Wolfson WM8960GEFL Schematic

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WM8960

Stereo CODEC with 1W Stereo Class D Speaker Drivers and

Headphone Drivers for Portable Audio Applications

DESCRIPTION

The WM8960 is a low power, high quality stereo codec designed for portable digital audio applications.

Stereo class D speaker drivers provide 1W per channel into 8Ω loads with a 5V supply. Low leakage, excellent PSRR and pop/click suppression mechanisms also allow direct battery connection to the speaker supply. Flexible speaker boost settings allow speaker output power to be maximised while minimising other analogue supply currents.

A highly flexible input configuration for up to three stereo sources is integrated, with a complete microphone interface. External component requirements are drastically reduced as no separate microphone, speaker or headphone amplifiers are required. Advanced on-chip digital signal processing performs automatic level control for the microphone or line input.

Stereo 24-bit sigma-delta ADCs and DACs are used with low power over-sampling digital interpolation and decimation filters and a flexible digital audio interface.

The master clock can be input directly or generated internally by an onboard PLL, supporting most commonly-used clocking schemes.

The WM8960 operates at analogue supply voltages down to

2.7V, although the digital supplies can operate at voltages down to 1.71V to save power. The speaker supply can operate at up to 5.5V, providing 1W per channel into 8Ω loads. Unused functions can be disabled using software control to save power.

The WM8960 is supplied in a very small and thin 5x5mm QFN package, ideal for use in hand-held and portable systems.

FEATURES

• DAC SNR 98dB (‘A’ weighted), THD -84dB at 48kHz, 3.3V

• ADC SNR 94dB (‘A’ weighted), THD -82dB at 48kHz, 3.3V

• Pop and click suppression

• 3D Enhancement

• Stereo Class D Speaker Driver

• <0.1% THD with 1W per channel into 8Ω BTL speakers

• 70dB PSRR @217Hz

• 87% efficiency (1W output)

• Flexible internal switching clock

• On-chip Headphone Driver

• 40mW output power into 16Ω at 3.3V

• Capless mode support

• THD -75dB at 20mW, SNR 90dB with 16Ω load

• Microphone Interface

• Pseudo differential for high noise immunity

• Integrated low noise MICBIAS

• Programmable ALC / Limiter & Noise Gate

• Low Power Consumption

• 10mW headphone playback (2.7V / 1.8V supplies)

• 20mW record and playback (2.7V / 1.8V supplies)

• Low Supply Voltages

• Analogue 2.7V to 3.6V (Speaker supply up to 5.5V)

• Digital core and I/O: 1.71V to 3.6V

• On-chip PLL provides flexible clocking scheme

• Sample rates: 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48

• 5x5x0.9mm QFN package

APPLICATIONS

• Games consoles

• Portable media / DVD players

• Mobile multimedia

WOLFSON MICROELECTRONICS plc

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Preliminary Technical Data, September 2006, Rev 2.0

Copyright 2006 Wolfson Microelectronics plc

WM8960 Preliminary Technical Data

PIN CONFIGURATION

2526272829303132

MICBIAS

LINPUT3/JD2

LINPUT2

LINPUT1

RINPUT1

RINPUT2

RINPUT3/JD3

DCVDD

1

2

3

4

5

6

7

8

TOP VIEW

24

SPKGND1

SPK_LN

23

22

SPK_RP

21

SPKVDD2

20

SPKGND2

19

SPK_RN

18

SDIN

17

SCLK

161514131211109

ORDERING INFORMATION

ORDER CODE TEMPERATURE RANGE PACKAGE MOISTURE

WM8960GEFL/V

-40°C to +85°C 32-pin QFN (5x5x0.9mm)

(Pb-free)

WM8960GEFL/RV

-40°C to +85°C 32-pin QFN (5x5x0.9mm)

(Pb-free, Tape and reel)

Note:

Reel quantity = 3500

SENSITIVITY LEVEL

MSL3 260°C

PEAK SOLDERING

TEMPERATURE

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WM8960 Preliminary Technical Data

PIN DESCRIPTION

PIN NO NAME TYPE DESCRIPTION

1 MICBIAS

2 LINPUT3 / JD2

3 LINPUT2

4 LINPUT1

5 RINPUT1

6 RINPUT2

7 RINPUT3 / JD3

8 DCVDD

9 DGND

10 DBVDD

11 MCLK

12 BCLK

13 DACLRC

14 DACDAT

15 ADCLRC / GPIO1

16 ADCDAT

17 SCLK

18 SDIN

19 SPK_RN

20 SPKGND2

21 SPKVDD2

22 SPK_RP

23 SPK_LN

24 SPKGND1

25 SPK_LP

26 SPKVDD1

27 VMID

28 AGND

29 HP_R

30 OUT3

31 HP_L

32 AVDD

33 GND_PADDLE

Note:

1. It is recommended that the QFN ground paddle should be connected to analogue ground on the application PCB.

2. Refer to the application note WAN_0118 on “Guidelines on How to Use QFN Packages and Create Associated PCB Footprints”

Analogue Output Microphone bias

Analogue Input Left channel line input /

Left channel positive differential MIC input / Jack detect input pin

Analogue Input Left channel line input /

Left channel positive differential MIC input

Analogue Input Left channel single-ended MIC input /

Left channel negative differential MIC input

Analogue Input Right channel single-ended MIC input /

Right channel negative differential MIC input

Analogue Input Right channel line input /

Right channel positive differential MIC input

Analogue Input Right channel line input /

Right channel positive differential MIC input / Jack detect input pin

Supply Digital core supply

Supply Digital ground (Return path for both DCVDD and DBVDD)

Supply Digital buffer (I/O) supply

Digital Input Master clock

Digital Input / Output Audio interface bit clock

Digital Input / Output Audio interface DAC left / right clock

Digital Input DAC digital audio data

Digital Input / Output Audio interface ADC left / right clock / GPIO1 pin

Digital Output ADC digital audio data

Digital Input Control interface clock input

Digital Input/Output Control interface data input / 2-wire acknowledge output

Analogue Output Right speaker negative output

Supply Ground for speaker drivers 2

Supply Supply for speaker drivers 2

Analogue Output Right speaker positive output

Analogue Output Left speaker negative output

Supply Ground for speaker drivers 1

Analogue Output Left speaker positive output

Supply Supply for speaker drivers 1

Analogue Output Midrail voltage decoupling capacitor

Supply Analogue ground (Return path for AVDD)

Analogue Output Right output (Line or headphone)

Analogue Output Mono, left, right or buffered midrail output for capless mode

Analogue Output Left output (Line or headphone)

Supply Analogue supply

Die Paddle (Note 1)

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Preliminary Technical Data WM8960

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

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.

Supply voltages (excluding SPKVDD1 and SPKVDD2)

SPKVDD1, SPKVDD2

Voltage range digital inputs

Voltage range analogue inputs

Operating temperature range, TA

Storage temperature after soldering

Notes

1. Analogue, digital and speaker grounds must always be within 0.3V of each other.

2. All digital and analogue supplies are completely independent from each other (i.e. not internally connected).

3. DCVDD must be less than or equal to AVDD and DBVDD.

4. AVDD must be less than or equal to SPKVDD1 and SPKVDD2.

5. SPKVDD1 and SPKVDD2 must be high enough to support the peak output voltage when using DCGAIN and ACGAIN functions, to avoid output waveform clipping. Peak output voltage is AVDD*(DCGAIN+ACGAIN)/2.

of this device.

CONDITION MIN MAX

-0.3V +4.5V

-0.3V +7V

DGND -0.3V DBVDD +0.3V

AGND -0.3V AVDD +0.3V

-40°C +85°C

-65°C +150°C

RECOMMENDED OPERATING CONDITIONS

PARAMETER SYMBOL MIN TYP MAX UNIT

Digital supply range (Core)

Digital supply range (Buffer)

Analogue supplies range

Speaker supply range

Ground

DGND, AGND, SPKGND1,

DCVDD 1.71 3.6 V

DBVDD 1.71 3.6 V

AVDD 2.7 3.6 V

SPKVDD1, SPKVDD2 2.7 5.5 V

SPKGND2

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V

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WM8960 Preliminary Technical Data

ELECTRICAL CHARACTERISTICS

Test Conditions

DCVDD = 1.8V, DBVDD = 3.3V, AVDD = SPKVDD1 = SPKVDD2 = 3.3V, T 0dB, 24-bit audio data unless otherwise stated.

PARAMETER SYMBOL TEST CONDITIONS MIN TYP MAX UNIT

Analogue Inputs (LINPUT1, RINPUT1, LINPUT2, LINPUT3, RINPUT2, RINPUT3)

Full-scale Input Signal Level – note this changes in proportion to AVDD

V

INFS

L/RINPUT1

Single-ended or

differential MIC

L/RINPUT2/3

Differential MIC

L/RINPUT2/3

Boost or bypass path

L/RINPUT3

Boost + bypass path

Mic PGA equivalent input noise

0 to 20kHz, +30dB gain

R

INPUT1

+30dB PGA gain

Differential or single-

ended MIC configuration

0dB PGA gain

Differential or single-

ended MIC configuration

-17.25dB PGA gain

Input resistance (Note that input boost and

bypass path resistances will be seen in parallel with PGA input resistance when these paths are enabled)

Differential or single-

ended MIC configuration

R

INPUT2,

R

INPUT3

(Constant for all gains)

Differential MIC

configuration

R

INPUT2,

R

INPUT3

R

INPUT2,

R

INPUT3

R

INPUT2,

R

INPUT3

R

INPUT3

Max boost gain

L/RINPUT2/3 to boost

0dB boost gain

L/RINPUT2/3 to boost

Min boost gain

L/RINPUT2/3 to boost

Max bypass gain

L/RINPUT2/3 to bypass

R

INPUT3

Min bypass gain

L/RINPUT2/3 to bypass

Input capacitance

10 pF

MIC Programmable Gain Amplifier (PGA)

Programmable Gain Min

Programmable Gain Max

Programmable Gain Step Size

Mute Attenuation

Guaranteed monotonic 0.75 dB

-17.25 dB

30 dB

LMIC2B = 0 and

RMIC2B = 0

Selectable Input Gain Boost

Gain Boost Steps

Input from PGA

Input from L/RINPUT2 or

L/RINPUT3

= +25oC, 1kHz signal, fs = 48kHz, PGA gain =

A

1.0 0 Vrms dBV

0.5

-6

0.5

-6

0.5

-6

Vrms

dBV

Vrms

dBV

Vrms

dBV

150 uV

3 kΩ

49

87

85 kΩ

7.5 kΩ

13 kΩ

37

20 kΩ

224

90

0, 13, 20,

29, MUTE

-12, -9, -6, -3 0, 3, 6,

MUTE

kΩ

dB

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Preliminary Technical Data WM8960

Test Conditions

DCVDD = 1.8V, DBVDD = 3.3V, AVDD = SPKVDD1 = SPKVDD2 = 3.3V, T 0dB, 24-bit audio data unless otherwise stated.

PARAMETER SYMBOL TEST CONDITIONS MIN TYP MAX UNIT

Analogue Inputs (LINPUT2, RINPUT2, LINPUT3, RINPUT3) to ADC out

Signal to Noise Ratio (A-weighted)

Total Harmonic Distortion Plus Noise

Total Harmonic Distortion

ADC Channel Separation

Line Input / MIC Separation (Quiescent input to ADC via

boost; Output on ADC; 1kHz on L/RINPUT3 to HP out

via bypass path)

Boost / Bypass Separation (Quiescent L/RINPUT3 to HP

outputs via bypass)

Channel Matching

Headphone Outputs (HP_L, HP_R)

0dB Full scale output voltage

Mute attenuation

Channel Separation

DAC to Line-Out (HP_L, HP_R or OUT3 with 10kΩΩΩΩ / 50pF load)

Signal to Noise Ratio (A-weighted)

Total Harmonic Distortion Plus Noise

Total Harmonic Distortion

Channel Separation

SNR

THD+N

THD

SNR

THD+N

THD

AVDD = 3.3V 94

AVDD = 2.7V 93

-3dBFs input, AVDD = 3.3V

-3dBFs input, AVDD = 2.7V

-3dBFs input, AVDD = 3.3V

-3dBFs input, AVDD = 2.7V

1kHz full scale signal into

ADC via L/RINPUT1, MIC

amp (single-ended) and

boost

1kHz full scale signal into

ADC via L/RINPUT1/2,

MIC amp (pseudo-

differential) and boost

1kHz full scale signal into

ADC via L/RINPUT2 and

boost

1kHz full scale signal into

ADC via L/RINPUT3 and

boost

Single-ended MIC input

on L/RINPUT1

Differential MIC input

using L/RINPUT2

1kHz on LINPUT2 to ADC

via boost only

1kHz on LINPUT1 to ADC

via single-ended MIC

PGA & boost

1kHz signal 0.2 dB

AVDD/3.3 Vrms

1kHz, full scale signal 90 dB

L/RINPUT3 to headphone

outputs via bypass

AVDD=3.3V 99

AVDD=2.7V 98

AVDD=3.3V -85

AVDD=2.7V -90

AVDD=3.3V -87

AVDD=2.7V -92

1kHz full scale signal 110 dB

= +25oC, 1kHz signal, fs = 48kHz, PGA gain =

A

-86

0.005

-80

-89

TBD

90 dB

85 dB

dB

%

dB

%

dB

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WM8960 Preliminary Technical Data

Test Conditions

DCVDD = 1.8V, DBVDD = 3.3V, AVDD = SPKVDD1 = SPKVDD2 = 3.3V, T 0dB, 24-bit audio data unless otherwise stated.

PARAMETER SYMBOL TEST CONDITIONS MIN TYP MAX UNIT

Headphone Output (HP_L, HP_R, using capacitors unless otherwise specified)

Output Power per channel

Total Harmonic Distortion Plus Noise

P

O

THD+N

Output power is very closely correlated with THD; see below.

AVDD=2.7V, RL=32Ω

=5mW

P

O

AVDD=2.7V, RL=16Ω

=5mW

P

O

AVDD=3.3V, RL=32Ω,

=20mW

P

O

AVDD=3.3V, RL=16Ω,

P

=20mW

O

AVDD=2.7V, RL=32Ω

P

=5mW; Capless mode

O

AVDD=2.7V, RL=16Ω

=5mW; Capless mode

P

O

AVDD=3.3V, RL=32Ω,

=20mW; Capless mode

O

Signal to Noise Ratio (A-weighted)

SNR

AVDD=3.3V, R

=20mW; Capless mode

O

AVDD = 3.3V 92 99

AVDD = 2.7V 98

=16Ω,

L

AVDD = 3.3V; Capless mode

AVDD = 2.7V; Capless mode

Speaker Outputs (DAC to SPK_LP, SPK_LN, SPK_RP, SPK_RN with 8Ω bridge tied load)

Output Power

Total Harmonic Distortion Plus Noise

(DAC to speaker outputs)

P

Output power is very closely correlated with THD; see below

O

THD+N

PO =200mW, RL = 8Ω, SPKVDD1=SPKVDD2

=3.3V; AVDD=3.3V

PO =320mW, RL = 8Ω, SPKVDD1=SPKVDD2

=3.3V; AVDD=3.3V

PO =500mW, RL = 8Ω, SPKVDD1=SPKVDD2

=5V; AVDD=3.3V

P

=1W, RL = 8Ω,

O

SPKVDD1=SPKVDD2

=5V; AVDD=3.3V

= +25oC, 1kHz signal, fs = 48kHz, PGA gain =

A

0.013

-78

0.018

-75

0.025

-72

0.032

-70

0.013

-78

0.018

-75

0.025

-72

0.032

-70

92 99

98

0.01

%

-80

0.03

%

-72

0.011

%

-79

0.03

%

-71

%

dB

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Preliminary Technical Data WM8960

Test Conditions

DCVDD = 1.8V, DBVDD = 3.3V, AVDD = SPKVDD1 = SPKVDD2 = 3.3V, T

= +25oC, 1kHz signal, fs = 48kHz, PGA gain =

A

0dB, 24-bit audio data unless otherwise stated.

PARAMETER SYMBOL TEST CONDITIONS MIN TYP MAX UNIT

Total Harmonic Distortion Plus Noise

(LINPUT3 and RINPUT3 to speaker outputs)

THD+N

PO =200mW, RL = 8Ω, SPKVDD1=SPKVDD2

=3.3V; AVDD=3.3V

PO =320mW, RL = 8Ω, SPKVDD1=SPKVDD2

0.01

-80

0.03

-72

%

dB

%

dB

=3.3V; AVDD=3.3V

PO =500mW, RL = 8Ω, SPKVDD1=SPKVDD2

0.011

-79

%

dB

=5V; AVDD=3.3V

P

=1W, RL = 8Ω,

O

SPKVDD1=SPKVDD2

0.03

-71

%

dB

=5V; AVDD=3.3V

Signal to Noise Ratio (A-weighted) (DAC to speaker outputs)

SNR

SPKVDD1=SPKVDD2

=3.3V; AVDD=3.3V;

R

= 8Ω, ref=2.0Vrms

L

SPKVDD1=SPKVDD2

90 dB

92 dB

=5V; AVDD=3.3V;

= 8Ω, ref=2.8Vrms

R

L

Signal to Noise Ratio (A-weighted) (LINNPUT3 and RINPUT3 to

speaker outputs)

SNR

SPKVDD1=SPKVDD2

=3.3V; AVDD=3.3V;

R

= 8Ω, ref=2.0Vrms

L

SPKVDD1=SPKVDD2

90 dB

92 dB

=5V; AVDD=3.3V;

R

= 8Ω, ref=2.8Vrms

L

Speaker Supply Leakage current

I

SPKVDD

SPKVDD1=SPKVDD2

1 uA

=5V;

All other supplies

disconnected

SPKVDD1=SPKVDD2

1 uA

=5V;

All other supplies 0V

Power Supply Rejection Ratio (100mV ripple on SPKVDD1/SPKVDD2 @217Hz)

PSRR

DAC to speaker playback 80 dB

L/RINPUT3 to speaker

80 dB

playback

Analogue Reference Levels

Midrail Reference Voltage

Buffered Reference Voltage

VMID –3% AVDD/2 +3% V

VREF –3% AVDD/2 +3% V

Microphone Bias

Bias Voltage

V

MICBIA S

3mA load current

–5% 0.9×AVDD + 5% V

MBSEL=1

3mA load current

–5% 0.65×AVDD + 5% V

MBSEL=0

Bias Current Source

Output Noise Voltage

I

3 mA

MICBIAS

Vn 1K to 20kHz 15 nV/√Hz

Digital Input / Output

Input HIGH Level

Input LOW Level

Output HIGH Level

Output LOW Level

Input capacitance

Input leakage

V

0.7×DBVDD V

IH

V

0.3×DBVDD V

IL

V

I

OH

V

I

OL

=1mA 0.9×DBVDD V

OL

=-1mA 0.1×DBVDD V

OH

10 pF

-0.9 0.9 uA

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WM8960 Preliminary Technical Data

OUTPUT PGA GAIN

Output PGA Gains (Target gain, not measured)

10

0

0 20 40 60 80 100 120 140

-10

-20

-30

-40

Gain (dB)

-50

-60

-70

-80

Volume Register Setting

Figure 1 Output PGA Gains (LOUT1VOL, ROUT1VOL, SPKLVOL, SPKRVOL)

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Preliminary Technical Data WM8960

TYPICAL POWER CONSUMPTION

Mode

Off (Default state at power-up, no clocks)

Off (Thermal sensor disabled, no clocks)

Sleep (Thermal sensor enabled , VMID enabled using 250k VMID resistors )

Stereo line record @8kHz (No signal )

Stereo line record @16kHz (No signal )

Stereo line record @44.1kHz (No s ignal)

Stereo line record @44.1kHz, PLL enabled, MCLK=12MHz, no signal, master m ode

DAC Playback to 16Ohm headphones @44.1kHz, (no signal)

DAC Playback to 16Ohm headphones @44.1kHz, (white noise 1Vrms)

DAC Playback to 16Ohm headphones @44.1kHz, (1kHz tone 100mVrms)

DAC Playback to 16Ohm headphones @44.1kHz, PLL enabled, MCLK=12MHz (no signal), mas ter mode

DAC Playback to 8Ohm speak ers @44.1kHz (no signal)

DAC Playback to 8Ohm speak ers @44.1kHz (1kHz tone, full scale)

DAC Playback to 8Ohm sp eakers @44.1kHz (white noise, 1Vrms)

DAC Playback to mono speaker @44.1kHz (1kHz tone, full scale)

Stereo line record @8kHz and stereo DAC playback to line output (load = 10kO hm) @48kHz

Notes:

1. Power in the load is included.

AVDD SPKVDD DBVDD DCVDD IAVDD ISPKVDD IDBVDD IDCVDD Total (V) (V) (V) (V) (mA) (mA) (mA) (mA) (mW)

