w WM8510
Mono CODEC with Speaker Driver
WOLFSON MICROELECTRONICS plc
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Production Data, September 2008, Rev 4.5
Copyright ©2008 Wolfson Microelectronics plc
DESCRIPTION
The WM8510 is a low power, high quality mono codec designed
for Voice over Internet Protocol (VoIP) and Digital Telephones.
The device integrates support for one pseudo-differential and
one single ended input (Handset Mic and Speaker Mic) and
includes drivers for speakers or headset, and mono line output,
making it ideal for Telephone designs. External component
requirements are reduced as no separate microphone or
earpiece amplifiers are required.
Advanced Sigma Delta Converters are used along with digital
decimation and interpolation filters to give high quality audio at
sample rates from 8 to 48kHz.
Additional digital filtering options are available in the ADC path,
to cater for application filtering such as ‘wind noise reduction’,
plus an advanced mixed signal ALC function with noise gate is
provided.
An on-chip PLL is provided to generate the required Master
Clock from an external reference clock. The PLL clock can also
be output if required elsewhere in the system.
The WM8510 operates at supply voltages from 2.5 to 3.6V,
although the digital supplies can operate at voltages down to
1.71V to save power. The speaker and mono outputs use a
separate supply of up to 5V which enables increased output
power if required. Different sections of the chip can also be
powered down under software control by way of the selectable
two or three wire control interface.
WM8510 is supplied in a convenient 28-lead SSOP package,
offering high levels of functionality in an easy to use package.
BLOCK DIAGRAM
FEATURES
• Mono Codec:
• Audio sample rates:8, 11.025, 16, 22.05, 24, 32, 44.1,
48kHz
• DAC SNR 93dB, THD -84dB (‘A’-weighted @ 8 – 48kHz)
• ADC SNR 90dB, THD -80dB (‘A’-weighted @ 8 – 48kHz)
• On-chip Headphone/Speaker Driver with ‘cap-less’ connect
- 40mW output power into 16Ω / 3.3V SPKVDD
- BTL speaker drive 0.8W into 8Ω / 5V SPKVDD
• Earpiece Line output
• Multiple analog inputs, plus analog bypass path (0 or -10dB)
• Mic Preamps:
• Two Microphone Interfaces
- One pseudo-differential input with common mode
rejection
- One single ended input
- Programmable preamp gain
- Programmable ALC / Noise Gate in ADC path
• Low-noise bias supplied for microphone
Other Features
• Digital Playback Limiter
• Programmable ADC High Pass Filter (wind noise reduction)
• Programmable ADC Notch Filter
• On-chip PLL
• Low power, low voltage
- 2.5V to 3.6V (digital supplies: 1.71V to 3.6V)
- power consumption <10mW all-on 48kHz mode
• 28 lead SSOP package
APPLICATIONS
• VoIP Telephones
• Digital Telephones
• Conference Speaker-phone
• Mobile Telephone Hands-free Kits
• General Purpose low power audio CODEC
CONTROL
INTERFACE
MICBIAS
250k250k
I2S or PCM
INTERFACE
ADC
DAC
DIGITAL
FILTERS
Volume
Digital
Limiter
ADC
DIGITAL
FILTERS
Volume
Limiter /
ALC
Wind Noise
Filters
PLL
25k 25k
4k 5k
SIDETONE SPKOUTP
SPKOUTN
DAC
-1
L - (-R)
= L+R
MONO OUT
Rbia s
Mic
NOISY
GND
Gains
: -12dB to
+35.25dB
MICN
MICP IP PGA IP BOOST/MIX
SPKR PGA
20k
20k
MIC2
-10dB or +0dB
-10dB or +0dB
W
WM8510
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TABLE OF CONTENTS
DESCRIPTION .......................................................................................................1
BLOCK DIAGRAM .................................................................................................1
FEATURES.............................................................................................................1
APPLICATIONS .....................................................................................................1
PIN CONFIGURATION...........................................................................................3
ORDERING INFORMATION ..................................................................................3
PIN DESCRIPTION ................................................................................................4
ABSOLUTE MAXIMUM RATINGS.........................................................................5
RECOMMENDED OPERATING CONDITIONS .....................................................5
ELECTRICAL CHARACTERISTICS ......................................................................6
TERMINOLOGY ............................................................................................................ 8
SIGNAL TIMING REQUIREMENTS .......................................................................9
SYSTEM CLOCK TIMING ............................................................................................. 9
AUDIO INTERFACE TIMING – MASTER MODE .......................................................... 9
AUDIO INTERFACE TIMING – SLAVE MODE............................................................ 10
CONTROL INTERFACE TIMING – 3-WIRE MODE .................................................... 11
CONTROL INTERFACE TIMING – 2-WIRE MODE .................................................... 12
DEVICE DESCRIPTION.......................................................................................13
INTRODUCTION......................................................................................................... 13
INPUT SIGNAL PATH ................................................................................................. 14
ANALOGUE TO DIGITAL CONVERTER (ADC).......................................................... 19
INPUT LIMITER / AUTOMATIC LEVEL CONTROL (ALC) .......................................... 23
OUTPUT SIGNAL PATH ............................................................................................. 36
ANALOGUE OUTPUTS............................................................................................... 41
OUTPUT SWITCH ...................................................................................................... 46
DIGITAL AUDIO INTERFACES................................................................................... 48
AUDIO SAMPLE RATES............................................................................................. 53
MASTER CLOCK AND PHASE LOCKED LOOP (PLL) ............................................... 54
GENERAL PURPOSE INPUT/OUTPUT...................................................................... 56
CONTROL INTERFACE.............................................................................................. 56
RESETTING THE CHIP .............................................................................................. 57
POWER SUPPLIES .................................................................................................... 58
POWER MANAGEMENT ............................................................................................ 61
REGISTER MAP...................................................................................................64
REGISTER BITS BY ADDRESS ................................................................................. 65
DIGITAL FILTER CHARACTERISTICS ...............................................................76
TERMINOLOGY .......................................................................................................... 76
DAC FILTER RESPONSES......................................................................................... 77
ADC FILTER RESPONSES......................................................................................... 77
DE-EMPHASIS FILTER RESPONSES........................................................................ 78
HIGHPASS FILTER..................................................................................................... 79
APPLICATIONS INFORMATION .........................................................................80
RECOMMENDED EXTERNAL COMPONENTS.......................................................... 80
IMPORTANT NOTICE ..........................................................................................82
ADDRESS ................................................................................................................... 82
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PIN CONFIGURATION
ORDERING INFORMATION
ORDER CODE TEMPERATURE
RANGE
PACKAGE MOISTURE SENSITIVITY
LEVEL
PACKAGE BODY
TEMPERATURE
WM8510GEDS/V -40°C to +85°C 28-lead SSOP
(Pb-free)
MSL3 260oC
WM8510GEDS/RV -40°C to +85°C 28-lead SSOP
(Pb-free, tape and reel)
MSL3 260oC
Note:
Reel Quantity = 2,000
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PIN DESCRIPTION
PIN NAME TYPE DESCRIPTION
1 VMID Reference Decoupling for midrail reference voltage
2 MICN Analog Input Microphone negative input
3 MICP Analog Input Microphone positive input (common mode)
4 MICBIAS Analog Output Microphone Bias
5 NC NC No Connect
6 AVDD Supply Analogue supply (feeds ADC, DAC and PLL)
7 AGND Supply Analogue ground (feeds ADC, DAC and PLL)
8 AGND Supply Analogue ground (feeds ADC, DAC and PLL)
9 DCVDD Supply Digital Core supply
10 DBVDD Supply Digital Buffer (Input/Output) supply
11 DGND Supply Digital ground
12 ADCDAT Digital Output ADC Digital Audio Data Output
13 DACDAT Digital Input DAC Digital Audio Data Input
14 FRAME Digital Input/Output DAC and ADC Sample Rate Clock or Frame synch
15 BCLK Digital Input/Output Digital Audio Port Clock
16 MCLK Digital Input Master Clock Input
17 CSB/GPIO Digital Input/Output 3-Wire MPU Chip Select or General Purpose Input/Output pin.
18 SCLK Digital Input 3-Wire MPU Clock Input / 2-Wire MPU Clock Input
19 SDIN Digital Input/Output 3-Wire MPU Data Input / 2-Wire MPU Data Input/Output
20 MODE Digital Input Control Interface Mode Selection Pin.
21 MONOOUT Analog Output Mono Audio Output
22 SPKOUTP Analog Output Speaker Output Positive
23 SPKGND Supply Speaker ground (feeds speaker and mono output amps only)
24 SPKGND Supply Speaker ground (feeds speaker and mono output amps only)
25 SPKOUTN Analog Output Speaker Output Negative
26 SPKVDD Supply Speaker supply (feeds speaker and mono output amps only)
27 SPKVDD Supply Speaker supply (feeds speaker and mono output amps only)
28 MIC2 Analog Input Second Analog Input
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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
DBVDD, DCVDD, AVDD supply voltages -0.3V +3.63V
SPKVDD supply voltage -0.3V +7V
Voltage range digital inputs DGND -0.3V DVDD +0.3V
Voltage range analogue inputs AGND -0.3V AVDD +0.3V
Operating temperature range, TA -40°C +85°C
Storage temperature after soldering -65°C +150°C
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. When using the PLL, DCVDD should be ≥ 1.9V
RECOMMENDED OPERATING CONDITIONS
PARAMETER SYMBOL TEST
CONDITIONS
MIN TYP MAX UNIT
Digital supply range (Core) DCVDD 1.71 3.6 V
Digital supply range (Buffer) DBVDD 1.71 3.6 V
Analogue supplies range AVDD 2.5 3.6 V
Speaker supply SPKVDD 2.5 5.5 V
Ground DGND,AGND, SPKGND 0 V
Notes
1. DCVDD ≤ DBVDD at all times.
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ELECTRICAL CHARACTERISTICS
Test Conditions
DCVDD = 1.8V, AVDD = DBVDD = 3.3V, SPKVDD = 3.3V, TA = +25oC, 1kHz signal, fs = 48kHz, 24-bit audio data unless
otherwise stated.
PARAMETER SYMBOL TEST CONDITIONS MIN TYP MAX UNIT
Microphone Inputs (MICN, MICP)
Full-scale Input Signal Level
(Note 1) – note this changes
with AVDD
VINFS PGABOOST = 0dB
INPPGAVOL = 0dB
1.0
0
Vrms
dBV
Mic PGA equivalent input noise At 35.25dB
gain
150 uV
Input resistance RMICIN Gain set to 35.25dB 1.6 kΩ
Input resistance RMICIN Gain set to 0dB 47 kΩ
Input resistance RMICIN Gain set to -12dB 75 kΩ
Input resistance RMICIP MICP2INPPGA = 1 94 kΩ
Input Capacitance CMICIN 10 pF
Recommended coupling cap CCOUP 220 pF
MIC Input Programmable Gain Amplifier (PGA)
Programmable Gain -12 35.25 dB
Programmable Gain Step Size Guaranteed monotonic 0.75 dB
Mute Attenuation 108 dB
Selectable Input Gain Boost (0/+20dB)
Gain Boost 0 20 dB
Automatic Level Control (ALC)/Limiter – ADC only
Target Record Level -28.5 -6 dB
Programmable Gain -12 35.25 dB
Programmable Gain Step Size Guaranteed Monotonic 0.75 dB
Gain Hold Time (Note 2) tHOLD MCLK=12.288MHz
(Note 4)
0, 2.67, 5.33, 10.67, … , 43691
(time doubles with each step)
ms
ALCMODE=0 (ALC),
MCLK=12.288MHz
(Note 4)
3.3, 6.6, 13.1, … , 3360
(time doubles with each step)
Gain Ramp-Up (Decay) Time
(Note 3)
tDCY
ALCMODE=1 (limiter),
MCLK=12.288MHz
(Note 4)
0.73, 1.45, 2.91, … , 744
(time doubles with each step)
ms
ALCMODE=0 (ALC),
MCLK=12.288MHz
(Note 4)
0.83, 1.66, 3.33, … , 852
(time doubles with each step)
Gain Ramp-Down (Attack) Time
(Note 3)
tATK
ALCMODE=1 (limiter),
MCLK=12.288MHz
(Note 4)
0.18, 0.36, 0.73, … , 186
(time doubles with each step)
ms
Analogue to Digital Converter (ADC)
Signal to Noise Ratio (Note 5) SNR A-weighted, 0dB PGA
gain
87 90 dB
Total Harmonic Distortion
(Note 6)
THD
-
1dBFS input, 0dB PGA
gain
-80 -65 dB
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Test Conditions
DCVDD = 1.8V, AVDD = DBVDD = 3.3V, SPKVDD = 3.3V, TA = +25oC, 1kHz signal, fs = 48kHz, 24-bit audio data unless
otherwise stated.
PARAMETER SYMBOL TEST CONDITIONS MIN TYP MAX UNIT
MIC2 Analogue Input
Full-scale Input Signal Level
(0dB) – note this scales with
AVDD
VINFS 1.0
0
Vrms
dBV
Input Resistance RMIC2IN MIC2MODE=0 20 kΩ
Input Capacitance CMIC2IN 10 pF
Digital to Analogue Converter (DAC) to MONO output (all data measured with 10kΩ / 50pF load)
Signal to Noise Ratio (Note 5) SNR A-weighted 90 93 dB
Total Harmonic Distortion
(Note 6)
THD RL = 10 kΩ
full-scale signal
-84 -70 dB
MONOBOOST=0 AVDD/3.3 0dB Full Scale output voltage
(Note 7)
MONOBOOST=1
1.5x
(AVDD/3.3)
V
RMS
Speaker Output PGA
Programmable Gain -57 6 dB
Programmable Gain Step Size Guaranteed monotonic 1 dB
BTL Speaker Output (SPKOUTP, SPKOUTN with 8Ω bridge tied load)
Output Power PO Output power is very closely correlated with THD; see below
PO =180mW, RL = 8Ω,
SPKVDD=3.3V
0.03
-70
%
dB
PO =400mW, RL = 8Ω,
SPKVDD=3.3V
5.0
-26
%
dB
PO =360mW, RL = 8Ω,
SPKVDD=5V
0.02
-75
%
dB
Total Harmonic Distortion THD
PO =800mW, RL = 8Ω,
SPKVDD=5V
0.06
-65
%
dB
SPKVDD=3.3V,
RL = 8Ω
90 dB
Signal to Noise Ratio SNR
SPKVDD=5V,
RL = 8Ω
90 dB
Power Supply Rejection Ratio 50 dB
‘Headphone’ output (SPKOUTP, SPKOUTN with resistive load to ground)
Signal to Noise Ratio SNR 93 dB
THD Po=20mW, RL = 16Ω,
SPKVDD=3.3V
0.02
-74
%
dB
Total Harmonic Distortion
Po=20mW, RL = 32Ω,
SPKVDD=3.3V
0.017
- 75
%
dB
Microphone Bias
Bias Voltage (MBVSEL=0) VMICBIAS 0.9*AVDD V
Bias Voltage (MBVSEL=1) VMICBIAS 0.65*AVDD V
Bias Current Source IMICBIAS 3 mA
Output Noise Voltage Vn 1kHz to 20kHz 15 nV/√Hz
Digital Input / Output
Input HIGH Level VIH 0.7×DVDD V
Input LOW Level VIL 0.3×DVDD V
Output HIGH Level VOH I
OL=1mA 0.9×DVDD V
Output LOW Level VOL I
OH-1mA 0.1xDVDD V
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TERMINOLOGY
1. MICN input only in single ended microphone configuration. Maximum input signal to MICP without distortion is -3dBV.
2. Hold Time is the length of time between a signal detected being too quiet and beginning to ramp up the gain. It does
not apply to ramping down the gain when the signal is too loud, which happens without a delay.
