[8:7] ADCLRM
[1:0]
00
Configures ADCLRC pin
00 = ADCLRC is ADC word clock input (slave
mode) or ADCLRC output (master mode)
01 = ADCLRC pin is MCLK output
10 = ADCLRC pin is MCLK / 5.5 output
11 = ADCLRC pin is MCLK / 6 output
Table 34 ADCLRC Clock Output
CLOCKING AND SAMPLE RATES
The WM8971L supports a wide range of master clock frequencies on the MCLK pin, and can
generate many commonly used audio sample rates directly from the master clock. The ADC and
DAC do not need to run at the same sample rate; several different combinations are possible.
There are two clocking modes:
`Normal' mode supports master clocks of 128f
s
, 192f
s
, 256f
s
, 384f
s
, and their multiples
(Note: f
s
refers to the ADC or DAC sample rate, whichever is faster)
USB mode supports 12MHz or 24MHz master clocks. This mode is intended for use in
systems with a USB interface, and eliminates the need for an external PLL to generate
another clock frequency for the audio codec.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
6
CLKDIV2
0
Master Clock Divide by 2
1 = MCLK is divided by 2
0 = MCLK is not divided
5:1
SR [4:0]
00000
Sample Rate Control
R8 (08h)
Clocking and
Sample Rate
Control
0
USB
0
Clocking Mode Select
1 = USB Mode
0 = `Normal' Mode
Table 35 Clocking and Sample Rate Control
The clocking of the WM8971L is controlled using the CLKDIV2, USB, and SR control bits. Setting the
CLKDIV2 bit divides MCLK by two internally. The USB bit selects between `Normal' and USB mode.
Each value of SR[4:0] selects one combination of MCLK division ratios and hence one combination
of sample rates (see next page). Since all sample rates are generated by dividing MCLK, their
accuracy depends on the accuracy of MCLK. If MCLK changes, the sample rates change
proportionately.
Note that some sample rates (e.g. 44.1kHz in USB mode) are approximated, i.e. they differ from their
target value by a very small amount. This is not audible, as the maximum deviation is only 0.27%
(8.0214kHz instead of 8kHz in USB mode). By comparison, a half-tone step corresponds to a 5.9%
change in pitch.
The SR[4:0] bits must be set to configure the appropriate ADC and DAC sample rates in both master
and slave mode.
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MCLK
CLKDIV2=0
MCLK
CLKDIV2=1
ADC SAMPLE RATE
(ADCLRC)
DAC SAMPLE RATE
(DACLRC)
USB SR
[4:0] FILTER
TYPE
BCLK
(MS=1)
`Normal' Clock Mode (`*' indicates backward compatibility with WM8731)
8 kHz (MCLK/1536)
8 kHz (MCLK/1536)
0
00110 *
1
MCLK/4
8 kHz (MCLK/1536)
48 kHz (MCLK/256)
0
00100 *
1
MCLK/4
12 kHz (MCLK/1024)
12 kHz (MCLK/1024)
0
01000
1
MCLK/4
16 kHz (MCLK/768)
16 kHz (MCLK/768)
0
01010
1
MCLK/4
24 kHz (MCLK/512)
24 kHz (MCLK/512)
0
11100
1
MCLK/4
32 kHz (MCLK/384)
32 kHz (MCLK/384)
0
01100 *
1
MCLK/4
48 kHz (MCLK/256)
8 kHz (MCLK/1536)
0
00010 *
1
MCLK/4
48 kHz (MCLK/256)
48 kHz (MCLK/256)
0
00000 *
1
MCLK/4
12.288 MHz 24.576 MHz
96 kHz (MCLK/128)
96 kHz (MCLK/128)
0
01110 *
3
MCLK/2
8.0182 kHz (MCLK/1408)
8.0182 kHz (MCLK/1408)
0
10110 *
1
MCLK/4
8.0182 kHz (MCLK/1408)
44.1 kHz (MCLK/256)
0
10100 *
1
MCLK/4
11.025 kHz (MCLK/1024)
11.025 kHz (MCLK/1024)
0
11000
1
MCLK/4
22.05 kHz (MCLK/512)
22.05 kHz (MCLK/512)
0
11010
1
MCLK/4
44.1 kHz (MCLK/256)
8.0182 kHz (MCLK/1408)
0
10010 *
1
MCLK/4
44.1 kHz (MCLK/256)
44.1 kHz (MCLK/256)
0
10000 *
1
MCLK/4
11.2896MHz
22.5792MHz
88.2 kHz (MCLK/128)
88.2 kHz (MCLK/128)
0
11110 *
3
MCLK/2
8 kHz (MCLK/2304)
8 kHz (MCLK/2304)
0
00111 *
1
MCLK/6
8 kHz (MCLK/2304)
48 kHz (MCLK/384)
0
00101 *
1
MCLK/6
12 kHz (MCLK/1536)
12 kHz (MCLK/1536)
0
01001
1
MCLK/6
16kHz (MCLK/1152)
16 kHz (MCLK/1152)
0
01011
1
MCLK/6
24kHz (MCLK/768)
24 kHz (MCLK/768)
0
11101
1
MCLK/6
32 kHz (MCLK/576)
32 kHz (MCLK/576)
0
01101 *
1
MCLK/6
48 kHz (MCLK/384)
48 kHz (MCLK/384)
0
00001 *
1
MCLK/6
48 kHz (MCLK/384)
8 kHz (MCLK/2304)
0
00011 *
1
MCLK/6
18.432MHz
36.864MHz
96 kHz (MCLK/192)
96 kHz (MCLK/192)
0
01111 *
3
MCLK/3
8.0182 kHz (MCLK/2112)
8.0182 kHz (MCLK/2112)
0
10111 *
1
MCLK/6
8.0182 kHz (MCLK/2112)
44.1 kHz (MCLK/384)
0
10101 *
1
MCLK/6
11.025 kHz (MCLK/1536)
11.025 kHz (MCLK/1536)
0
11001
1
MCLK/6
22.05 kHz (MCLK/768)
22.05 kHz (MCLK/768)
0
11011
1
MCLK/6
44.1 kHz (MCLK/384)
8.0182 kHz (MCLK/2112)
0
10011 *
1
MCLK/6
44.1 kHz (MCLK/384)
44.1 kHz (MCLK/384)
0
10001 *
1
MCLK/6
16.9344MHz
33.8688MHz
88.2 kHz (MCLK/192)
88.2 kHz (MCLK/192)
0
11111 *
3
MCLK/3
USB Mode (`*' indicates backward compatibility with WM8731)
8 kHz (MCLK/1500)
8 kHz (MCLK/1500)
1
00110 *
0
MCLK
8 kHz (MCLK/1500)
48 kHz (MCLK/250)
1
00100 *
0
MCLK
8.0214 kHz (MCLK/1496)
8.0214kHz (MCLK/1496)
1
10111 *
1
MCLK
8.0214 kHz (MCLK/1496)
44.118 kHz (MCLK/272)
1
10101 *
1
MCLK
11.0259 kHz (MCLK/1088)
11.0259kHz (MCLK/1088)
1
11001
1
MCLK
12 kHz (MCLK/1000)
12 kHz (MCLK/1000)
1
01000
0
MCLK
16kHz (MCLK/750)
16kHz (MCLK/750)
1
01010
0
MCLK
22.0588kHz (MCLK/544)
22.0588kHz (MCLK/544)
1
11011
1
MCLK
24kHz (MCLK/500)
24kHz (MCLK/500)
1
11100
0
MCLK
32 kHz (MCLK/375)
32 kHz (MCLK/375)
1
01100 *
0
MCLK
44.118 kHz (MCLK/272)
8.0214kHz (MCLK/1496)
1
10011 *
1
MCLK
44.118 kHz (MCLK/272)
44.118 kHz (MCLK/272)
1
10001 *
1
MCLK
48 kHz (MCLK/250)
8 kHz (MCLK/1500)
1
00010 *
0
MCLK
48 kHz (MCLK/250)
48 kHz (MCLK/250)
1
00000 *
0
MCLK
88.235kHz (MCLK/136)
88.235kHz (MCLK/136)
1
11111 *
3
MCLK
12.000MHz 24.000MHz
96 kHz (MCLK/125)
96 kHz (MCLK/125)
1
01110 *
2
MCLK
Table 36 Master Clock and Sample Rates
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CONTROL INTERFACE
SELECTION OF CONTROL MODE
The WM8971L is controlled by writing to registers through a serial control interface. A control word
consists of 16 bits. The first 7 bits (B15 to B9) are address bits that select which control register is
accessed. The remaining 9 bits (B8 to B0) are data bits, corresponding to the 9 bits in each control
register. The control interface can operate as either a 3-wire or 2-wire MPU interface. The MODE pin
selects the interface format.
MODE INTERFACE
FORMAT
Low 2
wire
High 3
wire
Table 37 Control Interface Mode Selection
3-WIRE SERIAL CONTROL MODE
In 3-wire mode, every rising edge of SCLK clocks in one data bit from the SDIN pin. A rising edge on
CSB latches in a complete control word consisting of the last 16 bits.
B15
B14
B13
B12
B11
B10
B9
B8
B7
B6
B5
B4
B3
B2
B1
B0
SDIN
SCLK
CSB
control register address
control register data bits
latch
Figure 27 3-Wire Serial Control Interface
2-WIRE SERIAL CONTROL MODE
The WM8971L supports software control via a 2-wire serial bus. Many devices can be controlled by
the same bus, and each device has a unique 7-bit address (this is not the same as the 7-bit address
of each register in the WM8971L).
The WM8971L operates as a slave device only. The controller indicates the start of data transfer with
a high to low transition on SDIN while SCLK remains high. This indicates that a device address and
data will follow. All devices on the 2-wire bus respond to the start condition and shift in the next eight
bits on SDIN (7-bit address + Read/Write bit, MSB first). If the device address received matches the
address of the WM8971L and the R/W bit is `0', indicating a write, then the WM8971L responds by
pulling SDIN low on the next clock pulse (ACK). If the address is not recognised or the R/W bit is `1',
the WM8971L returns to the idle condition and wait for a new start condition and valid address.