2.7 2.7 1.71 1.71 0.0314 0 0 0 0.085 3 3 1.8 1.8 0.0326 0 0 0 0.098

3.3 3.3 3.3 1.8 0. 033 0 0 0 0.109

3.6 5.5 3.6 3.6 0.0345 0 0 0 0.124

2.7 2.7 1.71 1.71 0.0086 0 0 0 0.023 3 3 1.8 1.8 0.0092 0 0 0 0.028

3.3 3.3 3.3 1.8 0.0096 0 0 0 0.032

3.6 5.5 3.6 3.6 0.0102 0 0 0 0.037

2.7 2.7 1.71 1.71 0.0537 0 0 0 0.145 3 3 1.8 1.8 0.0621 0 0 0 0.186

3.3 3.3 3.3 1.8 0.0674 0 0 0 0.222

3.6 5.5 3.6 3.6 0.0728 0 0 0 0.262

2.7 2.7 1.71 1.71 4.8 0 0.049 0.81 14.429 3 3 1.8 1.8 5.1 0 0.05 0.86 16.938

3.3 3.3 3.3 1.8 5.5 0 0.11 0.88 20.097

3.6 5.5 3.6 3.6 5.8 0 0.11 2.2 29.196

2.7 2.7 1.71 1.71 5.8 0 0.09 1.33 18.088 3 3 1.8 1.8 6.2 0 0.09 1.4 21.282

3.3 3.3 3.3 1.8 6.5 0 0.18 1.41 24.582

3.6 5.5 3.6 3.6 6.9 0 0.2 3. 6 38.520

2.7 2.7 1.71 1.71 5.93 0 0.02 3.1 21.346 3 3 1.8 1.8 6.3 0 0.02 3.3 24.876

3.3 3.3 3.3 1.8 6.8 0 0.04 3.3 28.512

3.6 5.5 3.6 3.6 7.1 0 0.05 8.7 57.060

2.7 2.7 1.71 1.71 6.4 0 0.25 3.5 23.693 3 3 1.8 1.8 6.9 0 0.26 3.78 27.972

3.3 3.3 3.3 1.8 7.5 0 0.5 3. 8 33.240

3.6 5.5 3.6 3.6 7.9 0 0.54 9.6 64.944

2.7 2.7 1.71 1.71 3.869 0 0.0029 3.38 16.231 3 3 1.8 1.8 4.35 0 0.0031 3.6 19.536

3.3 3.3 3.3 1.8 4.8 0 0.0098 3.78 22.676

3.6 5.5 3.6 3.6 5.33 0 0.0145 9.4 53.080

2.7 2.7 1.71 1.71 19.6 0 0.003 3.6 59.081 3 3 1.8 1.8 22.1 0 0.004 3.9 73.327

3.3 3.3 3.3 1.8 23.8 0 0.012 3.9 85.600

3.6 5.5 3.6 3.6 26 0 0.02 9. 9 129.312

2.7 2.7 1.71 1.71 7.8 0 0.003 3. 5 27.050 3 3 1.8 1.8 8.9 0 0.004 3.8 33.547

3.3 3.3 3.3 1.8 9.6 0 0.012 3.8 38.560

3.6 5.5 3.6 3.6 10.5 0 0.014 9.5 72.050

2.7 2.7 1.71 1.71 4.77 0 0.23 3.7 19.599 3 3 1.8 1.8 5.4 0 0.25 3.9 23.670

3.3 3.3 3.3 1.8 6.04 0 0.46 3.9 28.470

3.6 5.5 3.6 3.6 6.6 0 0.49 10.1 61.884

2.7 2.7 1.71 1.71 5.1 1.4 0.0032 3.57 23.660

3.3 3.3 3.3 1.8 6.3 1.79 0.01 3.84 33.642

3.3 5 3.3 1.8 6.3 2.9 0.01 3.8 42.163

3.3 5.5 3.3 1.8 6.9 3.2 0.0132 9.8 58.054

2.7 2.7 1.71 1.71 5.1 240 0.0032 3. 57 667.880

3.3 3.3 3.3 1.8 6.3 304 0.01 3.84 1030.935

3.3 5 3.3 1.8 6.3 450 0.01 3.8 2277.663

3.3 5.5 3.3 1.8 6.9 486 0.0132 9.8 2713.454

2.7 2.7 1.71 1.71 5.1 48 0.0032 3.57 149.480

3.3 3.3 3.3 1.8 6.3 56 0.01 3.84 212.535

3.3 5 3.3 1.8 6.3 82 0. 01 3.8 437.663

3.3 5.5 3.3 1.8 6.9 90 0.0132 9.8 535.454

2.7 2.7 1.71 1.71 3 125 0.0034 3.63 351.813

3.3 3.3 3.3 1.8 3.77 154 0.0126 3.89 527.685

3.3 5 3.3 1.8 3.79 229 0.0126 3.7 1164.209

3.6 5.5 3.6 3.6 4.2 250 0.0163 9.7 1425.099

2.7 2.7 1.71 1.71 8.64 0 0.009 4.5 31.038 3 3 1.8 1.8 9.44 0 0.01 4.7 36.798

3.3 3.3 3.3 1.8 10. 24 0 0.025 4.7 42.335

3.6 5.5 3.6 3.6 11.032 0 0.03 11.732 82. 058

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WM8960 Preliminary Technical Data

SIGNAL TIMING REQUIREMENTS

SYSTEM CLOCK TIMING

MCLK

t

MCLKY

Figure 2 System Clock Timing Requirements

Test Conditions

DCVDD=1.8V, DBVDD=AVDD=SPKVDD1=SPKVDD2=3.3V, DGND=AGND=SPKGND1=SPKGND2=0V, T

PARAMETER SYMBOL CONDITIONS MIN TYP MAX UNIT

System Clock Timing Information

MCLK cycle time

MCLK duty cycle

T

MCLKY

T

60:40 40:60

MCLKD S

33.33 ns

AUDIO INTERFACE TIMING – MASTER MODE

= +25oC

A

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Figure 2 Digital Audio Data Timing – Master Mode (see Control Interface)

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Preliminary Technical Data WM8960

Test Conditions

DCVDD=1.8V, DBVDD=AVDD=SPKVDD1=SPKVDD2=3.3V, DGND=AGND=SPKGND1=SPKGND2=0V, T

=+25oC, Slave

A

Mode, fs=48kHz, MCLK=256fs, 24-bit data, unless otherwise stated.

PARAMETER SYMBOL MIN TYP MAX UNIT

Audio Data Input Timing Information

ADCLRC/DACLRC propagation delay from BCLK falling edge

ADCDAT propagation delay from BCLK falling edge

DACDAT setup time to BCLK rising edge

DACDAT hold time from BCLK rising edge

t

10 ns

DL

t

10 ns

DDA

t

10 ns

DST

t

10 ns

DHT

AUDIO INTERFACE TIMING – SLAVE MODE

Figure 3 Digital Audio Data Timing – Slave Mode

Test Conditions

DCVDD=1.8V, DBVDD=AVDD=SPKVDD1=SPKVDD2=3.3V, DGND=AGND=SPKGND1=SPKGND2=0V, T

=+25oC, Slave

A

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

BCLK pulse width high

BCLK pulse width low

ADCLRC/DACLRC set-up time to BCLK rising edge

ADCLRC/DACLRC hold time from BCLK rising edge

DACDAT hold time from BCLK rising edge

ADCDAT propagation delay from BCLK falling edge

DACDAT set-up time to BCLK rising edge

t

50 ns

BCY

t

20 ns

BCH

t

20 ns

BCL

t

10 ns

LRSU

t

10 ns

LRH

t

10 ns

DH

t

10 ns

DD

t

10 ns

DS

Note:

BCLK period should always be greater than or equal to MCLK period.

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WM8960 Preliminary Technical Data

CONTROL INTERFACE TIMING – 2-WIRE MODE

SDIN

SCLK

t

3

t

6

t

1

t

9

t

5

t

2

t

7

t

3

t

4

Figure 4 Control Interface Timing – 2-Wire Serial Control Mode

Test Conditions

DCVDD=1.8V, DBVDD=AVDD=SPKVDD1=SPKVDD2=3.3V, DGND=AGND=SPKGND1=SPKGND2=0V, T Mode, fs=48kHz, MCLK = 256fs, 24-bit data, unless otherwise stated.

PARAMETER SYMBOL MIN TYP MAX UNIT

Program Register Input Information

SCLK Frequency

SCLK Low Pulse-Width

SCLK High Pulse-Width

Hold Time (Start Condition)

Setup Time (Start Condition)

Data Setup Time

SDIN, SCLK Rise Time

SDIN, SCLK Fall Time

Setup Time (Stop Condition)

Data Hold Time

Pulse width of spikes that will be suppressed

526 kHz

t

1.3 us

1

t

600 ns

2

t

600 ns

3

t

600 ns

4

t

100 ns

5

t

300 ns

6

t

300 ns

7

t

600 ns

8

t

900 ns

9

t

0 5 ns

ps

=+25oC, Slave

A

t

8

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Preliminary Technical Data WM8960

INTERNAL POWER ON RESET CIRCUIT

Figure 5 Internal Power on Reset Circuit Schematic

The WM8960 includes an internal Power-On-Reset Circuit, as shown in Figure 5, which is used to reset the digital logic into a default state after power up. The POR circuit is powered from AVDD and monitors DCVDD. It asserts PORB low if AVDD or DCVDD is below a minimum threshold.

Figure 6 Typical Power up sequence where AVDD is Powered before DCVDD

Figure 6 shows a typical power-up sequence where AVDD comes up first. When AVDD goes above the minimum threshold, V low and the chip is held in reset. In this condition, all writes to the control interface are ignored. Now AVDD is at full supply level. Next DCVDD rises to V registers are in their default state and writes to the control interface may take place.

On power down, where AVDD falls first, PORB is asserted low whenever AVDD drops below the minimum threshold V

, there is enough voltage for the circuit to guarantee PORB is asserted

pora

.

pora_off

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and PORB is released high and all

pord_on

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WM8960 Preliminary Technical Data

Figure 7 Typical Power up Sequence where DCVDD is Powered before AVDD

Figure 7 shows a typical power-up sequence where DCVDD comes up first. First it is assumed that DCVDD is already up to specified operating voltage. When AVDD goes above the minimum threshold, V chip is held in reset. In this condition, all writes to the control interface are ignored. When AVDD rises to V

pora_on

interface may take place.

, there is enough voltage for the circuit to guarantee PORB is asserted low and the

pora

, PORB is released high and all registers are in their default state and writes to the control

On power down, where DCVDD falls first, PORB is asserted low whenever DCVDD drops below the minimum threshold V

pord_off

.

SYMBOL MIN TYP MAX UNIT

V

pora_on

V

pora_off

V

pord_on

V

pord_off

pora

0.4 0.6 0.8 V

0.9 1.2 1.6 V

0.4 0.6 0.8 V

0.5 0.7 0.9 V

0.4 0.6 0.8 V

Table 1 Typical POR Operation (typical values, not tested)

Notes:

1. If AVDD and DCVDD suffer a brown-out (i.e. drop below the minimum recommended operating level but do not go below V

pora_off

or V

) then the chip will not reset and will resume normal

pord_off

operation when the voltage is back to the recommended level again.

2. The chip will enter reset at power down when AVDD or DCVDD falls below V

pora_off

or V

pord_off

This may be important if the supply is turned on and off frequently by a power management system.

3. The minimum t

period is maintained even if DCVDD and AVDD have zero rise time. This

por

specification is guaranteed by design rather than test.

.

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Preliminary Technical Data WM8960

DEVICE DESCRIPTION

INTRODUCTION

The WM8960 is a low power audio codec offering a combination of high quality audio, advanced features, low power and small size. These characteristics make it ideal for portable digital audio applications with stereo speaker and headphone outputs such as games consoles, portable media players and multimedia phones.

Stereo class D speaker drivers can provide >1W per channel into 8Ω loads. BTL configuration provides high power output and excellent PSRR. Low leakage and pop/click suppression mechanisms allow direct battery connection, reducing component count and power consumption in portable battery-powered applications. Highly flexible speaker boost settings provide fully internal level-shifting of analogue output signals, allowing speaker output power to be maximised while minimising other analogue supply currents, and requiring no additional components.

A flexible input configuration includes support for two stereo microphone interfaces (single-ended or pseudo-differential) and additional stereo line inputs. Up to three stereo analogue input sources are available, removing the need for external analogue switches in many applications. Boost amplifiers are available for additional gain on the microphone inputs and a programmable gain amplifier with a mixed signal automatic level control (ALC) keeps the recording volume constant.

The stereo ADC and DAC are of hi-fi quality using a 24-bit, low-order oversampling architecture to deliver optimum performance. A flexible clocking arrangement supports mixed ADC and DAC sample rates.

The DAC output signal can be mixed with analogue input signals from the line inputs or bypass paths. This mix is available on speaker and headphone/line outputs.

The WM8960 has a configurable digital audio interface where ADC data can be read and digital audio playback data fed to the DAC. It supports a number of audio data formats including I Mode (a burst mode in which frame sync plus two data packed words are transmitted), MSB-First, left justified and MSB-First, right justified, and can operate in master or slave modes. In PCM mode A-law and µ-law companding is supported.

The SYSCLK (system clock) provides clocking for the ADCs, DACs, DSP core, class D outputs and the digital audio interface. SYSCLK can be derived directly from the MCLK pin or via an integrated PLL, providing flexibility to support a wide range of clocking schemes. All MCLK frequencies typically used in portable systems are supported for sample rates between 8kHz and 48kHz. A flexible switching clock for the class D speaker drivers (synchronous with the audio DSP clocks for best performance) is also derived from SYSCLK.

To allow full software control over all its features, the W M8960 uses a 2 wire control interface. It is fully compatible and an ideal partner for a wide range of industry standard microprocessors, controllers and DSPs. Unused circuitry can be disabled via software to save power, while low leakage currents extend standby and off time in portable battery-powered applications.

2

S, DSP

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WM8960 Preliminary Technical Data

INPUT SIGNAL PATH

The WM8960 has three flexible stereo analogue input channels which can be configured as line inputs, differential microphone inputs or single-ended microphone inputs. Line inputs and microphone PGA outputs can be routed to the hi-fi ADCs or directly to the output mixers via a bypass path.

MICROPHONE INPUTS

Differential microphones can be connected between LINPUT1 and LINPUT2 or LINPUT3, and between RINPUT1 and RINPUT2 or RINPUT3. Alternatively single-ended microphones can be connected to LINPUT1 or RINPUT1.

In single-ended microphone input configuration the microphone signal should be input to LINPUT1 or RINPUT1 and the internal non-inverting input of the input PGA should be switched to VMID.

In differential mode the larger signal should be input to LINPUT2 or LINPUT3 on the left channel, or RINPUT2 or RINPUT3 on the right channel. The smaller (e.g. noisy ground connection) should be input to LINPUT1 or RINPUT1.

The gain of the microphone PGAs can be controlled directly via software, or using the ALC / Limiter.

The inputs LINPUT2, RINPUT2, LINPUT3 and RINPUT3 should not be connected to the boost mixer or bypass path while operating as the non-inverting input in differential microphone configuration.

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Preliminary Technical Data WM8960

Figure 8 Microphone Input PGA Circuit

The input PGAs and boost mixers are enabled by the AINL and AINR register bits. The microphone PGAs can be also be disabled independently of the boost mixer to save power, using LMIC and RMIC register bits.

REGISTER

ADDRESS

R25 (19h) Power

Management (1)

R47 (2Fh) Power

Management (3)

Table 2 Input PGA and Boost Enable Register Settings

The input PGAs can be configured as differential inputs, using LINPUT1/LINPUT2 or LINPUT1/LINPUT3, and RINPUT1/RINPUT2 or RINPUT1/RINPUT3. The input impedance to these non-inverting inputs is constant in this configuration. Differential configuration is controlled by LMP2, LMP3, RMP2 and RMP3 as shown in Table 3.

When single-ended configuration is selected, the non-inverting input of the PGA is connected to VMID.

BIT LABEL DEFAULT DESCRIPTION

5 AINL 0 Left channel input PGA and boost

stage enable 0 = PGA disabled, boost disabled 1 = PGA enabled (if LMIC = 1),

boost enabled

4 AINR 0 Right channel input PGA and boost

stage enable 0 = PGA disabled, boost disabled 1 = PGA enabled (if LMIC = 1),

boost enabled

5 LMIC 0 Left channel input PGA enable

0 = PGA disabled 1 = PGA enabled (if AINL = 1)

4 RMIC 0 Right channel input PGA enable

0 = PGA disabled 1 = PGA enabled (if AINR = 1)

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WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R32 (20h) ADCL Input

Signal Path

R33 (21h) ADCR Input

Signal Path

Table 3 Input PGA Control

BIT LABEL DEFAULT DESCRIPTION

3 LMIC2B 0 Connect Left Input PGA to Left Input

6 LMP2 0 Connect LINPUT2 to non-inverting input of

7 LMP3 0 Connect LINPUT3 to non-inverting input of

8 LMN1 1 Connect LINPUT1 to inverting input of Left

3 RMIC2B 0 Connect Right Input PGA to Right Input

6 RMP2 0 Connect RINPUT2 to non-inverting input

7 RMP3 0 Connect RINPUT3 to non-inverting input

8 RMN1 1 Connect RINPUT1 to inverting input of

Boost mixer 0 = Not connected 1 = Connected

Left Input PGA 0 = LINPUT2 not connected to PGA 1 = LINPUT2 connected to PGA (Constant

input impedance)

Left Input PGA 0 = LINPUT3 not connected to PGA 1 = LINPUT3 connected to PGA (Constant

input impedance)

Input PGA 0 = LINPUT1 not connected to PGA 1 = LINPUT1 connected to PGA

Boost mixer 0 = Not connected 1 = Connected

of Right Input PGA 0 = RINPUT2 not connected to PGA 1 = RINPUT2 connected to PGA

(Constant input impedance)

of Right Input PGA 0 = RINPUT3 not connected to PGA 1 = RINPUT3 connected to PGA

(Constant input impedance)

Right Input PGA 0 = RINPUT1 not connected to PGA 1 = RINPUT1 connected to PGA

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INPUT PGA VOLUME CONTROLS

The input PGAs have a gain range from -17.25dB to +30dB in 0.75dB steps. The gains from the inverting inputs (LINPUT1 and RINPUT1) to the PGA outputs and from the non-inverting inputs (LINPUT2/RINPUT2 and LINPUT3/RINPUT3) to the PGA output are always common in differential configuration and controlled by the register bits LINVOL[5:0] and RINVOL[5:0].

When the Automatic Level Control (ALC) is enabled the input PGA gains are controlled automatically and the LINVOL and RINVOL bits should not be used.

The left and right input PGAs can be independently muted using the LINMUTE and RINMUTE register bits.

To allow simultaneous volume updates of left and right channels, PGA gains are not altered until a 1 is written to the IPVU bit.

To prevent "zipper noise", a zero-cross function is provided, so that when enabled, volume updates will not take place until a zero-crossing is detected. In the event of a long period without zerocrossings, a timeout function is available. When this function is enabled (using the TOEN register bit), the volume will update automatically after a timeout. The timeout period is set by TOCLKSEL. Note that SYSCLK must be running to use this function.

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Preliminary Technical Data WM8960

REGISTER

ADDRESS

R0 (00h) Left Channel PGA

R1 (01h) Right Channel PGA

R23 (17h) Additional

Control (1)

Table 4 Input PGA Volume Control

See "Volume Updates" for more information on volume update bits, zero cross and timeout operation.

BIT LABEL DEFAULT DESCRIPTION

8

7

6

5:0

8 IPVU N/A Input PGA Volume Update

7 RINMUTE 1 Right Input PGA Analogue Mute

6 RIZC 0 Right Input PGA Zero Cross

5:0 RINVOL

0 TOEN 0 Timeout Enable (Also enables jack

1 TOCLKSEL 0 Slow Clock Selection (Used for

IPVU N/A Input PGA Volume Update

Writing a 1 to this bit will cause left and right input PGA volumes to be updated (LINVOL and RINVOL)

LINMUTE 1 Left Input PGA Analogue Mute

1 = Enable Mute 0 = Disable Mute Note: IPVU must be set to un-mute.

LIZC 0 Left Input PGA Zero Cross Detector

1 = Change gain on zero cross only 0 = Change gain immediately

LINVOL [5:0]

[5:0]

010111 ( 0dB )

Left Input PGA Volume Control 111111 = +30dB 111110 = +29.25dB . . 0.75dB steps down to 000000 = -17.25dB

Writing a 1 to this bit will cause left and right input PGA volumes to be updated (LINVOL and RINVOL)

1 = Enable Mute 0 = Disable Mute Note: IPVU must be set to un-mute.

Detector 1 = Change gain on zero cross only 0 = Change gain immediately

Right Input PGA Volume Control 111111 = +30dB 111110 = +29.25dB . . 0.75dB steps down to 000000 = -17.25dB

detect debounce clock) 0 = Timeout disabled 1 = Timeout enabled

volume update timeouts and for jack detect debounce)

0 = SYSCLK / 2 Response)

1 = SYSCLK / 2 Response)

21

(Slower

19

(Faster

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LINE INPUTS

Two pairs of stereo line inputs (LINPUT2 / RINPUT2 and LINPUT3 / RINPUT3) are available as analogue inputs into the ADC path. LINPUT3 and RINPUT3 can also be input directly to the output mixers via the bypass paths.

See "Output Signal Path" for more information on the bypass paths.