3. Ramp-up and Ramp-Down times are defined as the time it takes the PGA to change its gain by 6dB.
4. All hold, ramp-up and ramp-down times scale proportionally with MCLK
5. Signal-to-noise ratio (dB) – SNR is a measure of the difference in level between the full scale output and the output with
no signal applied. (No Auto-zero or Automute function is employed in achieving these results).
6. THD (dB) – THD is a ratio, of the rms values, of Noise Signal.
7. The maximum output voltage can be limited by the speaker power supply. If MONOBOOST=1 then SPKVDD should
be 1.5xAVDD or higher to prevent clipping taking place in the output stage.
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SIGNAL TIMING REQUIREMENTS
SYSTEM CLOCK TIMING
MCLK
t
MCLKL
t
MCLKH
t
MCLKY
Figure 1 System Clock Timing Requirements
Test Conditions
DCVDD=1.8V, DBVDD=AVDD=SPKVDD=3.3V, DGND=AGND=SPKGND=0V, TA = +25oC, Slave Mode
PARAMETER SYMBOL MIN TYP MAX UNIT
System Clock Timing Information
MCLK System clock cycle time TMCLKY Tbd ns
MCLK duty cycle TMCLKDS 60:40 40:60
AUDIO INTERFACE TIMING – MASTER MODE
Figure 2 Digital Audio Data Timing – Master Mode (see Control Interface)
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Test Conditions
DCVDD=1.8V, DBVDD=AVDD=SPKVDD=3.3V, DGND=AGND=SPKGND=0V, TA=+25oC, Master Mode, fs=48kHz,
MCLK=256fs, 24-bit data, unless otherwise stated.
PARAMETER SYMBOL MIN TYP MAX UNIT
Audio Data Input Timing Information
FRAME propagation delay from BCLK falling edge tDL 10 ns
ADCDAT propagation delay from BCLK falling edge tDDA 10 ns
DACDAT setup time to BCLK rising edge tDST 10 ns
DACDAT hold time from BCLK rising edge tDHT 10 ns
AUDIO INTERFACE TIMING – SLAVE MODE
Figure 3 Digital Audio Data Timing – Slave Mode
Test Conditions
DCVDD=1.8V, DBVDD=AVDD=SPKVDD=3.3V, DGND=AGND=SPKGND=0V, TA=+25oC, 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 tBCY 50 ns
BCLK pulse width high tBCH 20 ns
BCLK pulse width low tBCL 20 ns
FRAME set-up time to BCLK rising edge tLRSU 10 ns
FRAME hold time from BCLK rising edge tLRH 10 ns
DACDAT hold time from BCLK rising edge tDH 10 ns
DACDAT set-up time to BCLK rising edge tDS 10 ns
ADCDAT propagation delay from BCLK falling edge tDD 20 ns
Note:
BCLK period should always be greater than or equal to MCLK period.
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CONTROL INTERFACE TIMING – 3-WIRE MODE
Figure 4 Control Interface Timing – 3-Wire Serial Control Mode
Test Conditions
DCVDD = 1.8V, DBVDD = AVDD = SPKVDD = 3.3V, DGND = AGND = SPKGND = 0V, TA = +25oC, 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 tSCS 80 ns
SCLK pulse cycle time tSCY 200 ns
SCLK pulse width low tSCL 80 ns
SCLK pulse width high tSCH 80 ns
SDIN to SCLK set-up time tDSU 40 ns
SCLK to SDIN hold time tDHO 40 ns
CSB pulse width low tCSL 40 ns
CSB pulse width high tCSH 40 ns
CSB rising to SCLK rising tCSS 40 ns
Pulse width of spikes that will be suppressed tps 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 5 Control Interface Timing – 2-Wire Serial Control Mode
Test Conditions
DCVDD=1.8V, DBVDD=AVDD=SPKVDD=3.3V, DGND=AGND=SPKGND=0V, TA = +25oC, Slave Mode, fs = 48kHz, MCLK =
256fs, 24-bit data, unless otherwise stated.
PARAMETER SYMBOL MIN TYP MAX UNIT
Program Register Input Information
SCLK Frequency 0 526 kHz
SCLK Low Pulse-Width t1 1.3 us
SCLK High Pulse-Width t2 600 ns
Hold Time (Start Condition) t3 600 ns
Setup Time (Start Condition) t4 600 ns
Data Setup Time t5 100 ns
SDIN, SCLK Rise Time t6 300 ns
SDIN, SCLK Fall Time t7 300 ns
Setup Time (Stop Condition) t8 600 ns
Data Hold Time t9 900 ns
Pulse width of spikes that will be suppressed tps 0 5 ns
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DEVICE DESCRIPTION
INTRODUCTION
The WM8510 is a low power audio codec combining a high quality mono audio DAC and ADC, with
flexible line and microphone input and output processing. Applications for this device are anticipated
to include VoIP telephones, digital telephones, conference speaker phones and mobile hands-free
kits.
FEATURES
The chip offers great flexibility in use, and so can support many different modes of operation as
follows:
MICROPHONE INPUTS
Two microphone inputs are provided, allowing for either a differential microphone input or a single
ended microphone to be connected. These inputs have a user programmable gain range of -12dB to
+35.25dB using internal resistors. After the input PGA stage comes a boost stage which can add a
further 20dB of gain. A microphone bias is output from the chip which can be used to bias the
microphones. The signal routing can be configured to allow manual adjustment of mic levels, or to
allow the ALC loop to control the level of mic signal that is transmitted.
Total gain through the microphone paths of up to +55.25dB can be selected.
FLEXIBLE MIC2 INPUT
The flexible configuration of the mono input, MIC2, with integrated on-chip resistors allows several
analogue signals to be summed into the single input if required. This can be used as a microphone,
line input or an input for warning tones (beep) etc. The output from this circuit can be summed into
the mono output and/or the speaker output paths, so allowing for mixing of audio with ‘backing music’
etc as required.
SIDETONE ATTENUATION
A bypass path allows analog signals to travel directly to the outputs without passing through the ADC
and DAC. For side tone features in telephone handsets this analogue bypass can be attenuated.
PGA AND ALC OPERATION
A programmable gain amplifier is provided in the input path to the ADC. This may be used manually
or in conjunction with a mixed analogue/digital automatic level control (ALC) which keeps the
recording volume constant.
ADC
The mono ADC uses a multi-bit high-order oversampling architecture to deliver optimum
performance with low power consumption. Various sample rates are supported, from the 8ks/s rate
typically used in voice dictation, up to the 48ks/s rate used in high quality audio applications.
HI-FI DAC
The hi-fi DAC provides high quality audio playback suitable for all portable mono audio type
applications.
DIGITAL FILTERING
Advanced Sigma Delta Converters are used along with digital decimation and interpolation filters to
give high quality audio at sample rates from 8ks/s to 48ks/s.
Application specific digital filters are also available which help to reduce the effect of specific noise
sources such as ‘wind noise’. The filters include a programmable ADC high pass filter and a
programmable ADC notch filter.
OUTPUT MIXING AND VOLUME ADJUST
Flexible mixing is provided on the outputs of the device; a mixer is provided for the speaker outputs,
and an additional mono summer for the mono output. These mixers allow the output of the DAC, the
output of the ADC volume control and the MIC2 input to be combined. The output volume can be
adjusted using the integrated digital volume control and there is additional analogue gain adjustment
capability on the speaker output.
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AUDIO INTERFACES
The WM8510 has a standard audio interface, to support the transmission of audio data to and from
the chip. This interface is a 4 wire standard audio interface which supports a number of audio data
formats including I2S, DSP Mode, MSB-First, left justified and MSB-First, right justified, and can
operate in master or slave modes.
CONTROL INTERFACES
To allow full software control over all its features, the WM8510 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 selection between 2-wire mode and 3-wire mode is
determined by the state of the MODE pin. If MODE is high then 3-wire control mode is selected, if
MODE is low then 2-wire control mode is selected.
In 2 wire mode, only slave operation is supported, and the address of the device is fixed as 0011010.
CLOCKING SCHEMES
WM8510 offers the normal audio DAC clocking scheme operation, where 256fs MCLK is provided to
the DAC/ADC.
However, a PLL is also included which may be used to generate the internal master clock frequency
in the event that this is not available from the system controller. This PLL uses an input clock,
typically the 12MHz USB or ilink clock, to generate high quality audio clocks. If this PLL is not
required for generation of these clocks, it can be reconfigured to generate alternative clocks which
may then be output on the CSB/GPIO pin and used elsewhere in the system.
POWER CONTROL
The design of the WM8510 has given much attention to power consumption without compromising
performance. It operates at low supply voltages, and includes the facility to power off any unused
parts of the circuitry under software control, includes standby and power off modes.
INPUT SIGNAL PATH
The WM8510 has 3 flexible analogue inputs for two separate microphone inputs. These inputs can
be used in a variety of ways. The input signal path before the ADC has a flexible PGA block which
then feeds into a gain boost/mixer stage.
MICROPHONE INPUTS
The WM8510 can accommodate a variety of microphone configurations including single ended and
pseudo-differential inputs. The inputs through the MICN, MICP and optionally MIC2 pins are
amplified through the input PGA as shown in Figure 6.
A pseudo differential input is the preferential configuration where the positive terminal of the input
PGA is connected to the MICP input pin by setting MICP2INPPGA=1. The microphone ground
should then be connected to MICN (when MICN2INPPGA=1) or optionally to MIC2 (when
MIC2_2INPPGA=1) input pins.
Alternatively a single ended microphone can be connected to the MICN input with MICN2INPPGA set
to 1. The non-inverting terminal of the input PGA should be connected internally to VMID by setting
MICP2INPPGA to 0.
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Figure 6 Microphone Input PGA Circuit
(switch positions shown are for pseudo-differential mic input)
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
0 MICP2INPPGA 1 Connect input PGA amplifier positive
terminal to MICP or VMID.
0 = input PGA amplifier positive terminal
connected to VMID
1 = input PGA amplifier positive terminal
connected to MICP through variable
resistor string
1 MICN2INPPGA 1 Connect MICN to input PGA negative
terminal.
0=MICN not connected to input PGA
1=MICN connected to input PGA
amplifier negative terminal.
R44
Input
Control
2 MIC2_2INPPGA 0 Select MIC2 amplifier output as input
PGA signal source.
0=MIC2 not connected to input PGA
1=MIC2 connected to input PGA
amplifier negative terminal.
The input PGA is enabled by the INPGAEN register bit.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R2
Power
Management
2
2 INPGAEN 0 Input microphone PGA enable
0 = disabled
1 = enabled
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INPUT PGA VOLUME CONTROL
The input microphone PGA has a gain range from -12dB to +35.25dB in 0.75dB steps. The gain
from the MICN input to the PGA output and from the MIC2 amplifier to the PGA output are always
common and controlled by the register bits INPPGAVOL[5:0]. These register bits also affect the
MICP pin when MICP2INPPGA=1.
When the Automatic Level Control (ALC) is enabled the input PGA gain is then controlled
automatically and the INPPGAVOL bits should not be used.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
5:0 INPPGAVOL 010000 Input PGA volume
000000 = -12dB
000001 = -11.25db
.
010000 = 0dB
.
111111 = +35.25dB
6 INPPGAMUTE 0 Mute control for input PGA:
0=Input PGA not muted, normal operation
1=Input PGA muted (and disconnected
from the following input BOOST stage).
R45
Input PGA
volume
control
7 INPPGAZC 0 Input PGA zero cross enable:
0=Update gain when gain register
changes
1=Update gain on 1st zero cross after gain
register write.
R32
ALC control
1
8 ALCSEL 0 ALC function select:
0=ALC off (PGA gain set by INPPGAVOL
register bits)
1=ALC on (ALC controls PGA gain)
Table 1 Input PGA Volume Control
MIC 2 INPUT
A second mic input circuit, MIC2 (Figure 7) is provided which consists of an amplifier which can be
configured either as an inverting buffer for a single input signal or as a mixer/summer for multiple
inputs with the use of external resistors. The circuit is enabled by the register bit MIC2EN.
Figure 7 MIC2 Input Circuit
The MIC2MODE register bit controls the input mode of operation:
In buffer mode (MIC2MODE=0) the switch labelled MIC2SW in Figure 7 is open and the signal at the
MIC2 pin will be buffered and inverted through the MIC2 circuit using only the internal components.
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In mixer mode (MIC2MODE=1) the on-chip input resistor is bypassed, this allows the user to sum in
multiple inputs with the use of external resistors. When used in this mode there will be gain
variations through this path from part to part due to the variation of the internal 20kΩ resistors relative
to the higher tolerance external resistors.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R1
Power
management
1
6 MIC2EN 0 MIC2 input buffer enable
0 = OFF
1 = ON
R44
Input control
3 MIC2MODE 0 0 = inverting buffer
1 = mixer (on-chip input resistor bypassed)
Table 2 MIC2 Input Buffer Control
INPUT BOOST
The input BOOST circuit has 3 selectable inputs: the input microphone PGA output, the MIC2
amplifier output and the MICP input pin (when not using a differential microphone configuration).
These three inputs can be mixed together and have individual gain boost/adjust as shown in Figure
8.
Figure 8 Input Boost Stage
The input PGA path can have a +20dB boost (PGABOOST=1) a 0dB pass through (PGABOOST=0)
or be completely isolated from the input boost circuit (INPPGAMUTE=1).
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R45
Input PGA
gain control
6 INPPGAMUTE 0 Mute control for input PGA:
0=Input PGA not muted, normal
operation
1=Input PGA muted (and
disconnected from the following input
BOOST stage).
R47
Input BOOST
control
8 PGABOOST 1 0 = PGA output has +0dB gain
through input BOOST stage.
1 = PGA output has +20dB gain
through input BOOST stage.
Table 3 Input BOOST Stage Control
The MIC2 amplifier path to the BOOST stage is controlled by the MIC2_2BOOSTVOL[2:0] register
bits. When MIC2_2BOOSTVOL=000 this path is completely disconnected from the BOOST stage.
Settings 001 through to 111 control the gain in 3dB steps from -12dB to +6dB.
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The MICP path to the BOOST stage is controlled by the MICP2BOOSTVOL[2:0] register bits. When
MICP2BOOSTVOL=000 this input pin is completely disconnected from the BOOST stage. Settings
001 through to 111 control the gain in 3dB steps from -12dB to +6dB.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
2:0 MIC2_2BOOSTV
OL
000 Controls the MIC2 amplifier to the
input boost stage:
000=Path disabled (disconnected)
001=-12dB gain through boost stage
010=-9dB gain through boost stage
…
111=+6dB gain through boost stage
R47
Input BOOST
control
6:4 MICP2BOOSTVOL 000 Controls the MICP pin to the input
boost stage (NB, when using this
path set MICPZIUNPPGA=0):
000=Path disabled (disconnected)
001=-12dB gain through boost stage
010=-9dB gain through boost stage
…
111=+6dB gain through boost stage
Table 4 Input BOOST Stage Control
The BOOST stage is enabled under control of the BOOSTEN register bit.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R2
Power
management
2
4 BOOSTEN 0 Input BOOST enable
0 = Boost stage OFF
1 = Boost stage ON
Table 5 Input BOOST Enable Control
MICROPHONE BIASING CIRCUIT
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 MICBIAS voltage can be altered via
the MBVSEL register bit. When MBVSEL=0, MICBIAS=0.9*AVDD and when MBVSEL=1,
MICBIAS=0.75*AVDD. The output can be enabled or disabled using the MICBEN control bit.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R1
Power
management 1
4 MICBEN 0 Microphone Bias Enable
0 = OFF (high impedance output)
1 = ON
Table 6 Microphone Bias Enable
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R44
Input Control
8 MBVSEL 0 Microphone Bias Voltage Control
0 = 0.9 * AVDD
1 = 0.65 * AVDD
Table 7 Microphone Bias Voltage Control
The internal MICBIAS circuitry is shown in Figure 9. Note that the maximum source current
capability for MICBIAS is 3mA. The external biasing resistors therefore must be large enough to limit
the MICBIAS current to 3mA.