Once the WM8971L has acknowledged a correct address, the controller sends the first byte of
control data (B15 to B8, i.e. the WM8971L register address plus the first bit of register data). The
WM8971L then acknowledges the first data byte by pulling SDIN low for one clock pulse. The
controller then sends the second byte of control data (B7 to B0, i.e. the remaining 8 bits of register
data), and the WM8971L acknowledges again by pulling SDIN low.
The transfer of data is complete when there is a low to high transition on SDIN while SCLK is high.
After receiving a complete address and data sequence the WM8971L returns to the idle state and
waits for another start condition. If a start or stop condition is detected out of sequence at any point
during data transfer (i.e. SDIN changes while SCLK is high), the device jumps to the idle condition.
SDIN
SCLK
register address and
1st register data bit
DEVICE ADDRESS
(7 BITS)
RD / WR
BIT
ACK
(LOW)
CONTROL BYTE 1
(BITS 15 TO 8)
CONTROL BYTE 2
(BITS 7 TO 0)
remaining 8 bits of
register data
STOP
START
ACK
(LOW)
ACK
(LOW)
Figure 28 2-Wire Serial Control Interface
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The WM8971L has two possible device addresses, which can be selected using the CSB pin.
CSB STATE
DEVICE ADDRESS
Low
0011010 (0 x 34h)
High
0011011 (0 x 36h)
Table 38 2-Wire MPU Interface Address Selection
POWER SUPPLIES
The WM8971L can use up to four separate power supplies:
AVDD / AGND: Analogue supply, powers all analogue functions except the headphone drivers.
AVDD can range from 1.8V to 3.6V and has the most significant impact on overall power
consumption (except for power consumed in the headphone). A large AVDD slightly improves
audio quality.
HPVDD / HPGND: Headphone supply, powers analogue outputs L/ROUT1, L/ROUT2 and
MONOOUT. HPVDD is normally tied to AVDD, but requires separate layout and decoupling
capacitors to curb harmonic distortion. If HPVDD is lower than AVDD, the output signal may be
clipped.
DCVDD: Digital core supply, powers all digital functions except the audio and control
interfaces. DCVDD can range from 1.42V to 3.6V, and has no effect on audio quality. The
return path for DCVDD is DGND, which is shared with DBVDD.
DBVDD: Digital buffer supply, powers the audio and control interface buffers. This makes it
possible to run the digital core at very low voltages, saving power, while interfacing to other
digital devices using a higher voltage. DBVDD draws much less power than DCVDD, and has
no effect on audio quality. DBVDD can range from 1.8V to 3.6V. The return path for DBVDD is
DGND, which is shared with DCVDD.
It is possible to use the same supply voltage on all four. However, digital and analogue supplies
should be routed and decoupled separately to keep digital switching noise out of the analogue signal
paths.
POWER MANAGEMENT
The WM8971L has two control registers that allow users to select which functions are active. For
minimum power consumption, unused functions should be disabled. To avoid any pop or click noise,
it is important to enable or disable functions in the correct order (see Applications Information).
VMIDSEL is the enable for the Vmid reference, which defaults to disabled and can be enabled as a
50kOhm potential divider or, for low power maintenance of Vref when all other blocks are disabled,
as a 500kOhm potential divider.
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REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
8:7
VMIDSEL
00
Vmid divider enable and select
00 Vmid disabled (for OFF mode)
01 50k
divider enabled (for
playback/record)
10 500k
divider enabled (for low-power
standby)
11 5k
divider enabled (for fast start-up)
6
VREF
0
VREF (necessary for all other functions)
Power down = 0
Power up = 1
5
AINL
0
Analogue in PGA Left
Power down = 0
Power up = 1
4
AINR
0
Analogue in PGA Right
Power down = 0
Power up = 1
3 ADCL 0
ADC
Left
Power down = 0
Power up = 1
2 ADCR 0
ADC
Right
Power down = 0
Power up = 1
R25 (19h)
Power
Management
(1)
1 MICB 0
MICBIAS
Power down = 0
Power up = 1
8 DACL 0
DAC
Left
Power down = 0
Power up = 1
7 DACR 0
DAC
Right
Power down = 0
Power up = 1
6 LOUT1 0
LOUT1
Output
Buffer*
Power down = 0
Power up = 1
5 ROUT1 0
ROUT1
Output
Buffer*
Power down = 0
Power up = 1
4 LOUT2 0
LOUT2
Output
Buffer*
Power down = 0
Power up = 1
3 ROUT2 0
ROUT2
Output
Buffer*
Power down = 0
Power up = 1
R26 (1Ah)
Power
Management
(2)
2
MONO
0
MONOOUT Output Buffer and Mono Mixer
Power down = 0
Power up = 1
* The left mixer is enabled when LOUT1=1 or LOUT2=1. The right mixer is enabled when
ROUT1=1 or ROUT2=1.
Table 39 Power Management
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STOPPING THE MASTER CLOCK
In order to minimise power consumed in the digital core of the WM8971L, the master clock may be
stopped in Standby and OFF modes. If this cannot be done externally at the clock source, the
DIGENB bit (R25, bit 0) can be set to stop the MCLK signal from propagating into the device core. In
Standby mode, setting DIGENB will typically provide an additional power saving on DCVDD of 20uA.
However, since setting DIGENB has no effect on the power consumption of other system
components external to the WM8971L, it is preferable to disable the master clock at its source
wherever possible.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
R25 (19h)
Additional Control
(1)
0
DIGENB
0
Master clock disable
0: master clock enabled
1: master clock disabled
Table 40 ADC and DAC Oversampling Rate Selection
Note: Before DIGENB can be set, the control bits ADCL, ADCR, DACL and DACR must be set to
zero and a waiting time of 1ms must be observed. Any failure to follow this procedure may prevent
DACs and ADCs from re-starting correctly.
SAVING POWER BY REDUCING OVERSAMPLING RATE
The default mode of operation of the ADC and DAC digital filters is in 128x oversampling mode.
Under the control of ADCOSR and DACOSR the oversampling rate may be halved. This will result in
a slight decrease in noise performance but will also reduce the power consumption of the device.
REGISTER
ADDRESS
BIT LABEL DEFAULT
DESCRIPTION
1
ADCOSR
0
ADC oversample rate select
1 = 64x (lowest power)
0 = 128x (best SNR)
R24 (18h)
Additional Control
(2)
0
DACOSR
0
DAC oversample rate select
1 = 64x (lowest power)
0 = 128x (best SNR)
Table 41 ADC and DAC Oversampling Rate Selection
Note: ADCOSR set to 1, 64x oversample mode, is not supported in USB mode (USB=1)
SAVING POWER AT HIGHER SUPPLY VOLTAGES
The analogue supplies to the WM8971L can run from 1.8V to 3.6V. By default, all analogue circuitry
on the device is optimized to run at 3.3V. This set-up is also good for all other supply voltages down
to 1.8V.
At lower voltages, performance can be improved by increasing the bias current. If low
power operation is preferred the bias current can be left at the default setting. This is controlled as
shown below.
REGISTER
ADDRESS
BIT LABEL
DEFAULT
DESCRIPTION
R23 (17h)
Additional
Control(1)
7:6 VSEL
[1:0]
11
Analogue Bias optimization
00: Highest bias current, optimized for
AVDD=1.8V
01: Bias current, optimized for AVDD=2.5V
1X: Lowest bias current, optimized for AVDD=3.3V
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REGISTER MAP
REGISTER
ADDRESS
(Bit 15 9)
remarks
Bit[8] Bit[7] Bit[6] Bit[5] Bit[4] Bit[3] Bit[2] Bit[1] Bit[0]
default page
ref
R0 (00h)
0000000
Left Input volume
LIVU
LINMUTE
LIZC
LINVOL
010010111
21
R1 (01h)
0000001
Right Input volume
RIVU
RINMUTE
RIZC
RINVOL
010010111
21
R2 (02h)
0000010
LOUT1 volume
LO1VU
LO1ZC
LOUT1VOL[6:0]
001111001
33
R3 (03h)
0000011
ROUT1 volume
RO1VU
RO1ZC
ROUT1VOL[6:0]
001111001
33
R4
(04h)
0000100 Reserved
0 0 0 0 0 0 0 0 0 000000000 -
R5 (05h)
0000101
ADC and DAC Control ADCDIV2 DACDIV2
ADCPOL[1:0]
HPOR
DACMU
DEEMPH[1:0]
ADCHPD
000001000
21,27,30
R6
(06h)
0000110 Reserved
0 0 0 0 0 0 0 0 0 000000000 -
R7
(07h) 0000111
Audio
Interface
0 BCLKINV MS LRSWAP LRP
WL[1:0]
FORMAT[1:0]
000001010
42
R8 (08h)
0001000
Sample rate
BCM[1:0]
CLKDIV2
SR[4:0]
USB
000000000
44
R9
(09h)
0001001 Reserved
0 0 0 0 0 0 0 0 0 000000000 -
R10 (0Ah)
0001010
Left DAC volume
LDVU
LDACVOL[7:0]
011111111
28
R11 (0Bh)
0001011
Right DAC volume
RDVU
RDACVOL[7:0]
011111111
28
R12 (0Ch)
0001100
Bass control
0
BB
BC
0
0
BASS[3:0]
000001111
29
R13 (0Dh)
0001101
Treble control
0
0
TC
0
0
TRBL[3:0]
000001111
29
R15 (0Fh)
0001111
Reset
writing to this register resets all registers to their default state
not reset
-
R16
(10h)
0010000 Reserved
0 0 0 0 0 0 0 0 0 000000000 -
R17 (11h)
0010001
ALC1
ALCSEL[1:0]
MAXGAIN[2:0]
ALCL[3:0]
001111011
26
R18 (12h)
0010010
ALC2
0
ALCZC
0
0
0
HLD[3:0]
000000000
26
R19 (13h)
0010011
ALC3
0
DCY[3:0]
ATK[3:0]
000110010
26
R20 (14h)
0010100
Noise Gate
0
NGTH[4:0]
NGG[1:0]
NGAT
000000000
27
R21 (15h)
0010101
Left ADC volume
LAVU
LADCVOL[7:0]
011000011
24
R22 (16h)
0010110
Right ADC volume
RAVU
RADCVOL[7:0]
011000011
24
R23 (17h)
0010111
Additional control(1)
TSDEN
VSEL[1:0]
DMONOMIX[1:0]
DATSEL[1:0]
DACINV
TOEN
011000000
19,21,30,37
R24 (18h)
0011000
Additional control(2)
0
0
HPSWEN HPSWPOL ROUT2INV
TRI LRCM
ADCOSR
DACOSR
000000000
34, 36,50
R25 (19h)
0011001
Pwr Mgmt (1)
VMIDSEL[1:0]
VREF
AINL
AINR
ADCL
ADCR
MICB
DIGENB
000000000
48
R26 (1Ah)
0011010
Pwr Mgmt (2)
DACL
DACR
LOUT1
ROUT1
LOUT2
ROUT2
MONO
0
0
000000000
48
R27 (1Bh)
0011011
Additional Control (3)
ADCLRM[1:0]
VROI
HPFLREN
0
0
0
0
0
000000000
35
R31 (1Fh)
0011111
ADC input mode
DS
MONOMIX[1:0]
0
0
0
0
0
0
000000000
19
R32 (20h)
0100000
ADCL signal path
0
LINSEL[1:0]
LMICBOOST[1:0]
0
0
0
0
000000000
19
R33 (21h)
0100001
ADCR signal path
0
RINSEL[1:0]
RMICBOOST[1:0]
0
0
0
0
000000000
19
R34 (22h)
0100010
Left out Mix (1)
LD2LO
LI2LO
LI2LOVOL[2:0]
0
LMIXSEL[2:0]
001010000
31
R35 (23h)
0100011
Left out Mix (2)
RD2LO
RI2LO
RI2LOVOL[2:0]
0
0
0
0
001010000
31
R36 (24h)
0100100
Right out Mix (1)
LD2RO
LI2RO
LI2ROVOL[2:0]
0
RMIXSEL[2:0]
001010000
32
R37 (25h)
0100101
Right out Mix (2)
RD2RO
RI2RO
RI2ROVOL[2:0]
0
0
0
0
001010000
32
R38 (26h)
0100110
Mono out Mix (1)
LD2MO
LI2MO
LI2MOVOL[2:0]
0
0
0
0
001010000
32
R39 (27h)
0100111
Mono out Mix (2)
RD2MO
RI2MO
RI2MOVOL[2:0]
0
0
0
0
001010000
32
R40 (28h)
0101000
LOUT2 volume
LO2VU
LO2ZC
LOUT2VOL[6:0]
001111001
34
R41 (29h)
0101001
ROUT2 volume
RO2VU
RO2ZC
ROUT2VOL[6:0]
001111001
34
R42 (2Ah)
0101010
MONOOUT volume
0
MOZC
MOUTVOL[6:0]
001111001
34
Note:
1.
All unused register bits must be set to `0' when writing to WM8971L.
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DIGITAL FILTER CHARACTERISTICS
The ADC and DAC employ different digital filters. There are 4 types of digital filter, called Type 0, 1, 2
and 3. The performance of Types 0 and 1 is listed in the table below, the responses of all filters is
shown in the proceeding pages.
PARAMETER TEST
CONDITIONS
MIN
TYP
MAX
UNIT
ADC Filter Type 0 (USB Mode, 250fs operation)
+/- 0.05dB
0
0.416fs
Passband
-6dB
0.5fs
Passband Ripple
+/-
0.05
dB
Stopband
0.584fs
Stopband Attenuation
f > 0.584fs
-60
dB
ADC Filter Type 1 (USB mode, 272fs or Normal mode operation)
+/- 0.05dB
0
0.4535fs
Passband
-6dB
0.5fs
Passband Ripple
+/-
0.05
dB
Stopband
0.5465fs
Stopband Attenuation
f > 0.5465fs
-60
dB
-3dB
3.7
-0.5dB
10.4
High Pass Filter Corner
Frequency
-0.1dB
21.6
Hz
DAC Filter Type 0 (USB mode, 250fs operation)
+/- 0.03dB
0
0.416fs
Passband
-6dB
0.5fs
Passband Ripple
+/-0.03
dB
Stopband
0.584fs
Stopband Attenuation
f > 0.584fs
-50
dB
DAC Filter Type 1 (USB mode, 272fs or Normal mode operation)
+/- 0.03dB
0
0.4535fs
Passband
-6dB
0.5fs
Passband Ripple
+/-
0.03
dB
Stopband
0.5465fs
Stopband Attenuation
f > 0.5465fs
-50
dB
Table 42 Digital Filter Characteristics
DAC FILTERS
ADC FILTERS
Mode Group
Delay
Mode Group
Delay
0 (250 USB)
11/FS
0 (250 USB)
13/FS
1
(256/272) 16/FS
1
(256/272) 23/FS
2 (250 USB, 96k mode)
4/FS
2 (250 USB, 96k mode)
4/FS
3 (256/272, 88.2/96k mode)
3/FS
3 (256/272, 88.2/96k mode)
5/FS
Table 43 ADC/DAC Digital Filters Group Delay
TERMINOLOGY
1.
Stop Band Attenuation (dB) the degree to which the frequency spectrum is attenuated (outside audio band)
2.
Pass-band Ripple any variation of the frequency response in the pass-band region
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DAC FILTER RESPONSES
-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.06
-0.05
-0.04
-0.03
-0.02
-0.01
0
0.01
0.02
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0.4
0.45
0.5
Response (dB)
Frequency (Fs)
Figure 29 DAC Digital Filter Frequency Response Type 0 Figure 30 DAC Digital Filter Ripple Type 0
-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.06
-0.05
-0.04
-0.03
-0.02
-0.01
0
0.01
0.02
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0.4
0.45
0.5
Response (dB)
Frequency (Fs)
Figure 31 DAC Digital Filter Frequency Response Type 1 Figure 32 DAC Digital Filter Ripple Type 1
-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.06
-0.05
-0.04
-0.03
-0.02
-0.01
0
0.01
0.02
0
0.05
0.1
0.15
0.2
0.25
Response (dB)
Frequency (Fs)
Figure 33 DAC Digital Filter Frequency Response Type 2 Figure 34 DAC Digital Filter Ripple Type 2
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-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.25
-0.2
-0.15
-0.1
-0.05
0
0.05
0.1
0.15
0.2
0.25
0
0.05
0.1
0.15
0.2
0.25
Response (dB)
Frequency (Fs)
Figure 35 DAC Digital Filter Frequency Response Type 3 Figure 36 DAC Digital Filter Ripple Type 3
ADC FILTER RESPONSES
-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.04
-0.03
-0.02
-0.01
0
0.01
0.02
0.03
0.04
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0.4
0.45
0.5
Response (dB)
Frequency (Fs)
Figure 37 ADC Digital Filter Frequency Response Type 0
Figure 38 ADC Digital Filter Ripple Type 0
-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.06
-0.05
-0.04
-0.03
-0.02
-0.01
0
0.01
0.02
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0.4
0.45
0.5
Response (dB)
Frequency (Fs)
Figure 39 ADC Digital Filter Frequency Response Type 1
Figure 40 ADC Digital Filter Ripple Type 1
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-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.25
-0.2
-0.15
-0.1
-0.05
0
0.05
0.1
0.15
0.2
0.25
0
0.05
0.1
0.15
0.2
0.25
Response (dB)
Frequency (Fs)
Figure 41 ADC Digital Filter Frequency Response Type 2 Figure 42 ADC Digital Filter Ripple Type 2
-100
-80
-60
-40
-20
0
0
0.5
1
1.5
2
2.5
3
Response (dB)
Frequency (Fs)
-0.25
-0.2
-0.15
-0.1
-0.05
0
0.05
0.1
0.15
0.2
0.25
0
0.05
0.1
0.15
0.2
0.25
Response (dB)
Frequency (Fs)
Figure 43 ADC Digital Filter Frequency Response Type 2 Figure 44 ADC Digital Filter Ripple Type 3
DE-EMPHASIS FILTER RESPONSES
-10
-8
-6
-4
-2
0
0
2000
4000
6000
8000
10000
12000
14000
16000
Response (dB)
Frequency (Fs)
-0.4
-0.3
-0.2
-0.1
0
0.1
0.2
0.3
0.4
0
2000
4000
6000
8000
10000
12000
14000
16000
Response (dB)
Frequency (Fs)
Figure 45 De-emphasis Frequency Response (32kHz)
Figure 46 De-emphasis Error (32kHz)
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-10
-8
-6
-4
-2
0
0
5000
10000
15000
20000
Response (dB)
Frequency (Fs)
-0.4
-0.3
-0.2
-0.1
0
0.1
0.2
0.3
0.4
0
5000
10000
15000
20000
Response (dB)
Frequency (Fs)
Figure 47 De-emphasis Frequency Response (44.1kHz)
Figure 48 De-emphasis Error (44.1kHz)
-10
-8
-6
-4
-2
0
0
5000
10000
15000
20000
Response (dB)
Frequency (Fs)
-0.4
-0.3
-0.2
-0.1
0
0.1
0.2
0.3
0.4
0
5000
10000
15000
20000
Response (dB)
Frequency (Fs)
Figure 49 De-emphasis Frequency Response (48kHz)
Figure 50 De-emphasis Error (48kHz)
HIGHPASS FILTER
The WM8971L has a selectable digital highpass filter in the ADC filter path to remove DC offsets. The filter response is
characterised by the following polynomial:
Figure 51 ADC Highpass Filter Response
1 - z
-1
1 - 0.9995z
-1
H(z) =
-15
-10
-5
0
0
0.0005
0.001
0.0015
0.002
Response (dB)
Frequency (Fs)
WM8971L
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APPLICATIONS INFORMATION
RECOMMENDED EXTERNAL COMPONENTS
Figure 52 Recommended External Components Diagram
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LINE INPUT CONFIGURATION
When LINPUT1/RINPUT1 or MIC are used as line inputs, the microphone boost and ALC functions
should normally be disabled.
In order to avoid clipping, the user must ensure that the input signal does not exceed AVDD. This
may require a potential divider circuit in some applications. It is also recommended to remove RF
interference picked up on any cables using a simple first-order RC filter, as high-frequency
components in the input signal may otherwise cause aliasing distortion in the audio band. AC signals
with no DC bias should be fed to the WM8971L through a DC blocking capacitor, e.g. 1
F.
MICROPHONE INPUT CONFIGURATION
Figure 53 Recommended Circuit for Line Input
For interfacing to a microphone, the ALC function should be enabled and the microphone boost
switched on. Microphones held close to a speaker's mouth would normally use the 13dB gain setting,
while tabletop or room microphones would need a 29dB boost.
The recommended application circuit is shown above. R1 and R2 form part of the biasing network
(refer to Microphone Bias section). R1 connected to MICBIAS is necessary only for electret type
microphones that require a voltage bias. R2 should always be present to prevent the microphone
input from charging to a high voltage which may damage the microphone on connection. R1 and R2
should be large so as not to attenuate the signal from the microphone, which can have source
impedance greater than 2kOhm. C1 together with the source impedance of the microphone and the
WM8971L input impedance forms an RF filter. C2 is a DC blocking capacitor to allow the microphone
to be biased at a different DC voltage to the MICIN signal.
MINIMISING POP NOISE AT THE ANALOGUE OUTPUTS
To minimise any pop or click noise when the system is powered up or down, the following procedures
are recommended.
POWER UP
Switch on power supplies. By default the WM8971L is in Standby Mode, the DAC is
digitally muted and the Audio Interface, Line outputs and Headphone outputs are all OFF
(DACMU = 1 Power Management registers 1 and 2 are all zeros).
Enable Vmid and VREF.
Enable DACs as required
Enable line and / or headphone output buffers as required.
Set DACMU = 0 to soft-un-mute the audio DACs.
POWER DOWN
Set DACMU = 1 to soft-mute the audio DACs.
Disable all output buffers.
Switch off the power supplies.
R2
47kOhm
C1
220pF
C2
1uF
AGND
AGND AGND
L/RINPUT1
MIC
FROM
MICROPHONE
R1
680 Ohm to 2.2kOhm
check microphone's specification
MICBIAS
WM8971L
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POWER MANAGEMENT EXAMPLES
POWER MANAGEMENT (1)
POWER MANAGEMENT (2)
PGAs ADCs DACs
Output
Buffers
OPERATION MODE
VREF
AI
NL
/R
PGL PGR ADL ADR MBI DAL DAR LO1 RO1 LO2 RO2 MO
Stereo Headphone Playback
1 0 0 0 0 0 0 1 1 1 1 0 0 0
Stereo Line-in Record
1 1 1 1 1 1 0 0 0 0 0 0 0 0
Stereo Microphone Record
1 1 1 1 1 1 1 0 0 0 0 0 0 0
Mono Microphone Record
1 1 1 0 1 0 1 0 0 0 0 0 0 0
Stereo Line-in to Headphone Out
1 1 0 0 0 0 0 0 0 1 1 0 0 0
Phone Call
1 1 1 0 0 0 1 0 0 1 1 0 0 1
Speaker Phone Call [ROUT2INV = 1]
1 1 1 0 0 0 1 0 0 0 0 1 1 1
Record Phone Call [L channel = mic with
boost, R channel = RX, enable mono mix]
1 1 1 1 1 1 1 0 0 1 1 0 0 1
Table 44 Register Settings for Power Management
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PACKAGE DIMENSIONS
DM030.E
FL: 32 PIN QFN PLASTIC PACKAGE 5
X
5
X
0.9 mm BODY, 0.50 mm LEAD PITCH
NOTES:
1. DIMENSION b APPLIED TO METALLIZED TERMINAL AND IS MEASURED BETWEEN 0.25 mm AND 0.30 mm FROM TERMINAL TIP. DIMENSION L1 REPRESENTS TERMINAL PULL BACK FROM
PACKAGE SIDE WALL. MAXIMUM OF 0.1mm IS ACCEPTABLE. WHERE TERMINAL PULL BACK EXISTS, ONLY UPPER HALF OF LEAD IS VISIBLE ON PACKAGE SIDE WALL DUE TO HALF
ETCHING OF LEADFRAME.
2. FALLS WITHIN JEDEC, MO-220 WITH THE EXCEPTION OF D2, E2:
D2,E2: LARGER PAD SIZE CHOSEN WHICH IS JUST OUTSIDE JEDEC SPECIFICATION
3. ALL DIMENSIONS ARE IN MILLIMETRES
4. THIS DRAWING IS SUBJECT TO CHANGE WITHOUT NOTICE.
5. SHAPE AND SIZE OF CORNER TIE BAR MAY VARY WITH PACKAGE TERMINAL COUNT. CORNER TIE BAR IS CONNECTED TO EXPOSED PAD INTERNALLY.
6. REFER TO APPLICATION NOTE WAN_0118 FOR FURTHER INFORMATION REGARDING PCB FOOTPRINTS AND QFN PACKAGE SOLDERING.
SEE DETAIL B
E2
E2/2
b
B
16
15
A
8
9
e
5
CORNER
TIE BAR
B
D2
L
D2/2
SEE DETAIL A
INDEX AREA
(D/2 X E/2)
TOP VIEW
D
C
aaa
2 X
C
aaa
2 X
E
DETAIL B
TERMI
NAL T
I
P
R
DAT
U
M
e
e/2
L1
1
DETAIL A
B
C
bbb
M
A
32x b
L
3
2x K
L1
R
1
1
0.5
66
m
m
0.4
3 m
m
5
CORNER
TIE BAR
Symbols
Dimensions (mm)
MIN
NOM
MAX
NOTE
A
A1
A3
b
D
D2
E
E2
e
L
L1
R
0.85
0.90
1.00
0.05
0.02
0
0.2 REF
0.30
0.23
0.18
5.00
3.4
3.3
3.2
0.5 BSC
0.35
0.4
0.45
0.1
b(min)/2
1
2
2
1
K
0.20
aaa
bbb
ccc
REF:
0.15
0.10
0.10
JEDEC, MO-220, VARIATION VHHD-2
Tolerances of Form and Position
4.90
5.10
5.00
4.90
5.10
3.4
3.3
3.2
1
17
24
25
32
EXPOSED
GROUND
PADDLE
6
EXPOSED
GROUND
PADDLE
BOTTOM VIEW
C
0.08
C
ccc
A
A1
C
(A3)
SEATING PLANE
1
SIDE VIEW
WM8971L
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IMPORTANT NOTICE
Wolfson Microelectronics plc (WM) reserve the right to make changes to their products or to discontinue any product or
service without notice, and advise customers to obtain the latest version of relevant information to verify, before placing
orders, that information being relied on is current. All products are sold subject to the WM terms and conditions of sale
supplied at the time of order acknowledgement, including those pertaining to warranty, patent infringement, and limitation
of liability.
WM warrants performance of its products to the specifications applicable at the time of sale in accordance with WM's
standard warranty. Testing and other quality control techniques are utilised to the extent WM deems necessary to support
this warranty. Specific testing of all parameters of each device is not necessarily performed, except those mandated by
government requirements.
In order to minimise risks associated with customer applications, adequate design and operating safeguards must be used
by the customer to minimise inherent or procedural hazards. Wolfson products are not authorised for use as critical
components in life support devices or systems without the express written approval of an officer of the company. Life
support devices or systems are devices or systems that are intended for surgical implant into the body, or support or
sustain life, and whose failure to perform when properly used in accordance with instructions for use provided, can be
reasonably expected to result in a significant injury to the user. A critical component is any component of a life support
device or system whose failure to perform can be reasonably expected to cause the failure of the life support device or
system, or to affect its safety or effectiveness.
WM assumes no liability for applications assistance or customer product design. WM does not warrant or represent that
any license, either express or implied, is granted under any patent right, copyright, mask work right, or other intellectual
property right of WM covering or relating to any combination, machine, or process in which such products or services might
be or are used. WM's publication of information regarding any third party's products or services does not constitute WM's
approval, license, warranty or endorsement thereof.
Reproduction of information from the WM web site or datasheets is permissible only if reproduction is without alteration and
is accompanied by all associated warranties, conditions, limitations and notices. Representation or reproduction of this
information with alteration voids all warranties provided for an associated WM product or service, is an unfair and deceptive
business practice, and WM is not responsible nor liable for any such use.
Resale of WM's products or services with statements different from or beyond the parameters stated by WM for that
product or service voids all express and any implied warranties for the associated WM product or service, is an unfair and
deceptive business practice, and WM is not responsible nor liable for any such use.
ADDRESS:
Wolfson Microelectronics plc
26 Westfield Road
Edinburgh
EH11 2QB
United Kingdom
Tel :: +44 (0)131 272 7000
Fax :: +44 (0)131 272 7001
Email ::
sales@wolfsonmicro.com
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Revision History
DATE
RELEASE
DESCRIPTION OF CHANGES
PAGES
14/08/03 2.0
Initial
Release
30/03/04
3.0
Pin Configuration Pins 11, 24 and 26 renamed
Ordering Information Peak Soldering Temp added and order codes changed
Pin Description Pin Descriptions for 11, 24 and 26 changed
Speaker Output Power updated
4
5
8
12/05/04
4.0
Updated to Production Data
Block Diagram ROUT2 updated to show inverting buffer, buffer added to LD2MO
and RD2MO
Pin Diagram rotated
Pin Description, Ground padded note added
Recommended Operating Conditions, DBVDD to 1.7
Electrical Characteristics updated, Speaker Output
Speaker THD and Noise Versus Power added
Fig 3, Bit Clock Mode added
Control Interface Timing 3-wire mode test conditions table updated
Power on Reset added
Speaker Output THD Versus Power removed
PGA Control text updated
Automatic Level Control text updated
Note added to Noise Gate Control table
Output Mixers, mono mixer para added
Timing Diagrams, Figs, 17,18 and 19 updated LRP = 1 changed to LRP = 0
DSP diagrams updated, Figs 20, 21
Bit Clock added
Clock Output table and text updated
Updates to Table 38 Power Management
Register Map HPFLREN added to R27, R8, Bit 8 & 7 BCM[1:0] added
Group Delay added
Recommended External Component diagram updated to show component
values, ground paddle added
External component values table deleted
Ref updated in Package Drawing
1
4
5
6
8
11
14
16
17
21
25
27
31
39
41
43
44
48
50
51
56
57
59
16/08/05 4.2
Ordering information leaded parts removed
Pin description - Changed pin 19 description
Electrical Characteristics ADC THD max values removed
Updated Operation Mode table (Stereo Headphone playback, AINL/UR changed
from 1 to 0)
Package Drawing updated to DM030.E to include ground paddle info
4
5
7
58
59