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WM8960 Preliminary Technical Data

INPUT BOOST

The input path to the ADCs is via a boost stage, which can mix signals from the microphone PGAs and the line inputs.

The boost stage can provide up to +29dB additional gain from the microphone PGA output to the ADC input, providing a total maximum available analogue gain of +59dB from microphone to ADC. The microphone PGA path to the boost mixer is muted using LINMUTE and RINMUTE as shown in Table 4. Microphone PGA to boost gain settings are shown in Table 5.

REGISTER

ADDRESS

R32 (20h) ADCL Signal

path

R33 (21h) ADCR Signal

path

Table 5 Microphone PGA Boost Control

For line inputs, -12dB to +6dB gain is available on the boost mixer, with mute control, as shown in Table 6.

REGISTER

ADDRESS

R43 (2Bh) Input Boost

Mixer 1

R44 (2Ch) Input Boost

Mixer 2

Table 6 Line Input Boost Control

When all three input paths to the boost mixer are disabled, the boost mixer will automatically be muted.

BIT LABEL DEFAULT DESCRIPTION

5:4 LMICBOOST

[1:0]

5:4 RMICBOOST

[1:0]

BIT LABEL DEFAULT DESCRIPTION

6:4 LIN3BOOST

[2:0]

3:1 LIN2BOOST

[2:0]

6:4 RIN3BOOST

[2:0]

3:1 RIN2BOOST

[2:0]

00 Left Channel Input PGA Boost Gain

00 = +0dB 01 = +13dB 10 = +20dB 11 = +29dB

00 Right Channel Input PGA Boost

Gain 00 = +0dB 01 = +13dB 10 = +20dB 11 = +29dB

000 LINPUT3 to Boost Mixer gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

000 LINPUT2 to Boost Mixer gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

000 RINPUT3 to Boost Mixer gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

000 RINPUT2 to Boost Mixer gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

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Preliminary Technical Data WM8960

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 MBSEL register bit. W hen MBSEL=0, MICBIAS=0.9*AVDD and when MBSEL=1, MICBIAS=0.65*AVDD. The output can be enabled or disabled using the MICB control bit.

REGISTER

ADDRESS

R25 (19h) Power

management (1)

R48 (30h) Additional Control

(4)

Table 7 Microphone Bias Control

The internal MICBIAS circuitry is shown in Figure 9. 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.

BIT LABEL DEFAULT DESCRIPTION

1 MICB 0 Microphone Bias Enable

0 = OFF (high impedance output) 1 = ON

0 MBSEL 0 Microphone Bias Voltage Control

0 = 0.9 * AVDD 1 = 0.65 * AVDD

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Figure 9 Microphone Bias Schematic

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WM8960 Preliminary Technical Data

EXAMPLE INPUT CONFIGURATIONS

Some example input configurations are shown below.

Single-ended MIC configuration on left channel.

LINPUT2 and LINPUT3 unused

Pseudo-differential MIC configuration on left channel using LINPUT1 as ground connection and LINPUT2 as signal input.

LINPUT3 unused.

Single-ended MIC configuration on left channel.

LINPUT2 used as additional input to boost stage.

LINPUT3 unused.

Single-ended MIC configuration on left channel.

LINPUT3 used as input to bypass path.

LINPUT2 unused.

Figure 10 Example Microphone Input Configurations (See also "Recommended External Components")

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Preliminary Technical Data WM8960

ANALOGUE TO DIGITAL CONVERTER (ADC)

The WM8960 uses stereo 24-bit, 64x oversampled sigma-delta ADCs. The use of multi-bit feedback and high oversampling rates reduce 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 1.0V Any voltage greater than full scale may overload the ADC and cause distortion.

The ADCs are enabled by the ADCL/R register bit.

REGISTER

ADDRESS

R25 (19h) Power

management (2)

Table 8 ADC Enable Control

The polarity of the output signal can be changed under software control using the ADCPOL[1:0] register bits. The DATSEL bits are used to select which channel is used for the left and right ADC data.

REGISTER

ADDRESS

R5 (05h) ADC and DAC

Control (1)

R23 (17h) Additional Control

(1)

Table 9 ADC Control

BIT LABEL DEFAULT DESCRIPTION

3 ADCL 0 Enable ADC left channel:

0 = ADC disabled 1 = ADC enabled

2 ADCR 0 Enable ADC right channel:

0 = ADC disabled 1 = ADC enabled

BIT LABEL DEFAULT DESCRIPTION

6:5 ADCPOL[1:0] 00 ADC polarity control:

00 = Polarity not inverted 01 = ADC L inverted 10 = ADC R inverted 11 = ADC L and R inverted

3:2

DATSEL [1:0]

00 ADC Data Output Select

00: left data = left ADC;

right data =right ADC

01: left data = left ADC;

right data = left ADC

10: left data = right ADC;

right data =right ADC

11: left data = right ADC;

right data = left ADC

rms

.

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DIGITAL ADC VOLUME CONTROL

The output of the ADCs can be digitally amplified or attenuated over a range from –97dB to +30dB in

0.5dB steps. The volume of each channel can be controlled separately. The gain for a given eight-bit

code X is given by:

0.5 × (X-195) dB for 1 ≤ X ≤ 255; MUTE for X = 0

The ADCVU control bit controls the loading of digital volume control data. When ADCVU is set to 0, the LADCVOL or RADCVOL control data will be loaded into the respective control register, but will not actually change the digital gain setting. Both left and right gain settings are updated when ADCVU is set to 1. This makes it possible to update the gain of both channels simultaneously.

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WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R21 (15h) Left ADC

Digital Volume

R22 (16h) Right ADC

Digital Volume

Table 10 ADC Digital Volume Control

BIT LABEL DEFAULT DESCRIPTION

7:0 LADCVOL

[7:0]

8 ADCVU 0 ADC Volume Update

7:0 RADCVOL

[7:0]

8 ADCVU 0 ADC Volume Update

11000011 ( 0dB )

Left ADC Digital Volume Control 0000 0000 = Digital Mute 0000 0001 = -97dB 0000 0010 = -96.5dB ... 0.5dB steps up to 1111 1111 = +30dB

0 = Store LADCVOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = LADCVOL, right = intermediate latch)

Right ADC Digital Volume Control 0000 0000 = Digital Mute 0000 0001 = -97dB 0000 0010 = -96.5dB ... 0.5dB steps up to 1111 1111 = +30dB

0 = Store RADCVOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = intermediate latch, right = RADCVOL)

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.

HIGH PASS FILTER

A digital high pass filter is applied by default to the ADC path to remove DC offsets. This filter can be disabled using the ADCHPD register bit.

REGISTER

ADDRESS

R5 (05h) ADC and DAC

Control (1)

Table 11 ADC High Pass Filter

The high pass filter characteristics are shown in the Digital Filter Characteristics section.

BIT LABEL DEFAULT DESCRIPTION

0 ADCHPD 0 ADC High Pass Filter Disable

0 = Enable high pass filter on left and right channels

1 = Disable high pass filter on left and right channels

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Preliminary Technical Data WM8960

AUTOMATIC LEVEL CONTROL (ALC)

The WM8960 has an automatic level control that aims to keep a constant recording volume irrespective of the input signal level. This is achieved by continuously adjusting the PGA gain so that the signal level at the ADC input remains constant. A digital peak detector monitors the ADC output and changes the PGA gain if necessary. Note that when the ALC function is enabled, the settings of registers 0 and 1 (LINVOL, IPVU, LIZC, LINMUTE, RINVOL, RIZC and RINMUTE) are ignored.

input

signal

PGA

gain

signal

after ALC

hold time

decay

time

attack

time

ALC

target

level

Figure 11 ALC Operation

The ALC function is enabled using the ALCSEL control bits. When enabled, the recording volume can be programmed between –1.5dB and –22.5dB (relative to ADC full scale) using the ALCL register bits. An upper limit for the PGA gain can be imposed by setting the MAXGAIN control bits.

HLD, DCY and ATK control the hold, decay and attack times, respectively:

Hold time is the time delay between the peak level detected being below target and the PGA gain beginning to ramp up. It can be programmed in power-of-two (2

n

) steps, e.g. 2.67ms, 5.33ms,

10.67ms etc. up to 43.7s. Alternatively, the hold time can also be set to zero. The hold time only

applies to gain ramp-up, there is no delay before ramping the gain down when the signal level is above target.

Decay (Gain Ramp-Up) Time is the time that it takes for the PGA gain to ramp up across 90% of its range (e.g. from –15B up to 27.75dB). The time it takes for the recording level to return to its target value therefore depends on both the decay time and on the gain adjustment required. If the gain adjustment is small, it will be shorter than the decay time. The decay time can be programmed in power-of-two (2

n

) steps, from 24ms, 48ms, 96ms, etc. to 24.58s.

Attack (Gain Ramp-Down) Time is the time that it takes for the PGA gain to ramp down across 90% of its range (e.g. from 27.75dB down to -15B gain). The time it takes for the recording level to return to its target value therefore depends on both the attack time and on the gain adjustment required. If the gain adjustment is small, it will be shorter than the attack time. The attack time can be programmed in power-of-two (2

n

) steps, from 6ms, 12ms, 24ms, etc. to 6.14s.

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When operating in stereo, the peak detector takes the maximum of left and right channel peak values, and any new gain setting is applied to both left and right PGAs, so that the stereo image is preserved. However, the ALC function can also be enabled on one channel only. In this case, only one PGA is controlled by the ALC mechanism, while the other channel runs independently with its PGA gain set through the control register.

When one ADC channel is unused, the peak detector disregards that channel.

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REGISTER

ADDRESS

R17 (11h) ALC Control

(1)

R18 (12h) ALC Control

(2)

R19 (13h) ALC Control

(3)

BIT LABEL DEFAULT DESCRIPTION

8:7 ALCSEL

[1:0]

6:4 MAXGAIN

[2:0]

3:0 ALCL

[3:0]

6:4 MINGAIN

[2:0]

3:0 HLD

[3:0]

8 ALCMODE 0 Determines the ALC mode of operation:

7:4 DCY

[3:0]

3:0 ATK

[3:0]

00 (OFF)

111 (+30dB)

1011 (-12dB)

000 Set Minimum Gain of PGA

0000 (0ms)

0011 (192ms)

0010 (24ms)

ALC Function Select 00 = ALC off (PGA gain set by register) 01 = Right channel only 10 = Left channel only 11 = Stereo (PGA registers unused)

Note: ensure that LINVOL and RINVOL settings (reg. 0 and 1) are the same before entering this mode.

Set Maximum Gain of PGA 111 : +30dB 110 : +24dB ….(-6dB steps) 001 : -6dB 000 : -12dB

ALC Target (Sets signal level at ADC input)

0000 = -22.5dB FS 0001 = -21.0dB FS … (1.5dB steps) 1101 = -3.0dB FS 1110 = -1.5dB FS 1111 = -1.5dB FS

000 = -17.25dB 001 = -11.25dB 010 = -5.25dB 011 = +0.75dB 100 = +6.75dB 101 = +12.75dB 110 = +18.75dB 111 = +24.75dB

ALC hold time before gain is increased. 0000 = 0ms 0001 = 2.67ms 0010 = 5.33ms … (time doubles with every step) 1111 = 43.691s

0 = ALC mode 1 = Limiter mode

ALC decay (gain ramp-up) time 0000 = 24ms 0001 = 48ms 0010 = 96ms … (time doubles with every step) 1010 or higher = 24.58s

ALC attack (gain ramp-down) time 0000 = 6ms 0001 = 12ms 0010 = 24ms … (time doubles with every step) 1010 or higher = 6.14s

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R27 (1Bh) Additional

Control (3)

Table 12 ALC Control

2:0 ADC_ALC_SR

[2:0]

000 ALC Sample Rate

000 = 44.1k / 48k 001 = 32k 010 = 22.05k / 24k 011 = 16k 100 = 11.25k / 12k 101 = 8k 110 and 111 = Reserved

ALC SAMPLE RATE CONTROL

The register bits ADC_ALC_SR must be set correctly to ensure that the ALC attack, decay and hold times are correct for the chosen sample rate as shown in Table 12.

PEAK LIMITER

To prevent clipping when a large signal occurs just after a period of quiet, the ALC circuit includes a limiter function. If the ADC input signal exceeds 87.5% of full scale (–1.16dB), the PGA gain is ramped down at the maximum attack rate (as when ATK = 0000), until the signal level falls below

87.5% of full scale. This function is automatically enabled whenever the ALC is enabled.

Note:

If ATK = 0000, then the limiter makes no difference to the operation of the ALC. It is designed to prevent clipping when long attack times are used.

NOISE GATE

When the signal is very quiet and consists mainly of noise, the ALC function may cause “noise pumping”, i.e. loud hissing noise during silence periods. The WM8960 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 [dB] < NGTH [dB] + PGA gain [dB] + Mic Boost gain [dB]

This is equivalent to:

• Signal level at input pin [dB] < NGTH [dB]

The PGA gain will then be held constant (preventing it from ramping up as it normally would when the signal is quiet).

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The table below summarises the noise gate control register. The NGTH control bits set the noise gate threshold with respect to the ADC full-scale range. The threshold is adjusted in 1.5dB steps. Levels at the extremes of the range may cause inappropriate operation, so care should be taken with set–up of the function. Note that the noise gate only works in conjunction with the ALC function, and always operates on the same channel(s) as the ALC (left, right, both, or none).

REGISTER

ADDRESS

R20 (14h) Noise Gate Control

Table 13 Noise Gate Control

BIT LABEL DEFAULT DESCRIPTION

NGTH

7:3

[4:0]

NGAT 0 Noise gate function enable

0

00000 Noise gate threshold

00000 -76.5dBfs 00001 -75dBfs … 1.5 dB steps 11110 -31.5dBfs 11111 -30dBfs

0 = disable 1 = enable

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OUTPUT SIGNAL PATH

The hi-fi DACs and DAC digital filters are enabled by register bits DACL and DACR. 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 WM8960, irrespective of whether the DACs are enabled or not.

The W M8960 DACs receive digital input data on the DACDAT pin. The digital filter block processes the data to provide the following functions:

§ Digital volume control with soft mute and soft un-mute

§ Mono mix

§ 3D stereo enhancement

§ De-emphasis

§ Sigma-delta modulation

High performance sigma-delta 24-bit audio DAC converts the digital data into an analogue signal.

The analogue outputs from the DACs can then be mixed with the analogue line inputs and the ADC analogue inputs. This mix is fed to the output drivers for headphone or speaker output. OUT3 can provide a mono mix of left and right mixers or a pseudo-ground for capless headphone drive.

DIGITAL PLAYBACK (DAC) PATH

Digital data is passed to the WM8960 via the flexible audio interface to the hi-fi DACs. The DACs are enabled by the DACL and DACR register bits.

REGISTER

ADDRESS

R26 (1Ah) Power

Management (2)

Table 14 DAC Enable Control

BIT LABEL DEFAULT DESCRIPTION

8 DACL 0 Left Channel DAC Enable

0 = DAC disabled 1 = DAC enabled

7 DACR 0 Right Channel DAC Enable

0 = DAC disabled 1 = DAC enabled

DIGITAL DAC VOLUME CONTROL

The signal volume from each DAC can be controlled digitally, in the same way as the ADC volume (see Digital ADC Volume Control). The gain and attenuation range is –127dB to 0dB in 0.5dB steps. The level of attenuation for an eight-bit code X is given by:

0.5 × (X-255) dB for 1 ≤ X ≤ 255; MUTE for X = 0

The DACVU control bit controls the loading of digital volume control data. When DACVU is set to 0, the LDACVOL or RDACVOL control data is loaded into an intermediate register, but the actual gain does not change. Both left and right gain settings are updated simultaneously when DACVU is set to

1.

See "Volume Updates" for more information on volume update bits.

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REGISTER

ADDRESS

R10 (0Ah) Left Channel

Digital Volume

R11 (0Bh) Right Channel

Digital Volume

Table 15 Digital Volume Control

BIT LABEL DEFAULT DESCRIPTION

8 DACVU 0 DAC Volume Update

0 = Store LDACVOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = LDACVOL, right = intermediate latch)

7:0 LDACVOL

[7:0]

8 DACVU 0 DAC Volume Update

7:0 RDACVOL

[7:0]

11111111 ( 0dB )

Left DAC Digital Volume Control 0000 0000 = Digital Mute 0000 0001 = -127dB 0000 0010 = -126.5dB ... 0.5dB steps up to 1111 1111 = 0dB

0 = Store RDACVOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = intermediate latch, right = RDACVOL)

Right DAC Digital Volume Control similar to LDACVOL

DAC SOFT MUTE AND SOFT UN-MUTE

The WM8960 also has a soft mute function, which, when enabled, gradually attenuates the volume of the digital signal to zero. When soft mute is disabled, the gain will either gradually ramp back up to the digital gain setting, or return instantly to the digital gain setting, depending on the DACSMM register bit.

The DAC is soft-muted by default. To play back an audio signal, this function must first be disabled by setting the DACMU bit to zero.

DACSMM would typically be enabled when using soft mute during playback of audio data so that when soft mute is then disabled, the sudden volume increase will not create pop noise by jumping immediately to the previous volume level (e.g. resuming playback after pausing during a track).

DACSMM would typically be disabled when un-muting at the start of a digital music file, so that the first part of the track is not attenuated (e.g. when starting playback of a new track, or resuming playback after pausing between tracks).

DAC muting and un-muting using volume control bits LDACVOL and RDACVOL.

DAC muting and un-muting using soft mute bit DACMU.

Soft un-mute not enabled (DACSMM = 0).

DAC muting and un-muting using soft mute bit DACMU.

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Soft un-mute enabled (DACSMM = 1).

Figure 12 DAC Mute Control

The volume ramp rate during soft mute and un-mute is controlled by the DACMR bit. Ramp rates of fs/32 and fs/2 are selectable as shown in Table 16 (fs = DAC sample rate).

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REGISTER

ADDRESS

R5 (05h) ADC and DAC

Control (1)

R6 (06h) ADC and DAC

Control (2)

Table 16 DAC Soft-Mute Control

DAC DE-EMPHASIS

Digital de-emphasis can be applied to the DAC playback data (e.g. when the data comes from a CD with pre-emphasis used in the recording). De-emphasis filtering is available for sample rates of 48kHz, 44.1kHz and 32kHz.

REGISTER

ADDRESS

R5 (05h) ADC and DAC

Control (1)

Table 17 DAC De-Emphasis Control

BIT LABEL DEFAULT DESCRIPTION

DACMU 1 Digital Soft Mute

3

1 = Mute 0 = No mute (signal active)

DACSMM 0 DAC Soft Mute Mode

3

0 = Disabling soft-mute (DACMU=0) will cause the volume to change immediately to the LDACVOL / RDACVOL settings

1 = Disabling soft-mute (DACMU=0) will cause the volume to ramp up gradually to the LDACVOL / RDACVOL settings

DACMR 0 DAC Soft Mute Ramp Rate

2

0 = Fast ramp (fs/2 at fs=48k, providing maximum delay of 10.7ms)

1 = Slow ramp (fs/32 at fs=48k, providing maximum delay of 171ms)

BIT LABEL DEFAULT DESCRIPTION

2:1

DEEMPH [1:0]

00 De-Emphasis Control

11 = 48kHz sample rate 10 = 44.1kHz sample rate 01 = 32kHz sample rate 00 = No de-emphasis

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DAC OUTPUT PHASE AND MONO MIXING

The digital audio data is converted to oversampled bit streams in the on-chip, true 24-bit digital interpolation filters. The bitstream data enters two multi-bit, sigma-delta DACs, which convert them to high quality analogue audio signals. The multi-bit DAC architecture reduces high frequency noise and sensitivity to clock jitter. It also uses a Dynamic Element Matching technique for high linearity and low distortion.

In normal operation, the left and right channel digital audio data is converted to analogue in two separate DACs. There is a mono-mix mode where the two audio channels are mixed together digitally and then converted to analogue using only one DAC, while the other DAC is switched off. The mono-mix signal can be selected to appear on both analogue output channels. The mono mix is automatically attenuated by 6dB to prevent clipping.

The DAC output defaults to non-inverted. Setting DACPOL[0] bit will invert the left DAC output phase and setting DACPOL[1] bit will invert the right DAC output phase.

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REGISTER

ADDRESS

R6 (06h) ADC and DAC

Control (2)

R23 (17h) Additional

Control (1)

Table 18 DAC Mono Mix and Phase Invert Select

BIT LABEL DEFAULT DESCRIPTION

DACPOL[1:0] 00 DAC Polarity Control:

6:5

00 = Polarity not inverted 01 = DAC L inverted 10 = DAC R inverted 11 = DAC L and R inverted

DMONOMIX 0 DAC Mono Mix

4

0 = Stereo 1 = Mono (Mono MIX output on

enabled DACs)

3D STEREO ENHANCEMENT

The WM8960 has a digital 3D enhancement option to artificially increase the separation between the left and right channels. This effect can only be used for playback, not for record.

The 3D enhancement function is activated by the 3DEN bit, and the 3DDEPTH setting controls the degree of stereo expansion. Additionally, one of four filter characteristics can be selected for the 3D processing, using the 3DUC and 3DLC control bits.