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Figure 9 Microphone Bias Schematic
ANALOGUE TO DIGITAL CONVERTER (ADC)
The WM8510 uses a multi-bit, oversampled sigma-delta ADC 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 1.0Vrms.
Any voltage greater than full scale may overload the ADC and cause distortion.
ADC DIGITAL FILTERS
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.
Figure 10 ADC Digital Filter Path
A
GND
MBVSEL=0
MICBIAS
= 1.8 x VMID
= 0.9 X AVDD
VMID
internal
resistor
internal
resistor
MB
MBVSEL=1
MICBIAS
= 1.3 x VMID
= 0.65 X AVDD
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The ADC is enabled by the ADCEN register bit.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R2
Power
management 2
0 ADCEN 0 0 = ADC disabled
1 = ADC enabled
Table 8 ADC Enable
The polarity of the output signal can also be changed under software control using the ADCPOL
register bit. The oversampling rate of the ADC can be adjusted using the ADCOSR register bit.
With ADCOSR=0 the oversample rate is 64x which gives lowest power operation and when
ADCOSR=1 the oversample rate is 128x which gives best performance.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
3 ADCOSR 0 ADC oversample rate select:
0=64x (lower power)
1=128x (best performance)
R14
ADC Control
0 ADCPOL 0 0=normal
1=inverted
Table 9 ADC Oversample Rate Select
SELECTABLE HIGH PASS FILTER
A selectable high pass filter is provided. To disable this filter set HPFEN=0. The filter has two
modes controlled by HPFAPP. In Audio Mode (HPFAPP=0) the filter is first order, with a cut-off
frequency of 3.7Hz. In Application Mode (HPFAPP=1) the filter is second order, with a cut-off
frequency selectable via the HPFCUT register. The cut-off frequencies when HPFAPP=1 are shown
in Table 11
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
8 HPFEN 1 High Pass Filter Enable
0=disabled
1=enabled
7 HPFAPP 0 Select audio mode or application mode
0=Audio mode (1st order, fc = ~3.7Hz)
1=Application mode (2nd order, fc =
HPFCUT)
R14
ADC Control
6:4 HPFCUT 000 Application mode cut-off frequency
See Table 11 for details.
Table 10 ADC Filter Select
FS (KHZ)
SR=101/100 SR=011/010 SR=001/000
HPFCUT
8 11.025 12 16 22.05 24 32 44.1 48
000 82 113 122 82 113 122 82 113 122
001 102 141 153 102 141 153 102 141 153
010 131 180 196 131 180 196 131 180 196
011 163 225 245 163 225 245 163 225 245
100 204 281 306 204 281 306 204 281 306
101 261 360 392 261 360 392 261 360 392
110 327 450 490 327 450 490 327 450 490
111 408 563 612 408 563 612 408 563 612
Table 11 High Pass Filter Cut-off Frequencies (HPFAPP=1)
Note that the High Pass filter values (when HPFAPP=1) work on the basis that the SR register bits
are set correctly for the actual sample rate as shown in Table 11.
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PROGRAMMABLE NOTCH FILTER
A programmable notch filter is provided. This filter has a variable centre frequency and bandwidth,
programmable via two coefficients, a0 and a1. These coefficients should be converted to 2’s
complement numbers to determine the register values. a0 and a1 are represented by the register bits
NFA0[13:0] and NFA1[13:0]. Because these coefficient values require four register writes to setup
there is an NFU (Notch Filter Update) flag which should be set only when all four registers are setup.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
6:0 NFA0[13:7] 0 Notch Filter a0 coefficient, bits [13:7]
7 NFEN 0 Notch filter enable:
0=Disabled
1=Enabled
R27
Notch Filter 1
8 NFU 0 Notch filter update. The notch filter
values used internally only update when
one of the NFU bits is set high.
6:0 NFA0[6:0] 0 Notch Filter a0 coefficient, bits [6:0] R28
Notch Filter 2 8 NFU] 0 Notch filter update. The notch filter
values used internally only update when
one of the NFU bits is set high.
6:0 NFA1[13:7] 0 Notch Filter a1 coefficient, bits [13:7] R29
Notch Filter 3 8 NFU 0 Notch filter update. The notch filter
values used internally only update when
one of the NFU bits is set high.
6:0 NFA1[6:0] 0 Notch Filter a1 coefficient, bits [6:0] R30
Notch Filter 4 8 NFU 0 Notch filter update. The notch filter
values used internally only update when
one of the NFU bits is set high.
Table 12 Notch Filter Function
The coefficients are calculated as follows:
)2/tan(1
)2/tan(1
0
b
b
w
w
a+
−
=
)cos()1( 001 waa +−=
Where:
sc ffw /2
0
π
=
sbb ffw /2
π
=
fc = centre frequency in Hz, fb = -3dB bandwidth in Hz, fs = sample frequency in Hz
The coefficients are calculated as follows:
NFA0 = -a0 x 213
NFA1 = -a1 x 212
These values are then converted to 2’s complement notation to determine the register values.
NOTCH FILTER WORKED EXAMPLE
The following example illustrates how to calculate the a0 and a1 coefficients for a desired centre
frequency and -3dB bandwidth.
fc = 1000 Hz
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fb = 100 Hz
fs = 48000 Hz
sc0 f/f2w π= = π2 x (1000 / 48000) = 0.1308996939 rads
sbb f/f2w π= = π2 x (100 / 48000) = 0.01308996939 rads
)2/wtan(1
)2/wtan(1
a
b
b
0+
−
=
= )2/90130899693.0tan(1
)2/90130899693.0tan(1
+
−
= 0.9869949627
)wcos()a1(a 001 +−= = )1308996939.0cos()9869949627.01( +− = -1.969995945
NFn_A0 = -a0 x 213 = -8085 (rounded to nearest whole number)
NFn_A1 = -a1 x 212 = 8069 (rounded to nearest whole number)
These values are then converted to 2’s complement:
NFA0 = 14’h206B = 14’b10000001101011
NFA1 = 14’h1F85 = 14’b 01111110000101
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DIGITAL ADC VOLUME CONTROL
The output of the ADCs can be digitally attenuated over a range from –127dB to 0dB in 0.5dB steps.
The gain for a given eight-bit code X is given by:
Gain = 0.5 x (x–255) dB for 1 ≤ x ≤ 255, MUTE for x = 0
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R15
ADC Digital
Volume
7:0 ADCVOL
[7:0]
11111111
( 0dB )
ADC Digital Volume Control
0000 0000 = Digital Mute
0000 0001 = -127dB
0000 0010 = -126.5dB
... 0.5dB steps up to
1111 1111 = 0dB
Table 13 ADC Volume
INPUT LIMITER / AUTOMATIC LEVEL CONTROL (ALC)
The WM8510 has an automatic PGA gain control circuit, which can function as an input peak limiter
or as an automatic level control (ALC).
The Automatic Level Control (ALC) provides continuous adjustment of the input PGA in response to
the amplitude of the input signal. A digital peak detector monitors the input signal amplitude and
compares it to a register defined threshold level (ALCLVL).
If the signal is below the threshold, the ALC will increase the gain of the PGA at a rate set by
ALCDCY. If the signal is above the threshold, the ALC will reduce the gain of the PGA at a rate set
by ALCATK.
The ALC has two modes selected by the ALCMODE register: normal mode and peak limiter mode.
The ALC/limiter function is enabled by setting the register bit R32[8] ALCSEL.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
2:0 ALCMIN
[2:0]
000 (-12dB) Set minimum gain of PGA
000 = -12dB
001 = -6dB
010 = 0dB
011 = +6dB
100 = +12dB
101 = +18dB
110 = +24dB
111 = +30dB
5:3 ALCMAX
[2:0]
111
(+35.25dB)
Set Maximum Gain of PGA
111 = +35.25dB
110 = +29.25dB
101 = +23.25dB
100 = +17.25dB
011 = +11.25dB
010 = +5.25dB
001 = -0.75dB
000 = -6.75dB
R32 (20h)
ALC Control
1
8 ALCSEL 0 ALC function select
0 = ALC disabled
1 = ALC enabled
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
3:0 ALCLVL
[3:0]
1011
(-12dB)
ALC target – sets signal level at ADC
input
1111 = -6dBFS
1110 = -7.5dBFS
1101 = -9dBFS
1100 = -10.5dBFS
1011 = -12dBFS
1010 = -13.5dBFS
1001 = -15dBFS
1000 = -16.5dBFS
0111 = -18dBFS
0110 = -19.5dBFS
0101 = -21dBFS
0100 = -22.5dBFS
0011 = -24dBFS
0010 = -25.5dBFS
0001 = -27dBFS
0000 = -28.5dBFS
8 ALCZC 0 (zero
cross off)
ALC uses zero cross detection circuit.
0 = Disabled (recommended)
1 = Enabled
It is recommended that zero cross is not
used in conjunction with the ALC or
Limiter functions
R33 (21h)
ALC Control
2
7:4 ALCHLD
[3:0]
0000
(0ms)
ALC hold time before gain is increased.
0000 = 0ms
0001 = 2.67ms
0010 = 5.33ms
0011 = 10.66ms
0100 = 21.32ms
0101 = 42.64ms
0110 = 85.28ms
0111 = 0.17s
1000 = 0.34s
1001 = 0.68s
1010 or higher = 1.36s
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
8 ALCMODE 0 Determines the ALC mode of operation:
0 = ALC mode (Normal Operation)
1 = Limiter mode.
Decay (gain ramp-up) time
(ALCMODE ==0)
Per
step
Per
6dB
90% of
range
0000 410us 3.38ms 23.6ms
0001 820us 6.56ms 47.2ms
0010 1.64ms 13.1ms 94.5ms
… (time doubles with every step)
0011
(26ms/6dB)
1010
or
higher
420ms 3.36s 24.2s
Decay (gain ramp-up) time
(ALCMODE ==1)
Per
step
Per
6dB
90% of
range
0000 90.8us 726us 5.23ms
0001 182us 1.45ms 10.5ms
0010 363us 2.91ms 20.9ms
… (time doubles with every step)
7:4 ALCDCY
[3:0]
0011
(5.8ms/6dB)
1010 93ms 744ms 5.36s
ALC attack (gain ramp-down) time
(ALCMODE == 0)
Per
step
Per
6dB
90% of
range
0000 104us 832us 6ms
0001 208us 1.66ms 12ms
0010 416us 3.33ms 24ms
… (time doubles with every step)
0010
(3.3ms/6dB)
1010
or
higher
106ms 852ms 6.13s
ALC attack (gain ramp-down) time
(ALCMODE == 1)
Per
step
Per
6dB
90% of
range
0000 22.7us 182.4us 1.31ms
0001 45.4us 363us 2.62ms
0010 90.8us 726us 5.23ms
… (time doubles with every step)
R34 (22h)
ALC Control
3
3:0 ALCATK
[3:0]
0010
(726us/6dB)
1010
or
higher
23.2ms 186ms 1.34s
Table 14 ALC Control Registers
When the ALC is disabled, the input PGA remains at the last controlled value of the ALC. An input
gain update must be made by writing to the INPPGAVOLL/R register bits.
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NORMAL MODE
In normal mode, the ALC will attempt to maintain a constant signal level by increasing or decreasing
the gain of the PGA. The following diagram shows an example of this.
Figure 11 ALC Normal Mode Operation
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LIMITER MODE
In limiter mode, the ALC will reduce peaks that go above the threshold level, but will not increase the
PGA gain beyond the starting level. The starting level is the PGA gain setting when the ALC is
enabled in limiter mode. If the ALC is started in limiter mode, this is the gain setting of the PGA at
start-up. If the ALC is switched into limiter mode after running in ALC mode, the starting gain will be
the gain at switchover. The diagram below shows an example of limiter mode.
Figure 12 ALC Limiter Mode Operation
ATTACK AND DECAY TIMES
The attack and decay times set the update times for the PGA gain. The attack time is the time
constant used when the gain is reducing. The decay time is the time constant used when the gain is
increasing. In limiter mode, the time constants are faster than in ALC mode. The time constants are
shown below in terms of a single gain step, a change of 6dB and a change of 90% of the PGAs gain
range.
Note that, these times will vary slightly depending on the sample rate used (specified by the SR
register).
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NORMAL MODE
Table 15 ALC Normal Mode (Attack and Decay times)
ALCMODE = 0 (Normal Mode)
ALCATK tATK tATK6dB tATK90%
0000 104µs 832µs 6ms
0001 208µs 1.66ms 12ms
0010 416µs 3.33ms 24ms
0011 832µs 6.66ms 48ms
0100 1.66ms 13.3ms 96ms
0101 3.33ms 26.6ms 192ms
0110 6.66ms 53.2ms 384ms
0111 13.3ms 106ms 767ms
1000 26.6ms 213.2ms 1.53s
1001 53.2ms 426ms 3.07s
1010 106ms 852ms 6.13s
Attack Time (s)
ALCMODE = 0 (Normal Mode)
ALCDCY t
DCY
t
DCY6dB
t
DCY 90%
0000 410µs 3.28ms 23.6ms
0001 820µs 6.56ms 47.2ms
0010 1.64ms 13.1ms 94.5ms
0011 3.28ms 26.2ms 189ms
0100 6.56ms 52.5ms 378ms
0101 13.1ms 105ms 756ms
0110 26.2ms 210ms 1.51s
0111 52.5ms 420ms 3.02s
1000 105ms 840ms 6.05s
1001 210ms 1.68s 12.1s
1010 420ms 3.36s 24.2s
Decay Time (s)
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LIMITER MODE
Table 16 ALC Limiter Mode (Attack and Decay times)
ALCMODE = 1 (Limiter Mode)
ALCATK t
ATKLIM
t
ATKLIM6dB
t
ATKLIM90%
0000 22.7µs 182µs 1.31ms
0001 45.4µS 363µs 2.62ms
0010 90.8µS 726µs 5.23ms
0011 182µS 1.45ms 10.5ms
0100 363µS 2.91ms 20.9ms
0101 726µS 5.81ms 41.8ms
0110 1.45ms 11.6ms 83.7ms
0111 2.9ms 23.2ms 167ms
1000 5.81ms 46.5ms 335ms
1001 11.6ms 93ms 669ms
1010 23.2ms 186ms 1.34s
Attack Time (s)
ALCMODE = 1 (Limiter Mode)
ALCDCY t
DCYLIM
t
DC Y LIM6dB
t
DCY LIM90%
0000 90.8µs 726µs 5.23ms
0001 182µS 1.45ms 10.5ms
0010 363µS 2.91ms 20.9ms
0011 726µS 5.81ms 41.8ms
0100 1.45ms 11.6ms 83.7ms
0101 2.91ms 23.2ms 167ms
0110 5.81ms 46.5ms 335ms
0111 11.6ms 93ms 669ms
1000 23.2ms 186ms 1.34s
1001 46.5ms 372ms 2.68s
1010 93ms 744ms 5.36s
Attack Time (s)
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MINIMUM AND MAXIMUM GAIN
The ALCMIN and ALCMAX register bits set the minimum/maximum gain value that the PGA can be
set to whilst under the control of the ALC. This has no effect on the PGA when ALC is not enabled.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
5:3 ALCMAX 111 Set Maximum Gain of PGA R32
ALC Control
1
2:0 ALCMIN 000 Set minimum gain of PGA
Table 17 ALC Max/Min Gain
In normal mode, ALCMAX sets the maximum boost which can be applied to the signal. In limiter
mode, ALCMAX will normally have no effect (assuming the starting gain value is less than the
maximum gain specified by ALCMAX) because the maximum gain is set at the starting gain level.