REGISTER

ADDRESS

R16 (10h) 3D enhance

Table 19 3D Stereo Enhancement Function

BIT LABEL DEFAULT DESCRIPTION

6 3DUC 0 Upper Cut-Off Frequency

0 = High (Recommended for fs>=32kHz)

1 = Low (Recommended for fs<32kHz)

5 3DLC 0 Lower Cut-Off Frequency

0 = Low (Recommended for fs>=32kHz)

1 = High (Recommended for fs<32kHz)

4:1 3DDEPTH

[3:0]

0 3DEN 0 3D Stereo Enhancement Enable

0000 3D Stereo Depth

0000 = 0% (minimum 3D effect) 0001 = 6.67%

....

1110 = 93.3% 1111 = 100% (maximum 3D effect)

0 = Disabled 1 = Enabled

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When 3D enhancement is enabled it may be necessary to attenuate the signal by 6dB to avoid limiting. This is a user-selectable function, enabled by setting DACDIV2.

REGISTER

ADDRESS

R5 (05h) ADC and DAC

control (1)

Table 20 DAC 6dB Attenuation Select

BIT LABEL DEFAULT DESCRIPTION

7 DACDIV2 0 DAC 6dB attenuate enable

0 = disabled (0dB) 1 = -6dB enabled

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OUTPUT MIXERS

Left and right analogue mixers allow the DAC output and analogue bypass paths to be mixed. Programmable attenutation and mute is available on the analogue bypass paths from LINPUT3, RINPUT3 and from the input boost mixers as shown in Figure 13. A mono mix of left and right output mixers is also available on OUT3.

Figure 13 Output Mixer Path

Left and right mixers are enabled by the LOMIX and ROMIX register bits. The mono mixer is enabled by OUT3 register bit, which also enables the OUT3 driver.

REGISTER

ADDRESS

R47 (2Fh) Power

Management (3)

R26 (1Ah) Power

Management (2)

Table 21 Output Mixer Enable Control

Inputs to the mixers from the DAC and bypass paths can be individually muted. The bypass paths have programmable attenuation as shown in Table 22. To prevent pop noise, it is recommended not to change volume levels of these paths during playback.

BIT LABEL DEFAULT DESCRIPTION

3 LOMIX 0 Left Output Mixer Enable Control

0 = Disabled 1 = Enabled

4 ROMIX 0 Right Output Mixer Enable Control

0 = Disabled 1 = Enabled

1 OUT3 0 Mono Output and Mono Mixer Enable

Control 0 = Mono mixer and output disabled 1 = Mono mixer and output enabled

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REGISTER

ADDRESS

R34 (22h) Left Output

Mixer Control

R45 (2Dh) Bypass (1)

R37 (25h) Right Output

Mixer Control

R46 (2Eh) Bypass (2)

Table 22 Left and Right Output Mixer Mute and Volume Control

BIT LABEL DEFAULT DESCRIPTION

8 LD2LO 0 Left DAC to Left Output Mixer

0 = Disable (Mute) 1 = Enable Path

7 LI2LO 0 LINPUT3 to Left Output Mixer

0 = Disable (Mute) 1 = Enable Path

6:4 LI2LOVOL

[2:0]

7 LB2LO 0 Left Input Boost Mixer to Left Output

6:4 LB2LOVOL

[2:0]

8 RD2RO 0 Right DAC to Right Output Mixer

7 RI2RO 0 RINPUT3 to Right Output Mixer

6:4 RI2ROVOL

[2:0]

7 RB2RO 0 Right Input Boost Mixer to Right Output

6:4 RB2ROVOL

[2:0]

101 (-15dB)

LINPUT3 to Left Output Mixer Volume 000 = 0dB ...(3dB steps) 111 = -21dB

Mixer 0 = Disable (Mute) 1 = Enable Path

Left Input Boost Mixer to Left Output Mixer Volume

000 = 0dB ...(3dB steps) 111 = -21dB

0 = Disable (Mute) 1 = Enable Path

RINPUT3 to Right Output Mixer Volume 000 = 0dB ...(3dB steps) 111 = -21dB

Mixer 0 = Disable (Mute) 1 = Enable Path

Right Input Boost Mixer to Right Output Mixer Volume

000 = 0dB ...(3dB steps) 111 = -21dB

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The mono output mixer can output, left, right, left+right or a buffered VMID. 0dB or 6dB attenuation is selectable using MOUTVOL register bit. It is recommended to attenuate a mono mix of left and right channels by 6dB in order to prevent clipping. This attenuation control (MOUTVOL) should not be modified while OUT3 is enabled as this may cause an audible click noise.

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REGISTER

ADDRESS

R38 (26h) Mono Out Mix

(1)

R39 (27h) Mono Out Mix

(2)

R42 (2Ah) Mono Out

Volume

Table 23 Output Mixer Enable Control

When left and right inputs to the mono mixer are both disabled, the mono mixer will output VMID.

ANALOGUE OUTPUTS

HP_L AND HP_R OUTPUTS

The HP_L and HP_R pins can drive a 16Ω or 32Ω headphone or a line output (see Headphone Output and Line Output sections, respectively). The signal volume on HP_L and HP_R can be independently adjusted under software control by writing to LOUT1VOL and ROUT1VOL, respectively. Note that gains over 0dB may cause clipping if the signal is large. Any gain setting below 0101111 (minimum) mutes the output driver. The corresponding output pin remains at the same DC level (the reference voltage on the VREF pin), so that no click noise is produced when muting or un-muting.

BIT LABEL DEFAULT DESCRIPTION

7 L2MO 0 Left Output Mixer to Mono Output Mixer

Control 0 = Left channel mix disabled 1 = Left channel mix enabled

7 R2MO 0 Right Output Mixer to Mono Output

Mixer Control 0 = Right channel mix disabled 1 = Right channel mix enabled

6 MOUTVOL 1 Mono Output Mixer Volume Control

0 = 0dB 1 = -6dB

A zero cross detect on the analogue output may also be enabled when changing the gain setting to minimize audible clicks and zipper noise as the gain updates. If zero cross is enabled a timeout is also available to update the gain if a zero cross does not occur. This function may be enabled by setting TOEN in register R23 (17h). The timeout period is set by TOCLKSEL. Note: SYSCLK must be enabled to use this function.

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REGISTER

ADDRESS

R2 (02h) LOUT1 Volume

R3 (03h) ROUT1 Volume

Table 24 LOUT1/ROUT1 Volume Control

BIT LABEL DEFAULT DESCRIPTION

8 OUT1VU 0 Headphone Volume Update

0 = Store LOUT1VOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = LOUT1VOL, right = intermediate latch)

7 LO1ZC 0 Left zero cross enable

0 = Change gain immediately 1 = Change gain on zero cross only

6:0 LOUT1VOL

[6:0]

8 OUT1VU 0 Headphone Volume Update

7 RO1ZC 0 Right zero cross enable

6:0 ROUT1VOL

[6:0]

0000000 (MUTE)

LOUT1 Volume 1111111 = +6dB … 1dB steps down to 0110000 = -73dB 0101111 to 0000000 = Analogue

MUTE

0 = Store ROUT1VOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = intermediate latch, right = ROUT1VOL)

0 = Change gain immediately 1 = Change gain on zero cross only

ROUT1 Volume Similar to LOUT1VOL

See "Volume Updates" for more information on volume update bits, zero cross and timeout operation.

CLASS D SPEAKER OUTPUTS

The SPK_LP/SPK_LN and SPK_RP/SPK_RN output pins are class D speaker drivers. Each pair is independently controlled and can drive an 8Ω BTL speaker (see Speaker Output section). Output mixer volume is relative to AVDD, while an additional boost stage is available to accommodate higher SPKVDD1/SPKVDD2 supply voltages. This allows AVDD to be run at a lower voltage to save power, while maximum output power can be delivered to the load, utilising the full range of SPKVDD1/SPKVDD2. Note that the BTL speaker connection provides an additional +6dB gain at the output.

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Figure 14 Speaker Boost Operation

REGISTER

ADDRESS

R40 (28h) Left Speaker Volume

R41 (29h) Right Speaker Volume

R51 (33h) Class D

Control (3)

BIT LABEL DEFAULT DESCRIPTION

6:0 SPKLVOL

[6:0]

7 SPKLZC 0 Left Speaker Zero Cross Enable

8 SPKVU 0 Speaker Volume Update

6:0 SPKRVOL

[6:0]

7 SPKRZC 0 Right Speaker Zero Cross Enable

8 SPKVU 0 Speaker Volume Update

5:3 DCGAIN

[2:0]

0000000 (MUTE)

000 (1.0x)

SPK_LP/SPK_LN Volume 1111111 = +6dB … 1dB steps down to 0110000 = -73dB 0101111 to 0000000 = Analogue MUTE

1 = Change gain on zero cross only 0 = Change gain immediately

0 = Store SPKLVOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = SPKLVOL, right = intermediate latch)

SPK_RP/SPK_RN Volume 1111111 = +6dB … 1dB steps down to 0110000 = -73dB 0101111 to 0000000 = Analogue MUTE

1 = Change gain on zero cross only 0 = Change gain immediately

0 = Store SPKRVOL in intermediate latch (no gain change)

1 = Update left and right channel gains (left = intermediate latch, right = SPKRVOL)

DC Speaker Boost (Boosts speaker DC output level by up to 1.8 x on left and right channels)

000 = 1.00x boost (+0dB) 001 = 1.27x boost (+2.1dB) 010 = 1.40x boost (+2.9dB) 011 = 1.52x boost (+3.6dB) 100 = 1.67x boost (+4.5dB) 101 = 1.8x boost (+5.1dB) 110 to 111 = Reserved

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REGISTER

ADDRESS

2:0 ACGAIN

Table 25 SPK_L/SPK_R Volume and Speaker Boost Control

To prevent pop noise, DCGAIN and ACGAIN should not be modified while the speaker outputs are enabled.

To avoid clipping at speaker ground, ACGAIN should not be greater than DCGAIN.

To avoid clipping at speaker supply, SPKVDD1 and SPKVDD2 must be high enough to support the peak output voltage when using DCGAIN and ACGAIN functions. The peak output voltage is AVDD*(DCGAIN+ACGAIN)/2.

DCGAIN should normally be set to the same value as ACGAIN.

See "Volume Updates" for more information on volume update bits, zero cross and timeout operation.

See "Class D Speaker Outputs" for more information on class D speaker operation.

BIT LABEL DEFAULT DESCRIPTION

[2:0]

000 (1.0x)

AC Speaker Boost (Boosts speaker AC output signal by up to 1.8 x on left and right channels)

000 = 1.00x boost (+0dB) 001 = 1.27x boost (+2.1dB) 010 = 1.40x boost (+2.9dB) 011 = 1.52x boost (+3.6dB) 100 = 1.67x boost (+4.5dB) 101 = 1.8x boost (+5.1dB) 110 to 111 = Reserved

OUT3 OUTPUT

The OUT3 pin can drive a 16Ω or 32Ω headphone or a line output or be used as a pseudo-ground for capless headphone drive (see Headphone Output section). It can also drive out a mono mix of left and right output mixers (See Output Signal Path).

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ENABLING THE OUTPUTS

Each analogue output of the WM8960 can be independently enabled or disabled. The analogue mixer associated with each output is powered on or off along with the output pin. All outputs are disabled by default. To save power, unused outputs should remain disabled.

REGISTER

ADDRESS

R26 (1Ah) Power

Management (2)

R49 (31h) Class D

Control (1)

Note: All “Enable” bits are 1 = ON, 0 = OFF

Table 26 Analogue Output Control

The speaker output enable bits SPK_OP_EN[1:0] should not be enabled until there is a valid switching clock to drive the class D outputs. This means that SYSCLK must be active, and DCLKDIV set to an appropriate value to produce a class D clock of between 700kHz and 800kHz for best performance (See "Class D Speaker Outputs" and "Clocking and Sample Rates" sections for more information).

BIT LABEL DEFAULT DESCRIPTION

6 LOUT1 0 LOUT1 Output Enable

5 ROUT1 0 ROUT1 Output Enable

4 SPKL 0 SPK_LP and SPK_LN Volume Control

3 SPKR 0 SPK_RP and SPK_RN Volume Control

1 OUT3 0 OUT3 Enable

7:6 SPK_OP_EN

[1:0]

00 Enable Class D Speaker Outputs

Enable

00 = Off 01 = Left speaker only 10 = Right speaker only 11 = Left and right speakers enabled

Whenever an analogue output is disabled, it remains connected to VREF through a resistor. This helps to prevent pop noise when the output is re-enabled. The resistance between VREF and each output can be controlled using the VROI bit in register 27. If a high impedance is desired for disabled outputs, VROI can then be set to 1, increasing the resistance to about 20kΩ.

R27 (1Bh) Additional (1)

Table 27 Disabled Outputs to VREF Resistance

HEADPHONE OUTPUT

Analogue outputs HP_L/HP_R, and OUT3, 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

REGISTER

ADDRESS

BIT LABEL DEFAULT DESCRIPTION

6 VROI 0 VREF to Analogue Output Resistance

(Disabled Outputs) 0 = 500Ω VMID to output 1 = 20kΩ VMID to output

DC Coupled Headphone Output

(L2MO=0; R2MO=0)

Figure 15 Recommended Headphone Output Configurations

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When DC blocking capacitors are used, then their capacitance and the load resistance together determine the lower cut-off frequency, f response. Smaller capacitance values will diminish the bass response. Assuming a 32Ω load and C1, C2 = 100µF:

f

= 1 / 2π RLC1 = 1 / (2π x 32Ω x 100µF) = 50 Hz

c

In the DC coupled configuration, the headphone “ground” is connected to the OUT3 pin, which must be enabled by setting OUT3 = 1 and muted by setting L2MO=0 and R2MO=0. As the OUT3 pin produces a DC voltage of AVDD/2 (=VREF), there is no DC offset between HP_L/HP_R and OUT3, and therefore no DC blocking capacitors are required. This saves space and material cost in portable applications.

It is recommended to connect the DC coupled headphone 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.

CLASS D SPEAKER OUTPUTS

The class D speaker outputs SPK_LN/SPK_LP and SPK_RN/SPK_RP can drive 1W into 8Ω BTL speakers. Class D outputs reduce power consumption and maximise efficiency by reducing power dissipated in the output drivers, delivering most of the power directly to the load. This is achieved by pulse width modulation (PWM) of a high frequency square wave, allowing the audio signal level to be set by controlling the pulse width. The frequency of the output waveform is controlled by DCLKDIV, and is derived from SYSCLK.

When the speakers are close to the device (typically less than about 100mm), the internal filtering effects of the speaker can be used. Where signals are routed over longer distances, it is recommended to use additional passive filtering, positioned close to the WM8960, to reduce EMI. See "Applications Information" for more information on EMI reduction.

REGISTER

ADDRESS

R8 (08h) Clocking (2)

R49 (31h) Class D

Control (1)

Table 28 Class D Control Registers

. Increasing the capacitance lowers fc, improving the bass

c

BIT LABEL DEFAULT DESCRIPTION

8:6 DCLKDIV 111 Controls clock division from

7:6 SPK_OP_EN

[1:0]

00 Enable Class D Speaker Outputs

SYSCLK to generate suitable class D clock.

000 = SYSCLK / 1.5 001 = SYSCLK / 2 010 = SYSCLK / 3 011 = SYSCLK / 4 100 = SYSCLK / 6 101 = SYSCLK / 8 110 = SYSCLK / 12 111 = SYSCLK / 16

00 = Off 01 = Left speaker only 10 = Right speaker only 11 = Left and right speakers enabled

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The class D outputs require a PWM switching clock, which is derived from SYSCLK. This clock should not be altered or disabled while the class D outputs are enabled.

See "Clocking and Sample Rates" for more information.

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VOLUME UPDATES

Volume settings will not be applied to input or output PGAs until a '1' is written to one of the update bits (IPVU, OUT1VU, SPKVU bits). This is to allow left and right channels to be updated at the same time, as shown in Figure 16.

Figure 16 Simultaneous Left and Right Volume Updates

If the volume is adjusted while the signal is a non-zero value, an audible click can occur as shown in Figure 17.

Figure 17 Click Noise During Volume Update

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In order to prevent this click noise, a zero cross function is provided. When enabled, this will cause the PGA volume to update only when a zero crossing occurs, minimising click noise as shown in Figure 18.

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Figure 18 Volume Update Using Zero Cross Detection

If there is a long period where no zero-crossing occurs, a timeout circuit in the WM8960 will automatically update the volume. The volume updates will occur between one and two timeout periods, depending on when the volume update bit is set as shown in Figure 19. The TOEN register bit must be set to enable this timeout function. The timeout period is set by TOCLKSEL.

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Figure 19 Volume Update after Timeout

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HEADPHONE JACK DETECT

The ADCLRC/GPIO1, LINPUT3/JD2 and RINPUT3/JD3 pins can be selected as headphone jack detect inputs to automatically disable the speaker output and enable the headphone output e.g. when a headphone is plugged into a jack socket. In this mode, enabled by setting HPSWEN, the headphone detect input pin switches between headphone and speaker outputs (e.g. when the pin is connected to a mechanical switch in the headphone socket to detect plug-in). The HPSEL[1:0] bits select the input pin used for this function. The HPSWPOL bit reverses the pin’s polarity. Note that the LOUT1, ROUT1, SPKL and SPKR bits in register 26 must also be set for headphone and speaker output (see Table 29 and Table 30).

TOEN must also be set to enable the clock which is used for de-bouncing the jack detect input. TOCLKSEL selects a fast or slow de-bounce period. Note that SYSCLK must be enabled to use this function.

When using capless mode, the OUT3CAP bit should be enabled so that OUT3 is enabled/disabled at the same time as HP_L and HP_R to prevent pop noise.

The debounced headphone detect signal can also be output to the ADCLRC/GPIO1 pin (See GPIO section). This function is not available when using GPIO1 as an input or as ADCLRC.

When using the ADCLRC/GPIO1 pin as a headphone detect input, the ALRCGPIO register bit needs to be set to 1. In this mode, DACLRC is used for both ADC and DAC frame clocks. (See GPIO section for more information)

Note:

When LINPUT3 or RINPUT3 is used as the headphone detect input, the thresholds become CMOS levels (0.3 AVDD / 0.7 AVDD).

HPSWEN HPSWPOL HEADPHONE

DETECT PIN

(LINPUT3/JD2,

RINPUT3/JD3 OR

ADCLRC/GPIO1)

0 X X 0 0 no no

0 X X 0 1 no yes

0 X X 1 0 yes no

0 X X 1 1 yes yes

1 0 0 X 0 no no

1 0 0 X 1 no yes

1 0 1 0 X no no

1 0 1 1 X yes no

1 1 0 0 X no no

1 1 0 1 X yes no

1 1 1 X 0 no no

1 1 1 X 1 no yes

Table 29 Headphone Jack Detect Operation

L/ROUT1

(AND OUT3 IN

CAPLESS MODE)

(REG. 26)

SPKL/R

(REG. 26)

HEADPHONE

ENABLED

(AND OUT3 IN

CAPLESS MODE)

SPEAKER ENABLED

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REGISTER

ADDRESS

R24 (18h) Additional

Control (2)

BIT LABEL DEFAULT DESCRIPTION

6 HPSWEN 0

5 HPSWPOL 0 Headphone Switch Polarity

Headphone Switch Enable 0 = Headphone switch disabled 1 = Headphone switch enabled

0 = HPDETECT high = headphone 1 = HPDETECT high = speaker

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REGISTER

ADDRESS

R27 (1Bh) Additional

Control (3)

R48 (30h) Additional

Control (4)

R23 (17h) Additional

Control (1)

Table 30 Headphone Jack Detect

BIT LABEL DEFAULT DESCRIPTION

3 OUT3CAP 0 Capless Mode Headphone Switch

3:2 HPSEL[1:0] 00 Headphone Switch Input Select

0 TOEN 0 Slow Clock Enable (Must be enabled for

1 TOCLKSEL 0 Slow Clock Selection (Used for volume

Enable 0 = OUT3 unaffected by jack detect

events 1 = OUT3 enabled and disabled together

with HP_L and HP_R in response to jack detect events

0X = GPIO1 used for jack detect input (Requires ADCLRC pin to be configured as a GPIO)

10 = JD2 used for jack detect input 11 = JD3 used for jack detect input

jack detect de-bounce) 0 = Slow Clock Disabled 1 = Slow Clock Enabled

update timeouts and for jack detect debounce)

0 = SYSCLK / 2 1 = SYSCLK / 2

21

(Slower Response)

19

(Faster Response)

Figure 20 Example Headset Detection Circuit Using Normally-Open Switch

Figure 21 Example Headset Detection Circuit Using Normally-Closed Switch

THERMAL SHUTDOWN

The speaker and headphone outputs can drive very large currents. To protect the WM8960 from overheating a thermal shutdown circuit is included and is enabled by default. If the device temperature reaches approximately 150 TSENSEN = 1) the speaker and headphone amplifiers (HP_L, HP_R, SPK_LP, SPK_LN, SPK_RP, SPK_RN and OUT3) will be disabled. This feature can be disabled to save power when the device is in standby mode.

TSENSEN must be set to 1 to enable the temperature sensor when using the TSDEN thermal shutdown function. The output of the temperature sensor can also be output to the GPIO1 pin.

0

C and the thermal shutdown circuit is enabled (TSDEN = 1;

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WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R23 (17h) Additional

Control (1)

R48 (30h) Additional

Control (4)

Table 31 Thermal Shutdown

GENERAL PURPOSE INPUT/OUTPUT

The WM8960 has three dual purpose input/output pins.

• LINPUT3/JD2: Analogue input or headphone detect input.

• RINPUT3/JD3: Analogue input or headphone detect input.

• ADCLRC/GPIO1: ADC left/right frame clock or GPIO pin.

The ADCLRC/GPIO1 pin can be configured as a left/right frame clock for the ADC, a headphone detect input, or one of a number of GPIO output functions as shown in Table 32.

The default configuration for the LINPUT3 and RINPUT2 pins is to be analogue inputs. The default configuration for the ADCLRC/GPIO1 pin is to be the ADC left/right frame clock.

REGISTER

ADDRESS

R9 (09h) Audio

Interface (2)

R48 (30h) Additional Control (4)

R52 (34h) Clocking (2)

Table 32 GPIO Control

Slow clock must be enabled (TOEN = 1) when using the jack detect function. This slow clock is used to debounce the jack detect input. The debounce period can be selected using TOCLKSEL.

The temperature sensor must be enabled for the "Temperature ok" GPIO output to function properly.

BIT LABEL DEFAULT DESCRIPTION

8 TSDEN 1 Thermal Shutdown Enable

0 = Thermal shutdown disabled 1 = Thermal shutdown enabled (TSENSEN must be enabled for this

function to work)

1 TSENSEN 1 Temperature Sensor Enable

0 = Temperature sensor disabled 1 = Temperature sensor enabled

BIT LABEL DEFAULT DESCRIPTION

6 ALRCGPIO 0 ADCLRC/GPIO1 Pin Function Select

0 = ADCLRC frame clock for ADC 1 = GPIO pin

6:4 GPIOSEL

[2:0]

7 GPIOPOL 0 GPIO Polarity Invert

8:6 OPCLKDIV

[2:0]

000 ADCLRC/GPIO1 GPIO Function Select:

000 = Jack detect input 001 = Reserved 010 = Temperature ok 011 = Debounced jack detect output 100 = SYSCLK output 101 = PLL lock 110 = Logic 0 111 = Logic 1

0 = Non inverted 1 = Inverted

000 SYSCLK Output to GPIO Clock Division

ratio 000 = SYSCLK 001 = SYSCLK / 2 010 = SYSCLK / 3 011 = SYSCLK / 4 100 = SYSCLK / 5.5 101 = SYSCLK / 6

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For further details of the Jack detect operation see the Headphone Switch section.

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DIGITAL AUDIO INTERFACE

The digital audio interface is used for inputting DAC data into the WM8960 and outputting ADC data from it. It uses five pins:

• ADCDAT: ADC data output

• ADCLRC: ADC data alignment clock

• DACDAT: DAC data input

• DACLRC: DAC data alignment clock

• BCLK: Bit clock, for synchronisation

The clock signals BCLK, ADCLRC and DACLRC can be outputs when the WM8960 operates as a master, or inputs when it is a slave (see Master and Slave Mode Operation, below).

ADCLRC can also be configured as a GPIO pin. In this case, the ADC will use DACLRC as a frame clock. The ADCLRC/GPIO1 pin function should not be modified while the ADC is enabled.

Four different audio data formats are supported:

• Left justified

• Right justified

2

S

• I

• DSP mode

All four of these modes are MSB first. They are described in Audio Data Formats, below. Refer to the Electrical Characteristic section for timing information.

MASTER AND SLAVE MODE OPERATION

The WM8960 can be configured as either a master or slave mode device. As a master device the WM8960 generates BCLK, ADCLRC and DACLRC and thus controls sequencing of the data transfer on ADCDAT and DACDAT. In slave mode, the WM8960 responds with data to clocks it receives over the digital audio interface. The mode can be selected by writing to the MS bit. Master and slave modes are illustrated below.

BCLK ADCLRC DACLRC

WM8960

ADCDAT

CODEC

DACDAT

Note: The ADC and DAC can run at diff erent sample rat es

DSP ENCODER/ DECODER

WM8960 CODEC

Note: Th e ADC and DAC can run at diffe rent sample rate s

BCLK ADCLRC DACLRC ADCDAT DACDAT

DSP ENCODER/ DECODER

Figure 22 Master Mode Figure 23 Slave Mode

OPERATION WITH ADCLRC AS GPIO

When ALRCGPIO=1, the DACLRC pin is used as a frame clock for ADCs and DACs as shown below. The ADCs and DACs must operate at the same sample rate in this mode. See Table 32 for details of GPIO pin configuration.

Figure 24 Master Mode with ADCLRC as GPIO Figure 25 Slave Mode with ADCLRC as GPIO

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BCLK DIVIDE

The BCLK frequency in master mode is controlled by BCLKDIV[3:0]. When the ADCs and DACs are operating at different sample rates, BCLKDIV must be set appropriately to support the data rate of whichever is the faster.

Internal clock divide and phase control mechanisms ensure that the BCLK, ADCLRC and DACLRC edges will occur in a predictable and repeatable position relative to each other and to the data for a given combination of DAC sample rate, ADC sample rate and BCLKDIV settings.

See Clocking and Sample Rates section for more information.

AUDIO DATA FORMATS

In Left Justified mode, the MSB is available on the first rising edge of BCLK following a LRCLK transition. The other bits up to the LSB are then transmitted in order. Depending on word length, BCLK frequency and sample rate, there may be unused BCLK cycles before each LRCLK transition.

Figure 26 Left Justified Audio Interface (assuming n-bit word length)

In Right Justified mode, the LSB is available on the last rising edge of BCLK before a LRCLK transition. All other bits are transmitted before (MSB first). Depending on word length, BCLK frequency and sample rate, there may be unused BCLK cycles after each LRCLK transition.

Figure 27 Right Justified Audio Interface (assuming n-bit word length)

2

S mode, the MSB is available on the second rising edge of BCLK following a LRCLK transition.

In I 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.

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Preliminary Technical Data WM8960

2

Figure 28 I

In DSP/PCM mode, the left channel MSB is available on either the 1

S Justified Audio Interface (assuming n-bit word length)

st

(mode B) or 2nd (mode A) rising edge of BCLK (selectable by LRP) following a rising edge of LRC. Right channel data immediately follows left channel data. Depending on word length, BCLK frequency and sample rate, there may be unused BCLK cycles between the LSB of the right channel data and the next sample.

In device master mode, the LRC output will resemble the frame pulse shown in Figure 29 and Figure

30. In device slave mode, Figure 31 and Figure 32, it is possible to use any length of frame pulse less than 1/fs, providing the falling edge of the frame pulse occurs greater than one BCLK period before the rising edge of the next frame pulse.

1/fs

1 BCLK

LRCLK

BCLK

LEFT CHANNEL RIGHT CHANNEL

DACDAT /

ADCDAT

n-2 n-1

Input Word Length (WL)

n321

LSBMSB

n-2 n-1

n321

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Figure 29 DSP/PCM Mode Audio Interface (mode A, LRP=0, Master)

1/fs

1 BCLK

LRC

BCLK

RIGHT CHANNEL

n-2 n-1

n321

DACDAT /

ADCDAT

LEFT CHANNEL

n-2 n-1

Input Word Length (WL)

n321

LSBMSB

Figure 30 DSP/PCM Mode Audio Interface (mode B, LRP=1, Master)

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WM8960 Preliminary Technical Data

1/fs

1 BCLK

LRCLK

BCLK

DACDAT /

ADCDAT

LEFT CHANNEL RIGHT CHANNEL

Input Word Length (WL)

falling edge can occur anywhere i n this area

n321

n-2 n-1

LSBMSB

n-2 n-1

Figure 31 DSP/PCM Mode Audio Interface (mode A, LRP=0, Slave)

1/fs

1 BCLK

LRC

BCLK

falling edge can occur anywher e in this area

n321

1 BCLK

RIGHT CHANNEL

n-2 n-1

n321

DACDAT /

ADCDAT

LEFT CHANNEL

n-2 n-1

Input Word Length (WL)

n321

LSBMSB

Figure 32 DSP/PCM Mode Audio Interface (mode B, LRP=1, Slave)

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AUDIO INTERFACE CONTROL

The register bits controlling audio format, word length and master / slave mode are summarised in Table 33. MS selects audio interface operation in master or slave mode. In Master mode BCLK, ADCLRC and DACLRC are outputs. The frequency of ADCLRC and DACLRC is set by the bits ADCDIV and DACDIV and the frequency of BCLK is set by the bits BCLKDIV (See "Clocking and Sample Rates"). In Slave mode BCLK, ADCLRC and DACLRC are inputs.

REGISTER

ADDRESS

R7 (07h) Digital Audio

Interface Format

Table 33 Audio Data Format Control

BIT LABEL DEFAULT DESCRIPTION

8 ALRSWAP 0 Left/Right ADC channel swap

1 = Swap left and right ADC data in audio interface

0 = Output left and right data as normal

7 BCLKINV 0 BCLK invert bit (for master and slave

6 MS 0 Master / Slave Mode Control

5 DLRSWAP 0 Left/Right DAC Channel Swap

4 LRP 0

3:2 WL[1:0] 10 Audio Data Word Length

1:0 FORMAT[1:0] 10 Audio Data Format Select

modes) 0 = BCLK not inverted 1 = BCLK inverted

0 = Enable slave mode 1 = Enable master mode

0 = Output left and right data as normal 1 = Swap left and right DAC data in

audio interface

Right, left and I2S modes – LRCLK polarity

0 = normal LRCLK polarity 1 = invert LRCLK polarity

DSP Mode – mode A/B select 0 = MSB is available on 2nd BCLK rising

edge after LRC rising edge (mode A) 1 = MSB is available on 1st BCLK rising

edge after LRC rising edge (mode B)

00 = 16 bits 01 = 20 bits 10 = 24 bits 11 = 32 bits (see Note)

00 = Right justified 01 = Left justified

2

10 = I

S Format

11 = DSP Mode

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Note: Right Justified mode does not support 32-bit data.

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AUDIO INTERFACE OUTPUT TRISTATE

Register bit TRIS, register 24(18h) bit[3] can be used to tristate the ADCDAT pin and switch ADCLRC, DACLRC and BCLK to inputs. In Slave mode (MS=0) ADCLRC, DACLRC and BCLK are by default configured as inputs and only ADCDAT will be tri-stated, (see Table 34).

When the ADCLRC/GPIO1 pin is configured as a GPIO, this pin will not be tristated by the TRIS register bit.

REGISTER

ADDRESS

R24 (18h) Additional Control (2)

Table 34 Tri-stating the Audio Interface

MASTER MODE ADCLRC AND DACLRC ENABLE

In master mode, by default ADCLRC clock generator is disabled and will output a logic 0 when the ADCs are both disabled and DACLRC clock generator is disabled and will output a logic 0 when the DACs are both disabled.

BIT LABEL DEFAULT DESCRIPTION

3 TRIS 0

Tristates ADCDAT and switches ADCLRC, DACLRC and BCLK to inputs.

0 = ADCDAT is an output; ADCLRC, DACLRC and BCLK are inputs (slave mode) or outputs (master mode)

1 = ADCDAT is tristated; DACLRC and BCLK are inputs; ADCLRC is an input (when not configured as a GPIO)

Register bit LRCM, register 24 (18h) bit[2] changes the control so that the ADCLRC and DACLRC clock generators are both disabled only when both ADCs and both DACs are disabled. This enables the user to use e.g. ADCLRC for both ADC and DAC LRCLK and disable the ADC when DAC only operation is required, (see Table 35).

When ADCLRC is configured as a GPIO (using ALRCGPIO), DACLRC is used for the ADCs and the DACs and will only be disabled in master mode when both ADCs and both DACs are disabled.

Figure 33 Master Mode Clock Ouput Control

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REGISTER

ADDRESS

R24 (18h) Additional Control (2)

Table 35 ADCLRC/DACLRC Enable

BIT LABEL DEFAULT DESCRIPTION

2 LRCM 0

Selects disable mode for ADCLRC and DACLRC (Master mode)

0 = ADCLRC disabled when ADC (Left and

Right) disabled; DACLRC disabled when DAC (Left and Right) disabled.

1 = ADCLRC and DACLRC disabled only when

ADC (Left and Right) and DAC (Left and Right) are disabled.

COMPANDING

The WM8960 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 DACCOMP or ADCCOMP register bits respectively.

REGISTER

ADDRESS

R9 (09h) Audio

Interface (2)

Table 36 Companding Control

BIT LABEL DEFAULT DESCRIPTION

2:1 ADCCOMP 00 ADC companding

00 = off 01 = reserved 10 = µ-law 11 = A-law

4:3 DACCOMP 00 DAC companding

00 = off 01 = reserved 10 = µ-law 11 = A-law

5 WL8 0 0 = off

1 = device operates in 8-bit mode.

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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).

Setting the WL8 register bit allows the device to operate with 8-bit data. In this mode it is possible to use 8 BCLK cycles per LRC frame. When using DSP mode B, this allows 8-bit data words to be output consecutively every 8 BCLK cycles and can be used with 8-bit data words using the A-law and u-law companding functions.

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BIT7 BIT[6:4] BIT[3:0]

SIGN EXPONENT MANTISSA

Table 37 8-bit Companded Word Composition

u-law Companding

120

100

80

60

40

Companded Output

20

0

0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

Normalised Input

1

0.9

0.8

0.7

0.6

0.5

0.4

0.3

0.2

0.1

0

Normalised Output

Figure 34 µ-Law Companding

120

100

80

60

40

Companded Output

20

0

0 0.2 0.4 0.6 0.8 1

Figure 35 A-Law Companding

A-law Companding

Normalised Input

1

0.9

0.8

0.7

0.6

0.5

0.4

0.3

0.2

0.1

0

Normalised Output

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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.

The ADCs and DACs must both use DACLRC when loopback is enabled. This is enabled by setting register bit ALRCGPIO = 1.

REGISTER

ADDRESS

R9 Audio

Interface

Table 38 Loopback Control

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.

CLOCKING AND SAMPLE RATES

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Figure 36 Clocking Scheme

Clocks for the ADCs and DACs, the DSP core functions, the digital audio interface and the class D outputs are all derived from SYSCLK as show in Figure 36.

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WM8960 Preliminary Technical Data

SYSCLK can either be derived directly from MCLK, or generated from a PLL using MCLK as a reference. The clock source is selected by CLKSEL. Many commonly-used audio sample rates can be derived directly from MCLK, while the PLL provides additional flexibility.

The ADC and DAC sample rates are independently selectable, relative to SYSCLK, using ADCDIV and DACDIV. In master mode, BCLK is also derived from SYSCLK via a programmable clock divide (BCLKDIV).

When the ADCLRC/GPIO1 pin is configured as a GPIO, a clock derived from SYSCLK can be output on this pin to provide clocking for other parts of the system. The frequency of this output clock is set by OPCLKDIV.

A slow clock derived from SYSCLK is used to de-bounce the headphone detect function, and to set the timeout period for volume updates when zero-cross functions are used. This clock is enabled by TOEN and its frequency is set by TOCLKSEL.

The class D outputs require a clock, and this is also derived from SYSCLK via a programmable divider (DCLKDIV) as shown in Figure 36. The class D switching clock should be set between 700kHz and 800kHz.

The class D switching clock should not be disabled when the speaker outputs are active, as this would prevent the speaker outputs from functioning. The class D switching clock frequency should not be altered while the speaker outputs are active as this may generate an audible click.

Table 39 shows the clocking and sample rate controls for MCLK input, BITCLK output (in master mode), ADCs, DACs, class D outputs and GPIO clock output. Refer to Table 40 for example clocking configurations.

REGISTER

ADDRESS

R4 (04h) Clocking (1)

BIT LABEL DEFAULT DESCRIPTION

8:6 ADCDIV

[2:0]

5:3 DACDIV

[2:0]

2:1 SYSCLKDIV

[1:0]

0 CLKSEL 0 SYSCLK selection

000 ADC Sample rate divider (Also

determines ADCLRC in master mode) 000 = SYSCLK / (1.0 * 256) 001 = SYSCLK / (1.5 * 256) 010 = SYSCLK / (2 * 256) 011 = SYSCLK / (3 * 256) 100 = SYSCLK / (4 * 256) 101 = SYSCLK / (5.5 * 256) 110 = SYSCLK / (6 * 256) 111 = Reserved

000 DAC Sample rate divider (Also

determines DACLRC in master mode) 000 = SYSCLK / (1.0 * 256) 001 = SYSCLK / (1.5 * 256) 010 = SYSCLK / (2 * 256) 011 = SYSCLK / (3 * 256) 100 = SYSCLK / (4 * 256) 101 = SYSCLK / (5.5 * 256) 110 = SYSCLK / (6 * 256) 111 = Reserved

00 SYSCLK Pre-divider. Clock source

(MCLK or PLL output) will be divided by this value to generate SYSCLK.

00 = Divide SYSCLK by 1 01 = Reserved 10 = Divide SYSCLK by 2 11 = Reserved

0 = SYSCLK derived from MCLK 1 = SYSCLK derived from PLL output

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Preliminary Technical Data WM8960

REGISTER

ADDRESS

R8 (08h) Clocking (2)

Table 39 ADC, DAC and BCLK Control

SYSCLK

(=MCLK OR PLL OUTPUT)

11.2896

Table 40 ADC and DAC Sample Rates

BIT LABEL DEFAULT DESCRIPTION

8:6 DCLKDIV 111 Class D switching clock divider.

3:0 BCLKDIV[3:0] 0000 BCLK Frequency (Master Mode)

ADCDIV OR

DACDIV

(MHz)

000 (=1) 48

001 (=1.5) 32

010 (=2) 24

12.288

2.048

011 (=3) 16

100 (=4) 12

101 (=5.5) (Not used)

110 (=6) 8

111 Reserved

000 (=1) 44.1

001 (=1.5) (Not used)

010 (=2) 22.05

011 (=3) (Not used)

100 (=4) 11.025

101 (=5.5) 8.018

110 (=6) (Not used)

111 Reserved

000 (=1) 8

001 (=1.5) (Not used)

010 (=2) (Not used)

011 (=3) (Not used)

100 (=4) (Not used)

101 (=5.5) (Not used)

110 (=6) (Not used)

111 Reserved

000 = SYSCLK / 1.5 001 = SYSCLK / 2 010 = SYSCLK / 3 011 = SYSCLK / 4 100 = SYSCLK / 6 101 = SYSCLK / 8 110 = SYSCLK / 12 111 = SYSCLK / 16

0000 = SYSCLK 0001 = SYSCLK / 1.5 0010 = SYSCLK / 2 0011 = SYSCLK / 3 0100 = SYSCLK / 4 0101 = SYSCLK / 5.5 0110 = SYSCLK / 6 0111 = SYSCLK / 8 1000 = SYSCLK / 11 1001 = SYSCLK / 12 1010 = SYSCLK / 16 1011 = SYSCLK / 22 1100 = SYSCLK / 24 1101 to 1111 = SYSCLK / 32

ADC / DAC SAMPLE

RATE (kHz)

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Although the ADC and DAC can run at different sample rates, they share the same bit clock pin BCLK.

When operating in master mode, register bits BCLKDIV[3:0] should be set to an appropriate value to ensure that there are sufficient BCLK cycles to transfer the complete data word from the ADCs and to the DACs.

When operating in slave mode, the host device must provide sufficient BCLK cycles to transfer complete data words to the ADCs and DACs.

Table 41 shows the maximum word lengths supported for a given SYSCLK and BCLKDIV, assuming that either the ADCs or DACs are running at maximum rate (i.e. ADCDIV[2:0]=000 or DACDIV[2:0]=000).

SYSCLK

(=MCLK OR PLL OUTPUT)

(MHz)

12.288

11.2896

Table 41 BCLK Divider in Master Mode

BCLKDIV[3:0] BCLK RATE

0000 (=1) 12.288 32

0001 (=1.5) 8.192 32

0010 (=2) 6.144 32

0011 (=3) 4.096 32

0100 (=4) 3.072 32

0101 (=5.5) 2.2341818 20

0110 (=6) 2.048 20

0111 (=8) 1.536 16

1000 (=11) 1.117091 8

1001 (=12) 1.024 8

1010 (=16) 0.768 8

1011 (=22) 0.558545 N/A

1100 (=24) 0.512 N/A

1101 (=32) 0.384 N/A

1110 (=32) 0.384 N/A

1111 (=32) 0.384 N/A

0000 (=1) 11.2896 32

0001 (=1.5) 7.5264 32

0010 (=2) 5.6448 32

0011 (=3) 3.7632 32

0100 (=4) 2.8224 32

0101 (=5.5) 2.052655 20

0110 (=6) 1.8816 20

0111 (=8) 1.4112 16

1000 (=11) 1.026327 8

1001 (=12) 0.9408 8

1010 (=16) 0.7056 8

1011 (=22) 0.513164 N/A

1100 (=24) 0.4704 N/A

1101 (=32) 0.3528 N/A

1110 (=32) 0.3528 N/A

1111 (=32) 0.3528 N/A

(MASTER MODE)

(MHz)

MAXIMUM WORD LENGTH

(AT MAXIMUM ADC OR

DAC SAMPLE RATE)

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OTHER SAMPLE RATE CONTROL BITS

The ALC, de-emphasis filter and 3D stereo enhance functions all need to be configured for the chosen sample rate when in use, as show in Table 42.