ALCMIN sets the minimum gain value which can be applied to the signal.
Table 18 ALC Max Gain Values
ALCMAX Maximum Gain (dB)
111 35.25
110 29.25
101 23.25
100 17.25
011 11.25
010 5.25
001 -0.75
000 -6.75
Figure 13 ALC Min/Max Gain
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Table 19 ALC Min Gain Values
Note that if the ALC gain setting strays outside the ALC operating range, either by starting the ALC
outside of the range or changing the ALCMAX or ALCMIN settings during operation, the ALC will
immediately adjust the gain to return to the ALC operating range. It is recommended that the ALC
starting gain is set between the ALCMAX and ALCMIN limits.
ALC HOLD TIME (NORMAL MODE ONLY)
In Normal mode, the ALC has an adjustable hold time which sets a time delay before the ALC begins
its decay phase (gain increasing). The hold time is set by the ALCHLD register.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R33
ALC Control
2
7:4 ALCHLD 0000 ALC hold time before gain is increased.
Table 20 ALC Hold Time
If the hold time is exceeded this indicates that the signal has reached a new average level and the
ALC will increase the gain to adjust for that new average level. If the signal goes above the threshold
during the hold period, the hold phase is abandoned and the ALC returns to normal operation.
ALCMIN Minimum Gain (dB)
000 -12
001 -6
010 0
011 6
100 12
101 18
110 24
111 30
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Figure 14 ALCLVL
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Input
Signal
Output
of PGA ALCLVL
PGA
Gain
tHOLD
Figure 15 ALC Hold Time
Table 21 ALC Hold Time Values
ALCHLD t
HOLD
(s)
0000 0
0001 2.67ms
0010 5.34ms
0011 10.7ms
0100 21.4ms
0101 42.7ms
0110 85.4ms
0111 171ms
1000 342ms
1001 684ms
1010 1.37s
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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 ALCATK = 0000), until the signal level falls
below 87.5% of full scale. This function is automatically enabled whenever the ALC is enabled.
Note: If ALCATK = 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.
NOISE GATE (NORMAL MODE ONLY)
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 WM8510 has a noise gate function that
prevents noise pumping by comparing the signal level at the input pins against a noise gate
threshold, NGTH. The noise gate cuts in when:
Signal level at ADC [dBFS] < NGTH [dBFS] + PGA gain [dB] + Mic Boost gain [dB]
This is equivalent to:
Signal level at input pin [dBFS] < NGTH [dBFS]
The PGA gain is then 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 6dB steps.
Levels at the extremes of the range may cause inappropriate operation, so care should be taken with
set–up of the function. The noise gate only operates in conjunction with the ALC and cannot be used
in limiter mode.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
2:0 NGTH
000 Noise gate threshold:
000 = -39dB
001 = -45dB
010 = -51db
011 = -57dB
100 = -63dB
101 = -69dB
110 = -75dB
111 = -81dB
R35 (23h)
ALC Noise Gate
Control
3 NGATEN 0 Noise gate function enable
1 = enable
0 = disable
Table 22 ALC Noise Gate Control
The diagrams below show the response of the system to the same signal with and without noise
gate.
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Figure 16 ALC Operation Above Noise Gate Threshold
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Figure 17 Noise Gate Operation
OUTPUT SIGNAL PATH
The WM8510 output signal paths consist of digital application filters, up-sampling filters, a Hi-Fi DAC,
analogue mixers, speaker and mono output drivers. The digital filters and DAC are enabled by bit
DACEN. 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 WM8510, irrespective of whether the DACs are running or not.
The WM8510 DAC receives digital input data on the DACDAT pin. The digital filter block processes
the data to provide the following functions:
• Digital volume control
• A digital peak limiter
• Sigma-Delta Modulation
The high performance sigma-delta audio DAC converts the digital data into an analogue signal.
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Figure 18 DAC Digital Filter Path
The analogue output from the DAC can then be mixed with the MIC2 analogue input and the ADC
analogue input. The mix is fed to the output drivers, SPKOUTP/N, and MONOOUT.
MONOOUT: can drive a 16Ω or 32Ω headphone or line output or can be a buffered version of VMID
(When MONOMUTE=1).
SPKOUTP/N: can drive a 16Ω or 32Ω stereo headphone or stereo line output, or an 8Ω BTL mono
speaker.
DIGITAL HI-FI DAC VOLUME CONTROL
The signal volume from each Hi-Fi DAC can be controlled digitally. 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
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R11
DAC Digital
Volume
7:0 DACVOL
[7:0]
11111111
( 0dB )
DAC Digital Volume Control
0000 0000 = Unused
0000 0001 = -127dB = mute
0000 0010 = -126.5dB
... 0.5dB steps up to
1111 1111 = 0dB
Table 23 DAC Volume
HI-FI DIGITAL TO ANALOGUE CONVERTER (DAC)
Digital ‘de-emphasis’ can be applied to the audio data if necessary. De-emphasis filtering is available
for sample rates of 48kHz, 44.1kHz and 32kHz.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R10
DAC Control
5:4 DEEMPH 00 De-Emphasis Control
00 = No de-emphasis
01 = 32kHz sample rate
10 = 44.1kHz sample rate
11 = 48kHz sample rate
Table 24 De-Emphasis
The DAC is enabled by the DACEN register bit.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R3
Power
Management 3
0 DACEN 0 DAC enable
0 = DAC disabled
1 = DAC enabled
Table 25 DAC Enable
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The WM8510 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
R10
DAC Control
6 DACMU 0 DAC soft mute enable
0 = DACMU disabled
1 = DACMU enabled
Table 26 DAC Control Register
The digital audio data is converted to oversampled bit streams in the on-chip, true 24-bit digital
interpolation filters. The bitstream data enters a multi-bit, sigma-delta DAC, which converts it to a
high quality analogue audio signal. The multi-bit DAC architecture reduces high frequency noise and
sensitivity to clock jitter.
The DAC output defaults to non-inverted. Setting DACPOL will invert the DAC output phase.
AUTOMUTE
The DAC has an automute function which applies an analogue mute when 1024 consecutive zeros
are detected. The mute is release as soon as a non-zero sample is detected. Automute can be
disabled using the AMUTE control bit.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R10
DAC Control
2 AMUTE 0 DAC auto mute enable
0 = auto mute disabled
1 = auto mute enabled
Table 27 DAC Auto Mute Control Register
DAC OUTPUT LIMITER
The WM8510 has a digital output limiter function. The operation of this is shown in Figure 19. In this
diagram the upper graph shows the envelope of the input/output signals and the lower graph shows
the gain characteristic.
Figure 19 DAC Digital Limiter Operation
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The limiter has a programmable upper threshold which is close to 0dB. Referring to Table 30, in
normal operation (LIMBOOST=000 => limit only) signals below this threshold are unaffected by the
limiter. Signals above the upper threshold are attenuated at a specific attack rate (set by the
LIMATK register bits) until the signal falls below the threshold. The limiter also has a lower threshold
1dB below the upper threshold. When the signal falls below the lower threshold the signal is
amplified at a specific decay rate (controlled by LIMDCY register bits) until a gain of 0dB is reached.
Both threshold levels are controlled by the LIMLVL register bits. The upper threshold is 0.5dB above
the value programmed by LIMLVL and the lower threshold is 0.5dB below the LIMLVL value.
VOLUME BOOST
The limiter has programmable upper gain which boosts signals below the threshold to compress the
dynamic range of the signal and increase its perceived loudness. This operates as an ALC function
with limited boost capability. The volume boost is from 0dB to +12dB in 1dB steps, controlled by the
LIMBOOST register bits.
The output limiter volume boost can also be used as a stand alone digital gain boost when the limiter
is disabled.
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
3:0 LIMATK
0010 Limiter Attack time (per 6dB gain
change) for 44.1kHz sampling. Note
that these will scale with sample rate.
0000=94us
0001=188s
0010=375us
0011=750us
0100=1.5ms
0101=3ms
0110=6ms
0111=12ms
1000=24ms
1001=48ms
1010=96ms
1011 to 1111=192ms
7:4 LIMDCY 0011 Limiter Decay time (per 6dB gain
change) for 44.1kHz sampling. Note
that these will scale with sample rate:
0000=750us
0001=1.5ms
0010=3ms
0011=6ms
0100=12ms
0101=24ms
0110=48ms
0111=96ms
1000=192ms
1001=384ms
1010=768ms
1011 to 1111=1.536s
R24
DAC digital
limiter control
1
8 LIMEN 0 Enable the DAC digital limiter:
0=disabled
1=enabled
3:0 LIMBOOST 0000 Limiter volume boost (can be used as a
stand alone volume boost when
LIMEN=0):
0000=0dB
0001=+1dB
0010=+2dB
… (1dB steps)
1011=+11dB
1100=+12dB
1101 to 1111=reserved
R25
DAC digital
limiter control
2
6:4 LIMLVL 000 Programmable signal threshold level
(determines level at which the limiter
starts to operate)
000=-1dB
001=-2dB
010=-3dB
011=-4dB
100=-5dB
101 to 111=-6dB
Table 28 DAC Digital Limiter Control
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ANALOGUE OUTPUTS
The WM8510 has a single MONO output and two outputs SPKOUTP and SPOUTN for driving a
mono BTL speaker. These analogue output stages are supplied from SPKVDD and are capable of
driving up to 1.5V rms signals (equivalent to 3V rms into a bridge tied speaker) as shown in Figure
20.
Figure 20 Speaker and Mono Analogue Outputs
The Mono and speaker outputs have output driving stages which can be controlled by the register
bits MONOBOOST and SPKBOOST respectively. Each output stage has a selectable gain boost of
1.5x. When this boost is enabled the output DC level is also level shifted (from AVDD/2 to
1.5xAVDD/2) to prevent the signal from clipping. A dedicated amplifier, as shown in Figure 20, is
used to perform the DC level shift operation. This buffer must be enabled using the BUFDCOPEN
register bit for this operating mode. It should also be noted that if SPKVDD is not equal to or greater
than 1.5xAVDD this boost mode may result in signals clipping. Table 30 summarises the effect of
the SPKBOOST/MONOBOOST control bits.
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
2 SPKBOOST 0 Speaker output boost stage control
(see Table 30 for details)
0=No boost (outputs are inverting
buffers)
1 = 1.5x gain boost
R49
Output control
3 MONOBOOST 0 Mono output boost stage control (see
Table 30 for details)
0=No boost (output is inverting buffer)
1=1.5x gain boost
R1
Power
management 1
8 BUFDCOPEN 0 Dedicated buffer for DC level shifting
output stages when in 1.5x gain boost
configuration.
0=Buffer disabled
1=Buffer enabled (required for 1.5x
gain boost)
Table 29 Output Boost Control
SPKBOOST/
MONOBOOST
OUTPUT
STAGE GAIN
OUTPUT DC
LEVEL
OUTPUT STAGE
CONFIGURATION
0 1x AVDD/2 Inverting
1 1.5x 1.5xAVDD/2 Non-inverting
Table 30 Output Boost Stage Details
SPKOUTP/SPKOUTN OUTPUTS
The SPKOUT pins can drive a single bridge tied 8Ω speaker or two headphone loads of 16Ω or 32Ω
or a line output (see Headphone Output and Line Output sections, respectively). The signal to be
output on SKPKOUT comes from the Speaker Mixer circuit and can be any combination of the DAC
output, the Bypass path (output of the boost stage) and the MIC2 input. The Bypass path has the
option of 0dB or -10dB attenuation, selected by the SPKATTN register bit. The SPKOUTP/N volume
is controlled by the SPKVOL register bits. Note that gains over 0dB may cause clipping if the signal
is large. The SPKMUTE register bit causes the speaker outputs to be muted (the output DC level is
driven out). The output pins remains at the same DC level (VMIDOP), so that no click noise is
produced when muting or un-muting.
The SPKOUTN pin always drives out an inverted version of the SPKOUTP signal.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
0 DAC2SPK 1 Output of DAC to speaker mixer input
0 = not selected
1 = selected
1 BYP2SPK 0 Bypass path (output of input boost
stage) to speaker mixer input
0 = not selected
1 = selected
R50
Speaker mixer
control
5 MIC2_2SPK 0 Output of MIC2 amplifier to speaker
mixer input
0 = not selected
1 = selected
R40
Bypass path
attenuation
control
1 SPKATTN 0 Attenuation control for bypass path
(output of input boost stage) to
speaker mixer input
0 = 0dB
1 = -10dB
Table 31 Speaker Mixer Control
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
7 SPKZC 0 Speaker Volume control enable:
1 = Change gain on zero cross only
0 = Change gain immediately
6 SPKMUTE 0 Speaker output mute enable
0=Speaker output enabled
1=Speaker output muted (VMIDOP)
R54
Speaker
volume
control
5:0 SPKVOL
[5:0]
111001
(0dB)
Speaker Volume Adjust
111111 = +6dB
111110 = +5dB
… (1.0 dB steps)
111001=0dB
…
000000=-57dB
Table 32 SPKOUT Volume Control
ZERO CROSS TIMEOUT
A zero-cross timeout function is also provided so that if zero cross is enabled on the input or output
PGAs the gain will automatically update after a timeout period if a zero cross has not occurred. This
is enabled by setting SLOWCLKEN. The timeout period is dependent on the clock input to the digital
and is equal to 221 * input clock period.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R7
Additional
control
0 SLOWCLKEN 0 Slow clock enable. Used for both the
jack insert detect debounce circuit and
the zero cross timeout.
0 = slow clock disabled
1 = slow clock enabled
Table 33 Timeout Clock Enable Control
MONO MIXER AND OUTPUT
The MONOOUT pin can drive a 16Ω or 32Ω headphone or a line output or be used as a DC
reference for a headphone output (see Headphone Output section). It can be selected to drive out
any combination of DAC, Bypass (output of input BOOST stage) and MIC2. The Bypass path has
the option of 0dB or -10dB attenuation, selected by the MONOATTN register bit. This output is
enabled by setting bit MONOEN.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
1 SPKATTN 0 0=off
1=-10dB
R40 Attenuation
Control
2 MONOATTN 0 0=off
1=-10dB
Table 34 Sidetone Attenuation Control
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
0 DAC2MONO 1 Output of DAC to mono mixer input
0 = not selected
1 = selected
1 BYP2MONO 0 Bypass path (output of input boost
stage) to mono mixer input
0 = non selected
1 = selected
2 MIC2_2MONO 0 Output of MIC2 amplifier to mono
mixer input:
0 = not selected
1 = selected
R56
Mono mixer
control
6 MONOMUTE 0 0=No mute
1=Output muted. During mute the
mono output will output VMID which
can be used as a DC reference for a
headphone out.
R40
Bypass path
attenuation
control
2 MONOATTN 0 Attenuation control for bypass path
(output of input boost stage) to mono
mixer input
0 = 0dB
1 = -10dB
Table 35 Mono Mixer Control
ENABLING THE OUTPUTS
Each analogue output of the WM8510 can be separately enabled or disabled. The analogue mixer
associated with each output has a separate enable. All outputs are disabled by default. To save
power, unused parts of the WM8510 should remain disabled.
Outputs can be enabled at any time, but it is not recommended to do so when BUFIO is disabled
(BUFIOEN=0), as this may cause pop noise (see “Power Management” and “Applications
Information” sections).