ADC_ALC_SR should be configured to match the chosen ADC sample rate.

DEEMPH, 3DUC and 3DUC should be configured to match the chosen DAC sample rate.

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REGISTER

ADDRESS

R27 (1Bh) Additional Control

(3)

R5 (05h) ADC and DAC

Control (1)

R16 (10h) 3D Enhance

Table 42 Additional Sample Rate Controls

BIT LABEL DEFAULT DESCRIPTION

2:0 ADC_ALC_SR

[2:0]

DEEMPH

2:1

[1:0]

6 3DUC 0 Upper Cut-Off Frequency

5 3DLC 0 Lower Cut-Off Frequency

000 ALC Sample Rate

000 = 44.1k / 48k 001 = 32k 010 = 22.05k / 24k 011 = 16k 100 = 11.25k / 12k 101 = 8k 110 and 111 = Reserved

00 De-Emphasis Control

11 = 48kHz sample rate 10 = 44.1kHz sample rate 01 = 32kHz sample rate 00 = No de-emphasis

0 = High (Recommended for fs>=32kHz)

1 = Low (Recommended for fs<32kHz)

0 = Low (Recommended for fs>=32kHz)

1 = High (Recommended for fs<32kHz)

PLL

The integrated PLL can be used to generate SYSCLK for the WM8960 or provide clocking for external devices via the GPIO1 pin.

The PLL is enabled by the PLLEN register bit.

REGISTER

ADDRESS

R26 (1Ah) Power

management (2)

R52 (34h) PLL (1)

Table 43 PLLEN Control Bit

The PLL frequency ratio R = f

PLLN = int R

PLLK = int (2

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 f

R = 98.304 / 12 = 8.192

PLLN = int R = 8

k = int ( 2

BIT LABEL DEFAULT DESCRIPTION

0 PLLEN 0 PLL Enable

0 = PLL off 1 = PLL on

5 SDM 0 Enable Integer Mode

0 = Integer mode 1 = Fractional mode

(See Figure 36) can be set using the register bits PLLK and PLLN:

2/f1

24

(R-PLLN))

= 4 x 2 x 12.288MHz = 98.304MHz.

2

24

x (8.192 – 8)) = 3221225 = 3126E9h

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MCLK

(MHz)

(f1)

REGISTER

ADDRESS

R52 (34h) PLL N value

R53 (35h) PLL K value

(1)

R54 (36h) PLL K Value

(2)

R55 (37h) PLL K Value

(3)

Table 44 PLL Frequency Ratio Control

The PLL performs best when f example settings are shown in Table 45.

DESIRED OUTPUT

(SYSCLK)

(MHz)

BIT LABEL DEFAULT DESCRIPTION

4 PLLPRESCALE 0 Divide MCLK by 2 before input to

3:0 PLLN 8h Integer (N) part of PLL input/output

5:0 PLLK [23:16] 31h

8:0 PLLK [15:8] 26h

8:0 PLLK [7:0] E9h

is between 90MHz and 100MHz. Its stability peaks at N=8. Some

2

f2

(MHz)

PLL 0 = Divide by 1 1 = Divide by 2

frequency ratio. Use values greater than 5 and less than 13.

Fractional (K) part of PLL1 input/output frequency ratio (treat as one 24-digit binary number).

R N K

(PLLPRESCALE)

PRESCALE DIVIDE

12 11.2896 90.3168 1 2 4 7.5264 7h 86C226h

12 12.288 98.304 1 2 4 8.192 8h

13 11.2896 90.3168 1 2 4 6.947446 6h

13 12.288 98.304 1 2 4 7.561846 7h

14.4 11.2896 90.3168 1 2 4 6.272 6h

14.4 12.288 98.304 1 2 4 6.826667 6h

19.2 11.2896 90.3168 2 2 4 9.408 9h

19.2 12.288 98.304 2 2 4 10.24 Ah

19.68 11.2896 90.3168 2 2 4 9.178537 9h

19.68 12.288 98.304 2 2 4 9.990243 9h

19.8 11.2896 90.3168 2 2 4 9.122909 9h

19.8 12.288 98.304 2 2 4 9.929697 9h

24 11.2896 90.3168 2 2 4 7.5264 7h

24 12.288 98.304 2 2 4 8.192 8h

26 11.2896 90.3168 2 2 4 6.947446 6h

26 12.288 98.304 2 2 4 7.561846 7h

27 11.2896 90.3168 2 2 4 6.690133 6h

27 12.288 98.304 2 2 4 7.281778 7h

Table 45 PLL Frequency Examples

(SYSCLKDIV[1:0])

FIXED POST-DIVIDE

POSTSCALE DIVIDE

3126E8h

F28BD4h

8FD525h

45A1CAh

D3A06Eh

6872AFh

3D70A3h

2DB492h

FD809Fh

1F76F7h

EE009Eh

86C226h

3126E8h

F28BD4h

8FD525h

B0AC93h

482296h

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Device running in master mode with 24-bit data MCLK input at 12.288MHz ADC and DAC running at fs=48kHz BCLK running at 64fs

Device running in slave mode with 24-bit data MCLK input at 12.288MHz ADC and DAC running at fs=48kHz BCLK supplied from host at 64fs in this example

Device running in master mode with 24-bit data MCLK input at 11.2896MHz ADC running at fs=8.018kHz DAC running at fs=44.1kHz BCLK running at 64fs (relative to DAC sample rate, as

DAC is operating at a higher sample rate than ADC)

Table 46 Example Clocking Schemes

Device running in slave mode with 24-bit data MCLK input at 11.2896MHz ADC running at fs=8.018kHz DAC running at fs=44.1kHz BCLK supplied from host at 64fs in this example (relative

to DAC sample rate, as DAC is operating at a higher sample rate than ADC)

Device running in master mode with 24-bit data MCLK input at 12MHz PLL Enabled and configured for SYSCLK=11.2896MHz ADC running at fs=8.018kHz DAC running at fs=44.1kHz BCLK running at 64fs (relative to DAC sample rate, as

DAC is operating at a higher sample rate than ADC) Class D clocks running at 705.6kHz

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CONTROL INTERFACE

2-WIRE SERIAL CONTROL INTERFACE

The WM8960 is controlled by writing to registers through a 2-wire serial control interface. A control word consists of 16 bits. The first 7 bits (B15 to B9) are address bits that select which control register is accessed. The remaining 9 bits (B8 to B0) are data bits, corresponding to the 9 bits in each control register. Many devices can be controlled by the same bus, and each device has a unique 7-bit address (this is not the same as the 7-bit address of each register in the WM8960).

The device address is 0011010 (0x34h).

The WM8960 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 WM8960 and the R/W bit is ‘0’, indicating a write, then the WM8960 responds by pulling SDIN low on the next clock pulse (ACK). If the address is not recognised or the R/W bit is ‘1’, the WM8960 returns to the idle condition and wait for a new start condition and valid address.

Once the WM8960 has acknowledged a correct address, the controller sends the first byte of control data (B15 to B8, i.e. the WM8960 register address plus the first bit of register data). The WM8960 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 WM8960 acknowledges again by pulling SDIN low.

The transfer of data is complete when there is a low to high transition on SDIN while SCLK is high. After receiving a complete address and data sequence the WM8960 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.

Figure 37 2-Wire Serial Control Interface

POWER MANAGEMENT

The WM8960 has three control registers that allow users to select which functions are active. For minimum power consumption, unused functions should be disabled. To avoid any pop or click noise, it is important to enable or disable functions in the correct order (see Applications Information). VMIDSEL is the enable for the Vmid reference, which defaults to disabled and can be enabled as a 2x50kΩ potential divider or, for low power maintenance of Vref when all other blocks are disabled, as a 2x250kΩ potential divider.

SDIN

SCLK

DEVICE ADDRESS

(7 BITS)

RD / WR

BIT

ACK

(LOW)

CONTROL BYTE 1

(BITS 15 TO 8)

register address and

1st register data bit

ACK

(LOW)

CONTROL BYT E 2

(BITS 7 TO 0)

remaining 8 bits of

register data

ACK

(LOW)

STOPSTART

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REGISTER

ADDRESS

R25 (19h) Power

Management (1)

BIT LABEL DEFAULT DESCRIPTION

8:7 VMIDSEL 00 Vmid Divider Enable and Select

00 = Vmid disabled (for OFF mode) 01 = 2 x 50kΩ divider enabled (for playback

/ record) 10 = 2 x 250kΩ divider enabled (for low-

power standby) 11 = 2 x 5kΩ divider enabled (for fast start-

up)

6 VREF 0 VREF (necessary for all other functions)

0 = Power down 1 = Power up

5 AINL 0 Analogue Input PGA and Boost Left

0 = Power down 1 = Power up (Note: LMIC must also be set to enable the

PGA)

4 AINR 0 Analogue Input PGA and Boost Right

0 = Power down 1 = Power up (Note: RMIC must also be set to enable the

PGA)

3 ADCL 0 ADC Left

0 = Power down 1 = Power up

2 ADCR 0 ADC Right

0 = Power down 1 = Power up

1 MICB 0 MICBIAS

0 = Power down 1 = Power up

0 DIGENB 0 Master Clock Disable

0 = Master clock enabled 1 = Master clock disabled

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R26 (1Ah) Power

Management (2)

8 DACL 0 DAC Left

0 = Power down 1 = Power up

7 DACR 0 DAC Right

0 = Power down 1 = Power up

6 LOUT1 0 LOUT1 Output Buffer

0 = Power down 1 = Power up

5 ROUT1 0 ROUT1 Output Buffer

0 = Power down 1 = Power up

4 SPKL 0 SPK_LP/SPK_LN Output PGA.

0 = Power down 1 = Power up (Note: Speaker output also requires

SPK_OP_EN[0] to be set)

3 SPKR 0 SPK_RP/SPK_RN Output PGA

0 = Power down

1 = Power up (Note: Speaker output also requires

SPK_OP_EN[1] to be set)

1 OUT3 0 OUT3 Output Buffer

0 PLL_EN 0 PLL Enable

R47 (2Fh) Power

Management (3)

Table 47 Power Management

5 LMIC Left Input PGA Enable

4 RMIC RIght Input PGA Enable

3 LOMIX Left Output Mixer Enable

2 ROMIX Right Output Mixer Enable

0 = Power down 1 = Power up

0 = Power down 1 = Power up (Note: PGA also requires AINL to be set)

0 = Power down 1 = Power up (Note: PGA also requires AINR to be set)

0 = Power down 1 = Power up

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STOPPING THE MASTER CLOCK

In order to minimise power consumed in the digital core of the WM8960, the master clock may be stopped in Standby and OFF modes. If this cannot be done externally at the clock source, the DIGENB bit (R25, bit 0) can be set to stop the MCLK signal from propagating into the device core. In Standby mode, setting DIGENB will typically provide an additional power saving on DCVDD of 20uA. However, since setting DIGENB has no effect on the power consumption of other system components external to the WM8960, it is preferable to disable the master clock at its source wherever possible.

MCLK should not be stopped while the class D outputs are enabled, as this would prevent the outputs from functioning.

REGISTER

ADDRESS

R25 (19h) Additional Control

(1)

Table 48 Enabling the Master Clock

NOTE: Before DIGENB can be set, the control bits ADCL, ADCR, DACL and DACR must be set to zero and a waiting time of 1ms must be observed. Any failure to follow this procedure may prevent DACs and ADCs from re-starting correctly.

BIT LABEL DEFAULT DESCRIPTION

0 DIGENB 0 Master clock disable

0 = Master clock enabled 1 = Master clock disabled

SAVING POWER AT HIGHER SUPPLY VOLTAGE

The AVDD supply of the WM8960 can operate beteen 2.7V and 3.6V. By default, all analogue circuitry on the device is optimized to run at 3.3V. This set-up is also good for all other supply voltages down to 2.7V. current by setting VSEL[1:0] = 01. If low power operation is preferred the bias current can be left at the default setting. This is controlled as shown below.

REGISTER

ADDRESS

R23 (17h) Additional

Control (1)

Table 49 Bias Optimisation

At lower voltages, performance can be improved by increasing the bias

BIT LABEL DEFAULT DESCRIPTION

7:6 VSEL

[1:0]

11 Analogue Bias Optimisation

00 = Reserved 01 = Increased bias current optimized for

AVDD=2.7V 1X = Lowest bias current, optimized for

AVDD=3.3V

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REGISTER MAP

REGISTER remarks Bit[8] Bit[7] Bit[6] Bit[5] Bit[4] Bit[3] Bit[2] Bit[1] Bit[0] default

R0 (00h) Left Input volume IPVU LINMUTE LIZC 0_1001_0111

R1 (01h) Right Input volume IPVU RINMUTE RIZC 0_1001_0111

R2 (02h) LOUT1 volume OUT1VU LO1ZC 0_0000_0000

R3 (03h) ROUT1 volume OUT1VU RO1ZC 0_0000_0000

R4 (04h) Clocking (1) CLKSEL 0_0000_ 0000

R5 (05h) ADC & DAC Control (CTR1) 0 DACDIV2 0 DACMU ADCHPD 0_0000_1000

R6 (06h) ADC & DAC Control (CTR2) 0 0 0 DACSMM DACMR DACSLOPE 0 0_0000_0000 R7 (07h) Audio Interface ALRSWAP BCLKINV MS DLRSWAP LRP 0_0000_1010

R8 (08h) Clocking (2) 00 1_1100_0000 R9 (09h) Audio Interface 0 0 ALRCGPIO WL8 LOOPBACK 0_0000_0000

R10 (0Ah) Left DAC volume DACVU 0_1111_1111

R11 (0Bh) Right DAC volume DACVU 0_1111_1111

R12 (0Ch) Reserved 0 0 0 0 0 0 0 0 0 0_0000_0000

R13 (0Dh) Reserved 0 0 0 0 0 0 0 0 0 0_0000_0000

R14 (0Eh) Reserved 0 0 0 0 0 0 0 0 0 0_0000_0000

R15 (0Fh) Reset not reset

R16 (10h) 3D control 0 0 3DUC 3DLC 3DEN 0_0000_0 000

R17 (11h) ALC1 0_0111_1011

R18 (12h) ALC2 1 0 1_0000_0000

R19 (13h) ALC3 ALCMODE 0_0011_0010

R20 (14h) Noise Gate 0 0 0 NGAT 0_0000_0000

R21 (15h) Left ADC volume ADCVU 0_1100_0011

R22 (16h) Right ADC volume ADCVU 0_1100_0011

R23 (17h) Additional control(1) TSDEN 0 DMONOMIX T OCLKSEL T OEN 1_1100_0000 R24 (18h) Additional control(2) 0 0 HPSWEN HPSWPOL 0 T RIS LRCM 0 0 0_0000_0000

R25 (19h) Pwr Mgmt (1) VREF AINL AINR ADCL ADCR MICB DIGENB 0_0000_0000

R26 (1Ah) Pwr Mgmt (2) DACL DACR LOUT1 ROUT1 SPKL SPKR 0 OUT3 PLL_EN 0_0000_0000 R27 (1Bh) Additional Control (3) 0 0 VROI 0 0 OUT3CAP 0_0000_0000 R28 (1Ch) Anti-pop 1 0 POBCTRL 0 0 BUFDCOPEN BUFIOEN SOFT_ST 0 HPSTBY 0_0000_0000

R29 (1Dh) Anti-pop 2 0 0 DISOP 0 0 0 0 0_0000_0000

R30 (1Eh) Reserved 0 0 0 0 0 0 0 0 0 0_0000_0000

R31 (1Fh) Reserved 0 0 0 0 0 0 0 0 0 0_0000_0000

R32 (20h) ADCL signal path LMN1 LMP3 LMP2 LMIC2B 0 0 0 1_0000_0000

R33 (21h) ADCR signal path RMN1 RMP3 RMP2 RMIC2B 0 0 0 1_0000_0000

R34 (22h) Left out Mix (1) LD2LO LI2LO 0 0 0 0 0_0101_0000

R35 (23h) Reserved 0 0 1 0 1 0 0 0 0 0_0101_0000

R36 (24h) Reserved 0 0 1 0 1 0 0 0 0 0_0101_0000

R37 (25h) Right out Mix (2) RD2RO RI2RO 0 0 0 0 0_0101_0000

R38 (26h) Mono out Mix (1) 0 L2MO 0 0 0 0 0 0 0 0_0000_0000

R39 (27h) Mono out Mix (2) 0 R2MO 0 0 0 0 0 0 0 0_0000_0000

R40 (28h) LOUT2 volume SPKVU SPKLZC 0_0000_0000

R41 (29h) ROUT2 volume SPKVU SPKRZC 0_0000_0000 R42 (2Ah) MONOOUT volume 0 0 MOUTVOL 0 0 0 0 0 0 0_0100_0000

R43 (2Bh) Input boost mixer (1) 0 0 0 0_0000_0000

R44 (2Ch) Input boost mixer (2) 0 0 0 0_0000_0 000

R45 (2Dh) Bypass (1) 0 LB2LO 0 0 0 0 0_0101_0000

R46 (2Eh) Bypass (2) 0 RB2RO 0 0 0 0 0_0101_0000

R47 (2Fh) Pwr Mgmt (3) 0 0 0 LMIC RMIC LOMIX ROMIX 0 0 0_0000_0000

R48 (30h) Additional Control (4) 0 GPIOPOL TSENSEN MBSEL 0_0000_0010

R49 (31h) Class D Control (1) 0 1 1 0 1 1 1 0_0011_0111

R50 (32h) Reserved 0 0 1 0 0 1 1 0 1 0_0100_1101

R51 (33h) Class D Control (3) 0 1 0 0_1000_0000

R52 (34h) PLL N SDM PLLRESCALE 0_0000_1000

R53 (35h) PLL K 1 0 0_0011_0001

R54 (36h) PLL K 2 0 0_0010_0110

R55 (37h) PLL K 3 0 0_1110_1001

ADCDIV[2:0] DACDIV[2:0] SYSCLKDIV[1:0]

ADCPOL[1:0] DEEMPH[1:0]

DACPOL[1: 0]

DCLKDIV[2:0] BCLKDIV[3:0]

DACCOMP[1 :0] ADCC OMP[1:0]

LDACVOL[7:0]

RDACVOL[7:0]

writing to this register resets all registers to their default state

ALCSEL[1:0] MAXG AIN[2:0] ALCL[3:0]

MINGAIN[2:0] HLD[3:0]

DCY[3:0] ATK[3:0]

NGTH[4:0 ]

LADCVOL[7:0]

RADCVOL[7:0]

VSEL[1:0] DATSEL[1:0 ]

VMIDSEL[1:0]

DRES[1:0]

LMICBOOST[1:0]

RMICBOOST[1:0]

LI2LOVOL[2:0]

RI2ROVOL[2:0]

LIN3BOOST[2:0] LIN2BOOST[2:0]

RIN3BOOST[2:0] RIN2BOOST[2:0]

LB2LOVOL[2:0]

RB2ROVOL[2:0]

GPIOSEL[2:0 ] HPSEL[1:0]

SPK_OP_EN[1:0]

DCGAIN[2:0] ACGAIN[2:0]

OPCLKDIV[2:0] PLLN[3:0]

PLLK[23:16]

PLLK[15:8]

PLLK[7:0]

LINVOL[5:0]

RINVOL[5 :0]

LOUT1VOL[6:0]

ROUT1VOL [6:0]

WL[1:0] FORMAT[1:0]

3DDEPTH[3:0]

ADC_ALC_SR[2:0 ]

SPKLVOL[6:0]

SPKRVOL[6:0]

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REGISTER BITS BY ADDRESS

REGISTER

ADDRESS

R0 (00h) Left Input

Volume

R1 (01h) Right Input

Volume

R2 (02h) LOUT1

Volume

R3 (03h) ROUT1

Volume

BIT LABEL DEFAULT DESCRIPTION REFER TO

8 IPVU N/A Input PGA Volume Update

Writing a 1 to this bit will cause left and right input PGA volumes to be updated (LINVOL and RINVOL)

7 LINMUTE 1 Left Input PGA Analogue Mute

1 = Enable Mute 0 = Disable Mute Note: IPVU must be set to un-mute.

6 LIZC 0 Left Input PGA Zero Cross Detector

1 = Change gain on zero cross only 0 = Change gain immediately

5:0 LINVOL[5:0] 010111 Left Input PGA Volume Control

111111 = +30dB 111110 = +29.25dB . . 0.75dB steps down to 000000 = -17.25dB

8 IPVU N/A Input PGA Volume Update

Writing a 1 to this bit will cause left and right input PGA volumes to be updated (LINVOL and RINVOL)

7 RINMUTE 1 Right Input PGA Analogue Mute

1 = Enable Mute 0 = Disable Mute Note: IPVU must be set to un-mute.