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
2 BUFIOEN 0 Unused input/output tie off buffer enable
8 BUFDCOPEN 0 Output stage 1.5xAVDD/2 driver enable
R1
Power
management
1 3 BIASEN 0 Analogue amplifiers bias enable
2 SPKMIXEN 0 Speaker Mixer enable
3 MONOMIXEN 0 Mono mixer enable
5 SPKPEN 0 SPKOUTP enable
6 SPKNEN 0 SPKOUTN enable
R3
Power
management
3
7 MONOEN 0 MONOOUT enable
Note: All “Enable” bits are 1 = ON, 0 = OFF
Table 36 Output Stages Power Management Control
UNUSED ANALOGUE INPUTS/OUTPUTS
Whenever an analogue input/output is disabled, it remains connected to a voltage source (either
AVDD/2 or 1.5xAVDD/2 as appropriate) through a resistor. This helps to prevent pop noise when the
output is re-enabled. The resistance between the voltage buffer and the output pins can be controlled
using the VROI control bit. The default impedance is low, 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 30kΩ.
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R49
0 VROI 0 VREF (AVDD/2 or 1.5xAVDD/2) to
analogue output resistance
0: approx 1kΩ
1: approx 30 kΩ
Table 37 Disabled Outputs to VREF Resistance
A dedicated buffer is available for tying off unused analogue I/O pins as shown in Figure 21. This
buffer can be enabled using the BUFIOEN register bit.
If the SPKBOOST or MONOBOOST bits are set then the relevant outputs will be tied to the output of
the DC level shift buffer at 1.5xAVDD/2 when disabled.
Table 38 summarises the tie-off options for the speaker and mono output pins.
Figure 21 Unused Input/Output Pin Tie-off Buffers
MONOEN/
SPKN/PEN
MONOBOOST/
SPKBOOST
VROI OUTPUT CONFIGURATION
0 0 0 1kΩ tieoff to AVDD/2
0 0 1 30kΩ tieoff to AVDD/2
0 1 0
1kΩ tieoff to 1.5xAVDD/2
0 1 1
30kΩ tieoff to 1.5xAVDD/2
1 0 X
Output enabled (DC level=AVDD/2)
1 1 X
Output enabled (DC level=1.5xAVDD/2)
Table 38 Unused Output Pin Tie-off Options
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OUTPUT SWITCH
When the device is configured with a 2-wire interface the CSB/GPIO pin can be used as a switch
control input to automatically disable the speaker outputs and enable the mono output. For example
when a line is plugged into a jack socket. In this mode, enabled by setting GPIOSEL=001, pin
CSB/GPIO switches between mono and speaker outputs (e.g. when pin 12 is connected to a
mechanical switch in the headphone socket to detect plug-in). The GPIOPOL bit reverses the polarity
of the CSB/GPIO input pin.
Note that the speaker outputs and the mono output must be enabled for this function to work (see
Table 39). The CSB/GPIO pin has an internal de-bounce circuit when in this mode in order to prevent
the output enables from toggling multiple times due to input glitches. This debounce circuit is
clocked from a slow clock with period 221 x MCLK, enabled using the SLOWCLKEN register bit.
GPIOPOL CSB/GPIO SPKNEN/
SPKPEN
MONOEN SPEAKER
ENABLED
MONO
OUTPUT
ENABLED
0 0 X 0 No No
0 0 X 1 No Yes
0 1 0 X No No
0 1 1 X Yes No
1 0 X 0 No No
1 0 X 1 No Yes
1 1 0 X No No
1 1 1 X Yes No
Table 39 Output Switch Operation (GPIOSEL=001)
THERMAL SHUTDOWN
The speaker outputs can drive very large currents. To protect the WM8510 from overheating a
thermal shutdown circuit is included. The thermal shutdown can be configured to produce an interrupt
when the device temperature reaches approximately 125oC. See the General Purpose Input/Output
section for details.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R49
Output control
1 TSDEN 1
Thermal Shutdown Enable
0 : thermal shutdown disabled
1 : thermal shutdown enabled
Table 40 Thermal Shutdown
SPEAKER OUTPUT
SPKOUTP/N can differentially drive a mono 8Ω Bridge Tied Load (BTL) speaker as shown below.
Figure 22 Speaker Output Connection
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HEADPHONE OUTPUT
The speaker outputs can drive a 16Ω or 32Ω headphone load, either through DC blocking capacitors,
or DC coupled without any capacitor.
Headphone Output using DC Blocking Capacitors: DC Coupled Headphone Output:
Figure 23 Recommended Headphone Output Configurations
When DC blocking capacitors are used, then their capacitance and the load resistance together
determine the lower cut-off frequency, fc. Increasing the capacitance lowers fc, improving the bass
response. Smaller capacitance values will diminish the bass response. Assuming a 16Ω load and
C1, C2 = 220µF:
fc = 1 / 2π RLC1 = 1 / (2π x 16Ω x 220µF) = 45 Hz
In the DC coupled configuration, the headphone “ground” is connected to the MONOOUT pin. The
MONOOUT pin can be configured as a DC output driver by setting the MONOMUTE register bit. The
DC voltage on MONOOUT in this configuration is equal to the DC offset on the SPKOUTP and
SPKOUTN pins therefore no DC blocking capacitors are required. This saves space and material
cost in portable applications.
It is recommended to connect the DC coupled outputs only to headphones, and not to the line input
of another device. Although the built-in short circuit protection will prevent any damage to the
headphone outputs, such a connection may be noisy, and may not function properly if the other
device is grounded.
MONO OUTPUT
The mono output, can be used as a line output, a headphone output or as a pseudo ground for cap-
less driving of loads by SPKOUT. Recommended external components are shown below.
Figure 24 Recommended Circuit for Line Output
The DC blocking capacitors and the load resistance together determine the lower cut-off frequency,
fc. Assuming a 10 kΩ load and C1 = 1µF:
fc = 1 / 2π (RL+R1) C1 = 1 / (2π x 10.1kΩ x 1µF) = 16 Hz
Increasing the capacitance lowers fc, improving the bass response. Smaller values of C1 will diminish
the bass response. The function of R1 is to protect the line outputs from damage when used
improperly.
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DIGITAL AUDIO INTERFACES
The audio interface has four pins:
• ADCDAT: ADC data output
• DACDAT: DAC data input
• FRAME: Data alignment clock
• BCLK: Bit clock, for synchronisation
The clock signals BCLK, and FRAME can be outputs when the WM8510 operates as a master, or
inputs when it is a slave (see Master and Slave Mode Operation, below).
Five different audio data formats are supported:
• Left justified
• Right justified
• I
2S
• DSP mode A
All 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 WM8510 audio interface may be configured as either master or slave. As a master interface
device the WM8510 generates BCLK and FRAME and thus controls sequencing of the data transfer
on ADCDAT and DACDAT. To set the device to master mode register bit MS should be set high. In
slave mode (MS=0), the WM8510 responds with data to clocks it receives over the digital audio
interfaces.
AUDIO DATA FORMATS
In Left Justified mode, the MSB is available on the first rising edge of BCLK following an FRAME
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 FRAME transition.
Figure 25 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 FRAME
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 FRAME transition.
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Figure 26 Right Justified Audio Interface (assuming n-bit word length)
In I2S mode, the MSB is available on the second rising edge of BCLK following a FRAME 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 27 I2S Audio Interface (assuming n-bit word length)
In DSP/PCM mode, the left channel MSB is available on the 2nd (mode A) rising edge of BCLK
(selectable by FRAMEP) following a rising edge of FRAME. 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.
Figure 28 DSP/PCM Mode Audio Interface (mode A)
When using ADCLRSWAP = 1 or DACLRSWAP = 1 in DSP/PCM mode, the data will appear in the
Right Phase of the FRAME, which will be 16/20/24/32 bits after the FRAME pulse.
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
1 ADCLRSWAP 0 Controls whether ADC data appears in
‘right’ or ‘left’ phases of FRAME clock:
0=ADC data appear in ‘left’ phase of
FRAME
1=ADC data appears in ‘right’ phase of
FRAME
2 DACLRSWAP 0 Controls whether DAC data appears in
‘right’ or ‘left’ phases of FRAME clock:
0=DAC data appear in ‘left’ phase of
FRAME
1=DAC data appears in ‘right’ phase of
FRAME
4:3 FMT 10 Audio interface Data Format Select:
00=Right Justified
01=Left Justified
10=I2S format
11= DSP/PCM mode
6:5 WL 10 Word length
00=16 bits
01=20 bits
10=24 bits
11=32 bits (see note)
7 FRAMEP 0 Frame clock polarity
0=normal
1=inverted
R4
Audio
interface
control
8 BCP 0 BCLK polarity
0=normal
1=inverted
Table 41 Audio Interface Control
AUDIO INTERFACE CONTROL
The register bits controlling audio format, word length and master / slave mode are summarised
below. Each audio interface can be controlled individually.
Register bit MS selects audio interface operation in master or slave mode. In Master mode BCLK,
and FRAME are outputs. The frequency of BCLK and FRAME in master mode are controlled with
BCLKDIV. These are divided down versions of master clock. This may result in short BCLK pulses
at the end of a frame if there is a non-integer ratio of BCLKs to FRAME clocks.
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
0 MS 0 Sets the chip to be master over FRAME
and BCLK
0=BCLK and FRAME clock are inputs
1=BCLK and FRAME clock are outputs
generated by the WM8510 (MASTER)
4:2 BCLKDIV 000 Configures the BCLK and FRAME
output frequency, for use when the chip
is master over BCLK.
000=divide by 1 (BCLK=MCLK)
001=divide by 2 (BCLK=MCLK/2)
010=divide by 4
011=divide by 8
100=divide by 16
101=divide by 32
110=reserved
111=reserved
7:5 MCLKDIV 010 Sets the scaling for either the MCLK or
PLL clock output (under control of
CLKSEL)
000=divide by 1
001=divide by 1.5
010=divide by 2
011=divide by 3
100=divide by 4
101=divide by 6
110=divide by 8
111=divide by 12
R6
Clock
generation
control
8 CLKSEL 1 Controls the source of the clock for all
internal operation:
0=MCLK
1=PLL output
Table 42 Clock Control
LOOPBACK
Setting the LOOPBACK register bit enables digital loopback. When this bit is set the output data
from the ADC audio interface is fed directly into the DAC data input.
COMPANDING
The WM8510 supports A-law and µ-law companding on both transmit (ADC) and receive (DAC)
sides. Companding can be enabled on the DAC or ADC audio interfaces by writing the appropriate
value to the DAC_COMP or ADC_COMP register bits respectively.
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
0 LOOPBACK 0 Digital loopback function
0=No loopback
1=Loopback enabled, ADC data output is
fed directly into DAC data input.
2:1 ADC_COMP 0 ADC companding
00=off
01=reserved
10=µ-law
11=A-law
R5
Companding
control
4:3 DAC_COMP 0 DAC companding
00=off
01=reserved
10=µ-law
11=A-law
Table 43 Companding Control
Companding involves using a piecewise linear approximation of the following equations (as set out by
ITU-T G.711 standard) for data compression:
µ-law (where µ=255 for the U.S. and Japan):
F(x) = ln( 1 + µ|x|) / ln( 1 + µ) -1 ≤ x ≤ 1
A-law (where A=87.6 for Europe):
F(x) = A|x| / ( 1 + lnA) } for x ≤ 1/A
F(x) = ( 1 + lnA|x|) / (1 + lnA) } for 1/A ≤ x ≤ 1
The companded data is also inverted as recommended by the G.711 standard (all 8 bits are inverted
for µ-law, all even data bits are inverted for A-law). The data will be transmitted as the first 8 MSB’s
of data.
Companding converts 13 bits (µ-law) or 12 bits (A-law) to 8 bits using non-linear quantization. The
input data range is separated into 8 levels, allowing low amplitude signals better precision than that
of high amplitude signals. This is to exploit the operation of the human auditory system, where
louder sounds do not require as much resolution as quieter sounds. The companded signal is an 8-
bit word containing sign (1-bit), exponent (3-bits) and mantissa (4-bits).
BIT7 BIT[6:4] BIT[3:0]
SIGN EXPONENT MANTISSA
Table 44 8-bit Companded Word Composition
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u-law Companding
0
20
40
60
80
100
120
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalised Input
Companded Output
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Normalised Output
Figure 29 u-Law Companding
A-law Companding
0
20
40
60
80
100
120
0 0.2 0.4 0.6 0.8 1
Normalised Input
Companded Output
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Normalised Output
Figure 30 A-Law Companding
AUDIO SAMPLE RATES
The WM8510 sample rates for the ADC and the DAC are set using the SR register bits. The cutoffs
for the digital filters and the ALC attack/decay times stated are determined using these values and
assume a 256fs master clock rate.
If a sample rate that is not explicitly supported by the SR register settings is required then the closest
SR value to that sample rate should be chosen, the filter characteristics and the ALC attack, decay
and hold times will scale appropriately.
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R7
Additional
control
3:1 SR
000 Approximate sample rate (configures the
coefficients for the internal digital filters):
000=48kHz
001=32kHz
010=24kHz
011=16kHz
100=12kHz
101=8kHz
110-111=reserved
Table 45 Sample Rate Control
MASTER CLOCK AND PHASE LOCKED LOOP (PLL)
The WM8510 has an on-chip phase-locked loop (PLL) circuit that can be used to:
Generate master clocks for the WM8510 audio functions from another external clock, e.g. in
telecoms applications.
Generate and output (on pin CSB/GPIO) a clock for another part of the system that is derived from
an existing audio master clock.
Figure 31 shows the PLL and internal clocking arrangement on the WM8510.
The PLL can be enabled or disabled by the PLLEN register bit.
Note: In order to minimise current consumption, the PLL is disabled when the VMIDSEL[1:0] bits are
set to 00b. VMIDSEL[1:0] must be set to a value other than 00b to enable the PLL.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R1
Power
management 1
5 PLLEN 0 PLL enable
0=PLL off
1=PLL on
Table 46 PLLEN Control Bit
Figure 31 PLL and Clock Select Circuit
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The PLL frequency ratio R = f2/f1 (see Figure 31) can be set using the register bits PLLK and PLLN:
PLLN = int R
PLLK = int (224 (R-PLLN))
EXAMPLE:
MCLK=12MHz, required clock = 12.288MHz.
R should be chosen to ensure 5 < PLLN < 13. There is a fixed divide by 4 in the PLL and a
selectable divide by N after the PLL which should be set to divide by 2 to meet this requirement.
Enabling the divide by 2 sets the required f2 = 4 x 2 x 12.288MHz = 98.304MHz.
R = 98.304 / 12 = 8.192
PLLN = int R = 8
k = int ( 224 x (8.192 – 8)) = 3221225 = 3126E9h
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
4 PLLPRESCALE 0 0 = MCLK input not divided (default)
1= Divide MCLK by 2 before input to
PLL
R36
PLL N value
3:0 PLLN 1000 Integer (N) part of PLL input/output
frequency ratio. Use values greater
than 5 and less than 13.
R37
PLL K value 1
5:0 PLLK [23:18] 0Ch
R38
PLL K Value 2
8:0 PLLK [17:9] 093h
R39
PLL K Value 3
8:0 PLLK [8:0] 0E9h
Fractional (K) part of PLL1
input/output frequency ratio (treat as
one 24-digit binary number).
Table 47 PLL Frequency Ratio Control
The PLL performs best when f2 is around 90MHz. Its stability peaks at N=8. Some example settings
are shown in Table 48.
MCLK
(MHz)
(F1)
DESIRED
OUTPUT
(MHz)
F2
(MHz)
PRESCALE
DIVIDE
POSTSCALE
DIVIDE
R N
(Hex)
K
(Hex)
12 11.2896 90.3168 1 2 7.5264 7 86C226
12 12.288 98.304 1 2 8.192 8 3126E8
13 11.2896 90.3168 1 2 6.947446 6 F28BD4
13 12.288 98.304 1 2 7.561846 7 8FD525
14.4 11.2896 90.3168 1 2 6.272 6 45A1CA
14.4 12.288 98.304 1 2 6.826667 6 D3A06E
19.2 11.2896 90.3168 2 2 9.408 9 6872AF
19.2 12.288 98.304 2 2 10.24 A 3D70A3
19.68 11.2896 90.3168 2 2 9.178537 9 2DB492
19.68 12.288 98.304 2 2 9.990243 9 FD809F
19.8 11.2896 90.3168 2 2 9.122909 9 1F76F7
19.8 12.288 98.304 2 2 9.929697 9 EE009E
24 11.2896 90.3168 2 2 7.5264 7 86C226
24 12.288 98.304 2 2 8.192 8 3126E8
26 11.2896 90.3168 2 2 6.947446 6 F28BD4
26 12.288 98.304 2 2 7.561846 7 8FD525
27 11.2896 90.3168 2 2 6.690133 6 BOAC93
27 12.288 98.304 2 2 7.281778 7 482296
Table 48 PLL Frequency Examples
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GENERAL PURPOSE INPUT/OUTPUT
The CSB/GPIO pin can be configured to perform a variety of useful tasks by setting the GPIOSEL
register bits. The GPIO is only available in 2 wire mode.
Note that SLOWCLKEN must be enabled when using the Jack Detect function
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
2:0 GPIOSEL 000 CSB/GPIO pin function select:
000=CSB input
001= Jack insert detect
010=Temp ok
011=Amute active
100=PLL clk o/p
101=PLL lock
110=Reserved
111=Reserved
3 GPIOPOL 0 GPIO Polarity invert
0=Non inverted
1=Inverted
R8
GPIO
control
5:4 OPCLKDIV 00 PLL Output clock division ratio
00=divide by 1
01=divide by 2
10=divide by 3
11=divide by 4
Table 49 CSB/GPIO Control
CONTROL INTERFACE
SELECTION OF CONTROL MODE AND 2-WIRE MODE ADDRESS
The control interface can operate as either a 3-wire or 2-wire MPU interface. The MODE pin
determines the 2 or 3 wire mode as shown in Table 50.
The WM8510 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 register bits, corresponding to the 9 bits in each
control register.
MODE INTERFACE FORMAT
Low 2 wire
High 3 wire
Table 50 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/GPIO latches in a complete control word consisting of the last 16 bits.
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Figure 32 3-Wire Serial Control Interface
2-WIRE SERIAL CONTROL MODE
The WM8510 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 device address (this is not the same as the 7-bit
address of each register in the WM8510).
The WM8510 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 WM8510, then the WM8510 responds by pulling SDIN low on the next clock pulse
(ACK). If the address is not recognised or the R/W bit is ‘1’ when operating in write only mode, the
WM8510 returns to the idle condition and wait for a new start condition and valid address.
During a write, once the WM8510 has acknowledged a correct address, the controller sends the first
byte of control data (B15 to B8, i.e. the WM8510 register address plus the first bit of register data).
The WM8510 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 WM8510 acknowledges again by pulling SDIN low.
Transfers are complete when there is a low to high transition on SDIN while SCLK is high. After a
complete sequence the WM8510 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 1
(BITS 7 TO 0)
remaining 8 bits of
register data
STOPSTART
ACK
(LOW)
ACK
(LOW)
Figure 33 2-Wire Serial Control Interface
In 2-wire mode the WM8510 has a fixed device address, 0011010.
RESETTING THE CHIP
The WM8510 can be reset by performing a write of any value to the software reset register (address
0 hex). This will cause all register values to be reset to their default values. In addition to this there
is a Power-On Reset (POR) circuit which ensures that the registers are set to default when the
device is powered up.
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POWER SUPPLIES
The WM8510 can use up to four separate power supplies:
AVDD and AGND: Analogue supply, powers all analogue functions except the speaker output and
mono output drivers. AVDD can range from 2.5V 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.
SPKVDD and SPKGND: Headphone and Speaker supplies, power the speaker and mono output
drivers. SPKVDD can range from 2.5V to 5.5V. SPKVDD can be tied to AVDD, but it requires
separate layout and decoupling capacitors to curb harmonic distortion. With a larger SPKVDD, louder
headphone and speaker outputs can be achieved with lower distortion. If SPKVDD is lower than
AVDD (or 1.5 x AVDD for BOOST mode), the output signal may be clipped.
DCVDD: Digital core supply, powers all digital functions except the audio and control interfaces.
DCVDD can range from 1.71V to 3.6V, and has no effect on audio quality. The return path for
DCVDD is DGND, which is shared with DBVDD.
DCVDD should be greater than or equal to 1.9V when using the PLL.
DBVDD can range from 1.71V to 3.6V. DBVDD return path is through DGND.
It is possible to use the same supply voltage for all four supplies. However, digital and analogue
supplies should be routed and decoupled separately on the PCB to keep digital switching noise out of
the analogue signal paths.
Note:
DCVDD should be greater than or equal to 1.9V when using the PLL.
RECOMMENDED POWER UP/DOWN SEQUENCE
In order to minimise output pop and click noise, it is recommended that the WM8510 device is
powered up and down using one of the following sequences:
Power Up When NOT Using the Output 1.5x Boost Stage:
1. Turn on external power supplies. Wait for supply voltage to settle.
2. Set BIASEN = 1, BUFIOEN = 1 and also the VMIDSEL[1:0] bits in the Power Management
1 register. * Notes 1 and 2.
3. Wait for the VMID supply to settle. * Note 2.
4. Enable DAC by setting DACEN = 1.
5. Enable mixers as required.
6. Enable output stages as required.
7. Unmute DAC by setting DACMU = 0.
Power Up When Using the Output 1.5x Boost Stage:
1. Turn on external power supplies. Wait for supply voltage to settle.
2. Enable 1.5x output boost. Set MONOBOOST = 1 and SPKBOOST = 1 as required.
3. Set BIASEN = 1, BUFIOEN = 1, BUFDCOPEN = 1 and also the VMIDSEL[1:0] bits in the
Power Management 1 register. * Notes 1 and 2.
4. Wait for the VMID supply to settle. * Note 2.
5. Enable DAC by setting DACEN = 1.
6. Enable mixers as required.
7. Enable output stages as required.
8. Unmute DAC by setting DACMU = 0.
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Power Down (all cases):
1. Soft mute DAC by setting DACMU = 1.
2. Disable power management register 1 by setting R1[8:0]=0x000
3. Disable all other output stages.
4. Turn off external power supplies.
Notes:
1. This step enables the internal device bias buffer and the VMID buffer for unassigned
inputs/outputs. This will provide a startup reference voltage for all inputs and outputs. This will
cause the inputs and outputs to ramp towards VMID (NOT using output 1.5x boost) or 1.5 x
(AVDD/2) (using output 1.5x boost) in a way that is controlled and predictable (see note 2).
2. Choose the value of the VMIDSEL bits based on the startup time (VMIDSEL=10 for slowest
startup, VMIDSEL=11 for fastest startup). Startup time is defined by the value of the VMIDSEL
bits (the reference impedance) and the external decoupling capacitor on VMID.
In addition to the power on sequence, it is recommended that the zero cross functions are used when
changing the volume in the PGAs to avoid any audible pops or clicks.
Vpora
DGND
Internal POR active
Device Ready
No Power
Vpor_off
Power Supply
POR
I2S Clocks
ADC Internal
State
tmidrail_on
Analogue Inputs
ADCDAT pin
GD
ADCEN bit
Power down Init Normal Operation Normal OperationInitPD Power down
ADC enabled ADC enabledADC off
tadcint
DNC
INPPGAEN bit
tadcint
INPPGA enabled
DNC
GD GD GD
POR
POR Undefined
VMID enabled
VMIDSEL/
BIASEN bits
AVDD/2
tmidrail_off
(Note 1) (Note 2)
(Note 3)
(Note 4)
Vpor_on
Figure 34 ADC Power Up and Down Sequence (not to scale)
SYMBOL MIN TYPICAL MAX UNIT
tmidrail_on 500 ms
tmidrail_off >10 s
tadcint 2/fs n/fs
Table 51 Typical POR Operation (typical values, not tested)
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Notes:
1. The analogue input pin charge time, tmidrail_on, is determined by the VMID pin charge time. This
time is dependent upon the value of VMID decoupling capacitor and VMID pin input resistance
and AVDD power supply rise time.
2. The analogue input pin discharge time, tmidrail_off, is determined by the analogue input coupling
capacitor discharge time. The time, tmidrail_off, is measured using a 1µF capacitor on the
analogue input but will vary dependent upon the value of input coupling capacitor.
3. While the ADC is enabled there will be LSB data bit activity on the ADCDAT pin due to system
noise but no significant digital output will be present.
4. The VMIDSEL and BIASEN bits must be set to enable analogue input midrail voltage and for
normal ADC operation.
5. ADCDAT data output delay from power up - with power supplies starting from 0V - is determined
primarily by the VMID charge time. ADC initialisation and power management bits may be set
immediately after POR is released; VMID charge time will be significantly longer and will dictate
when the device is stabilised for analogue input.
6. ADCDAT data output delay at power up from device standby (power supplies already applied) is
determined by ADC initialisation time, 2/fs.
Figure 35 DAC Power Up and Down Sequence (not to scale)
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SYMBOL MIN TYPICAL MAX UNIT
tline_midrail_on 500 ms
tline_midrail_off 1 s
thp_midrail_on 500 ms
thp__midrail_off 6 s
tdacint 2/fs n/fs
Table 52 Typical POR Operation (typical values, not tested)
Notes:
1. The lineout charge time, tline_midrail_on, is mainly determined by the VMID pin charge time. This
time is dependent upon the value of VMID decoupling capacitor and VMID pin input resistance
and AVDD power supply rise time. The values above were measured using a 4.7µF capacitor.
2. It is not advisable to allow DACDAT data input during initialisation of the DAC. If the DAC data
value is not zero at point of initialisation, then this is likely to cause a pop noise on the analogue
outputs. The same is also true if the DACDAT is removed at a non-zero value, and no mute
function has been applied to the signal beforehand.
3. The lineout discharge time, tline_midrail_off, is dependent upon the value of the lineout coupling
capacitor and the leakage resistance path to ground. The values above were measured using a
10µF output capacitor.
4. The headphone charge time, thp_midrail_on, is dependent upon the value of VMID decoupling
capacitor and VMID pin input resistance and AVDD power supply rise time. The values above
were measured using a 4.7µF VMID decoupling capacitor.
5. The headphone discharge time, thp_midrail_off, is dependent upon the value of the headphone
coupling capacitor and the leakage resistance path to ground. The values above were
measured using a 100µF capacitor.
The VMIDSEL and BIASEN bits must be set to enable analogue output midrail voltage and for
normal DAC operation.
POWER MANAGEMENT
SAVING POWER BY REDUCING OVERSAMPLING RATE
The default mode of operation of the ADC and DAC digital filters is in 64x oversampling mode. Under
the control of ADCOSR and DACOSR the oversampling rate may be doubled. 64x oversampling
results in a slight decrease in noise performance compared to 128x but lowers the power
consumption of the device.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R10
DAC control
3 DACOSR128 0 DAC oversample rate select
0 = 64x (lowest power)
1 = 128x (best SNR)
R14
ADC control
3 ADCOSR128 0 ADC oversample rate select
0 = 64x (lowest power)
1 = 128x (best SNR)
Table 53 ADC and DAC Oversampling Rate Selection
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VMID
The analogue circuitry will not work when VMID is disabled (VMIDSEL[1:0] = 00b). The impedance of
the VMID resistor string, together with the decoupling capacitor on the VMID pin will determine the
startup time of the VMID circuit.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R1
Power
management 1
1:0 VMIDSEL 00 Reference string impedance to VMID pin
(determines startup time):
00=off (open circuit)
01=50kΩ
10=500kΩ
11=5kΩ (for fastest startup)
Table 54 VMID Impedance Control
BIASEN
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION
R1
Power
management 1
3 BIASEN 0 Analogue amplifier bias control
0=Disabled
1=Enabled
Table 55 BIASEN Control
ESTIMATED SUPPLY CURRENTS
When either the DAC or ADC are enabled it is estimated that approximately 4mA will be drawn from
DCVDD when DCVDD=1.8V and fs=48kHz (This will be lower at lower sample rates). When the PLL
is enabled an additional 700 microamps will be drawn from DCVDD.
Table 45 shows the estimated 3.3V AVDD current drawn by various circuits, by register bit.
REGISTER BIT AVDD CURRENT (MILLIAMPS)
BUFDCOPEN 0.1
MONOEN 0.2
PLLEN 1.4 (with clocks applied)
MICBEN 0.5
BIASEN 0.3
BUFIOEN 0.1
VMIDSEL 10K=>0.3, less than 0.1 for 50k/500k
BOOSTEN 0.2
INPPGAEN 0.2
ADCEN x64 (ADCOSR=0)=>2.6, x128 (ADCOSR=1)=>4.9
MONOEN 0.2
SPKPEN 1mA from SPKVDD + 0.2mA from AVDD in 5V mode
SPKNEN 1mA from SPKVDD + 0.2mA from AVDD in 5V mode
MONOMIXEN 0.2
SPKMIXEN 0.2
DACEN x64 (DACOSR=0)=>1.8, x128(DACOSR=1)=>1.9
Table 56 AVDD Supply Current
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POWER SAVING
For minimum power consumption in standby mode, VMIDSEL should not be set to default. Instead,
the following sequence of writes should be implemented:
1. R10[6] = 1 (DACMU=1).
2. R1 = 0x00.
3. R2 = 0x00.
4. R3 = 0x00
5. R1 = 0x02 (VMIDSEL[1:0] = 10).
After reset, all register values are set to default. In order to achieve minimum power consumption,
the following sequence of writes should be implemented.
1. R10[6] = 1 (DACMU=1).
2. R1 = 0x00.
3. R0 = 0xFF.
4. R1 = 0x02 (VMIDSEL[1:0] = 10).
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REGISTER MAP
ADDR
B[15:9]
DEF’T
VAL
DEC
HEX
REGISTER
NAME
B8 B7 B6 B5 B4 B3 B2 B1 B0
(HEX)
0 00 Software Reset Software reset
1 01 Power manage’t 1 BUFDCOP
EN 0 MIC2EN PLLEN MICBEN BIASEN BUFIOEN
VMIDSEL 000
2 02 Power manage’t 2 0 0 0 0 BOOSTEN
0 INPPGAEN
0 ADCEN 000
3 03 Power manage’t 3 0 MONOEN SPKNEN SPKPEN 0 MONO
MIXEN SPK
MIXEN 0 DACEN 000
4 04 Audio Interface BCP FRAMEP WL FMT DACLRSW
AP ADCLRSW
AP 0 050
5 05 Companding ctrl 0 0 0 0 DAC_COMP ADC_COMP LOOPBACK 000
6 06 Clock Gen ctrl CLKSEL MCLKDIV BCLKDIV 0 MS 140
7 07 Additional ctrl 0 0 0 0 0 SR SLOWCLK
EN 000
8 08 GPIO Stuff 0 0 0 OPCLKDIV GPIOPOL
GPIOSEL 000
10 0A DAC Control 0 0 DACMU DEEMPH DACOSR
128 AMUTE 0 DACPOL 000
11 0B DAC digital Vol 0 DACVOL 0FF
14 0E ADC Control HPFEN HPFAPP HPFCUT ADCOSR
128 0 0 ADCPOL 100
15 0F ADC Digital Vol 0 ADCVOL 0FF
24 18 DAC Limiter 1 LIMEN LIMDCY LIMATK 032
25 19 DAC Limiter 2 0 0 LIMLVL LIMBOOST 000
27 1B Notch Filter 1 NFU NFEN NFA0[13:7] 000
28 1C Notch Filter 2 NFU 0 NFA0[6:0] 000
29 1D Notch Filter 3 NFU 0 NFA1[13:7] 000
30 1E Notch Filter 4 NFU 0 NFA1[6:0] 000
32 20 ALC control 1 ALCSEL 0 0 ALCMAX ALCMIN 038
33 21 ALC control 2 ALCZC ALCHLD ALCLVL 00B
34 22 ALC control 3 ALCMODE ALCDCY ALCATK 032
35 23 Noise Gate 0 0 0 0 0 NGEN NGTH 000
36 24 PLL N 0 0 0 0 PLL_PRE
SCALE PLLN[3:0] 008
37 25 PLL K 1 0 0 0 PLLK[23:18] 00C
38 26 PLL K 2 PLLK[17:9] 093
39 27 PLL K 3 PLLK[8:0] 0E9
40 28 Attenuation ctrl 0 0 0 0 0 0 MONOATTN
SPKATTN
0 000
44 2C Input ctrl MBVSEL 0 0 0 0 MIC2MOD
E MIC2_2
INPPGA MICN2
INPPGA MICP2
INPPGA
003
45 2D INP PGA gain ctrl 0 INPPGAZC
INPPGA
MUTE INPPGAVOL 010
47 2F ADC Boost ctrl PGABOOST
0 MICP2BOOSTVOL 0 MIC2_2BOOSTVOL 100
49 31 Output ctrl 0 0 0 0 0 MONO
BOOST SPK
BOOST TSDEN VROI 002
50 32 SPK mixer ctrl 0 0 0 MI C2_2SP
K 0 0 0 BYP2SPK
DAC2SPK
001
54 36 SPK volume ctrl 0 SPKZC SPKMUTE
SPKVOL 039
56 38 MONO mixer ctrl 0 0 MONO
MUTE 0 0 0 MIC2_2
MONO BYP2
MONO DAC2
MONO
001
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REGISTER BITS BY ADDRESS
Notes:
1. Default values of N/A indicate non-latched data bits (e.g. software reset or volume update bits).
2. Register bits marked as "Reserved" should not be changed from the default.
REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
0 (00h) [8:0] RESET N/A Software reset Resetting the
Chip
8 BUFDCOPEN 0 Dedicated buffer for DC level shifting output stages
when in 1.5x gain boost configuration.
0=Buffer disabled
1=Buffer enabled (required for 1.5x gain boost)
Analogue
Outputs
7 0 Reserved
6 MIC2EN 0 MIC2 input buffer enable
0 = OFF
1 = ON
MIC Inputs
5 PLLEN 0 PLL enable
0=PLL off
1=PLL on
Master Clock
and Phase
Locked Loop
(PLL)
4 MICBEN 0 Microphone Bias Enable
0 = OFF (high impedance output)
1 = ON
Microphone
Biasing Circuit
3 BIASEN 0 Analogue amplifier bias control
0=Disabled
1=Enabled
Power
Management
2 BUFIOEN 0 Unused input/output tie off buffer enable
0=Disabled
1=Enabled
Enabling the
Outputs
1 (01h)
1:0 VMIDSEL 00 Reference string impedance to VMID pin:
00=off (open circuit)
01=50kΩ
10=500kΩ
11=5kΩ
Power
Management
8:5 0000 Reserved
4 BOOSTEN 0 Input BOOST enable
0 = Boost stage OFF
1 = Boost stage ON
Input Boost
3 0 Reserved
2 INPPGAEN 0 Input microphone PGA enable
0 = disabled
1 = enabled
Input Signal
Path
1 0 Reserved
2 (02h)
0 ADCEN 0 ADC Enable Control
0 = ADC disabled
1 = ADC enabled
Analogue to
Digital Converter
(ADC)
8 0 Reserved
7 MONOEN 0 MONOOUT enable
0 = disabled
1 = enabled
Analogue
Outputs
3 (03h)
6 SPKNEN 0 SPKOUTN enable
0 = disabled
1 = enabled
Analogue
Outputs
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
5 SPKPEN 0 SPKOUTP enable
0 = disabled
1 = enabled
Analogue
Outputs
4 0 Reserved
3 MONOMIXEN 0 Mono Mixer Enable
0 = disabled
1 = enabled
Analogue
Outputs
2 SPKMIXEN 0 Speaker Mixer Enable
0 = disabled
1 = enabled
Analogue
Outputs
1 0 Reserved
0 DACEN 0 DAC enable
0 = DAC disabled
1 = DAC enabled
Analogue
Outputs
8 BCP 0 BCLK polarity
0=normal
1=inverted
Digital Audio
Interfaces
7 FRAMEP 0 Frame clock polarity
0=normal
1=inverted
Digital Audio
Interfaces
6:5 WL 10 Word length
00=16 bits
01=20 bits
10=24 bits
11=32 bits
Digital Audio
Interfaces
4:3 FMT 10 Audio interface Data Format Select:
00=Right Justified
01=Left Justified
10=I2S format
11= DSP/PCM mode
Digital Audio
Interfaces
2 DACLRSWAP 0 Controls whether DAC data appears in ‘right’ or ‘left’
phases of FRAME clock:
0=DAC data appear in ‘left’ phase of FRAME
1=DAC data appears in ‘right’ phase of FRAME
Digital Audio
Interfaces
1 ADCLRSWAP 0 Controls whether ADC data appears in ‘right’ or ‘left’
phases of FRAME clock:
0=ADC data appear in ‘left’ phase of FRAME
1=ADC data appears in ‘right’ phase of FRAME
Digital Audio
Interfaces
4 (04h)
0 0 Reserved
8:5 0000 Reserved
4:3 DAC_COMP 00 DAC companding
00=off
01=reserved
10=µ-law
11=A-law
Digital Audio
Interfaces
5 (05h)
2:1 ADC_COMP 00 ADC companding
00=off
01=reserved
10=µ-law
11=A-law
Digital Audio
Interfaces
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
0 LOOPBACK 0 Digital loopback function
0=No loopback
1=Loopback enabled, ADC data output is fed directly
into DAC data input.
Digital Audio
Interfaces
8 CLKSEL 1 Controls the source of the clock for all internal
operation:
0=MCLK
1=PLL output
Digital Audio
Interfaces
7:5 MCLKDIV 010 Sets the scaling for either the MCLK or PLL clock
output (under control of CLKSEL)
000=divide by 1
001=divide by 1.5
010=divide by 2
011=divide by 3
100=divide by 4
101=divide by 6
110=divide by 8
111=divide by 12
Digital Audio
Interfaces
4:2 BCLKDIV 000 Configures the BCLK and FRAME output frequency,
for use when the chip is master over BCLK.
000=divide by 1 (BCLK=MCLK)
001=divide by 2 (BCLK=MCLK/2)
010=divide by 4
011=divide by 8
100=divide by 16
101=divide by 32
110=reserved
111=reserved
Digital Audio
Interfaces
1 0 Reserved
6 (06h)
0 MS 0 Sets the chip to be master over FRAME and BCLK
0=BCLK and FRAME clock are inputs
1=BCLK and FRAME clock are outputs generated by
the WM8510 (MASTER)
Digital Audio
Interfaces
8:4 00000 Reserved
3:1 SR 000 Approximate sample rate (configures the coefficients
for the internal digital filters):
000=48kHz
001=32kHz
010=24kHz
011=16kHz
100=12kHz
101=8kHz
110-111=reserved
Audio Sample
Rates
7 (07h)
0 SLOWCLKEN 0 Slow clock enable. Used for both the jack insert
detect debounce circuit and the zero cross timeout.
0 = slow clock disabled
1 = slow clock enabled
Audio Sample
Rates
8:6 000 Reserved 8 (08h)
5:4 OPCLKDIV 00 PLL Output clock division ratio
00=divide by 1
01=divide by 2
10=divide by 3
11=divide by 4
General
Purpose Input
Output
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
3 GPIOPOL 0 GPIO Polarity invert
0=Non inverted
1=Inverted
General
Purpose Input
Output
2:0 GPIOSEL 000 CSB/GPIO pin function select:
000=CSB input
001= Jack insert detect
010=Temp ok
011=Amute active
100=PLL clk o/p
101=PLL lock
110=Reserved
111=Reserved
General
Purpose Input
Output
9 (09h) 8:0 Reserved
8:7 00 Reserved
6 DACMU 0 DAC soft mute enable
0 = DACMU disabled
1 = DACMU enabled
Output Signal
Path
5:4 DEEMPH 00 De-Emphasis Control
00 = No de-emphasis
01 = 32kHz sample rate
10 = 44.1kHz sample rate
11 = 48kHz sample rate
Output Signal
Path
3 DACOSR128 0 DAC oversample rate select
0 = 64x (lowest power)
1 = 128x (best SNR)
Power
Management
2 AMUTE 0 DAC auto mute enable
0 = auto mute disabled
1 = auto mute enabled
Output Signal
Path
1 0 Reserved
10 (0Ah)
0 DACPOL 0 DAC Polarity Invert
0 = No inversion
1 = DAC output inverted
Output Signal
Path
8 0 Reserved 11 (0Bh)
7:0 DACVOL 11111111 DAC Digital Volume Control
0000 0000 = Unused
0000 0001 = -127dB = mute
0000 0010 = -126.5dB
... 0.5dB steps up to
1111 1111 = 0dB
Output Signal
Path
12 (0Ch) 8:0 Reserved
13 (0Dh) 8:0 Reserved
8 HPFEN 1 High Pass Filter Enable
0=disabled
1=enabled
Analogue to
Digital Converter
(ADC)
7 HPFAPP 0 Select audio mode or application mode
0=Audio mode (1st order, fc = ~3.7Hz)
1=Application mode (2nd order, fc = HPFCUT)
Analogue to
Digital Converter
(ADC)
6:4 HPFCUT 000 Application mode cut-off frequency
See Table 11 details.
Analogue to
Digital Converter
(ADC)
14 (0Eh)
3 ADCOSR128 0 ADC oversample rate select
0 = 64x (lowest power)
1 = 128x (best SNR)
Power
Management
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
2:1 00 Reserved
0 ADCPOL 0 ADC Polarity
0=normal
1=inverted
Analogue to
Digital Converter
(ADC)
8 0 Reserved 15 (0Fh)
7:0 ADCVOL 11111111 ADC Digital Volume Control
0000 0000 = Digital Mute
0000 0001 = -127dB
0000 0010 = -126.5dB
... 0.5dB steps up to
1111 1111 = 0dB
Analogue to
Digital Converter
(ADC)
8 LIMEN 0 Enable the DAC digital limiter:
0=disabled
1=enabled
Output Signal
Path
7:4 LIMDCY 0011 DAC Limiter Decay time (per 6dB gain change) for
44.1kHz sampling. Note that these will scale with
sample rate:
0000=750us
0001=1.5ms
0010=3ms
0011=6ms
0100=12ms
0101=24ms
0110=48ms
0111=96ms
1000=192ms
1001=384ms
1010=768ms
1011 to 1111=1.536s
Output Signal
Path
24 (18h)
3:0 LIMATK 0010 DAC Limiter Attack time (per 6dB gain change) for
44.1kHz sampling. Note that these will scale with
sample rate.
0000=94us
0001=188s
0010=375us
0011=750us
0100=1.5ms
0101=3ms
0110=6ms
0111=12ms
1000=24ms
1001=48ms
1010=96ms
1011 to 1111=192ms
Output Signal
Path
8:7 00 Reserved 25 (19h)
6:4 LIMLVL 000 DAC Limiter Programmable signal threshold level
(determines level at which the limiter starts to operate)
000=-1dB
001=-2dB
010=-3dB
011=-4dB
100=-5dB
101 to 111=-6dB
Output Signal
Path
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
3:0 LIMBOOST 0000 DAC Limiter volume boost (can be used as a stand
alone volume boost when LIMEN=0):
0000=0dB
0001=+1dB
0010=+2dB
… (1dB steps)
1011=+11dB
1100=+12dB
1101 to 1111=reserved
Output Signal
Path
8 NFU 0 Notch filter update. The notch filter values used
internally only update when one of the NFU bits is set
high.
Analogue to
Digital Converter
(ADC)
7 NFEN 0 Notch filter enable:
0=Disabled
1=Enabled
Analogue to
Digital Converter
(ADC)
27 (1Bh)
6:0 NFA0[13:7] 0000000 Notch Filter a0 coefficient, bits [13:7] Analogue to
Digital Converter
(ADC)
8 NFU 0 Notch filter update. The notch filter values used
internally only update when one of the NFU bits is set
high.
Analogue to
Digital Converter
(ADC)
7 0 Reserved
28 (1Ch)
6:0 NFA0[6:0] 0000000 Notch Filter a0 coefficient, bits [6:0] Analogue to
Digital Converter
(ADC)
8 NFU 0 Notch filter update. The notch filter values used
internally only update when one of the NFU bits is set
high.
Analogue to
Digital Converter
(ADC)
7 0 Reserved
29 (1Dh)
6:0 NFA1[13:7] 0000000 Notch Filter a1 coefficient, bits [13:7] Analogue to
Digital Converter
(ADC)
8 NFU 0 Notch filter update. The notch filter values used
internally only update when one of the NFU bits is set
high.
Analogue to
Digital Converter
(ADC)
7 0 Reserved
30 (1Eh)
6:0 NFA1[6:0] 0000000 Notch Filter a1 coefficient, bits [6:0] Analogue to
Digital Converter
(ADC)
8 ALCSEL 0 ALC function select:
0=ALC off (PGA gain set by INPPGAVOL register bits)
1=ALC on (ALC controls PGA gain)
Input Limiter /
Automatic Level
Control (ALC)
7:6 Reserved
32 (20h)
5:3 ALCMAX 111 Set Maximum Gain of PGA when using ALC:
111=+35.25dB
110=+29.25dB
101=+23.25dB
100=+17.25dB
011=+11.25dB
010=+5.25dB
001=-0.75dB
000=-6.75dB
Input Limiter /
Automatic Level
Control (ALC)
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
2:0 ALCMIN 000 Set minimum gain of PGA when using ALC:
000=-12dB
001=-6dB
010=0dB
011=+6dB
100=+12dB
101=+18dB
110=+24dB
111=+30dB
Input Limiter /
Automatic Level
Control (ALC)
8 ALCZC 0 ALC zero cross detection.
0 = disabled (recommended)
1 = enabled
It is recommended that zero cross is not used in
conjunction with the ALC or Limiter functions
Input Limiter /
Automatic Level
Control (ALC)
7:4 ALCHLD 000 ALC hold time before gain is increased.
0000 = 0ms
0001 = 2.67ms
0010 = 5.33ms
… (time doubles with every step)
1111 = 43.691s
Input Limiter /
Automatic Level
Control (ALC)
33 (21h)
3:0 ALCLVL 1011 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
Input Limiter /
Automatic Level
Control (ALC)
8 ALCMODE 0 Determines the ALC mode of operation:
0=ALC mode
1=Limiter mode.
Input Limiter /
Automatic Level
Control (ALC)
Decay (gain ramp-up) time (ALCMODE =0)
Per step Per 6dB 90% of
range
0000 410us 3.38ms 23.6ms
0001 820us 6.6ms 47.2ms
0010 1.64ms 13.1ms 94.5
… (time doubles with every step)
0011
1010 or
higher
420ms 3.36s 24.2s
Decay (gain ramp-up) time (ALCMODE =1)
Per step Per 6dB 90% of
range
0000 90.8us 726us 5.23ms
0001 182us 1.45ms 10.5ms
0010 363us 2.91ms 20.9ms
… (time doubles with every step)
34 (22h)
7:4 ALCDCY
0011
1010 93ms 744ms 5.36s
Input Limiter /
Automatic Level
Control (ALC)
ALC attack (gain ramp-down) time
(ALCMODE = 0)
Per step Per 6dB 90% of
range
0000 104us 832us 6ms
0001 208us 1.664ms 12ms
3:0 ALCATK 0010
0010 416us 3.33ms 24ms
Input Limiter /
Automatic Level
Control (ALC)
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
… (time doubles with every step)
1010 or
higher
106ms 852ms 6.18s
ALC attack (gain ramp-down) time
(ALCMODE = 1)
Per step Per 6dB 90% of
range
0000 22.7us 182.4us 1.31ms
0001 45.4us 363us 2.62ms
0010 90.8us 726us 5.23ms
… (time doubles with every step)
0010
1010 23.2ms 186ms 1.34s
8:4 00000 Reserved
3 NGEN 0 ALC Noise gate function enable
1 = enable
0 = disable
Input Limiter /
Automatic Level
Control (ALC)
35 (23h)
2:0 NGTH 000 ALC Noise gate threshold:
000=-39dB
001=-45dB
010=-51db
… (6dB steps)
111=-81dB
Input Limiter /
Automatic Level
Control (ALC)
8:5 0000 Reserved
4 PLLPRESCALE 0 0 = MCLK input not divided (default)
1 = Divide MCLK by 2 before input PLL
Master Clock
and Phase
Locked Loop
(PLL)
36 (24h)
3:0 PLLN[3:0] 1000 Integer (N) part of PLL input/output frequency ratio.
Use values greater than 5 and less than 13.
Master Clock
and Phase
Locked Loop
(PLL)
8:6 000 Reserved 37 (25h)
5:0 PLLK[23:18] 001100 Fractional (K) part of PLL1 input/output frequency ratio
(treat as one 24-digit binary number).
Master Clock
and Phase
Locked Loop
(PLL)
38 (26h) 8:0 PLLK[17:9] 01001001
1
Fractional (K) part of PLL1 input/output frequency ratio
(treat as one 24-digit binary number).
Master Clock
and Phase
Locked Loop
(PLL)
39 (27h) 8:0 PLLK[8:0] 01110100
1
Fractional (K) part of PLL1 input/output frequency ratio
(treat as one 24-digit binary number).
Master Clock
and Phase
Locked Loop
(PLL)
8:3 000000 Reserved
2 MONOATTN 0 Attenuation control for bypass path (output of input
boost stage) to mono mixer input
0 = 0dB
1 = -10dB
Analogue
Outputs
1 SPKATTN 0 Attenuation control for bypass path (output of input
boost stage) to speaker mixer input
0 = 0dB
1 = -10dB
Analogue
Outputs
40 (28h)
0 0 Reserved
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
8 MBVSEL 0 Microphone Bias Voltage Control
0 = 0.9 * AVDD
1 = 0.75 * AVDD
Input Signal
Path
7:4 0000 Reserved
3 MIC2MODE 0 Auxiliary Input Mode
0 = inverting buffer
1 = mixer (on-chip input resistor bypassed)
Input Signal
Path
2 MIC2_2INPP
GA
0 Select AUX amplifier output as input PGA signal
source.
0=AUX not connected to input PGA
1=AUX connected to input PGA amplifier negative
terminal.
Input Signal
Path
1 MICN2INPPGA 1 Connect MICN to input PGA negative terminal.
0=MICN not connected to input PGA
1=MICN connected to input PGA amplifier negative
terminal.
Input Signal
Path
44 (2Ch)
0 MICP2INPPGA 1 Connect input PGA amplifier positive terminal to MICP
or VMID.
0 = input PGA amplifier positive terminal connected to
VMID
1 = input PGA amplifier positive terminal connected to
MICP through variable resistor string
Input Signal
Path
8 0 Reserved
7 INPPGAZC 0 Input PGA zero cross enable:
0=Update gain when gain register changes
1=Update gain on 1st zero cross after gain register
write.
Input Signal
Path
6 INPPGAMUTE 0 Mute control for input PGA:
0=Input PGA not muted, normal operation
1=Input PGA muted (and disconnected from the
following input BOOST stage).
Input Signal
Path
45 (2Dh)
5:0 INPPGAVOL 010000 Input PGA volume
000000 = -12dB
000001 = -11.25db
.
010000 = 0dB
.
111111 = 35.25dB
Input Signal
Path
8 PGABOOST 1 Input Boost
0 = PGA output has +0dB gain through input BOOST
stage.
1 = PGA output has +20dB gain through input BOOST
stage.
Input Signal
Path
7 0 Reserved
6:4 MICP2BOOST
VOL
000 Controls the MICP pin to the input boost stage (NB,
when using this path set MICP2INPPGA=0):
000=Path disabled (disconnected)
001=-12dB gain through boost stage
010=-9dB gain through boost stage
…
111=+6dB gain through boost stage
Input Signal
Path
47 (2Fh)
3 0 Reserved
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
2:0 MIC2_2BOOST
VOL
000 Controls the auxilliary amplifer to the input boost
stage:
000=Path disabled (disconnected)
001=-12dB gain through boost stage
010=-9dB gain through boost stage
…
111=+6dB gain through boost stage
Input Signal
Path
8:4 00000 Reserved
3 MONOBOOST 0 Mono output boost stage control (see Table 30 for
details)
0=No boost (output is inverting buffer)
1=1.5x gain boost
Analogue
Outputs
2 SPKBOOST 0 Speaker output boost stage control (see Table 30 for
details)
0=No boost (outputs are inverting buffers)
1 = 1.5x gain boost
Analogue
Outputs
1 TSDEN 1 Thermal Shutdown Enable
0 : thermal shutdown disabled
1 : thermal shutdown enabled
Output Switch
49 (31h)
0 VROI 0 VREF (AVDD/2 or 1.5xAVDD/2) to analogue output
resistance
0: approx 1kΩ
1: approx 30 kΩ
Analogue
Outputs
8:6 000 Reserved
5 MIC2_2SPK 0 Output of auxiliary amplifier to speaker mixer input
0 = not selected
1 = selected
Analogue
Outputs
4:2 000 Reserved
1 BYP2SPK 0 Bypass path (output of input boost stage) to speaker
mixer input
0 = not selected
1 = selected
Analogue
Outputs
50 (32h)
0 DAC2SPK 1 Output of DAC to speaker mixer input
0 = not selected
1 = selected
Analogue
Outputs
8
7 SPKZC 0 Speaker Volume control zero cross enable:
1 = Change gain on zero cross only
0 = Change gain immediately
Analogue
Outputs
6 SPKMUTE 0 Speaker output mute enable
0=Speaker output enabled
1=Speaker output muted (VMIDOP)
Analogue
Outputs
54 (36h)
5:0 SPKVOL 111001 Speaker Volume Adjust
111111 = +6dB
111110 = +5dB
… (1.0 dB steps)
111001=0dB
…
000000=-57dB
Analogue
Outputs
56 (38h) 8:7 0 Reserved
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REGISTER
ADDRESS
BIT LABEL DEFAULT DESCRIPTION REFER TO
6 MONOMUTE 0 MONOOUT Mute Control
0=No mute
1=Output muted. During mute the mono output will
output VMID which can be used as a DC reference for
a headphone out.
Analogue
Outputs
5:3 0 Reserved
2 MIC2_2MONO 0 Output of Auxillary amplifier to mono mixer input:
0 = not selected
1 = selected
Analogue
Outputs
1 BYP2MONO 0 Bypass path (output of input boost stage) to mono
mixer input
0 = non selected
1 = selected
Analogue
Outputs
0 DAC2MONO 1 Output of DAC to mono mixer input
0 = not selected
1 = selected
Analogue
Outputs
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DIGITAL FILTER CHARACTERISTICS
PARAMETER TEST CONDITIONS MIN TYP MAX UNIT
ADC Filter
+/- 0.025dB 0 0.454fs
Passband
-6dB 0.5fs
Passband Ripple +/- 0.025 dB
Stopband 0.546fs
Stopband Attenuation f > 0.546fs -60 dB
Group Delay3 21/fs
ADC High Pass Filter
-3dB 3.7
-0.5dB 10.4
High Pass Filter Corner
Frequency
-0.1dB 21.6
Hz
DAC Filter
+/- 0.035dB 0 0.454fs
Passband
-6dB 0.5fs
Passband Ripple +/-0.035 dB
Stopband 0.546fs
Stopband Attenuation f > 0.546fs -55 dB
Group Delay3 29/fs
Table 57 Digital Filter Characteristics
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
3. Note that this delay applies only to the filters and does not include additional delays through other digital circuits.
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DAC FILTER RESPONSES
-120
-100
-80
-60
-40
-20
0
00.511.522.53
Frequency (Fs)
Response (dB)
-0.2
-0.15
-0.1
-0.05
0
0.05
0.1
0.15
0.2
0 0.1 0.2 0.3 0.4 0.5
Frequency (Fs)
Response (dB)
Figure 36 DAC Digital Filter Frequency Response Figure 37 DAC Digital Filter Ripple
ADC FILTER RESPONSES
-120
-100
-80
-60
-40
-20
0
0 0.5 1 1.5 2 2.5 3
Frequency (Fs)
Response (dB)
-0.2
-0.15
-0.1
-0.05
0
0.05
0.1
0.15
0.2
0 0.1 0.2 0.3 0.4 0.5
Frequency (Fs)
Response (dB)
Figure 38 ADC Digital Filter Frequency Response Figure 39 ADC Digital Filter Ripple
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DE-EMPHASIS FILTER RESPONSES
-10
-9
-8
-7
-6
-5
-4
-3
-2
-1
0
0 2000 4000 6000 8000 10000 12000 14000 16000
Frequency (Hz)
Response (dB)
-0.15
-0.10
-0.05
0.00
0.05
0.10
0.15
0.20
0.25
0.30
0 2000 4000 6000 8000 10000 12000 14000 16000
Frequency (Hz)
Response (dB)
Figure 40 De-emphasis Frequency Response (32kHz) Figure 41 De-emphasis Error (32kHz)
-10
-9
-8
-7
-6
-5
-4
-3
-2
-1
0
0 5000 10000 15000 20000
Frequency (Hz)
Response (dB)
-0.20
-0.15
-0.10
-0.05
0.00
0.05
0.10
0 5000 10000 15000 20000
Frequency (Hz)
Response (dB)
Figure 42 De-emphasis Frequency Response (44.1kHz) Figure 43 De-emphasis Error (44.1kHz)
-10
-9
-8
-7
-6
-5
-4
-3
-2
-1
0
0 5000 10000 15000 20000
Frequency (Hz)
Response (dB)
-0.10
-0.08
-0.06
-0.04
-0.02
0.00
0.02
0.04
0.06
0.08
0.10
0 5000 10000 15000 20000
Frequency (Hz)
Response (dB)
Figure 44 De-emphasis Frequency Response (48kHz) Figure 45 De-emphasis Error (48kHz)
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HIGHPASS FILTER
The WM8510 has a selectable digital highpass filter in the ADC filter path. This filter has two modes,
audio and applications. In audio mode the filter is a 1st order IIR with a cut-off of around 3.7Hz. In
applications mode the filter is a 2nd order high pass filter with a selectable cut-off frequency.
-40
-35
-30
-25
-20
-15
-10
-5
0
5
0 5 10 15 20 25 30 35 40 45
Frequenc y (Hz)
Response (dB)
Figure 46 ADC Highpass Filter Response, HPFAPP=0
-60
-50
-40
-30
-20
-10
0
10
0 200 400 600 800 1000 1200
Frequency (Hz)
Response (dB)
-80
-70
-60
-50
-40
-30
-20
-10
0
10
0 200 400 600 800 1000 1200
Frequency (Hz)
Response (dB)
Figure 47 ADC Highpass Filter Responses (48kHz),
HPFAPP=1, all cut-off settings shown.
Figure 48 ADC Highpass Filter Responses (24kHz),
HPFAPP=1, all cut-off settings shown.
-90
-80
-70
-60
-50
-40
-30
-20
-10
0
10
0 200 400 600 800 1000 1200
Frequency (Hz)
Response (dB)
Figure 49 ADC Highpass Filter Responses (12kHz),
HPFAPP=1, all cut-off settings shown.
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APPLICATIONS INFORMATION
RECOMMENDED EXTERNAL COMPONENTS
Figure 50 Recommended External Components
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PACKAGE DIAGRAM
NOTES:
A. ALL LINEAR DIMENSIONS ARE IN MILLIMETERS.
B. THIS DRAWING IS SUBJECT TO CHANGE WITHOUT NOTICE.
C. BODY DIMENSIONS DO NOT INCLUDE MOLD FLASH OR PROTRUSION, NOT TO EXCEED 0.20MM.
D. MEETS JEDEC.95 MO-150, VARIATION = AH. REFER TO THIS SPECIFICATION FOR FURTHER DETAILS.
DM007.EDS: 28 PIN SSOP (10.2 x 5.3 x 1.75 mm)
Symbols
Dimensions
(mm)
MIN NOM MAX
A----- ----- 2.0
A10.05 ----- 0.25
A21.65 1.75 1.85
b0.22 0.30 0.38
c0.09 ----- 0.25
D9.90 10.20 10.50
e
E7.40 7.80 8.20
5.00 5.30 5.60
L0.55 0.75 0.95
θ
AA2 A1
14
1
15
28
E1 E
Θ
cL
GAUGE
PLANE
0.25
e
b
D
SEATING PLANE
-C-
0.10 C
REF: JEDEC.95, MO-150
E1
L11.25 REF
0.65 BSC
L1
0o4o8o
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IMPORTANT NOTICE
Wolfson Microelectronics plc (“Wolfson”) products and services are sold subject to Wolfson’s terms and conditions of sale,
delivery and payment supplied at the time of order acknowledgement.
Wolfson warrants performance of its products to the specifications in effect at the date of shipment. Wolfson reserves the
right to make changes to its products and specifications or to discontinue any product or service without notice. Customers
should therefore obtain the latest version of relevant information from Wolfson to verify that the information is current.
Testing and other quality control techniques are utilised to the extent Wolfson deems necessary to support its warranty.
Specific testing of all parameters of each device is not necessarily performed unless required by law or regulation.
In order to minimise risks associated with customer applications, the customer must use adequate design and operating
safeguards to minimise inherent or procedural hazards. Wolfson is not liable for applications assistance or customer
product design. The customer is solely responsible for its selection and use of Wolfson products. Wolfson is not liable for
such selection or use nor for use of any circuitry other than circuitry entirely embodied in a Wolfson product.
Wolfson’s products are not intended for use in life support systems, appliances, nuclear systems or systems where
malfunction can reasonably be expected to result in personal injury, death or severe property or environmental damage.
Any use of products by the customer for such purposes is at the customer’s own risk.
Wolfson does not grant any licence (express or implied) under any patent right, copyright, mask work right or other
intellectual property right of Wolfson covering or relating to any combination, machine, or process in which its products or
services might be or are used. Any provision or publication of any third party’s products or services does not constitute
Wolfson’s approval, licence, warranty or endorsement thereof. Any third party trade marks contained in this document
belong to the respective third party owner.
Reproduction of information from Wolfson datasheets is permissible only if reproduction is without alteration and is
accompanied by all associated copyright, proprietary and other notices (including this notice) and conditions. Wolfson is
not liable for any unauthorised alteration of such information or for any reliance placed thereon.
Any representations made, warranties given, and/or liabilities accepted by any person which differ from those contained in
this datasheet or in Wolfson’s standard terms and conditions of sale, delivery and payment are made, given and/or
accepted at that person’s own risk. Wolfson is not liable for any such representations, warranties or liabilities or for any
reliance placed thereon by any person.
ADDRESS
Wolfson Microelectronics plc
Westfield House
26 Westfield Road
Edinburgh
EH11 2QB
United Kingdom
Tel :: +44 (0)131 272 7000
Fax :: +44 (0)131 272 7001
Email :: sales@wolfsonmicro.com