6 RIZC 0 Right Input PGA Zero Cross Detector

1 = Change gain on zero cross only 0 = Change gain immediately

5:0 RINVOL[5:0] 010111 Right Input PGA Volume Control

111111 = +30dB 111110 = +29.25dB . . 0.75dB steps down to 000000 = -17.25dB

8 OUT1VU N/A Headphone Output PGA Volume Update

Writing a 1 to this bit will cause left and right headphone output volumes to be updated (LOUT1VOL and ROUT1VOL)

7 LO1ZC 0 Left Headphone Output Zero Cross Enable

0 = Change gain immediately 1 = Change gain on zero cross only

6:0 LOUT1VOL[6:0] 0000000 LOUT1 Volume

1111111 = +6dB … 1dB steps down to 0110000 = -73dB 0101111 to 0000000 = Analogue MUTE

8 OUT1VU N/A Headphone Output PGA Volume Update

Writing a 1 to this bit will cause left and right headphone output volumes to be updated (LOUT1VOL and ROUT1VOL)

7 RO1ZC 0 Right Headphone Output Zero Cross Enable

0 = Change gain immediately 1 = Change gain on zero cross only

Input Signal Path

Analogue Outputs

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REGISTER

ADDRESS

6:0 ROUT1VOL[6:0] 0000000 ROUT1 Volume

R4 (04h) Clocking

(1)

R5 (05h) ADC and

DAC Control (1)

BIT LABEL DEFAULT DESCRIPTION REFER TO

1111111 = +6dB … 1dB steps down to 0110000 = -73dB 0101111 to 0000000 = Analogue MUTE

8:6 ADCDIV[2:0] 000 ADC Sample rate divider (Also determines

ADCLRC in master mode) 000 = SYSCLK / (1.0 * 256) 001 = SYSCLK / (1.5 * 256) 010 = SYSCLK / (2 * 256) 011 = SYSCLK / (3 * 256) 100 = SYSCLK / (4 * 256) 101 = SYSCLK / (5.5 * 256) 110 = SYSCLK / (6 * 256) 111 = Reserved

5:3 DACDIV[2:0] 000 DAC Sample rate divider (Also determines

DACLRC in master mode) 000 = SYSCLK / (1.0 * 256) 001 = SYSCLK / (1.5 * 256) 010 = SYSCLK / (2 * 256) 011 = SYSCLK / (3 * 256) 100 = SYSCLK / (4 * 256) 101 = SYSCLK / (5.5 * 256) 110 = SYSCLK / (6 * 256) 111 = Reserved

2:1 SYSCLKDIV[1:0] 00 SYSCLK Pre-divider. Clock source (MCLK or

PLL output) will be divided by this value to generate SYSCLK.

00 = Divide SYSCLK by 1 01 = Reserved 10 = Divide SYSCLK by 2 11 = Reserved

0 CLKSEL 0 SYSCLK Selection

0 = SYSCLK derived from MCLK 1 = SYSCLK derived from PLL output

8 0 Reserved

7 DACDIV2 0 DAC 6dB Attenuate Enable

0 = Disabled (0dB) 1 = -6dB Enabled

6:5 ADCPOL[1:0] 00 ADC polarity control:

00 = Polarity not inverted 01 = ADC L inverted 10 = ADC R inverted 11 = ADC L and R inverted

4 0 Reserved

3 DACMU 1 DAC Digital Soft Mute

1 = Mute 0 = No mute (signal active)

2:1 DEEMPH[1:0] 00 De-emphasis Control

11 = 48kHz sample rate 10 = 44.1kHz sample rate 01 = 32kHz sample rate 00 = No de-emphasis

Analogue Outputs

Clocking and Sample Rates

Output Signal Path

Analogue to Digital Converter

Output Signal Path

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REGISTER

ADDRESS

0 ADCHPD 0 ADC High Pass Filter Disable

R6 (06h) ADC and

DAC Control (2)

R7 (07h) Audio

Interface

BIT LABEL DEFAULT DESCRIPTION REFER TO

0 = Enable high pass filter on left and right channels

1 = Disable high pass filter on left and right channels

8:7 00 Reserved

6:5 DACPOL[1:0] 00 DAC polarity control:

00 = Polarity not inverted 01 = DAC L inverted 10 = DAC R inverted 11 = DAC L and R inverted

4 0 Reserved

3 DACSMM 0 DAC Soft Mute Mode

0 = Disabling soft-mute (DACMU=0) will cause the volume to change immediately to the LDACVOL / RDACVOL settings

1 = Disabling soft-mute (DACMU=0) will cause the volume to ramp up gradually to the LDACVOL / RDACVOL settings

2 DACMR 0 DAC Soft Mute Ramp Rate

0 = Fast ramp (24kHz at fs=48k, providing maximum delay of 10.7ms)

1 = Slow ramp (1.5kHz at fs=48k, providing maximum delay of 171ms)

1 DACSLOPE 0 Selects DAC filter characteristics

0 = Normal mode 1 = Sloping stopband

0 0 Reserved

8 ALRSWAP 0 Left/Right ADC Channel Swap

1 = Swap left and right ADC data in audio interface

0 = Output left and right data as normal

7 BCLKINV 0 BCLK invert bit (for master and slave modes)

0 = BCLK not inverted 1 = BCLK inverted

6 MS 0 Master / Slave Mode Control

0 = Enable slave mode 1 = Enable master mode

5 DLRSWAP 0 Left/Right DAC Channel Swap

0 = Output left and right data as normal 1 = Swap left and right DAC data in audio

interface

4 LRP 0

3:2 WL[1:0] 10 Audio Data Word Length

Right, left and I2S modes – LRCLK polarity 0 = normal LRCLK polarity 1 = invert LRCLK polarity

DSP Mode – mode A/B select 0 = MSB is available on 2nd BCLK rising edge

after LRC rising edge (mode A) 1 = MSB is available on 1st BCLK rising edge

after LRC rising edge (mode B)

00 = 16 bits 01 = 20 bits 10 = 24 bits 11 = 32 bits (see Note)

Analogue to Digital Converter

Output Signal Path

Audio Interface Control

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REGISTER

ADDRESS

1:0 FORMAT[1:0] 10 00 = Right justified

R8 (08h) Clocking

(2)

R9 (09h) Audio

Interface

BIT LABEL DEFAULT DESCRIPTION REFER TO

01 = Left justified

2

10 = I

S Format

11 = DSP Mode

8:6 DCLKDIV[2:0] 111 Class D switching clock divider.

000 = SYSCLK / 1.5 (Not recommended) 001 = SYSCLK / 2 010 = SYSCLK / 3 011 = SYSCLK / 4 100 = SYSCLK / 6 101 = SYSCLK / 8 110 = SYSCLK / 12 111 = SYSCLK / 16

5:4 00 Reserved

3:0 BCLKDIV[3:0] 0000 BCLK Frequency (Master Mode)

0000 = SYSCLK 0001 = SYSCLK / 1.5 0010 = SYSCLK / 2 0011 = SYSCLK / 3 0100 = SYSCLK / 4 0101 = SYSCLK / 5.5 0110 = SYSCLK / 6 0111 = SYSCLK / 8 1000 = SYSCLK / 11 1001 = SYSCLK / 12 1010 = SYSCLK / 16 1011 = SYSCLK / 22 1100 = SYSCLK / 24 1101 to 1111 = SYSCLK / 32

8:7 00 Reserved

6 ALRCGPIO 0 ADCLRC/GPIO1 Pin Function Select

0 = ADCLRC frame clock for ADC 1 = GPIO pin

5 WL8 0 8-Bit Word Length Select (Used with

4:3 DACCOMP[1:0] 00 DAC companding

2:1 ADCCOMP[1:0] 00 ADC companding

0 LOOPBACK 0 Digital Loopback Function

companding) 0 = Off 1 = Device operates in 8-bit mode.

00 = off 01 = reserved 10 = µ-law 11 = A-law

0 = No loopback. 1 = Loopback enabled, ADC data output is fed

directly into DAC data input.

Audio Interface Control

Class D Speaker Outputs;

Clocking and Sample Rates

General Purpose Input / Output;

Digital Audio Interface

Audio Interface Control

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REGISTER

ADDRESS

R10 (0Ah) Left DAC

Volume

R11 (0Bh) Right DAC

Volume

R12 (0Ch) 8:0 000000000 Reserved

R13 (0Dh) 8:0 000000000 Reserved

R14 (0Eh) 8:0 000000000 Reserved

R15 (0Fh) Reset

R16 {10h) 3D Control

R17 (11h) ALC (1)

BIT LABEL DEFAULT DESCRIPTION REFER TO

8 DACVU N/A DAC Volume Update

Writing a 1 to this bit will cause left and right DAC volumes to be updated (LDACVOL and RDACVOL)

7:0 LDACVOL[7:0] 11111111 Left DAC Digital Volume Control

0000 0000 = Digital Mute 0000 0001 = -127dB 0000 0010 = -126.5dB ... 0.5dB steps up to 1111 1111 = 0dB

8 DACVU N/A DAC Volume Update

Writing a 1 to this bit will cause left and right DAC volumes to be updated (LDACVOL and RDACVOL)

7:0 RDACVOL[7:0] 11111111 Right DAC Digital Volume Control

0000 0000 = Digital Mute 0000 0001 = -127dB 0000 0010 = -126.5dB ... 0.5dB steps up to 1111 1111 = 0dB

8:0 Reset N/A Writing to this register resets all registers to their

default state.

8 0 Reserved

7 0 Reserved

6 3DUC 0 3D Enhance Filter Upper Cut-Off Frequency

0 = High (Recommended for fs>=32kHz) 1 = Low (Recommended for fs<32kHz)

5 3DLC 0 3D Enhance Filter Lower Cut-Off Frequency

0 = Low (Recommended for fs>=32kHz) 1 = High (Recommended for fs<32kHz)

4:1 3DDEPTH[3:0] 0000 3D Stereo Depth

0000 = 0% (minimum 3D effect) 0001 = 6.67%

....

1110 = 93.3% 1111 = 100% (maximum 3D effect)

0 3DEN 0 3D Stereo Enhancement Enable

0 = Disabled 1 = Enabled

8:7 ALCSEL[1:0] 00 ALC Function Select

00 = ALC off (PGA gain set by register) 01 = Right channel only 10 = Left channel only 11 = Stereo (PGA registers unused) Note:

ensure that LINVOL and RINVOL settings (reg. 0 and 1) are the same before entering this mode.

Output Signal Path

Automatic Level Control

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WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R18 (12h) ALC (2)

R19 (13h) ALC (3)

R20 (14h) Noise

Gate

BIT LABEL DEFAULT DESCRIPTION REFER TO

6:4 MAXGAIN[2:0] 0000 Set Maximum Gain of PGA (During ALC

3:0 ALCL[3:0] 1011 ALC Target (Sets signal level at ADC input)

8 1 Reserved

7 0 Reserved

6:4 MINGAIN[2:0] 000 Set Minimum Gain of PGA (During ALC

3:0 HLD[3:0] 0000 ALC hold time before gain is increased.

8 ALCMODE 0 Determines the ALC mode of operation:

7:4 DCY[3:0] 0011 ALC decay (gain ramp-up) time

3:0 ATK[3:0] 0010 ALC attack (gain ramp-down) time

8 0 Reserved

7:3 NGTH[4:0] 00000 Noise gate threshold

2:1 00 Reserved

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operation) 111 : +30dB 110 : +24dB ….(-6dB steps) 001 : -6dB 000 : -12dB

0000 = -22.5dB FS 0001 = -21.0dB FS … (1.5dB steps) 1101 = -3.0dB FS 1110 = -1.5dB FS 1111 = -1.5dB FS

operation) 000 = -17.25dB 001 = -11.25dB 010 = -5.25dB 011 = +0.75dB 100 = +6.75dB 101 = +12.75dB 110 = +18.75dB 111 = +24.75dB

0000 = 0ms 0001 = 2.67ms 0010 = 5.33ms … (time doubles with every step) 1111 = 43.691s

0 = ALC mode 1 = Limiter mode

0000 = 24ms 0001 = 48ms 0010 = 96ms … (time doubles with every step) 1010 or higher = 24.58s

0000 = 6ms 0001 = 12ms 0010 = 24ms … (time doubles with every step) 1010 or higher = 6.14s

00000 -76.5dBfs 00001 -75dBfs … 1.5 dB steps 11110 -31.5dBfs 11111 -30dBfs

Automatic Level Control

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Preliminary Technical Data WM8960

REGISTER

ADDRESS

0 NGAT 0 Noise gate function enable

R21 (15h) Left ADC

Volume

R22 (16h) Right ADC

Volume

R23 (17h) Additional

Control (1)

Additional Control (2)

BIT LABEL DEFAULT DESCRIPTION REFER TO

0 = disable 1 = enable

8 ADCVU N/A ADC Volume Update

Writing a 1 to this bit will cause left and right ADC volumes to be updated (LADCVOL and RADCVOL)

7:0 LADCVOL[7:0] 11000011 Left ADC Digital Volume Control

0000 0000 = Digital Mute 0000 0001 = -97dB 0000 0010 = -96.5dB ... 0.5dB steps up to 1111 1111 = +30dB

8 ADCVU N/A ADC Volume Update

Writing a 1 to this bit will cause left and right ADC volumes to be updated (LADCVOL and RADCVOL)

7:0 RADCVOL[7:0] 11000011 Right ADC Digital Volume Control

0000 0000 = Digital Mute 0000 0001 = -97dB 0000 0010 = -96.5dB ... 0.5dB steps up to 1111 1111 = +30dB

8 TSDEN 1 Thermal Shutdown Enable

0 = Thermal shutdown disabled 1 = Thermal shutdown enabled (TSENSEN must be enabled for this function to

work)

7:6 VSEL[1:0] 11 Analogue Bias Optimisation

00 = Reserved 01 = Increased bias current optimized for

AVDD=2.7V 1X = Lowest bias current, optimized for

AVDD=3.3V

5 0 Reserved

4 DMONOMIX 0 DAC Mono Mix

0 = Stereo 1 = Mono (Mono MIX output on enabled DACs)

3:2 DATSEL[1:0] 00 ADC Data Output Select

00: left data = left ADC; right data =right ADC 01: left data = left ADC; right data = left ADC 10: left data = right ADC; right data =right ADC 11: left data = right ADC; right data = left ADC

1 TOCLKSEL 0 Slow Clock Select (Used for volume update

0 TOEN 0 Enables Slow Clock for Volume Update Timeout

8:7 00 Reserved R24 (18h)

6 HPSWEN 0 Headphone Switch Enable

timeouts and for jack detect debounce) 0 = SYSCLK / 2 1 = SYSCLK / 2

and Jack Detect Debounce 0 = Slow clock disabled 1 = Slow clock enabled

0 = Headphone switch disabled 1 = Headphone switch enabled

21

(Slower Response)

19

(Faster Response)

Automatic Level Control

Analogue to Digital Converter

Thermal Shutdown

Power Management

Output Signal Path

Analogue to Digital Converter

Volume Updates;

Headphone Jack Detect

Volume Updates;

Headphone Jack Detect

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WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R25 (19h) Power

Mgmt (1)

R26 (1Ah) Power

Mgmt (2)

BIT LABEL DEFAULT DESCRIPTION REFER TO

5 HPSWPOL 0 Headphone Switch Polarity

0 = HPDETECT high = headphone 1 = HPDETECT high = speaker

4 Reserved

3 TRIS 0 Tristates ADCDAT and switches ADCLRC,

DACLRC and BCLK to inputs. 0 = ADCDAT is an output; ADCLRC, DACLRC

and BCLK are inputs (slave mode) or outputs (master mode)

1 = ADCDAT is tristated; DACLRC and BCLK are inputs; ADCLRC is an input (when not configured as a GPIO)

2 LRCM 0 Selects disable mode for ADCLRC and

DACLRC (Master mode) 0 = ADCLRC disabled when ADC (Left and

Right) disabled; DACLRC disabled when DAC (Left and Right) disabled.

1 = ADCLRC and DACLRC disabled only when ADC (Left and Right) and DAC (Left and Right) are disabled.

1:0 0 Reserved

8:7 VMIDSEL[1:0] 00 Vmid Divider Enable and Select

00 = Vmid disabled (for OFF mode) 01 = 2 x 50kΩ divider enabled (for playback /

record) 10 = 2 x 250kΩ divider enabled (for low-power

standby) 11 = 2 x 5kΩ divider enabled (for fast start-up)

6 VREF 0 VREF (necessary for all other functions)

0 = Power down 1 = Power up

5 AINL 0 Analogue in PGA Left

0 = Power down 1 = Power up

4 AINR 0 Analogue in PGA Right

0 = Power down 1 = Power up

3 ADCL 0 ADC Left

0 = Power down 1 = Power up

2 ADCR 0 ADC Right

0 = Power down 1 = Power up

1 MICB 0 MICBIAS

0 = Power down 1 = Power up

0 DIGENB 0 Master Clock Disable

0 = Master clock enabled 1 = Master clock disabled

8 DACL 0 DAC Left

0 = Power down 1 = Power up

7 DACR 0 DAC Right

0 = Power down 1 = Power up

Headphone Jack Detect

Audio Interface Control

Power Management

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Preliminary Technical Data WM8960

REGISTER

ADDRESS

R27 (1Bh) Additional

Control (3)

R28 (1Ch) Anti-Pop 1

BIT LABEL DEFAULT DESCRIPTION REFER TO

6 LOUT1 0 LOUT1 Output Buffer

5 ROUT1 0 ROUT1 Output Buffer

4 SPKL 0 SPK_LP/SPK_LN Output Buffers

3 SPKR 0 SPK_RP/SPK_RN Output Buffers

2 0 Reserved

1 OUT3 0 OUT3 Output Buffer

0 PLL_EN 0 PLL Enable

8:7 00 Reserved

6 VROI 0 VREF to Analogue Output Resistance (Disabled

5 0 Reserved

4 0 Reserved

3 OUT3CAP 0 Capless Mode Headphone Switch Enable

2:0 ADC_ALC_SR 000 ALC Sample Rate

8 0 Reserved

7 POBCTRL 0 Selects the bias current source for output

6:5 00 Reserved

4 BUFDCOPEN 0 Enables the VGS / R current generator

3 BUFIOEN 0 Enables the VGS / R current generator and the

2 SOFT_ST 0 Enables VMID soft start

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0 = Power down 1 = Power up

Outputs) 0 = 500Ω VMID to output 1 = 20kΩ VMID to output

0 = OUT3 unaffected by jack detect events 1 = OUT3 enabled and disabled together with

HP_L and HP_R in response to jack detect events

000 = 44.1k / 48k 001 = 32k 010 = 22.05k / 24k 011 = 16k 100 = 11.25k / 12k 101 = 8k 110 and 111 = Reserved

amplifiers and VMID buffer 0 = VMID / R bias 1 = VGS / R bias

0 = Disabled 1 = Enabled

analogue input and output bias 0 = Disabled 1 = Enabled

0 = Disabled 1 = Enabled

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Power Management

Enabling the Outputs

Headphone Jack Detect

Automatic Level Control

75

WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R29 (1Dh) Anti-pop 2

R30 (1Eh) 8:0 000000000 Reserved

R31 (1Fh) 8:0 000000000 Reserved

R32 (20h) ADCL

Signal Path

R33 (21h) ADCR

Signal Path

BIT LABEL DEFAULT DESCRIPTION REFER TO

1 0 Reserved

0 HPSTBY 0 Headphone Amplifier Standby

0 = Standby mode disabled (Normal operation) 1 = Standby mode enabled

8:7 00 Reserved

6 DISOP 0 Discharges the DC-blocking headphone

5:4 DRES[1:0] 00

3:0 0000 Reserved

8 LMN1 1 Connect LINPUT1 to inverting input of Left Input

7 LMP3 0 Connect LINPUT3 to non-inverting input of Left

6 LMP2 0 Connect LINPUT2 to non-inverting input of Left

5:4 LMICBOOST[1:0] 00 Left Channel Input PGA Boost Gain

3 LMIC2B 0 Connect Left Input PGA to Left Input Boost

2:0 000 Reserved

8 RMN1 1 Connect RINPUT1 to inverting input of Right

7 RMP3 0 Connect RINPUT3 to non-inverting input of

capacitors on HP_L and HP_R 0 = Disabled 1 = Enabled

DRES determines the value of the resistors used to discharge the DC-blocking headphone capacitors when DISOP=1

DRES[1:0] Resistance (Ohms)

0 0 400

0 1 200

1 0 600

1 1 150

PGA 0 = LINPUT1 not connected to PGA 1 = LINPUT1 connected to PGA

Input PGA 0 = LINPUT3 not connected to PGA 1 = LINPUT3 connected to PGA (Constant input

impedance)

Input PGA 0 = LINPUT2 not connected to PGA 1 = LINPUT2 connected to PGA (Constant input

impedance)

00 = +0dB 01 = +13dB 10 = +20dB 11 = +29dB

Mixer 0 = Not connected 1 = Connected

Input PGA 0 = RINPUT1 not connected to PGA 1 = RINPUT1 connected to PGA

Right Input PGA 0 = RINPUT3 not connected to PGA 1 = RINPUT3 connected to PGA (Constant input

impedance)

Input Signal Path

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Preliminary Technical Data WM8960

REGISTER

ADDRESS

R34 (22h) Left Out

Mix

R35 (23h) 8:0 001010000 Reserved

R36 (24h) 8:0 001010000 Reserved

R37 (25h) Right Out

Mix

R38 (26h) Mono Out

Mix (1)

R39 (27h) Mono Out

Mix (2)

R40 (28h) Left

Speaker Volume

BIT LABEL DEFAULT DESCRIPTION REFER TO

6 RMP2 0 Connect RINPUT2 to non-inverting input of

Right Input PGA 0 = RINPUT2 not connected to PGA 1 = RINPUT2 connected to PGA (Constant input

impedance)

5:4 RMICBOOST[1:0] 00 Right Channel Input PGA Boost Gain

00 = +0dB 01 = +13dB 10 = +20dB 11 = +29dB

3 RMIC2B 0 Connect Right Input PGA to Right Input Boost

Mixer 0 = Not connected 1 = Connected

2:0 000 Reserved

8 LD2LO 0 Left DAC to Left Output Mixer

0 = Disable (Mute) 1 = Enable Path

7 LI2LO 0 LINPUT3 to Left Output Mixer

0 = Disable (Mute) 1 = Enable Path

6:4 LI2LOVOL[2:0] 101 LINPUT3 to Left Output Mixer Volume

000 = 0dB ...(3dB steps) 111 = -21dB

3:0 0000 Reserved

8 RD2RO 0 Right DAC to Right Output Mixer

0 = Disable (Mute) 1 = Enable Path

7 RI2RO 0 RINPUT3 to Right Output Mixer

0 = Disable (Mute) 1 = Enable Path

6:4 RI2ROVOL[2:0] 101 RINPUT3 to Right Output Mixer Volume

000 = 0dB ...(3dB steps) 111 = -21dB

3:0 0000 Reserved

8 0 Reserved

7 L2MO 0 Left Output Mixer to Mono Output Mixer Control

0 = Left channel mix disabled 1 = Left channel mix enabled

6:0 0000000 Reserved

8 0 Reserved

7 R2MO 0 Right Output Mixer to Mono Output Mixer

Control 0 = Right channel mix disabled 1 = Right channel mix enabled

6:0 0000000 Reserved

8 SPKVU N/A Speaker Volume Update

Writing a 1 to this bit will cause left and right speaker volumes to be updated (SPKLVOL and SPKRVOL)

Input Signal Path

Output Signal Path

Analogue Outputs

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WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R41 (29h) Right

Speaker Volume

R42 (2Ah) OUT3

Volume

R43 (2Bh) Left Input

Boost Mixer

R44 (2Ch) Right Input

Boost Mixer

Left Bypass

BIT LABEL DEFAULT DESCRIPTION REFER TO

7 SPKLZC 0 Left Speaker Zero Cross Enable

1 = Change gain on zero cross only 0 = Change gain immediately

6:0 SPKLVOL[6:0] 0000000 SPK_LP/SPK_LN Volume

1111111 = +6dB … 1dB steps down to 0110000 = -73dB 0101111 to 0000000 = Analogue MUTE

8 SPKVU N/A Speaker Volume Update

Writing a 1 to this bit will cause left and right speaker volumes to be updated (SPKLVOL and SPKRVOL)

7 SPKRZC 0 Right Speaker Zero Cross Enable

1 = Change gain on zero cross only 0 = Change gain immediately

6:0 SPKRVOL[6:0] 0000000 SPK_RP/SPK_RN Volume

1111111 = +6dB … 1dB steps down to 0110000 = -73dB 0101111 to 0000000 = Analogue MUTE

8:7 00 Reserved

6 MOUTVOL 1 Mono Output Mixer Volume Control

0 = 0dB 1 = -6dB

5:0 000000 Reserved

8:7 00 Reserved

6:4 LIN3BOOST[2:0] 000 LINPUT3 to Boost Mixer Gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

3:1 LIN2BOOST[2:0] 000 LINPUT2 to Boost Mixer Gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

0 0 Reserved

8:7 00 Reserved

6:4 RIN3BOOST[2:0] 000 RINPUT3 to Boost Mixer Gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

3:1 RIN2BOOST[2:0] 000 RINPUT2 to Boost Mixer Gain

000 = Mute 001 = -12dB ...3dB steps up to 111 = +6dB

0 0 Reserved

8 0 Reserved R45 (2Dh)

7 LB2LO 0 Left Input Boost Mixer to Left Output Mixer

0 = Disable (Mute) 1 = Enable Path

Analogue Outputs

Output Signal Path

Input Signal Path

Output Signal Path

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Preliminary Technical Data WM8960

REGISTER

ADDRESS

R46 (2Eh) Right

Bypass

R47 (2Fh) Power

Mgmt (3)

R48 (30h) Additional

Control (4)

BIT LABEL DEFAULT DESCRIPTION REFER TO

6:4 LB2LOVOL[2:0] 101 Left Input Boost Mixer to Left Output Mixer

Volume 000 = 0dB ...(3dB steps) 111 = -21dB

3:0 0000 Reserved

8 0 Reserved

7 RB2RO 0 Right Input Boost Mixer to Right Output Mixer

0 = Disable (Mute) 1 = Enable Path

6:4 RB2ROVOL[2:0] 101 Right Input Boost Mixer to Right Output Mixer

Volume 000 = 0dB ...(3dB steps) 111 = -21dB

3:0 0000 Reserved

8:6 000 Reserved

5 LMIC 0 Left Channel Input PGA Enable

0 = PGA disabled 1 = PGA enabled (if AINL = 1)

4 RMIC 0 Right Channel Input PGA Enable

0 = PGA disabled 1 = PGA enabled (if AINR = 1)

3 LOMIX 0 Left Output Mixer Enable Control

0 = Disabled 1 = Enabled

2 ROMIX 0 Right Output Mixer Enable Control

0 = Disabled 1 = Enabled

1:0 00 Reserved

8 0 Reserved

7 GPIOPOL 0 GPIO Polarity Invert

0 = Non inverted 1 = Inverted

6:4 GPIOSEL[2:0] 000 ADCLRC/GPIO1 GPIO Function Select:

000 = Jack detect input 001 = Reserved 010 = Temperature ok 011 = Debounced jack detect output 100 = SYSCLK output 101 = PLL lock 110 = Logic 0 111 = Logic 1

3:2 HPSEL[1:0] 00 Headphone Switch Input Select

0X = GPIO1 used for jack detect input (Requires ADCLRC pin to be configured as a GPIO)

10 = JD2 used for jack detect input 11 = JD3 used for jack detect input

1 TSENSEN 1 Temperature Sensor Enable

0 = Temperature sensor disabled 1 = Temperature sensor enabled

0 MBSEL 0 Microphone Bias Voltage Control

0 = 0.9 * AVDD 1 = 0.65 * AVDD

Output Signal Path

Input Signal Path

Output Signal Path

General Purpose Input / Output

Headphone Jack Detect

Thermal Shutdown

Input Signal Path

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WM8960 Preliminary Technical Data

REGISTER

ADDRESS

R49 (31h) Class D

Control (1)

R50 (32h) 8:0 001001101 Reserved

R51 (33h) Class D

Control (2)

R52 (34h) PLL (1)

PLL (2)

PLL (3)

PLL (4)

BIT LABEL DEFAULT DESCRIPTION REFER TO

8 0 Reserved

7:6 SPK_OP_EN[1:0] 00 Enable Class D Speaker Outputs

00 = Off 01 = Left speaker only 10 = Right speaker only 11 = Left and right speakers enabled

5:0 110111 Reserved

8:6 010 Reserved

5:3 DCGAIN[2:0] 000 DC Speaker Boost (Boosts speaker DC output

level by up to 1.8 x on left and right channels) 000 = 1.00x boost (+0dB) 001 = 1.27x boost (+2.1dB) 010 = 1.40x boost (+2.9dB) 011 = 1.52x boost (+3.6dB) 100 = 1.67x boost (+4.5dB) 101 = 1.8x boost (+5.1dB) 110 to 111 = Reserved

2:0 ACGAIN[2:0] 000 AC Speaker Boost (Boosts speaker AC output

signal by up to 1.8 x on left and right channels) 000 = 1.00x boost (+0dB) 001 = 1.27x boost (+2.1dB) 010 = 1.40x boost (+2.9dB) 011 = 1.52x boost (+3.6dB) 100 = 1.67x boost (+4.5dB) 101 = 1.8x boost (+5.1dB) 110 to 111 = Reserved

8:6 OPCLKDIV[2:0] 000 SYSCLK Output to GPIO Clock Division ratio

000 = SYSCLK 001 = SYSCLK / 2 010 = SYSCLK / 3 011 = SYSCLK / 4 100 = SYSCLK / 5.5 101 = SYSCLK / 6

5 SDM 0 Enable Integer Mode

0 = Integer mode 1 = Fractional mode

4 PLLPRESCALE 0 Divide MCLK by 2 before input to PLL

0 = Divide by 1 1 = Divide by 2

3:0 PLLN[3:0] 1000 Integer (N) part of PLL input/output frequency

ratio. Use values greater than 5 and less than

13.

8 0 Reserved R53 (35h)

7:0 PLLK[23:16] 00110001 Fractional (K) part of PLL1 input/output

frequency ratio (treat as one 24-digit binary number).

8 0 Reserved R54 (36h)

7:0 PLLK[15:8] 00100110 Fractional (K) part of PLL1 input/output

frequency ratio (treat as one 24-digit binary number).

8 0 Reserved R55 (37h)

7:0 PLLK[7:0] 11101001 Fractional (K) part of PLL1 input/output

frequency ratio (treat as one 24-digit binary number).

PTD, September 2006, Rev 2.0

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Enabling the Outputs

Analogue Outputs

General Purpose Input / Output

Clocking and Sample Rates

80

Preliminary Technical Data WM8960

DIGITAL FILTER CHARACTERISTICS

PARAMETER TEST CONDITIONS MIN TYP MAX UNIT

ADC Filter

Passband

Passband Ripple +/- 0.05 dB

Stopband 0.546s

Stopband Attenuation f > 0.546 fs -60 dB

DAC Normal Filter

Passband

Passband Ripple 0.454 fs +/- 0.03 dB

Stopband 0.546 fs

Stopband Attenuation F > 0.546 fs -50 dB

DAC Sloping Stopband Filter

Passband

Passband Ripple 0.25 fs +/- 0.03 dB

Stopband 1 0.546 fs 0.7 fs

Stopband 1 Attenuation f > 0.546 fs -60 dB

Stopband 2 0.7 fs 1.4 fs

Stopband 2 Attenuation f > 0.7 fs -85 dB

Stopband 3 1.4 fs

Stopband 3 Attenuation F > 1.4 fs -55 dB

DAC FILTERS ADC FILTERS

Mode Group Delay Mode Group Delay

Normal 18 / fs Normal 18 / fs

Sloping Stopband 18 / fs

+/- 0.05dB 0 0.454 fs

-6dB 0.5fs

+/- 0.03dB 0 0.454 fs

-6dB 0.5 fs

+/- 0.03dB 0 0.25 fs

+/- 1dB 0.25 fs 0.454 fs

-6dB 0.5 fs

ADC FILTER RESPONSES

10

0

0.1

0.2

0.3

0.4

0.51

0.61

0.71

0.81

0.91

1.01

1.11

1.21

1.31

1.42

1.52

1.62

1.72

1.82

1.92

2.02

2.12

2.22

2.33

2.43

2.53

2.63

2.73

2.83

2.93

3.03

3.13

3.24

3.34

3.44

3.54

3.64

3.74

3.84

-10

-30

-50

-70

Magnitude (dB)

-90

-110

-130

-150

Frequency (fs)

3.94

0.1

0.08

0.06

0.04

0.02

0

0.00 0.25

-0.02

-0.04

-0.06

-0.08

-0.1

Figure 38 ADC Digital Filter Frequency Response Figure 39 ADC Digital Filter Ripple

Magnitude (dB): Passband Ripple

Frequency

PTD, September 2006, Rev 2.0

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81

WM8960 Preliminary Technical Data

DAC FILTER RESPONSES

DAC STOPBAND ATTENUATION

The DAC digital filter type is selected by the DACSLOPE register bit as shown in Table 50.

REGISTER

ADDRESS

R6 (06h) ADC and DAC

Control (2)

Table 50 DAC Filter Selection

BIT LABEL DEFAULT DESCRIPTION

1 DACSLOPE 0 Selects DAC filter characteristics

0 = Normal mode 1 = Sloping stopband mode

MAGNITUDE(dB)

10

-10

0 0.5 1 1.5 2 2.5 3

-30

-50

-70

-90

-110

-130

-150

Frequency (fs)

Figure 40 DAC Digital Filter Frequency Response (Normal Mode)

MAGNITUDE(dB)

10

-10

0 0.5 1 1.5 2 2.5 3

-30

-50

-70

-90

-110

-130

-150

Frequency (fs)

MAGNITUDE(dB)

0.04

0.035

0.03

0.025

0.02

0.015

0.01

0.005

0

0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5

-0.005

Frequency (fs)

Figure 41 DAC Digital Filter Ripple (Normal Mode)

MAGNITUDE(dB)

0.05

0

0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5

-0.05

-0.1

-0.15

-0.2

-0.25

-0.3

-0.35

-0.4

-0.45

-0.5

Freque ncy (fs)

Figure 42 DAC Digital Filter Frequency Response (Sloping Stopband Mode)

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Figure 43 DAC Digital Filter Ripple (Sloping Stopband Mode)

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Preliminary Technical Data WM8960

DE-EMPHASIS FILTER RESPONSES

MAG NITUDE(dB)

0

0 5000 10000 15000 20000

-1

-2

-3

-4

-5

-6

-7

-8

0.3

0.25

0.2

0.15

0.1

0.05

0

0 2000 4000 6000 8000 10000 12000 14000 16000 18000

-0.05

-0.1

-0.15

MAGNITUDE(dB)

Frequency (Hz)

-9

-10

Frequen cy (Hz)

Figure 44 De-Emhpasis Digital Filter Response (32kHz) Figure 45 De-Emphasis Error (32kHz)

MAGNITUDE(dB)

0

0 500 0 100 00 15000 20000 25000

-1

-2

-3

-4

-5

-6

-7

-8

-9

-10

Freque ncy (Hz)

0.2

0.15

0.1

0.05

0

0 5000 10000 15000 20000 25000

-0.05

-0.1

MAGNITUDE(dB)

Frequen cy (Hz)

Figure 46 De-Emhpasis Digital Filter Response (44.1kHz) Figure 47 De-Emphasis Error (44.1kHz)

MAGNITUDE(dB)

0

0 5000 10000 15000 20000 25000 30000

-2

-4

-6

-8

-10

-12

Frequency (Hz)

0.15

0.1

0.05

0

0 5000 10000 15000 20000 25000 30000

-0.05

-0.1

-0.15

MAGNITUDE(dB)

Frequency (Hz)

Figure 48 De-Emhpasis Digital Filter Response (48kHz) Figure 49 De-Emphasis Error (48kHz)

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83

WM8960 Preliminary Technical Data

APPLICATIONS INFORMATION

RECOMMENDED EXTERNAL COMPONENTS

SPEAKER SELECTION

For filterless operation, it is important to select a speaker with appropriate internal inductance. The internal inductance and the speaker's load resistance create a low-pass filter with a cut-off frequency of:

f

e.g. for an 8Ω speaker and required cut-off frequency of 20kHz, the speaker should be chosen to have an inductance of:

L = R

8Ω speakers typically have an inductance in the range 20µH to 100µH. Care should be taken to ensure that the cut-off frequency of the speaker's internal filtering is low enough to prevent speaker damage. The class D outputs of the WM8960 operate at much higher frequencies than is recommended for most speakers, and the cut-off frequency of the filter should be low enough to protect the speaker.

= RL / 2πL

c

/ 2πfc = 8Ω / 2π * 20kHz = 64µH

L

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Preliminary Technical Data WM8960

Figure 50 Speaker Equivalent Circuit

PCB LAYOUT CONSIDERATIONS

The efficiency of the speaker drivers is affected by the series resistance between the WM8960 and the speaker (e.g. inductor ESR) as shown in Figure 51. This resistance should be as low as possible to maximise efficiency.

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Figure 51 Speaker Connection Losses

The distance between the WM8960 and the speakers should be kept to a minimum to reduce series resistance, and also to reduce EMI. Further reductions in EMI can be achieved by additional passive filtering and/or shielding as shown in Figure 52. When additional passive filtering is used, low ESR components should be chosen to minimise series resistance between the WM8960 and the speaker, maximising efficiency.

LC passive filtering will usually be effective at reducing EMI at frequencies up to around 30MHz. To reduce emissions at higher frequencies, ferrite beads placed as close to the device as possible will be more effective.

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WM8960 Preliminary Technical Data

Figure 52 EMI Reduction Techniques

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Preliminary Technical Data WM8960

PACKAGE DIMENSIONS

FL: 32 PIN QFN PLASTIC PACKAGE 5

A3

C

25

24

A

17

SEATING PLANE

A3

D2

EXPOSED GROUND PADDLE

16 15

B

e

BOTTOM VIEW

SIDE VIEW

b

Exposed lead Half etch ti e bar

32

6

9

W

H

X 5 X 0.9 mm BODY, 0.50 mm LEAD PITCH

DETAIL 1

L

1

8

1

b

bbb

DETAIL 2

T

G

DETAIL 2

E2

M

A

BC

A

A1

INDEX AREA

4

(D/2 X E/2)

Caaa

2 X

Caaa

2 X

Cccc

5

C0.08

EXPOSED GROUND PADDLE

DM033.D

D

E

TOP VIEW

R = 0.3MM

DETAIL 1

NOM

0.90

0.02

0.20 REF

0.25

5.00

3.45

5.00

0.50 BSC

0.213

0.1

Dimensions (mm)

MAX NOTE

1.00

0.05

0.300.18

3.55

3.553.453.30

1

2

Symbols

A A1 A3

b

D

D2

E E2

e

G

H

L

T

MIN

0.80

0

3.30

0.30 0.40 0.50

W 0.2

Tolerances of Form and Position

aaa bbb ccc

REF:

NOTES:

1. DIMENSION b APPLIES TO METALLIZED TERMINAL AND IS MEASURED BETWEEN 0.15 mm AND 0.30 mm FROM TERMINAL TIP.

2. FALLS WITHIN JEDEC, MO-220, VARIATION VHHD-5.

3. ALL DIMENSIONS ARE IN MILLIMETRES.

4. THE TERMINAL #1 IDENTIFIER AND TERMINAL NUMBERING CONVENTION SHALL CONFORM TO JEDEC 95-1 SPP-002.

5. COPLANARITY APPLIES TO THE EXPOSED HEAT SINK SLUG AS WELL AS THE TERMINALS.

6. REFER TO APPLICATION NOTE WAN_0118 FOR FURTHER INFORMATION REGARDING PCB FOOTPRINTS AND QFN PACKAGE SOLDERING.

7. THIS DRAWING IS SUBJECT TO CHANGE WITHOUT NOTICE.

JEDEC, MO-220, VARIATION VHHD-5.

0.15

0.10

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WM8960 Preliminary Technical Data

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 W olfson 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

26 Westfield Road

Edinburgh

EH11 2QB

United Kingdom

Tel :: +44 (0)131 272 7000

Fax :: +44 (0)131 272 7001

Email :: [email protected]

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Wolfson WM8960GEFL Schematic

WM8960

Stereo CODEC with 1W Stereo Class D Speaker Drivers and

DESCRIPTION

FEATURES

APPLICATIONS

TABLE OF CONTENTS

PIN CONFIGURATION

ORDERING INFORMATION

PIN DESCRIPTION

ABSOLUTE MAXIMUM RATINGS

RECOMMENDED OPERATING CONDITIONS

ELECTRICAL CHARACTERISTICS

OUTPUT PGA GAIN

TYPICAL POWER CONSUMPTION

SIGNAL TIMING REQUIREMENTS

SYSTEM CLOCK TIMING

AUDIO INTERFACE TIMING – MASTER MODE

AUDIO INTERFACE TIMING – SLAVE MODE

CONTROL INTERFACE TIMING – 2-WIRE MODE

INTERNAL POWER ON RESET CIRCUIT

DEVICE DESCRIPTION

INTRODUCTION

INPUT SIGNAL PATH

ANALOGUE TO DIGITAL CONVERTER (ADC)

AUTOMATIC LEVEL CONTROL (ALC)

OUTPUT SIGNAL PATH

ANALOGUE OUTPUTS

ENABLING THE OUTPUTS

HEADPHONE OUTPUT

CLASS D SPEAKER OUTPUTS

VOLUME UPDATES

HEADPHONE JACK DETECT

THERMAL SHUTDOWN

GENERAL PURPOSE INPUT/OUTPUT

DIGITAL AUDIO INTERFACE

AUDIO INTERFACE CONTROL

CLOCKING AND SAMPLE RATES

CONTROL INTERFACE

POWER MANAGEMENT

REGISTER MAP

REGISTER BITS BY ADDRESS

DIGITAL FILTER CHARACTERISTICS

ADC FILTER RESPONSES

DAC FILTER RESPONSES

DE-EMPHASIS FILTER RESPONSES

APPLICATIONS INFORMATION

RECOMMENDED EXTERNAL COMPONENTS

PACKAGE DIMENSIONS

IMPORTANT NOTICE

ADDRESS: