DA7219-02VB6

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Audio codec with Advanced Accessory Detect

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DA7219 DIALOG

Document overview

Manufacturer or author: Provided By www.digicamel.com(FREE DATASHEET DOWNLOAD SITE)
PDF pages: 123

Technical content

Datasheet sections

1 Terms and definitions
2 References
3 Block diagram
4 Pinout
4.1 Microphone pins
4.2 Accessory detect pins
4.3 Interface input pins
4.4 Interface output pins
4.5 Interface bidirectional pins
4.6 Headphone output pins
4.7 Charge pump pins
4.8 References
4.9 Supply pins
4.10 Ground pins
5 Absolute maximum ratings
6 Recommended operating conditions
7 Electrical characteristics
8 Timing characteristics
9 Functional description
9.1 Device operating modes
9.2 Input paths
9.2.1 Microphone input
9.2.1.1 Microphone bias
9.2.1.2 Microphone amplifier
9.2.1.3 Input amplifiers
9.2.2 Analogue to Digital Converter (ADC)
9.3 Digital engine
9.3.1 Input processing
9.3.1.1 ADC digital gain
9.3.1.2 High-pass filter
9.3.1.3 Automatic Level Control (ALC)
9.3.2 Sidetone processing
9.3.3 Tone generator
9.3.4 Digital router
9.3.5 System controller
9.3.6 Output processing
9.3.6.1 DAC digital gain
9.3.6.2 High-pass filter

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 1 of 123 © 2015 Dialog Semiconductor General description The DA7219 is an audio codec with Advanced Accessory Detection (AAD). DA7219 contains a mono microphone to ADC path, and a stereo DAC to HP path. The AAD block supports three-pole (headphone) and four-pole (headset) jacks, and allows the automatic pin order switching of MIC/GND on CTIA or OMTP headsets. It also supports automatic button detection. Key features ■ High performance mono microphone to ADC record path with 90 dB SNR □ ADC digital filters with audio and voice mode high-pass characteristics □ A low -noise microphone bias regulator with programmable output ■ High performance stereo DAC to headphone playback path with 100 dB SNR □ DAC digital filters with audio and v oice mode high -pass cut -off and 5 -band equaliser ■ Advanced Accessory Detection supports □ Three/four pole jack detection □ MIC/GND polarity switching □ Multiple button detection □ Headphone impedance testing ■ A microphone input with ALC (automatic level control) ■ Digital Sidetone path with gain ■ Digital tone generator ■ System controller for simplified pop -free start-up and shutdown ■ Single interface 24 kHz ADC/48 kHz DAC mixed sample rates supported ■ Shutdown mode for very low current consumption during standby ■ Phase l ocked loop with sample rate tracking to generate system clock ■ 4-wire digital audio interface with support for I2S, TDM and other audio formats ■ 2-wire I2C compatible with support for High Speed mode up to 3.4 MHz ■ WL-CSP RouteEasy™ package for low cost PCB manufacture

Applications

■ Chromebooks ■ Portable audio applications ■ Tablets and eBooks ■ Headphone accessories ■ Remote controllers ■ Gaming controllers

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 2 of 123 © 2015 Dialog Semiconductor System diagram DA7219 Audio Jack Audio switch HP Amp Codec SoC Digital Class DDigital Digital Analog Analog Stereo Speaker Headset DMICDigital Class D Figure 1: DA7219 in a digital distributed system

1 Terms and definitions

ADC Analogue to Digital Converter ALC Automatic Level Control CTIA Cellular Telecommunications Industry Association, (now known as ‘The Wireless Association’) DAC Digital to Analogue Converter DAI Digital Audio Interface DMIC Digital Microphone DTMF Dual Tone Multi-Frequency FS Sample Rate I2C Inter-Integrated Circuit interface I2S Inter-IC Sound LDO Low Dropout Regulator MCLK Master Clock OMTP Open Mobile Terminals Platform PC Program Counter PGA Programmable Gain Amplifier PLL Phase Locked Loop PSRR Power Supply Rejection Ratio RC Resistance-Capacitance SC System Controller SDM Sigma Delta Modulator SNR Signal to Noise Ratio SRM Sample Rate Matching SWG Sine Wave Generator TDM Time Division Multiplexing THD+N Total Harmonic Distortion plus Noise VCO Voltage-Controlled Oscillator

2 References

[1] Android wired audio headset specification (v1.1) (https://source.android.com/accessories/headset/specification.html )

3 Block diagram

GND_CP MCLK DACREF VMID GND_HP MIC_N MIC_P VREF HP_R MICBIAS VDD_MIC HPCSP HPCSN HPCFN HPCFP GND MIC SLEEVE RING2 HP_L Input Filters (High-Pass, ALC) Output Filters (High-Pass, 5-BandEQ)System Controller JACKDET Tone Generator Accessory Detect Sidetone Path I2C CONTROL INTERFACE VDD SDA SCL nIRQ BCLK WCLK DATIN DATOUT VDD_IO Impedance Detection DIGITAL ENGINE SLEEVE_SENSE RING2_SENSE Figure 2: DA7219 block diagram

4 Pinout

MIC_N HPCFP VDD_IO HPCFN VDD VDD_IO WCLK DATIN DATOUT nIRQ SCL SCL SDA MICHPCSN GND_CP RING2_ SENSE HP_L HP_R SLEEVE_ SENSE DACREF VREF VMID GND RING2 SLEEVE VDD_MIC MICBIAS MIC_P JACKDET HPCSP 1 2 3 4 5 6 8 9 10 11 12 13 14 15 16 A B C D 1 2 3 4 5 6 8 9 10 11 12 13 14 15 167 MCLK Analogue Digital Power Ground View from above “Live Bug” BCLK RING2 GND_HP Figure 3: DA7219 ballout diagram Table 1: DA7219 pin description Pin no. Pin name Type (Table 2)

Description

B16 MIC_P AI Differential analogue microphone 1 input (Pos) A15 MIC_N AI Differential analogue microphone 1 input (Neg) B14 MICBIAS AO Microphone bias output Accessory Detect D16 JACKDET DI Jack Detect Input from socket A11 SLEEVE AIO Socket Sleeve (configured as MIC or GND) C13 RING2 AIO Socket Ring 2 (configured as GND or MIC) B6 SLEEVE_ SENSE AIO Socket Sleeve (Sense) B4 RING2_ SENSE AIO Socket Ring 2 (Sense) C15 MIC AIO Microphone DC input Headphone Outputs A3 HP_L AO Single-ended headphone output (Left) A5 HP_R AO Single-ended headphone output (Right) Charge Pump A1 HPCSP AIO Charge pump reservoir capacitor (Positive) C1 HPCSN AIO Charge pump reservoir capacitor (Negative) D2 HPCFP AIO Charge pump flying capacitor (Positive) C3 HPCFN AIO Charge pump flying capacitor (Negative)

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 9 of 123 © 2015 Dialog Semiconductor Pin no. Pin name Type (Table 2) D14 SDA DIOD I2C bidirectional data D12 SCL DI I2C clock D10 nIRQ DIOD Interrupt output (open drain active low) C7 DATIN DIO DAI data input to DA7219 C9 DATOUT DIO DAI data output from DA7219 D6 BCLK DIO DAI bit clock D8 WCLK DIO DAI word clock C11 MCLK DI Master clock input References B8 DACREF AIO DAC reference decoupling capacitor A9 VMID AIO Mid-rail reference decoupling capacitor A7 VREF AIO Bandgap reference decoupling capacitor Supplies C5 VDD AI Main analogue and digital supply A13 VDD_MIC AI Supply for MICBIAS LDO D4 VDD_IO AI Supply for digital interfaces Grounds B2 GND_CP AI Ground B10 GND AI Ground B12 GND_HP AI Ground Table 2: Pin type definition Pin type Description Pin type Description DI Digital Input AI Analogue Input DIO Digital Input/Output AO Analogue Output DIOD Digital Input/Output open Drain AIO Analogue Input/Output

4.1 Microphone pins

Pin MIC_P MIC_P is the positive differential input for the analogue microphone channel. It can be used as a single-ended input if MIC_N is grounded (see Figure 7). Pin MIC_N MIC_N is the negative differential input for the analogue microphone channel. It should be grounded when using a single-ended analogue microphone configuration. Pin MICBIAS MICBIAS is the internally generated microphone supply. This must be decoupled with a 1 µF capacitor.

4.2 Accessory detect pins

JACKDET is used to signal to the device when the Jack is fully inserted into the 3.5 mm jack (or alternative) socket If not required it should be left unconnected. Pin SLEEVE Sleeve is tested during the sense stage and then configured as either the headset microphone input or the headset ground connection Pin RING2 RING2 is tested during the sense stage and then configured as either the headset microphone input or the headset ground connection. Pin SLEEVE_SENSE SLEEVE sense line to guarantee accuracy over distance, cables and connectors. Pin RING2_SENSE RING2 sense line to guarantee accuracy over distance, cables and connectors . Pin MIC MIC is the DC input for the analogue accessory detect.

4.3 Interface input pins

MCLK is the master clock input pin. It is used as the main system clock either directly or via the PLL. Pin SCL SCL is the Control Interface (I2C) clock input and is used in conjunction with SDA to control the device. Pin DATIN DATIN is the data input pin which forms part of the Digital Audio Interface. It is used to present audio playback data to the device.

4.4 Interface output pins

nIRQ is the open drain active-low interrupt output to alert the host to either an accessory or a level-detect event. Pin DATOUT DATOUT is the data output pin which forms part of the Digital Audio Interface. It is used to present audio record data to the host.

4.5 Interface bidirectional pins

SDA is the Control Interface (I2C) data input/output and is used in conjunction with SCL to control the device. Pin BCLK BCLK is the bit clock input/output pin which forms part of the Digital Audio Interface (DAI). It is used to clock audio data bits into or out from the device or both. Pin WCLK WCLK is the word clock input/output pin which forms part of the DAI. It is used to indicate whether the data bits belong to the left or right audio channel.

4.6 Headphone output pins

Pin HP_L 3.6.2 HP_L is the left-channel headphone output. It is ground-centred so the headphone speaker can be connected directly between HP_L and ground. Pin HP_R HP_R is the right-channel single ended headphone output. It is ground-centred so the headphone speaker can be connected directly between HP_R and ground.

4.7 Charge pump pins

HPCSP is the positive output from the headphone charge pump. It should be connected to GND_CP via a reservoir capacitor. Pin HPCSN HPCSN is the negative output from the headphone charge pump. It must be connected to GND_CP via a reservoir capacitor. Pin HPCFP HPCFP is one of the flying capacitor connections required by the headphone charge pump. It must be connected to HPCFN via a capacitor. Pin HPCFN HPCFN is one of the flying capacitor connections required by the headphone charge pump. It must be connected to HPCFP via a capacitor.

4.8 References

VMID is mid-rail reference decoupling capacitor connection. Pin DACREF DACREF is the DAC reference decoupling capacitor connection. Pin VREF VREF is the bandgap reference decoupling capacitor connection.

4.9 Supply pins

VDD is main analogue supply pin. It supplies all the analogue circuits except the MICBIAS output and the HPAMP outputs. Pin VDD_IO VDD_IO is the supply pin for the digital input/output signals. Pin VDD_MIC VDD_MIC is the supply pin for the MICBIAS.

4.10 Ground pins

GND is the main analogue Ground pin. It is the Ground connection for all analogue circuits with the exception of the charge pump. Pin GND_CP GND_CP is the Ground pin for the charge pump and the digital engine. Pin GND_HP GND_HP is the ground point for the headset. When a headset is connected this pin is automatically connected internally to either RING2 or SLEEVE.

5 Absolute maximum ratings

Table 3: Absolute maximum ratings Parameter Description Conditions (Note 1) Min Max Unit Storage Temperature –65 +165 °C Operating Temperature –40 +85 °C VDD Main supply voltage –0.3 +2.75 V VDD_IO Digital IO supply voltage –0.3 +5.5 V VDD_MIC Microphone bias supply voltage –0.3 +5.5 V Digital IO pins SDA, SCL, BCLK, WCLK, DATIN, DATOUT, MCLK –0.3 VDD_IO + 0.3 V Accessory detect pins JACKDET –0.3 VDD + 0.3 V RING2, SLEEVE, MIC, RING2_SENSE, SLEEVE_SENSE –0.3 VDD_MIC + 0.3 V Analogue input pins MIC_P, MIC_N –0.3 VDD + 0.3 V Package thermal resistance °C/W ESD susceptibility Human body model (HBM) 2 kV Charged device model (CDM) 500 V Note 1 Stresses beyond those listed under ‘Absolute maximum ratings’ may cause permanent damage to the device. These are stress ratings only, so functional operation of the device at these or any other conditions beyond those indicated in the operational sections of the specification are not implied. Exposure to absolute maximum rating conditions for extended periods may affect device reliability. Note 2 All figures are to JEDEC specifications

6 Recommended operating conditions

Table 4: Recommended operating conditions Parameter Description Conditions Min Typ Max Unit Operating temperature –25 +85 °C VDD Main supply voltage 1.7 +2.5 V VDD_IO Digital IO supply voltage 1.5 +3.6 V VDD_MIC Microphone bias supply voltage 1.8 +3.6 V

7 Electrical characteristics

Unless otherwise stated, test conditions are as follows: VDD = VDD_IO = 1.8 V, VDD_MIC = 3.3 V, MCLK = 12.288 MHz, SR = 48 kHz, PLL = Bypass Mode. Table 5: Power consumption Description Conditions Min Typ Max Unit Powerdown mode 10 µA Digital playback to Headphone, no load DAC to HP_L/R, quiescent 3.4 mW Digital playback to Headphone, with load DAC to HP_L/R, 16Ω load, 0.1 mW at 0 dBFS 7.5 mW Microphone stereo record MIC P/N to ADCL/R 2.75 mW Microphone stereo record and digital playback to Headphone, no load MIC P/N to ADCL/R and DACL/R to HP_L/R, quiescent 4.8 mW Microphone stereo record and digital playback to Headphone, with load MIC P/N to ADCL/R and DACL/R to HP_L/R, 16 Ω load, 0.1 mW at 0 dBFS 8.9 mW Table 6: Electrical characteristics: Microphone bias Parameter Description Conditions Min Typ Max Unit VMICBIAS Bias voltage No load, VDD_MIC > VMICBIAS + 200 mV 1.8 2.9 V IBIAS Output current Output voltage drop < 50 mV 2 mA PSRR Power supply rejection ratio

20 Hz to 2 kHz

VMICBIAS ≤ 2.2 V 5 µVRMS Table 7: Electrical characteristics: micamp Parameter Description Conditions Min Typ Max Unit Full-scale input signal 0 dB, singled-ended 0 dB gain, differential 0.8×VDD 1.6×VDD VPP Input resistance Single-ended 12 15 18 kΩ Programmable gain −6 36 dB Gain step size 6 dB Absolute gain accuracy 0 dB @ 1 kHz -1.0 1.0 dB Gain step error 20 Hz to 20 kHz -0.1 0.1 dB Input noise level Inputs connected to GND, 24 dB gain, input-referred, A-weighted 5 µVRMS Amplitude ripple 20 Hz to 20 kHz -0.5 0.5 dB PSRR Power supply rejection ratio

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 15 of 123 © 2015 Dialog Semiconductor Table 8: Electrical characteristics: mixinamp Parameter Description Conditions Min Typ Max Unit VMAX Full-scale input signal 0 dB gain 1.6×VDD VPP Programmable gain −4.5 18 dB Gain step size 1.5 dB Absolute gain accuracy 0 dB @ 1 kHz -1.0 1.0 dB Gain step error 20 Hz to 20 kHz -0.1 0.1 dB Amplitude ripple 20 Hz to 20 kHz -0.5 0.5 dB PSRR Power supply rejection ratio Table 9: Electrical characteristics: adc_mono Parameter Description Conditions Min Typ Max Unit VMAX Full-scale input signal 0 dBFS digital output level 1.6×VDD VPP SNR Signal to Noise Ratio A-weighted 90 dB THD+N Total harmonic distortion plus noise -1 dBFS analogue input level -85 dB In-band spurious 0 dBFS analogue input level -85 dB PSRR Power supply rejection ratio Table 10: Electrical characteristics: dac_stereo Parameter Description Conditions Min Typ Max Unit VMAX Full-scale output signal 0 dBFS digital input level 1.6×VDD VPP SNR Signal to Noise Ratio A-weighted 100 dB THD+N Total harmonic distortion plus noise -1 dBFS digital input level -90 dB PSRR Power supply rejection ratio

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 16 of 123 © 2015 Dialog Semiconductor Table 11: Electrical characteristics: audio_hpamp_stereo Parameter Description Conditions Min Typ Max Unit VMAX Full-scale output signal No load 1.6×VDD VPP DC output offset −30 dB gain 250 µV Maximum output power per channel VDD = 1.8 V, THD < 0.1%, RLOAD = 16 Ω, 1 kHz 27 mWRMS Maximum output power per channel VDD = 2.5 V, THD < 0.1%, RLOAD = 16 Ω, 1 kHz 45 mWRMS Load resistance Single-ended mode 13 16 Ω Load capacitance 500 pF Load inductance 400 µH SNR Signal to Noise Ratio VDD = 1.8 V, 0 dB gain 98 dB VDD = 2.5 V, 0 dB gain 100 dB VNOISE Output noise level 20 Hz to 20 kHz, <20 dB gain 2.5 µVRMS THD+N Total harmonic distortion plus noise VDD = 1.8 V, RLOAD = 16 Ω, -5 dBFS, 1 kHz −75 dB Programmable gain −57 6 dB Gain step size 1.0 dB Absolute gain accuracy 0 dB @ 1 kHz −0.8 0.8 dB Left/right gain mismatch 20 Hz to 20 kHz −0.1 0.1 dB Gain step error 20 Hz to 20 kHz −0.1 0.1 dB Amplitude ripple 20 Hz to 20 kHz −0.5 0.5 dB Mute attenuation −70 dB PSRR Power supply rejection ratio Table 12: Electrical characteristics: Input filters Parameter Description Conditions Min Typ Max Unit BPASS Pass band 0.45×FS Hz Pass band ripple Voice mode Music mode ±0.3 ±0.1 dB BSTOP Stop band FS ≤ 48 kHz FS = 88.2 or 96 kHz 0.56×FS 7×FS 3.5×FS Hz Stop band attenuation Voice mode Music mode dB Group delay Voice mode Music mode FS = 88.2 or 96 kHz 4.3/FS 18/FS 9/FS s

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 17 of 123 © 2015 Dialog Semiconductor Table 13: DAC filter specifications Symbol Parameter Conditions Min Typ Max Unit BPASS Pass band 0.45×FS Hz Pass band ripple Voice mode Music mode ±0.3 ±0.1 dB BSTOP Stop band FS ≤ 48 kHz FS = 88.2 or 96 kHz 0.56×FS 7×FS 3.5×FS Hz Stop band attenuation Voice mode Music mode 55 dB Group delay Voice mode Music mode FS = 88.2 or 96 kHz 4.3/FS 18/FS 9/FS s Group delay variation 20 Hz to 20 kHz 1 µs Left/right channel group delay mismatch 2 µs Table 14: Electrical characteristics: ALC Parameter Description Conditions Min Typ Max Unit Attack rate FS = 48 kHz 1.6 6500 dB/s Release rate FS = 48 kHz 1.6 1675 dB/s Hold time FS = 48 kHz 1.3 42300 ms Maximum threshold −94.5 0 dBFS Minimum threshold −94.5 0 dBFS Noise threshold −94.5 0 dBFS Threshold step size 1.5 dB Maximum overall gain 0 90 dB Maximum overall attenuation 0 90 dB Maximum analogue gain 0 36 dB Minimum analogue gain 0 36 dB Gain step size 1.5 dB Table 15: Electrical characteristics: Accessory detect Parameter Description Conditions Min Typ Max Unit Ring2 ground switch resistance 50 mΩ Sleeve ground switch resistance 50 mΩ

8 Timing characteristics

Table 16: Timing I/O voltage characteristics Parameter Description Conditions Min Typ Max Unit VIH SCL, SDA, Input High Voltage 0.7*VDD_IO V VIL SCL, SDA, Input Low Voltage 0.3*VDD_IO V VIH MCLK, BCLK, WCLK, DATIN, DATOUT Input High Voltage 0.7*VDD_IO V VIL MCLK, BCLK, WCLK, DATIN, DATOUT Input Low Voltage 0.3*VDD_IO V VOL @3 mA SDA Output Low Voltage 0.24 V

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 19 of 123 © 2015 Dialog Semiconductor SCL SDA STH CLKL CLKH DST TSSDHT Figure 4: I2C bus timing Table 17: I2C control bus (VDD_IO = 1.8 V) Parameter Description Conditions Min Typ Max Unit Bus free time STOP to START 500 ns Bus line capacitive load 150 pF Standard/Fast Mode SCL clock frequency 0 1000 kHz Start condition setup time 260 ns STH Start condition hold time 260 ns CLKL SCL low time 500 ns CLKH SCL high time 260 ns SCL rise/fall time Input requirement 1000 ns SDA rise/fall time Input requirement 300 ns DST SDA setup time 50 ns DHT SDA hold time 0 ns TSS Stop condition setup time 260 ns High-Speed Mode SCL clock frequency 0 3400 kHz Start condition setup time 160 ns STH Start condition hold time 160 ns CLKL SCL low time 160 ns CLKH SCL high time 60 ns SCL rise/fall time Input requirement 160 ns SDA rise/fall time Input requirement 160 ns DST SDA setup time 10 ns DHT SDA hold time 0 ns TSS Stop condition setup time 160 ns

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 20 of 123 © 2015 Dialog Semiconductor T tlCthC BCLK DATIN tf tsD DATOUT WCLK tr tdCW tdCDtdWD thD tsW thW Figure 5: Digital audio interface timing diagram Note 3 Diagram shown is valid for all modes except DSP. For DSP mode the BCLK signal is inverted Table 18: Digital audio interface timing (I2S/DSP in master/slave mode) Parameter Description Conditions (VDD_IO = 1.8 V) Min Typ Max Unit Input impedance DC impedance > 10 MΩ 300 1.0 2.5 Ω pF T BCLK period 75 ns tr BCLK rise time 8 ns tf BLCK fall time 8 ns thC BCLK high period 40 % 60 % T tlC BCLK low period 40 % 60 % T tdCW BCLK to WCLK delay -30 % +30 % T tdCD BCLK to DATOUT delay -30 % +30 % T thW WCLK high time DSP mode 100 % T Non-DSP mode Word length (Note 4) T tlW WCLK low time DSP mode 100 % T Non-DSP mode Word length (Note 5) T tsW WCLK setup time Slave mode 7 ns thW WCLK hold time Slave mode 2 ns tsD DATIN setup time 7 ns thD DATIN hold time 2 ns tdWD DATOUT to WCLK delay DATOUT is synchronised to BCLK Note 4 WCLK must be high for at least the word length number of BCLK periods Note 5 WCLK must be low for at least the word length number of BCLK periods

9 Functional description

DA7219 is a high-performance, low-power audio codec with in-built Advanced Accessory Detection (AAD). The AAD supports the detection of three-pole (headphone or lineout) or four-pole (headset) jacks, with automatic pin order switching of MIC/GND on CTIA and OMTP headsets. The DA7219 contains a mono analogue microphone-to-ADC path and a digital audio interface (DAI) for input and output. The DAC to headphone path has a ground centred, single ended stereo headphone output. The digital core has an input filter with a high pass filter, and automatic level control (ALC), while the output filter has a high pass filter, and a 5-band EQ. There is also a sidetone path with gain and a tone generator that supports Dual Tone Multi- Frequency (DTMF).

9.1 Device operating modes

The DA7219 codec has three operating modes:

DEEP SLEEP – There is no clocking in DEEP SLEEP mode and consequently no functionality available and no accessory detection is performed. The system will awake when system_active = 1.
SLEEP – In SLEEP mode and with micbias OFF, AAD performs jack detection and jack configuration detection. Any button press is detected, but identification of the button cannot be performed until micbias is ON. No clocking is performed in SLEEP mode, and playback and record are not supported.
ON – AAD performs full-function accessory detection. Playback and record are supported. All modes, their maximum current consumption, and functionality are listed in Table 19.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec-2015 22 of 123 © 2015 Dialog Semiconductor Table 19: System states, configuration, and current consumption Mode Configuration CODEC Typical current consumption Comments VDD + VDD_IO VDD_MIC system_ active MCLK PLL mode AAD config Internal ref osc Internal PLL Audio path AAD VDD + VDD_IO VDD_MIC OFF OFF OFF N/A N/A N/A N/A N/A N/A N/A N/A None None N/A DEEP SLEEP ON ON 0 N/A N/A N/A OFF OFF OFF OFF < 10 µA 0 Wake on system_active = 1. SLEEP ON ON 1 OFF OFF Buttons OFF ON OFF OFF (playback/ record not supported) Jack insertion, jack type and pin order <200 µA 0 All ON As above plus Button detection (without identification) <500 µA 0 All ON + micbias_en FULL DETECTION (as above but with button Identification) <300 µA <2 mA ON ON ON 1 OFF SRM - locks to WCLK if DAI is in slave mode ALL ON + micbias_en ON ON ON (Playback/ record etc) FULL DETECTION Dependent on use case (playback level into load etc) AAD uses clock on demand to save power ON (11.8 MHz or

12.288 MHz)

(Valid Freq 2-80 MHz) Normal - lock to MCLK OFF ON

9.2 Input paths

9.2.1 Microphone input

The DA7219 analogue input consists of one set of amplifiers and an ADC as shown in Figure 6. ADC MIC_N MIC_P to ADC filter MIC_1_AMP MIXIN_L_AMP Figure 6: Analogue inputs block diagram

9.2.1.1 Microphone bias

The device has a microphone bias output, which is a programmable voltage source that can be used to supply analogue microphones. The bias output can be independently programmed from 1.8 V to 2.9 V in 0.2 V steps using micbias1_level. The microphone bias level can only be changed while the associated micbias circuit is disabled (micbias1_en = 0). Table 20: Microphone bias settings micbias1_level Output voltage (V) in low noise mode 000 reserved 001 1.8 010 2.0 011 2.2 100 2.4 101 2.6 110 2.8 111 2.9

9.2.1.2 Microphone amplifier

MIC_P MIC_N MICBIAS MIC_P MIC_N MICBIAS MIC_P (b) Differential (c) Pseudo-differential (d) Single-ended MIC_N MICBIAS MIC_P MIC_N (a) Headset RING2_SENSE MIC SLEEVE_SENSE Figure 7: ECM microphone configurations The microphone amplifier can be configured in a) Headset mode, microphone amplifier is fully differential b) fully differential mode for improved common-mode noise rejection c) pseudo-differential mode d) single-ended mode (MIC_P or MIC_N) All configurations are illustrated in Figure 7. The configuration of the first microphone amplifier is specified using the MIC_1_CTRL register. It is enabled by setting the mic_1_amp_en bit, and is muted by setting the mic_1_amp_mute_en bit. The gain of the amplifier can be set in the range of –6 dB to +36 dB in 6 dB steps using mic_1_amp_gain (see Table 21:). Table 21: MIC_1_GAIN gain settings mic_1_amp_gain Amplifier gain (dB) 000 -6 001 0 010 6 011 12 100 18 101 24 110 30 111 36

9.2.1.3 Input amplifiers

The input amplifier provides an additional gain stage between the microphone amplifier (see section 0 and Figure 6) and the ADC input. The input amplifier is enabled by setting mixin_l_amp_ramp_en = 1. The gain can be set in the range of –4.5 dB to +18 dB in 1.5 dB steps using mixin_l_amp_gain. Gain updates can be synchronised with signal zero-crossings by setting mixin_l_amp_zc_en = 1. If no zero-crossing is detected within the timeout period of approximately 100 ms, the update is applied unconditionally.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 25 of 123 © 2015 Dialog Semiconductor As an alternative to zero-cross synchronisation, gain updates can be ramped through all intermediate values by setting mixin_l_amp_ramp_en = 1. This ramp setting overrides the settings of mixin_l_amp_zc_en. The amplifier can be muted using mixin_l_amp_mute_en. Table 22: MIXIN_L_GAIN settings mixin_l_amp_gain Amplifier gain (dB) 0000 -4.5 0001 -3.0 0010 -1.5 0011 0.0 0100 1.5 0101 3.0 0110 4.5 0111 6.0 1000 7.5 1001 9.0 1010 10.5 1011 12.0 1100 13.5 1101 15.0 1110 16.5 1111 18.0

9.2.2 Analogue to Digital Converter (ADC)

The DA7219 codec contains a high quality audio ADC. The ADC is clocked at a fixed rate of either 3.072 MHz or 2.8224 MHz, depending on the required input sample rate (SR). The DA7219 includes a low power 24-bit high quality audio ADC that supports sampling rates from 8 kHz to 96 kHz. The sample rate is specified using the SR register. The ADC can be enabled and disabled using adc_l_en. The ADC channels offer a configurable digital gain from -83.25 dB to +12 dB in 0.75 dB steps after the digital conversion. Individual gain settings can be programmed via the adc_l_digital_gain_status control. The currently active gain settings are stored in the ADC_L_GAIN_STATUS register. Muting, and the ramping of digital gain changes, can be controlled using the dedicated ADC_L_CTRL register. If the ramping is enabled using the control bit adc_l_ramp_en, the rate of the ramping is controlled using gain_ramp_rate in the GAIN_RAMP_CTRL register.

9.3 Digital engine

The DA7219 chip contains a digital engine that performs the signal processing and also provides overall system control. The input signals from the ADCs are passed to the input filter block. The filter block includes a high-pass filter for DC offset removal and wind noise suppression, and an automatic level control. The signals from the input filters are sent to the digital mixer where they can be combined with signals from the tone generator and the digital audio interface (DAI), and routed to the output filters and the DAI. The output filters contain a high-pass filter for DC offset removal, and a fixed 5-band equaliser to adjust the sound of the output signals. There is also a low latency sidetone path that can take one signal from the ADC and apply gain before passing the signal straight to the output filters. The filter paths are shown in more detail in Figure 9. Finally a system controller module is included to ensure correct sequencing of the e vents required to bring up and shut down signal paths without creating pops and clicks. DIGITAL ENGINE Accessory Detect Logic System Control Output Filters High Pass 5-Band EQ Sidetone Path with Gain I2C DAI Tone Generator and S-Ramp Input Filters High Pass MUX Charge Pump Control Clock Generation ALC Accessory Detect Analogue Audio Input PLL Charge Pumps Analogue Outputs DAI Pads I2C Pads Figure 8: Digital engine block diagram OUT_1_FILTER IN_1_FILTER CIC 5BEQ SDM DWA SDM DWA GAIN STAGE TONE GEN + SRAMP FSDM 4×FS 2×FS FS 4×FS FS 2×FS 4×FS 8×FS FSDMMUX DAI ADC_Audio_DATA DAC_1L_DATA DAC_1R_DATA DAI 5BEQ Accessory Detect ADC_AccDet_DATA JACKDET ALC X to Mic Gain ZeroCross from Mic Figure 9: Digital filters block diagram

9.3.1 Input processing

9.3.1.1 ADC digital gain

The ADC channels offer a configurable digital gain from -83.25 dB to +12 dB in 0.75 dB steps after the digital conversion. Individual gain settings can be programmed via the adc_l_digital_gain control. The currently active gain settings are stored in the ADC_L_GAIN_STATUS register.

9.3.1.2 High-pass filter

Any DC offset from the input path is removed via IIR filters (typically <2 Hz roll-off, configurable). After reset the filters for both channels are enabled by default, but can be disabled by clear ing adc_hpf_en. The cut-off frequency of the filters can be programmed using adc_audio_hpf_corner. To improve the quality of microphone recordings, the DA7219 provides a programmable high pass filter engine, enabled via adc_voice_en in the ADC_FILTERS1 register. For the first filter, in music mode adc_voice_en must be set to 0 and the HPF corner frequency is set using adc_audio_hpf_corner. In ADC voice mode, adc_voice_en must = 1 and adc_hpf_en must = 1 in which case the HPF corner frequency is set using adc_voice_hpf_corner. The low frequency roll off is configured over a wide range using the adc_voice_hpf_corner control. This allows for flexible removal of wind and pop noise. The input high-pass filter is controlled using ADC_FILTERS1. For the first filter, in music mode adc_voice_en must be set to 0 and the HPF corner frequency is set using adc_audio_hpf_corner. In voice mode, adc_voice_en must = 1 in which case the HPF corner frequency is set using adc_audio_hpf_corner. The value of the HPF corner frequency also depends on the input sample rate ( SR) as shown in Table 23. The sample rates available in the different ADC power modes are summarised in Table 23. Table 23: Input high-pass filter settings adc_voice_en adc_voice_hpf_ corner adc_audio_hpf_ corner SR Sample Rate (kHz) 8 11.025 12 16 22.05 24 32 44.1 48 88.2 96 14.7 16 29.4 32 000 2.5 3.45 3.75 5 Voice HPF not available for sample rates above 16 kHz. 001 25 34.5 37.5 50 010 50 68.9 75 100 011 100 137.8 150 200 100 150 206.7 225 300 101 200 275.6 300 400 110 300 413.4 450 600 111 400 551.3 600 800

9.3.1.3 Automatic Level Control (ALC)

For improved sound recordings of signals with a large volume range, the DA7219 offers a fully- configurable automatic recording level control (ALC) for microphone inputs. This is enabled via the alc_en control. The ALC monitors the digital signal after the ADC and adjusts the microphones’ analogue and digital gain to maintain a constant recording level, whatever the analogue input signal level. Operation of ALC is illustrated in Figure 10. When the input signal volume is high, the ALC system will reduce the overall gain until the output volume is below the specified maximum value. When the input signal volume is low, the ALC will increase the gain until the output volume increases above the specified minimum value. If the output signal is within the desired signal level (between the specified minimum and maximum levels), the ALC does nothing. The maximum and the minimum thresholds that trigger a gain change of the ALC are programmed by the alc_threshold_min and alc_threshold_max controls. ALC Input ALC Gain ALC Output ALC MAX level ALC MIN level Release time Attack time Figure 10: Principle of operation of the ALC In hybrid mode, the total gain is made up of an analogue gain, which is appl ied to the microphone amplifier, and a digital gain, which is implemented in the filtering stage. The ALC block monitors and controls the gain of the microphone and the ADC. Although the ALC is controlling the gain, it does not modify any of the registers MIXIN_L_GAIN or ADC_L_GAIN, nor does it modify the digital gain register ADC_L_GAIN. These registers are ignored while the ALC is in operation. In digital-only mode only the digital gain in the ADC is altered. Although the ALC is controlling the gain, it does not modify adc_l_digital_gain in the ADC_L_GAIN register. This register is ignored while the ALC is in operation. Hybrid mode should be used whenever analogue microphones are being used. The hybrid analogue/digital gain mode (hybrid mode) can be enabled using. alc_sync_mode. The minimum and maximum levels of digital gain that can be applied by the ALC are controlled using alc_atten_max and alc_gain_max. Similarly the minimum and maximum levels of analogue gain are controlled b y alc_ana_gain_min and alc_ana_gain_max. The rates at which the gain is changed are defined by the attack and decay rates in register ALC_CTRL2. When attacking, the gain decreases with alc_attack rate. When decaying, the gain increases with alc_release rate.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 29 of 123 © 2015 Dialog Semiconductor The hold-time is defined by alc_hold in the ALC_CTRL3 register. This controls the length of time that the system maintains the current gain level before starting to decay. This prevents unwanted changes in the recording level when there is a short-lived ‘spike’ in input volume, for example when recording speech. Typically the attack rate should be much faster than the decay rate, as it is necessary to reduce rapidly increasing waveforms as quickly as possible, whereas fast release times will result in the signal appearing to ‘pump’. The ALC also has an anti clipping function that applies a very fast attack rate when the input signal is close to full-range. This prevents clipping of the signal by reducing the signal gain at a faster rate than would normally be applied. The anti clip function is enabled using alc_anticlip_en, and the threshold above which it is activated is set in the range 0.034 dB/fs to 0.272 dB/fs using alc_anticlip_step. A recording Noise-Gate feature is provided to avoid increasing the gain of the channel when there is no signal, or when only a noise signal is present. Boosting a signal on which only noise is present is known as ‘noise pumping’. The Noise-Gate prevents this. Whenever the level of the input signal drops below the noise threshold configured in alc_noise, the channel gain remains constant. Figure 11: Attack, delay and hold parameters

9.3.2 Sidetone processing

There is a low latency filter channel between inputs and outputs for implementing a sidetone path. The gain is controlled using sidetone_gain and provides gain in the range -42 dB to +0 dB in +3 dB steps. The sidetone path is enabled using sidetone_en. It is muted using sidetone_mute_en. The output from the sidetone channel can be added to left or right (or both) output filters using outfilt_st_1l_src and outfilt_st_1r_src. The sidetone path is enabled using sidetone_en. It is muted using sidetone_mute_en. max min atk dcyhld input signal gain level atk rate dcy rate time time

9.3.3 Tone generator

Parameter Conditions Min Typ Max Unit Single-tone frequency FS = 8,12,16,24,32,48,96kHz FS = 11.025, 22.05, 44.1, 88.2kHz 12000 11025 Hz Single-tone frequency step 0.2 Hz Dual-tone modulation frequency A 697 770 852 941 Hz Dual-tone modulation frequency B 1209 1336 1477 1633 Hz Output signal level 0 dBFS On/off pulse duration 10 2000 ms On/off pulse step size 10 to 200ms duration 200 to 2000ms duration ms On/off pulse repeat Programmable Continuous 1,2,3,4, 5,6∞ Cycles The tone generator contains two independent Sine Wave Generators, SWG1 and SWG2. Each SWG can generate a sine wave at a frequency (FREQ) from approximately 10 Hz to 12 kHz according to the programmed 16-bit value: FREQ[15:0] = 2^16 × fSWG/12000 – 1, for SR2 = 8, 12, 16, 24, 32, 48,96 kHz FREQ[15:0] = 2^16 × fSWG/11025 – 1, for SR2 = 11.025, 22.05, 44.1, 88.2 kHz For SWG1, the FREQ value is stored in two 8-bit registers as freq1_u = FREQ[15:8] and freq1_l = FREQ[7:0]. The SWG2 frequency is programmed in the same way using freq2_u and freq2_l. The output of the tone generator can come from either of the SWGs , or from a combination of both of them as specified by swg_sel. In addition the tone generator can produce standard Dual Tone Multi-Frequency (DTMF) tones using the two SWGs if dtmf_reg = 1 and the required key pad value is programmed in dtmf_reg as shown in Table 24. Table 24: DTMF tones corresponding to the dtmf_reg value SWG2 Freq (Hz) SWG1 Frequency (Hz) 1209 1336 1477 1633 697 0x1 0x2 0x3 0xA 770 0x4 0x5 0x6 0xB 852 0x7 0x8 0x9 0xC 941 0xE 0x0 0xF 0xD The tone generator can produce 1, 2, 3, 4, 8, 16, or 32 beeps, or a continuous beep, as determined by beep_cycles. Each beep has an ‘On’ period from 10 ms to 2 s as programmed in beep_on_per. and an ‘Off’ period from 10 ms to 2 s as programmed in beep_off_per. The tone generator is started by setting the start_stopn bit, and is halted by clearing this bit. If start_stopn is cleared, the tone generator stops at the completion of the current beep cycle or at the next zero-cross if the number of beeps is set to continuous (beep_cycles = 110 or = 111). The start_stopn bit is automatically cleared once the programmed number of beep cycles has been completed.

9.3.4 Digital router

There is a digital router block which is configured by registers DIG_ROUTING_DAI and DIG_ROUTING_DAC. The router options are illustrated in Figure 12. dac_l_src DIG_ROUTING_DAC ADC_L TONEGEN DAI_IN_L DAI_IN_R DAC_L DAC_R DAI_OUT_L dai_l_src dai_r_src dac_r_src DIG_ROUTING_DAI DA7219 Digital Router DAI_OUT_R 2E[5:4] 2E[1:0] 2A[5:4] 2A[1:0] [01] [10] [11] [00] Figure 12: DA7219 digital router DIG_ROUTING_DAC is used to select the inputs to go into the DAC filter chain from the router. DIG_ROUTING_DAI is used to select the inputs to go into the DAI from the router. For example, for dac_l_src, data selection to the DAC_L path is 00 = ADC left output 01 = Tone generator 10 = DAI input left / dai mono mix 11 = DAI input right / dai mono mix The same 2-bit code (00, 01, 10, 11) is used for dac_r_src, dai_l_src and dai_r_src.

9.3.5 System controller

The System Controller (SC) automates the sequencing of the multiple blocks required to set up one or more particular audio paths. It is an optional feature, and operates by performing register writes with optimal sequencing and timing, thus eliminating pops and clicks. System control for the inputs is controlled using SYSTEM_MODES_INPUT, and for the outputs by using SYSTEM_MODES_OUTPUT. Writing to the mode_submit field either of these registers will cause the system controller to process both input and output paths.

9.3.6 Output processing

9.3.6.1 DAC digital gain

Each channel includes individual gain settings that are controllable in 0.75 dB steps ranging from -78 dB (= mute) to 12 dB using dac_l_digital_gain and dac_r_digital_gain. The currently active gain settings are stored in DAC_L_GAIN_STATUS and DAC_R_GAIN_STATUS registers.

9.3.6.2 High-pass filter

Any DC offset from the input path is removed via IIR filters (typically <2 Hz roll-off, configurable). After reset the filters for both channels are enabled by default, but can be disabled by clearing dac_hpf_en. The cut-off frequency of the filters can be programmed using dac_audio_hpf_corner. During playback, dedicated voiceband filtering can be enabled using dac_voice_en in the DAC_FILTERS1 register. In DAC voice mode, dac_voice_en must = 1 and dac_hpf_en must = 1 in which case the HPF corner frequency is set using dac_voice_hpf_corner. The low frequency roll off is configured over a wide range using the dac_voice_hpf_corner control. In voice mode, the wind noise high-pass filter cut-off frequency is determined by the settings of the adc_voice_hpf_corner and the dac_voice_hpf_corner register bits, These cut-off frequencies are not fixed and vary with the sample rate being used. Table 25 shows the cut-off frequencies for all valid settings of adc_voice_hpf_corner and dac_voice_hpf_corner, at all sample rates of 16 kHz and below. Table 25: Output high-pass filter settings dac_voice_en dac_voice_hpf_corner dac_audio_hpf_corner SR Sample Rate (kHz) 8 11.025 12 16 22.05 24 32 44.1 48 88.2 96 14.7 16 29.4 32 000 2.5 3.45 3.75 5 Voice HPF not available for sample rates above 16 kHz. 001 25 34.5 37.5 50 010 50 68.9 75 100 011 100 137.8 150 200 100 150 206.7 225 300 101 200 275.6 300 400 110 300 413.4 450 600 111 400 551.3 600 800

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 33 of 123 © 2015 Dialog Semiconductor 9.3.6.3 5-band equaliser The gains of each band can be individually configured, from -10.5 to 12.0 dB in 1.5 dB steps, using the dac_eq_band1, dac_eq_band2, dac_eq_band3, dac_eq_band4, dac_eq_band5 controls. The output filters can provide gain or attenuation in each of five separate (fixed) frequency bands using the 5-band equaliser. The equaliser, for both left and right channels, is enabled using dac_eq_en. The centre or cut-off frequency of each of the five bands depends on the output sample rate as shown in Table 26. Table 26: Output 5-band equaliser centre/cut-off frequencies FS (kHz) Centre frequency (Hz) at programmed setting Band 1 Band 2 Band 3 Band 4 Band 5 8 0 99 493 1528 4000 11.025 0 136 680 2106 5512 12 0 148 740 2293 6000 16 0 96 440 2128 8000 22.05 0 133 607 2933 11025 24 0 145 660 3191 12000 32 0 95 418 1797 16000 44.1 0 131 576 2386 22050 48 0 143 627 2596 24000

88.2 N/A N/A N/A N/A N/A

96 N/A N/A N/A N/A N/A

Figure 13: Equaliser filter band 1 frequency response at FS = 48 kHz

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 35 of 123 © 2015 Dialog Semiconductor Figure 16: Equaliser filter band 4 frequency response at FS = 48 kHz Figure 17: Equaliser filter band 5 frequency response at FS = 48 kHz

9.3.6.4 DAC soft mute

To improve the user’s perception of audio reconfigurations, the DAC channel signals may be soft muted by asserting the control dac_softmute_en (in register. DAC_FILTERS5).The soft mute function attenuates the digital input to the DAC, ramping the gain down in steps of 0.1875 dB from its current level to -77.25 dB, then completely muting the channel. When dac_softmute_en is released, the attenuation is set to -77.25 dB, and then ramped up to the previous gain level. Both left and right channels of soft mute enabled output amplifiers are muted simultaneously. The ramping up and down rate is dependent on the audio sample rate and can be individually configured using control dac_softmute_en. Setting dac_softmute_en = 1 enables a soft mute on both channels. During active soft muting, the digital gain of the DAC will be different to the value programmed inside controls dac_l_digital_gain_status and dac_r_digital_gain_status.

9.4 Output paths

9.4.1 Digital to Analogue Converter (DAC)

The DA7219 codec includes a stereo audio DAC. Left and right channels of the DAC are independently and automatically enabled whenever the corresponding output filter channel is enabled. The DAC is clocked at 3.072 MHz or 2.8224 MHz depending on the output sample rate (SR). Left and right channels of the DAC are independently and automatically enabled whenever the corresponding output filter channel is enabled. The integrated stereo DAC is suitable for high quality audio playback by MP3 player s and by portable multimedia players of all kinds. The left and right channels of the DAC can be individually enabled using controls dac_l_en and dac_r_en. Each channel includes individual gain settings that are controllable in 0.75 dB steps from -78 dB to 12 dB using dac_l_digital_gain_status and dac_r_digital_gain_status. The currently active gain settings are stored in DAC_L_GAIN_STATUS and DAC_R_GAIN_STATUS registers. On the dedicated DAC_L_CTRL and DAC_R_CTRL registers, settings such as mute and ramping of gain changes can be configured. If ramping is enabled using the control bits dac_l_ramp_en or dac_r_ramp_en, the rate of the ramping can be controlled using gain_ramp_rate i n the GAIN_RAMP_CTRL register. A digital high-pass filter for each DAC channel is implemented with a 3 dB cut-off frequency controlled in the DAC_FILTERS1 register by dac_audio_hpf_corner. The high-pass filter is enabled by control dac_hpf_en. After Reset, the high pass filters for both channels are enabled by default.

9.4.2 Headphone outputs

Output Filters @ SR (High-Pass, 5-Band EQ) HP_L HP_R HP_L_AMP -57: 1.0: +6 dB HP_R_AMP -57: 1.0: +6 dB DA7219 MIXOUT_L MIXOUT_R Figure 18: Headphone output paths showing the two amplifiers

9.4.2.1 Buffer amplifier

The left-channel buffer amplifier MIXOUT_L, is enabled by setting mixout_l_amp_en = 1. The right channel output buffer MIXOUT_R is controlled in the same manner.

9.4.2.2 Headphone amplifiers

Each headphone path has one headphone amplifier stage providing a gain of –57 dB to +6 dB in 1.0 dB steps. The amplifiers are configured to operate in single ended mode. The headphone loads are connected between HP_L and HP_R and the internal ground set by the MIC/GND switches set during the detect sequence. The headphone amplifiers are configured to operate in true-ground (charge pump) mode. In true- ground supply mode, the charge pump must be enabled to generate the ground-centred supply rails for the amplifiers. The left-channel headphone amplifier (HP_L_CTRL) is enabled by setting hp_l_amp_en = 1. The output stage is enabled independently by setting hp_l_amp_oe = 1. The amplifier gain can be set in the range of –57 dB to +6 dB in 1.0 dB steps using hp_l_amp_gain. Gain updates can be ramped through all intermediate values by setting hp_l_amp_zc_en = 1. This ramp setting overrides the settings of hp_l_amp_zc_en. To prevent zipper noise when gain ramping is selected, the gain is ramped through additional sub-range gain steps. As an alternative to gain ramping, gain updates can be synchronised with signal zero -crossings by setting hp_l_amp_zc_en = 1. If no zero-crossing is detected within the timeout period, then the gain update is applied unconditionally. The timeout period is approximately 0.1 second, and is hard-coded into the device. It is not user-configurable. The amplifier can be muted by setting hp_l_amp_mute_en = 1. The amplifier can be put in its minimum gain configuration by setting hp_l_amp_min_gain_en = 1. If either zero-crossing or ramping are enabled when minimum gain is set, the ramping or the zero crossing or both will be performed while activating the minimum gain. The right-channel headphone amplifier (HP_R_CTRL) is controlled in the same manner.

9.4.3 Charge pump control

The charge pump is enabled by asserting cp_en in the CP_CTRL register. Once enabled, the charge pump can be controlled manually or automatically. When under manual control ( cp_mchange = 00), the output voltage level is directly determined by cp_mod. The amount of charge stored, and therefore the voltage generated, by the charge pump is controlled by the charge pump controller. As the power consumed by devices such as amplifiers is proportional to Voltage2, significant power savings are available by matching the charge pump’s output with the system’s power requirement. There are three modes of operation that are determined by the cp_mchange setting. All three modes are described in Table 27.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 38 of 123 © 2015 Dialog Semiconductor Table 27: Charge pump output voltage control Charge pump tracking mode cp_mchange Charge pump output voltage Details

00 Reserved Reserved

01 Voltage level depends on the

The charge pump controller monitors the PGA volume settings, and generates the minimum voltage that is high enough to drive a full-scale signal at the current gain level.

10 Voltage level depends on the DAC

The charge pump controller monitors the DAC signal, and generates a voltage that is high enough to drive a full-scale output at the current DAC signal volume level Voltage level depends on the signal magnitude and the programmed gain setting The charge pump monitors both the programmed volume settings and the actual signal size, and generates the appropriate output voltage. This is the most power-efficient mode of operation. When cp_mchange is set to 10 (tracking DAC signal size, described in Table 27) or cp_mchange is set to 11 (tracking the output signal size), the charge pump switches its supply between the VDD_A rail and the VDD_A/2 rail depending on its power requirements. When low output voltages are needed, the charge pump saves power by using the lower -voltage VDD_A /2 rail. The switching point between using the VDD_A rail and the VDD_A/2 rail is determined by the cp_thresh_vdd2 register setting. The switching points determined by cp_thresh_vdd2 vary between the two cp_mchange modes, and are summarised in Table 28 and Table 29. When the charge pump output voltage is controlled manually (cp_mchange = 00) or when it is tracking the PGA gain settings (cp_mchange = 01), the charge pump always takes its supply from VDD_CP. Table 28: cp_thresh_vdd2 settings in DAC_VOL mode (cp_mchange = 10) cp_thresh_vdd2 setting Approximate switching point (Note 6) Notes 0x01 -30 dBFS Do not use. Very power-inefficient as nearly always VDD/1 0x03 -24 dBFS Not recommended. Very power-inefficient as nearly always VDD/1 0x07 -18 dBFS Good to use but not power efficient 0x0E -12 dBFS Good to use 0x10 -10 dBFS Recommended setting 0x3F – 0x13 Not recommended

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 39 of 123 © 2015 Dialog Semiconductor Table 29: cp_thresh_vdd2 settings in signal size mode (cp_mchange = 11) cp_thresh_vdd2 setting Approximate switching point (Note 6) Notes 0x00 Never Not recommended. Always VDD/1 mode 0x01 Never Not recommended. Always VDD/1 mode 0x02 -32 dBFS Not recommended. Very power-inefficient as nearly always VDD/1 0x03 -24 dBFS Good to use 0x04 -20 dBFS Good to use 0x05 -17 dBFS Good to use 0x06 -15 dBFS Recommended setting 0x07 -13 dBFS Good to use 0x08 -12 dBFS Good to use 0x09 -11 dBFS Good to use 0x0A -10 dBFS Good to use 0x0B -9 dBFS Not recommended. VDD/2 begins to clip 0x0C Never Not recommended. Always VDD/2 mode 0x0D Never Not recommended. Always VDD/2 mode 0x0E Never Not recommended. Always VDD/2 mode 0x0F Never Not recommended. Always VDD/2 mode Note 6 Full Scale (FS) = 1.6 * VDD_A

9.4.4 Tracking the demands on the charge pump output

There are three points at which the demands on the charge pump can be tracked. These tracking points are determined by cp_mchange. 9.4.4.1 cp_mchange = 01 (tracking the PGA gain setting) If cp_mchange = 01, it is the PGA gain setting that is tracked, and which provides the feedback to boost the clock frequency when necessary. 9.4.4.2 cp_mchange = 10 (tracking the DAC signal setting) If cp_mchange = 10, it is the size of the DAC signal that is tracked, and which provides the feedback to boost the clock frequency when necessary. 9.4.4.3 cp_mchange = 11 (tracking the output signal magnitude) If cp_mchange = 11, it is the magnitude of the output signal that is tracked, and which pro vides the feedback to boost the clock frequency when necessary.

9.5 Advanced Accessory Detection (AAD)

If the DA7219 is configured for Advanced Accessory Detection (AAD), the insertion of a jack wakes the system up. No external clocking is required to detect a jack insertion, and the clock is only requested if the input is changing and if debouncing is required. This ensures the lowest possible power consumption with no digital leakage. Once a jack has been inserted, the AAD differentiates between a three-pole jack (used on headphones and lineouts) and a four-pole jack (used on headsets). Two-pole jacks are detected as a three-pole jack, and will work as designed as a mono output. There are two combinations of four-pole jack available in the market, both of which are supported by the AAD. The jack configurations are shown in Figure 19. Jack Type SLEEVE RING2 RING1 TIP Position CTIA OMTP SLEEVE MIC GND RING2 GND MIC RING1 HP_R HP_R TIP HP_L HP_L Figure 19: Jack socket variants On detecting the insertion of a jack, the DA7219 moves to the microphone-detect state where it drives current down the SLEEVE pin and measures the impedance to the RING2 pin, which will be connected to GND_HP by the internal GND switch. If the impedance measured between the SLEEVE and RING2 pins is below 500 Ω (configurable), the DA7219 detects the connected accessory as a three-pole jack. The DA7219 then returns to the jack detection state to poll for removal, but continues to periodically pulse current down the SLEEVE pin to verify that the connected accessory is three-pole. This continued polling avoids an incorrect detection, for example, if a four-pole accessory is inserted with a button depressed. Note that a two-pole jack is detected as three-pole jack, and will work as designed as a mono output. If the impedance measured between the SLEEVE and RING2 pins is above 500 Ω (configurable), the DA7219 detects the connected accessory as a four-pole jack. The DA7219 then moves to the pin order-detect state, where it first drives current down the SLEEVE and then the RING2 pins, and compares the voltages measured in each case. The pin that develops the largest voltage will be deemed to be the accessory’s microphone, and the other pin as the ground. The headphone impedance can also be to determine whether the output is to a headphone or to a line output. Impedance measurements below a pre-set threshold are deemed to be headphones, and impedances above the threshold are line outputs. The DA7219 will then move to the button detect state, where polling is carried out to detect button presses. If a button is pressed while MICBIAS is off, the DA7219 can only detect that a button has been pressed, but cannot distinguish between the buttons. To distinguish which one of up to four buttons was pressed, the MICBIAS rail must be enabled so that the impedance can be measured between the MIC and GND pins. See section 9.5.6 for further details.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 41 of 123 © 2015 Dialog Semiconductor While any of the four possible buttons is being pressed, any further button presses are ignored. Only once the first button has been released can a second or subsequent button press be detected. Detection of the jack type and its configuration, detection of the number of buttons, detec tion of a mic input, and detection of headphone or line outputs are all performed automatically when the AAD block is enabled. On detecting a button press, the DA7219 can identify all buttons as defined in the Android wired headset specification (v1.1) when MICBIAS is present. The DA7219 also offers the possibility of overriding the automatically detected accessories, and of setting them manually. A full cross-reference of the DA7219’s functionality and power consumption in different modes is listed in Table 19.

9.5.1 Configuring Advanced Accessory Detection (AAD)

AAD is enabled by setting accdet_en = 1. Within the AAD block, all individual accessory detection measurements can be enabled or disabled, and all accessory detect interrupt signals can be masked. All accessory detection measurements can be manually overridden, and the current statuses of all measurements can be interrogated from the status register fields. Jack type detection, jack configuration detection, and button detection are all based on measurements of resistance between different pins. The resistance thresholds for every measurement type are all configurable by using the relevant register fields. A signal timing diagram is illustrated in Figure 20. These features are all summarised in Table 30, and are described in greater detail in the following sections.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 42 of 123 © 2015 Dialog Semiconductor Table 30: DA7219 Advanced Accessory Detection (AAD) feature summary Feature DA7219 support Configuration Host reporting Jack insertion/removal detection Yes Enabled when system_active and accdet_en are both 1. Jack insertion latency set by jackdet_debounce (1 ms <--> 1 s). Host notifiaction via e_jack_inserted and e_jack_removed IRQ events. Jack type detection Yes - 3-Pole / 4-Pole Jack Type detection runs on insertion, Duration set by jack_detect_rate . Host configurable microphone detection impendance threshold mic_det_thresh Optional Host manual type override provided. Host notification via e_jack_detect_complete IRQ event. jack_type_sts register available for host readback ( 3-Pole or 4- Pole reported), data is qualified by e_jack_detect_complete. Pin order detection Yes - CTIA/OMTP Detection with GND switching runs on insertion if a 4-pole jack is detected. Optional Host manual pin Order override provided. Host notification via e_jack_detect_complete IRQ event. jack_pin_order_sts register available for host readback (LRGM or LRMG reported ), data is qualified by e_jack_detect_complete. Button press detection Yes - Press / Release detection for A,B,C and D button impedances with +/- 1% accuracy, as per Google Chromebook Headset Accessory Electrical Specification. Button detection enable and frequency (2 ms<--> 500 ms) set by button_config. Host controlled a_d_button_thresh, d_b_button_thresh, b_c_button_thresh, and c_mic_button_thresh set the ADC voltage thresholds for button press impedance measurements = (RLOAD / (RLOAD + RMICBIAS)) Host notification via e_button_*_pressed and e_button_*_released IRQ events, (where * = a/b/c/d). button_type_sts 8 bit adc measurement result also available for host readback. Interrupt reporting Yes - Single dedicated h/w interrupt line. All events are maskable and are 'Write 1 to clear'. Interrupt line asserted to host when any unmasked events are captured. Interrupt line is de asserted when all unmasked events have been cleared by host. MICBIAS isolation Yes- Both on insertion and removal. Host control when the MICBIAS rail can be enabled (requires VDD_MIC) with micbias1_en. Accdet will automatically enable the MICBIAS LDO following e_jack_detect_complete if jack_type_sts reports '4 pole'. MICBIAS is auto disabled, discharged and isolated on e_jack_removed to prevent audible artefacts on HPs during a fast jack removal. micbias_up_sts available for host readback to report MICBIAS rail is up.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 43 of 123 © 2015 Dialog Semiconductor Feature DA7219 support Configuration Host reporting HP_L impedance measurement Yes -supported by DA7219 using s/w controlled sequence following insertion. N/A: This feature does not live in the accdet block. N/A: This feature does not live in the accdet block. HP_L / HP_R to GND when device unpowered Yes-supported by DA7219 using pulldown on HPs. N/A: This feature does not live in the accdet block. N/A Host IRQ clear Host IRQ clear Host IRQ clear Host IRQ clear Host IRQ clear DET Accdet State NO_JACK INS_DEB JACK_DETECT JACK_IN BTN_CHECKJACK_IN BTN_CHECK JACK_IN REM_DEB NO_JACK nIRQ e_jack_insertion e_jack_detect_complete JACK_TYPE_STS DEFAULT 3-Pole / 4-Pole DEFAULT LRGM – CTIA / LRMG - OMTPJACK_PIN_ORDER_STS Host type check MICBIAS e_jack_removal e_d_button_pressed e_d_button_released DEFAULT DEFAULT D Button Press Jack insertion Detection Complete Button Pressed Button Released Jack removal MIC MICBIAS OFF MICBIAS ON MICBIAS OFFMICBIAS_UP_STS RED: Device PIN BLUE: Device Internal Signal Figure 20: Signal timing diagram for the AAD function Note 7 VDD_MIC must always be greater than VDD

9.5.2 Detection of jack insertion or removal

Whenever a jack is inserted, a jack insertion event is flagged by e_jack_inserted getting set to1. Similarly, a jack removal event is flagged by e_jack_removed getting set to1. The presence or absence of a jack is recorded in jack_insertion_sts. Any jack insertion will be detected, and is recorded by the setting of jack_insertion_sts. The register field jack_insertion_sts = 1 if a jack has been inserted, and jack_insertion_sts = 0 if no jack has been inserted. Jack detection latency, that is, the time from an e_jack_insertion event to the point where e_jack_detect_complete is asserted, is configurable using jack_detect_rate. The jack detection latency times are different for three-pole and four-pole jacks, and are listed in Table 22: Jack detection latency timings controlled by jack_detect_rate jack_detect_rate value Three-pole jacks (ms) Four-pole jacks (ms) 00 32 64 01 64 128 10 128 256 11 256 512 Debouncing is avaliable on jack insertion and removal events. Debounce on jack insertion is specified using jackdet_debounce, and on jack removal using jackdet_rem_deb. The debounce times are listed in Table 31 and Table 32. Table 31: Debounce settings for jack insertion events jackdet_debounce Debounce time (ms) 000 5 001 10 010 20 (default) 011 50 100 100 101 200 110 500 111 1 Table 32: Debounce settings for jack removal events jackdet_rem_deb Debounce time (ms) 00 1 (default) 01 5 10 10 11 20 The jack insertion, jack removal, and jack complete interrupts can be masked using the register fields in the ACCDET_IRQ_MASK_A register. The jack insertion interrupt is masked by setting m_jack_inserted = 1, the jack removal interrupt is masked by setting m_jack_removed = 1, and the jack detection complete interrupt is masked by setting m_jack_detect_complete = 1. These masking fields mask the interrupt signals, but do not prevent updating of the event fields or the status fields previously described.

9.5.3 Three-pole or four-pole jack insertion

The type of jack inserted can be determined automatically by setting jack_type_det_en = 1. Once the jack insertion measurement has been completed as indicated by e_jack_detect_complete = 1, the AAD determines whether a three-pole or a four-pole jack has been inserted. This is done by measuring the resistance between the SLEEVE and the RING2 pins. If the measured impedance is below the threshold setting, a three-pole jack is deemed to have been inserted. If the resistance is above this threshold setting, a four-pole jack is deemed to have been inserted. Note that if a mono two-pole jack is inserted, the AAD will detect this as a three-pole jack, but the two-pole jack will work as designed, that is, as a mono output. The threshold setting used to determine whether a three-pole or a four-pole jack has been inserted is set using mic_det_thresh. The settings are listed in Table 33. Table 33: Resistance threshold settings for three-pole and four-pole jack determination mic_det_thresh Resistance threshold (Ω) 00 200 01 500 (default) 10 750 11 1000 Once the jack type has been successfully determined, the type of jack is recorded in jack_type_sts. A three-pole jack is indicated by jack_type_sts = 0, and a four-pole jack by jack_type_sts = 1. The jack type status recorded in the register field jack_type_sts is not valid until the measurement has been completed. Measurement completion is indicated when e_jack_detect_complete = 1. Note that e_jack_detect_complete = 1 indicates the completion of both the jack_type_sts measurement and the jack_pin_order_sts measurement. The measurement of the type of jack, which is performed automatically when jack_type_det_en = 1, can be overridden if required. To do this, set jack_type_det_en = 0 to prevent the measurement taking place, and then use the jack_type_force register field to force the jack type. A three-pole jack is specified by setting jack_type_force = 0, and a four-pole jack by setting jack_type_force = 1.

9.5.4 Jack pin order detection with four-pole jacks

Two different polarities are widely used with four-pole jacks. These are the CTIA tip-ring-ring-sleeve configuration of GND-MIC-RIGHT-LEFT, and the OMTP configuration of MIC-GND-RIGHT-LEFT. If pin_order_det_en = 1, the detection of jack configuration is performed automatically. The measurement of jack configuration can be overridden by setting pin_order_det_en = 0, and using pin_order_force = 0 to specify the CTIA configurations. Setting pin_order_force = 1 specifies the OMTP configuration. The jack configuration status recorded in the register field jack_pin_order_sts is not valid until the measurement has been completed. Measurement completion is indicated when e_jack_detect_complete = 1. Note that e_jack_detect_complete = 1 indicates the completion of both jack_type_sts measurement and jack_pin_order_sts measurement.

9.5.5 Headphone output and line output

The DA7219 can detect whether the output is a headphone or a line output. Headphone or lineout detection is enabled by setting hptest_en = 1. The impedance is measured between HP_L (or HP_R) and the local GND connection on either SLEEVE or RING2 (depending on the jack configuration) to determine whether the output is to a headphone or to a lineout. Impedance measurements below a pre-set threshold are deemed to be headphones, and impedances above the threshold are line outputs. The threshold value is set between 1 and 10 kΩ by setting hptest_res_sel appropriately. The threshold settings available are listed in Table 34. The Tone Generator is used to develop a slow S-ramp of the signal amplitude at a frequency below the audible range on the HP outputs. The S-ramp profile is configurable for maximum flexibility. The device monitors the current drawn by the HP amps during this process, and reports back the load as either above or below the threshold level. The accuracy of the measurement is ±40%. The host AP must control the test:

Program the S-ramp profile
Initialise the DA7219 signal path and outputs
Initiate the S-ramp function
Read back the impedance detection status register for the load condition. Table 34: Resistance threshold settings for headphone and lineout determination hptest_res_sel setting Test threshold impedance (kΩ) 00 1.0 01 2.5 10 5.0 11 10.0 The result of the headphone threshold test is stored in hptest_comp.

9.5.6 Detection of buttons

After successful detection of the insertion of a four-pole jack (e_jack_detect_compete = 1 and jack_type_sts = 1), the DA7219 will move to the button detect state, where polling is carried out for button presses. If a button is pressed while MICBIAS is off, the DA7219 can only detect that a button has been pressed, but cannot distinguish between the buttons. On detecting a button press, the DA7219 can identify all buttons as defined in the Android wired headset specification (v1.1) when MICBIAS is present. The Android wired audio headset specification (v1.1) specifies the impedance associated with any button press. The impedance is measured between the MIC and GND. Note that when measuring the impedance of a button press, the measured impedance includes both the impedance of the resistor associated with a button and the impedance of the microphone. This is illustrated in Figure 21. While any of the four possible buttons is being pressed, any further button presses are ignored. Only after the first button has been released can a second or subsequent button press be detected.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 47 of 123 © 2015 Dialog Semiconductor MIC or GND LEFT RIGHT GND or MIC Button A Button B Button C Button D Measured impedance Microphone NOTE: The impedance for any button press is measured between MIC and GND. Note that the circuit includes both the resistor associated with the button AND the MIC’s impedance in parallel. Figure 21: Measuring the impedance of a button press The Android wired audio headset specification (v1.1) stipulates the functions and names of the four possible buttons on a headset. These are listed in Table 35. Table 35: Button names and functions in Android devices Button Function Function A Play, pause, or hook (short press) Trigger assist (long press) Next (double-press) Function B Volume + Function C Volume - Function D Google voice search feature Headsets with only one button must implement Function A. Headsets with multiple buttons must implement functions according to the following patterns:

Two functions: Functions A and D
Three functions: Functions A, B, and C
Four functions: Functions A, B, C, and D Whenever a button is pressed, a button press event is flagged by e_button_ <a|b|c|d>_pressed getting set = 1. Similarly, a button release event is flagged by e_button_<a|b|c|d>_released getting set = 1.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 48 of 123 © 2015 Dialog Semiconductor The measured impedance of the last button press is recorded in button_type_sts. The impedance measurements can be averaged using button_average. Averaging in this manner provides greater immunity to spurious measurements caused my noise, but at a cost of consuming more power and of increasing the measurement latency (every extra measurement used in the averaging takes approximately 1 ms to perform). The number of measurements that are used in the averaging are listed in Table 36. Table 36: Setting the number of measurements used in averaging button_average setting Number of measurements used in averaging 00 1 01 2 10 4 11 8 The button press interrupts can be masked by asserting the register fields m_button_<a|b|c|d>_pressed. The button released interrupts can be masked by setting m_button_<a|b|c|d>_released = 1. These masking fields mask the interrupt signals, but do not prevent updating of the event fields or the status fields previously described. The impedance threshold between Button A and Button D is specified using a_d_button_thresh, and is specified as (RLOAD / RLOAD + RMICBIAS). Similarly, the impedance threshold between Button D and Button B is specified using d_b_button_thresh, and is specified in the same manner (RLOAD / RLOAD + RMICBIAS). The impedance threshold between Button B and Button C is specified using b_c_button_thresh, and is specified in the same manner (RLOAD / RLOAD + RMICBIAS). The impedance threshold between Button C and MIC is specified using c_mic_button_thresh and is again specified in the same manner (RLOAD / RLOAD + RMICBIAS). The time between the periodic button press measurements is specified using button_config. The inter-measurement period can be between 2 ms and 500 ms. The button_config register is only active after a four-pole jack has been detected. Setting button_config = 0 also disables the button measurements.

9.6 Clocking

The internal system clock (SYSCLK) from which all other clocks are derived is always at one of two possible frequencies:

12.288 MHz for SR from the 48 kHz family (8, 12, 16, 24, 32, 48, 96 kHz) The DA7219 can run with or without an applied MCLK. If no MCLK is applied, the internal reference oscillator will clock the device. However when using the DAI an MCLK must be provided and correctly configured. The DA7219 contains a phase-locked loop (PLL), which supports a range of clocking modes and input clock (MCLK) frequencies.

9.6.1 MCLK input

MCLK is the master clock input which must be in the range of 2 to 54 MHz. MCLK can be applied as a full-amplitude square wave, or as a low-amplitude sine wave if the MCLK squarer circuit has been enabled. The clock squarer circuit is enabled by writing pll_mclk_sqr_en = 1. This clock squarer allows a sine wave or other low amplitude clock (down to 300 mVpp) to be applied to the chip. The MCLK input is AC coupled on chip when using the clock squarer m ode.

9.6.1.1 MCLK detection

A clock detection circuit will set bit [0] of pll_srm_status = 1 whenever the applied MCLK frequency is above the minimum detection frequency of approximately 1 MHz. Whenever this bit is high, the MCLK signal is selected as the clock input to the PLL.

9.6.2 Audio reference oscillator

For best audio performance, a system clock within the specified range is required. The DA7219 codec has an internal reference oscillator that provides the system clock when there is no valid MCLK signal. The reference oscillator is automatically enabled whenever the codec is in ACTIVE mode and the MCLK frequency is below the minimum frequency of 1 MHz. When the codec enters STANDBY mode, the oscillator is automatically disabled to save power.

9.6.3 PLL bypass mode

available, the PLL is not required and should be disabled to save power. PLL bypass mode is activated by setting pll_mode = 00. In this mode the PLL is bypassed and an audio frequency clock is applied to the MCLK pin of the codec. The required clock frequency depends on the sample rate at which the audio DACs and ADCs are operating. These clock frequencies are summarized in Table 37 for the range of DAC and ADC sample rates that can be configured using the SR register.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 50 of 123 © 2015 Dialog Semiconductor Table 37: Sample rate control register and corresponding system clock frequency Sample rate, FS (kHz) SR register System clock frequency (MHz) 8 0001 12.288 11.025 0010 11.2896 12 0011 12.288 16 0101 12.288 22.05 0110 11.2896 24 0111 12.288 32 1001 12.288 44.1 1010 11.2896 48 1011 12.288 88.2 1110 11.2896 96 1111 12.288 If digital playback or record is required in bypass mode then the MCLK frequency should be set to programmed accordingly. If no valid MCLK is detected, the output of the internal reference oscillator is used instead. However in this case only analogue bypass paths may be used.

9.6.4 Normal PLL mode (DAI master)

The DA7212 contains a Phase Locked Loop (PLL) that can be used to generate the required 11.2896 MHz or 12.288 MHz internal system clock when a frequency of between 2 and 54 MHz is applied to MCLK. This allows sharing of clocks between devices in an application, reducing total system cost. For example, the codec may operate from common 13 MHz or 19.2 MHz system clock frequency. The PLL is enabled by asserting pll_mode = 01. Once the PLL is enabled and has achieved phase lock, PLL bypass mode is disabled, and the output of the PLL is used as the system clock. The PLL input divider register (pll_indiv) is used to reduce the PLL reference frequency to the usable range of 2 to 54 MHz as shown in Table 38, this reduces the PLL reference frequency according to the following equation: FREF = FMCLK ÷ N Table 38: PLL input divider MCLK input frequency (MHz) Input divider, (÷N) pll_indiv register (0x27 [3:2]) 2 – 5 ÷1 000 5 – 10 ÷2 001 10 – 20 ÷4 010 20 – 40 ÷8 011 40 – 54 ÷16 100 The value of the PLL feedback divider is used to set the voltage controlled oscillator (VCO) frequency to 8 times the required system clock frequency (see Table 37). FVCO = FREF × PLL feedback divider

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 51 of 123 © 2015 Dialog Semiconductor The value of the PLL feedback divider is an unsigned number in the range of 0 to 128. It consists of seven integer bits and 13 fractional bits split across three registers:

PLL_INTEGER holds the seven integer bits
PLL_FRAC_BOT holds the top bits (MSB) of the fractional part of the divisor
PLL_FRAC_BOT holds the bottom bits (LSB) of the fractional part of the divisor

9.6.5 Example calculation of the feedback divider setting:

We will use as an example a codec operating with Fs (sample rate) = 48 kHz and a reference input clock frequency of 12.288 MHz. The required output frequency is 98.304 MHz. The reference clock input = 12.288 MHz, which falls in the range 10-20 MHz so pll_indiv be set to 0b010 (dividing the reference input frequency by 4 – see Table 38. The formula for calculating the feedback divider is: Feedback divider (F) = VCO output frequency * input divider (pll_indiv) / reference input clock Feedback divider = (98.304 * 4) / 12.288 = 32 So

pll_fbdiv_integer (holding the seven integer bits) = 0x20
pll_fbdiv_frac_top (holding the top bits (MSB) of the fractional part of the divisor) = 0x00
pll_fbdiv_frac_bot (holding the bottom bits (LSB) of the fractional part of the divisor) = 0x00 Table 39 shows example register settings that will configure the PLL when using a 13 MHz, 15 MHz or 19.2 MHz clock. Note that any MCLK input frequency between 2 and 54 MHz is supported. pll_indiv must be used to reduce the PLL reference frequency to the usable range of 2 to 5 MHz as shown in Table 39. Table 39: Example PLL configurations MCLK input frequency (MHz) System clock frequency (MHz) pll_mode register PLL_FRAC_TOP register PLL_FRAC_BOT register PLL_INTEGER register 13 11.2896 0x01 0x19 0x45 0x1B 13 12.288 0x01 0x07 0xEA 0x1E 15 11.2896 0x01 0x02 0xB4 0x18 15 12.288 0x01 0x06 0xDC 0x1A 19.2 11.2896 0x01 0x1A 0x1C 0x12 19.2 12.288 0x01 0x0F 0x5C 0x14

9.6.6 SRM PLL mode (DAI slave)

SRM mode enables the PLL output clock to be synchronized to the incoming WCLK signal on the DAI. The SRM PLL mode is enabled by setting pll_mode = 10. When using the digital audio interface in slave mode with the SRM enabled, removing and re - applying the DAI interface word clock WCLK may cause the PLL lock to be lost. To re -lock the PLL it is recommended that you disable the SRM (pll_mode = 00), reset the PLL by re-writing to register PLL_INTEGER, and then re-enable the SRM (pll_mode = 10) after the DAI WCLK has been reapplied. When switching sample rates between 44.1 kHz and 48 kHz (or between the multiples of these sample rates), SRM must be disabled and then re enabled using register bit pll_mode.

9.7 Reference generation

9.7.1 Voltage references

The audio circuits use supply-derived references of 0.45×VDD (VMID) and 0.9×VDD (DACREF). There is also bandgap-derived fixed voltage reference of 1.2 V (VREF). All three voltage references require off-chip decoupling capacitors (see Appendix B.6). Both VREF and VMID are automatically enabled whenever the device enters ACTIVE mode. They are automatically disabled when entering STANDBY mode. The VMID reference comes from a high-resistance voltage divider, which combines with the decoupling capacitor to create a large RC (resistance-capacitance) time constant. This ensures a noise-free VMID reference. To minimise startup time, set vmid_fast_charge = 1. This enables a low resistance path to charge the decoupling capacitor faster. Fast charge (vmid_fast_charge) must be disabled after start-up as it will increase the noise on the VMID reference. The bandgap reference VREF also takes time to charge its decoupling capacitor, but an internal timer ensures that no circuit that requires VREF is enabled until VREF has reached 1.2 V. The DACREF voltage reference is produced from VMID by a times-two buffer so is capable of charging its decoupling capacitor quickly.

9.7.2 Bias currents

DA7219 has a master bias current generation block that is enabled by default us ing the bias_en bit. Master bias current generation is set to ‘On’ by default. Each sub system has its own local current generation block, which is automatically enabled whenever any of its sub-blocks are enabled.

9.7.3 Voltage levels

9.7.4 IO voltage level

The digital input/output pins can be set to operate in either a high voltage (2.5 to 3.6 V) or low voltage (1.5 to 2.5 V) range using the io_voltage_level bit. See Table 40. Table 40: IO voltage level setting io_voltage_level setting Digital I/O voltage range (V) 0 2.5 to 3.6 1 1.2 to 2.5

9.8 I2C control interface

The DA7219 is completely software-controlled from the host by registers. The DA7219 provides an I2C compliant serial control interface to access these registers. Data is shifted into or out of the DA7219 under the control of the host processor, which also provides the serial clock. The I2C clock is supplied by the SCL line and the bi-directional I2C data is carried by the SDA line. The I2C interface is open-drain supporting multiple devices on a single line. The bus lines have to be pulled HIGH by external pull-up resistors (1 kΩ to 20 kΩ range). The attached devices only drive the bus lines LOW by connecting them to ground. This means that two devices cannot conflict if they drive the bus simultaneously. Table 41: Device I2C slave addresses Register cif_i2c_addr_cfg Device I2C address 00 0x18 01 0x19 10 0x1A (default) 11 0x1B In standard/fast mode the highest frequency of the bus is 1 MHz. The exact frequency can be determined by the application and does not have any relation to the DA7219 internal clock signals. DA7219 will follow the host clock speed within the described limitations and does not initiate any clock arbitration or slow down. In high-speed mode the maximum frequency of the bus can be increased up to 3.4 MHz. This mode is supported if the SCL line is driven with a push-pull stage from the host and if the host enables an external 3 mA pull-up at the SDA pin to decrease the rise time of the data. In this mode the SDA line on DA7219 is able to sink up to 12 mA. In all other respects the high speed mode behaves as the standard/fast mode. Communication on the I2C bus always takes place between two devices, one acting as the master and the other as the slave. The DA7219 will only operate as a slave. The I2C interface has direct access to the whole register map of the DA7219. Host Processor CodecSDA SCL Peripheral Device SDA SCL SCL SDA VDD_IO VDD_IO Figure 22: Schematic of the I2C control interface bus All data is transmitted across the I2C bus in groups of eight bits. To send a bit the SDA line is driven to the intended state while the SDA is LOW (a LOW on SDA indicates a zero bit). Once the SDA has settled, the SCL line is brought HIGH and then LOW. This pulse on SCL clocks the SDA bit into the receiver’s shift register. A two byte serial protocol is used containing one byte for address and one byte for data. Data and address transfer is transmitted MSB first for both read and write operations. All transm ission begins with the START condition from the master while the bus is in the IDLE state (the bus is free). It is initiated by a high to low transition on the SDA line while the SCL is in the high state (a STOP condition is indicated by a low to high transition on the SDA line while the SCL line is in the high state).

9.9 Digital Audio Interface (DAI)

DA7219 provides one Digital Audio Interface (DAI) to input DAC data or to output ADC data. It is enabled by asserting dai_en. The DSP provides flexible routing options allowing each interface to be connected to different signal paths as desired in each application. The DAI consists of a four-wire serial interface, with bit clock (BCLK), word clock (WCLK), data-in (DATIN) and data-out (DATOUT) pins. Both master and slave clock modes are supported by the DA7219. Master mode is enabled by setting register dai_clk_en = 1. In master mode, the bit clock and word clock signals are outputs from the codec. In slave mode these are inputs to the codec. DA7219 Codec Processor BCLK WCLK DATIN DATOUT Figure 29: Master mode (dai_clk_en = 1) DA7219 Codec Processor BCLK WCLK DATIN DATOUT Figure 30: Slave mode (dai_clk_en = 0) The internal serialised DAI data is 24 bits wide. Serial data that is not 24 bits wide is either shortened or zero-filled at input to, or at output from, the DAI’s internal 24-bit data width. The serial data word length can be configured to be 16, 20, 24 or 32 bits wide using the dai_word_length register bits. Four different data formats are supported by the digital audio interface. The data format is determined by the setting of the dai_format register bits.

I2S mode
Left Justified mode
Right Justified mode
DSP mode Time division multiplexing (TDM) is available in any of these modes to support the case where multiple devices are communicating simultaneously on the same bus. TDM is enabled by asserting the dai_tdm_mode_en bit.

9.9.1 DAI channels

The DAI supports one to two channels, even in non-TDM modes. The number of channels required is specified by setting dai_ch_num which controls the position of the channels. In TDM mode, each of the two channels can be individually enabled using the dai_tdm_ch_en register.

9.9.2 I2S mode

In I2S mode (dai_format = 0), the MSB of the left channel is valid on the second rising edge of the bit clock after the falling edge of the word clock. The MSB of the right channel is valid on the second rising edge of the bit clock after the rising edge of the word clock. BCLK WCLK msb Right Channel lsb msb Left Channel lsbDATIN/DATOUT msb WCLK 1 = RIGHT CHANNEL DATA WCLK 0 = LEFT CHANNEL DATA Figure 31: I2S mode

9.9.3 Left justified mode

In left-justified mode (dai_format = 1), the MSB of the right channel is valid on the rising edge of the bit clock following the falling edge of the word clock. The MSB of the left channel is valid on the rising edge of the bit clock following the rising edge of the word clock. BCLK WCLK msb Left Channel lsb msb Right Channel lsbDATIN/DATOUT msb WCLK 1 = LEFT CHANNEL DATA WCLK 0 = RIGHT CHANNEL DATA Figure 32: Left justified mode

9.9.4 Right justified mode

In right-justified mode (dai_format = 2), the LSB of the left channel is valid on the rising edge of the bit clock preceding the falling edge of word clock. The LSB of the right channel is valid on the rising edge of the bit clock preceding the rising edge of the word clock. BCLK WCLK msb Left Channel lsb msb Right Channel lsbDATIN/DATOUT lsb WCLK 1 = LEFT CHANNEL DATA WCLK 0 = RIGHT CHANNEL DATA Figure 33: Right justified mode

9.9.5 DSP mode

In DSP mode (dai_format = 3), the rising edge of the word clock starts the data transfer with the left channel data first and immediately followed by the right channel data. Each data bit is valid on the falling edge of the bit clock. BCLK WCLK msb Left Channel lsb msb Right Channel lsbDATIN/DATOUT msb The falling edge of WCLK can occur anywhere in this area The falling edge of WCLK can occur anywhere in this area BCLK WCLK msb Left Channel lsb msb Right Channel lsbDATIN/DATOUT msb Offset Figure 34: DSP mode

10 Register maps and definitions

Table 42: Register map accdet_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x000000C0 ACCDET_STA TUS_A Reserved micbias_up_ sts jack_pin_ord er_sts jack_type_sts jack_insertio n_sts 0x000000C1 ACCDET_STA TUS_B button_type_sts 0x000000C2 ACCDET_IRQ _EVENT_A Reserved e_jack_detec t_complete e_jack_remo ved e_jack_insert ed 0x000000C3 ACCDET_IRQ _EVENT_B e_button_a_r eleased e_button_b_r eleased e_button_c_r eleased e_button_d_r eleased e_button_d_ pressed e_button_c_p ressed e_button_b_ pressed e_button_a_ pressed 0x000000C4 ACCDET_IRQ _MASK_A Reserved m_jack_dete ct_complete m_jack_remo ved m_jack_inser ted 0x000000C5 ACCDET_IRQ _MASK_B m_button_a_ released m_button_b_ released m_button_c_ released m_button_d_ released m_button_d_ pressed m_button_c_ pressed m_button_b_ pressed m_button_a_ pressed 0x000000C6 ACCDET_CO NFIG_1 pin_order_de t_en jack_type_de t_en mic_det_thresh button_config accdet_en 0x000000C7 ACCDET_CO NFIG_2 jackdet_rem_deb jack_detect_rate jackdet_debounce accdet_paus e 0x000000C8 ACCDET_CO NFIG_3 A_D_BUTTON_THRESH 0x000000C9 ACCDET_CO NFIG_4 D_B_BUTTON_THRESH 0x000000CA ACCDET_CO NFIG_5 B_C_BUTTON_THRESH 0x000000CB ACCDET_CO NFIG_6 C_MIC_BUTTON_THRESH 0x000000CC ACCDET_CO NFIG_7 Reserved jack_type_for ce pin_order_for ce adc_1_bit_repeat button_average 0x000000CD ACCDET_CO NFIG_8 Reserved hptest_comp Reserved hptest_res_sel hptest_en

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 60 of 123 © 2015 Dialog Semiconductor Table 43: ACCDET_STATUS_A (Page 0: 0x000000C0) Bit Mode Symbol Description Reset

3 R micbias_up_sts Status of the microphone supply rail MICBIAS

0 = MICBIAS off 1 = MICBIAS On MICBIAS is enabled automatically when a four-pole jack is inserted 0x0 2 R jack_pin_order_sts Status of the jack pin-order detection. Pins are measured in Tip-Ring1-Ring2-Sleeve order. 0 = LRGM (CTIA format) 1 = LRMG (OMTP format) The data in this bit field is only valid after the e_jack_detect_complete event has fired, that is, once e_jack_detect_complete = 1 0x0 1 R jack_type_sts Status of the jack-type detection. 0 = 3-pole jack detected 1 = 4-pole jack detected The data in this bit field is only valid after the e_jack_detect_complete event has fired, that is, once e_jack_detect_complete = 1 0x0

0 R jack_insertion_sts Jack insertion status

0 = No jack is present 1 = Jack is present 0x0 Table 44: ACCDET_STATUS_B (Page 0: 0x000000C1) Bit Mode Symbol Description Reset 7:0 R button_type_sts The last measured 8-bit button impendance value from the ADC. 0x0 Table 45: ACCDET_IRQ_EVENT_A (Page 0: 0x000000C2) Bit Mode Symbol Description Reset

2 R e_jack_detect_compl

Jack detection IRQ event field. This is asserted once the jack detection has completed. This is a 'write 1 to clear' field. jack_type_sts and jack_pin_order_sts status bits are only valid after this event has been asserted. 0x0 1 R e_jack_removed Jack removal IRQ event field. This is asserted when a jack is removed. This is a 'write 1 to clear' field. 0x0 0 R e_jack_inserted Jack insertion IRQ event field. This is asserted when a jack is inserted. This is a 'write 1 to clear' field. 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 61 of 123 © 2015 Dialog Semiconductor Table 46: ACCDET_IRQ_EVENT_B (Page 0: 0x000000C3) Bit Mode Symbol Description Reset 7 R e_button_a_released Button A release IRQ event field. This is asserted when Button A is released. This is a 'write 1 to clear' field. 0x0 6 R e_button_b_released Button B release IRQ event field. This is asserted when Button B is released. This is a 'write 1 to clear' field. 0x0 5 R e_button_c_released Button C release IRQ event field. This is asserted when Button C is released. This is a 'write 1 to clear' field. 0x0 4 R e_button_d_released Button D release IRQ event field. This is asserted when Button D is released. This is a 'write 1 to clear' field. 0x0 3 R e_button_d_pressed Button A press IRQ event field. This is asserted when Button A is pressed. This is a 'write 1 to clear' field. 0x0 2 R e_button_c_pressed Button B press IRQ event field. This is asserted when Button B is pressed. This is a 'write 1 to clear' field. 0x0 1 R e_button_b_pressed Button C press IRQ event field. This is asserted when Button C is pressed. This is a 'write 1 to clear' field. 0x0 0 R e_button_a_pressed Button D press IRQ event field. This is asserted when Button D is pressed. This is a 'write 1 to clear' field. 0x0 Table 47: ACCDET_IRQ_MASK_A (Page 0: 0x000000C4) Bit Mode Symbol Description Reset

2 R/W m_jack_detect_comp

Interrupt mask for e_jack_detect_complete 0 = Jack detection IRQ is not masked 1 = Jack detection IRQ is masked 0x0

1 R/W m_jack_removed Interrupt mask for e_jack_removed

0 = Jack removal IRQ is not masked 1 = Jack removal IRQ is masked 0x0

0 R/W m_jack_inserted Interrupt mask for e_jack_inserted

0 = Jack insertion IRQ is not masked 1 = Jack insertion IRQ is masked 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 62 of 123 © 2015 Dialog Semiconductor Table 48: ACCDET_IRQ_MASK_B (Page 0: 0x000000C5) Bit Mode Symbol Description Reset

7 R/W m_button_a_release

d Interrupt mask for e_button_a_released 0 = Button A release IRQ is not masked 1 = Button A release IRQ is masked 0x0

6 R/W m_button_b_release

d Interrupt mask for e_button_b_released 0 = Button B release IRQ is not masked 1 = Button B release IRQ is masked 0x0

5 R/W m_button_c_release

d Interrupt mask for e_button_c_released 0 = Button C release IRQ is not masked 1 = Button C release IRQ is masked 0x0

4 R/W m_button_d_release

d Interrupt mask for e_button_d_released 0 = Button D release IRQ is not masked 1 = Button D release IRQ is masked 0x0

3 R/W m_button_d_pressed Interrupt mask for e_button_a_pressed

0 = Button A press IRQ is not masked 1 = Button A press IRQ is masked 0x0

2 R/W m_button_c_pressed Interrupt mask for e_button_b_pressed

0 = Button B press IRQ is not masked 1 = Button B press IRQ is masked 0x0

1 R/W m_button_b_pressed Interrupt mask for e_button_c_pressed

0 = Button C press IRQ is not masked 1 = Button C press IRQ is masked 0x0

0 R/W m_button_a_pressed Interrupt mask for e_button_d_pressed

0 = Button D press IRQ is not masked 1 = Button D press IRQ is masked 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 63 of 123 © 2015 Dialog Semiconductor Table 49: ACCDET_CONFIG_1 (Page 0: 0x000000C6) Bit Mode Symbol Description Reset

7 R/W pin_order_det_en Controls detection of the pin order on insertion of a

0 = Pin order is determined by the setting of the pin_order_force register field when jack_type_sts = 4-pole 1 = Pin order detection ( LRGM / LRMG ) runs on insertion of a 4-pole Jack 0x1

6 R/W jack_type_det_en Controls detection of the type of jack (3-pole without

a mic or 4-pole with a mic) when a jack is inserted 0 = The type of jack (3-pole or 4-pole) is determined by the setting of the jack_type_force register field 1 = Jack type detection runs on jack insertion to determine Jack Type (3-pole with no mic, or 4-pole with a mic) 0x1 5:4 R/W mic_det_thresh Impedance threshold for MIC detection measurement. If SLEEVE to RING2 impedance is below the threshold specified here, jack_type_sts is set to 0 (3-pole). If SLEEVE to RING2 impedance is above the threshold specified here, jack_type_sts is set to 1 (4-pole). 00 = 200 Ω 01 = 500 Ω (default) 10 = 750 Ω 11 = 1000 Ω 0x1 3:1 R/W button_config Specifies the time between the periodic button- press measurements when jack_type_sts = 1 (4- Pole). 000 = Disabled 001 = 2 ms 010 = 5 ms 011 = 10 ms (default) 100 = 50 ms 101 = 100 ms 110 = 200 ms 111 = 500 ms 0x3

0 R/W accdet_en Controls Accessory Detection

0 = Accessory Detection is disabled 1 = Accessory Detection is enabled The ACCDET analogue components require master bias to be enabled before enabling the digital block 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 64 of 123 © 2015 Dialog Semiconductor Table 50: ACCDET_CONFIG_2 (Page 0: 0x000000C7) Bit Mode Symbol Description Reset 7:6 R/W jackdet_rem_deb Control of the JACKDET deassertion debounce 00 = 1 ms (default) 01 = 5 ms 10 = 10 ms 11 = 20 ms 0x0 5:4 R/W jack_detect_rate Controls the jack-detection latency time, that is, the time from assertion of e_jack_insertion to assertion of e_jack_detect_complete 3-pole jack: 00 = 32 ms 01 = 64 ms 10 = 128 ms 11 = 256 ms (default) 4-pole jack: 00 = 64 ms 01 = 128 ms 10 = 256 ms 11 = 512 ms (default) Latency time is altered by changing the ramp rate of the MICDET current during jack type and pin order detection. 0x3 3:1 R/W jackdet_debounce Control of the JACKDET assertion debounce 0 = 5 ms 1 = 10 ms 2 = 20 ms (default) 3 = 50 ms 4 = 100 ms 5 = 200 ms 6 = 500 ms 7 = 1 s 0x2

0 R/W accdet_pause Pauses the periodic button checking within the

accessory detection function. This allows you to reconfigure the button measurements or to change MICBIAS or both. 0 = No effect 1 = Pauses the periodic button checking within the accessory detection block The difference between pausing by asserting this register field and disabling the accessory detection function entirely (accdet_en = 0) is that pausing allows for dynamic reconfiguration of the button measurements. When paused, the DA7219 chip will still respond to new removal or insertion events whereas when disabled, no insertion or removal events are detected. 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 65 of 123 © 2015 Dialog Semiconductor Table 51: ACCDET_CONFIG_3 (Page 0: 0x000000C8) Bit Mode Symbol Description Reset 7:0 R/W A_D_BUTTON_THR ESH Sets the impedance threshold between Button A and Button D. If the measured impedance of a button-press is lower than the threshold value specified here, the button that was pressed is Button A. The value of this register field is a calculated value. It is calculated as: 256 * Required threshold in Ω/(Required threshold in Ω + MICBIAS resistance in Ω) Example calculation: Assuming that MICBIAS resistance = 2200 Ω and the required threshold = 89 Ω, the bit value of this register field = 256 * 89/(89 + 2200) = 10 [or 0x0A]. So, in this example, setting this register field = 0x0A will give you a threshold value of 89 Ω. 0xA Table 52: ACCDET_CONFIG_4 (Page 0: 0x000000C9) Bit Mode Symbol Description Reset 7:0 R/W D_B_BUTTON_THR ESH Sets the impedance threshold between Button D and Button B. If the measured impedance of a button-press is lower than the threshold value specified here, the button that was pressed is Button D. The value of this register field is a calculated value. It is calculated as: 256 * Required threshold in Ω/(Required threshold in Ω + MICBIAS resistance in Ω) Example calculation: Assuming that MICBIAS resistance = 2200 Ω and the required threshold = 195 Ω, the bit value of this register field = 256 * 195/(195 + 2200) = 21 [or 0x15]. So, in this example, setting this register field = 0x15 will give you a threshold value of 195 Ω. 0x16

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 66 of 123 © 2015 Dialog Semiconductor Table 53: ACCDET_CONFIG_5 (Page 0: 0x000000CA) Bit Mode Symbol Description Reset 7:0 R/W B_C_BUTTON_THR ESH Sets the impedance threshold between Button B and Button C. If the measured impedance of a button-press is lower than the threshold value specified here, the button that was pressed is Button B. The value of this register field is a calculated value. It is calculated as: 256 * Required threshold in Ω/(Required threshold in Ω + MICBIAS resistance in Ω) Example calculation: Assuming that MICBIAS resistance = 2200 Ω and the required threshold = 325 Ω, the bit value of this register field = 256 * 325/(325 + 2200) = 33 [or 0x21]. So, in this example, setting this register field = 0x21 will give you a threshold value of 325 Ω. 0x21 Table 54: ACCDET_CONFIG_6 (Page 0: 0x000000CB) Bit Mode Symbol Description Reset 7:0 R/W C_MIC_BUTTON_T HRESH Sets the impedance threshold between Button C and the microphone. If the measured impedance of a button-press is lower than the threshold value specified here, the button that was pressed is Button C. The value of this register field is a calculated value. It is calculated as: 256 * Required threshold in Ω/(Required threshold in Ω + MICBIAS resistance in Ω) Example calculation: Assuming that MICBIAS resistance = 2200 Ω and the required threshold = 688 Ω, the bit value of this register field = 256 * 688/(688 + 2200) = 61 [or 0x3D]. So, in this example, setting this register field = 0x3D will give you a threshold value of 688 Ω. 0x3E

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 67 of 123 © 2015 Dialog Semiconductor Table 55: ACCDET_CONFIG_7 (Page 0: 0x000000CC) Bit Mode Symbol Description Reset

5 R/W jack_type_force Specifies the Jack type when jack type detection is

disabled (jack_type_det_en is 0) 0 = 3-pole jack is specified 1 = 4-pole jack is specified 0x0

4 R/W pin_order_force Specifies the jack pin order for 4-Pole Jacks when

pin order detection is disabled (pin_order_det_en = 0 = LRGM (CTIA format) 1 = LRMG (OMTP format) 0x0 3:2 R/W adc_1_bit_repeat Sets the number of repeated 1-bit measurements. Repeating the 1-bit measurements multiple times gives greater noise immunity but adds latency, and possible distortion, during periodic button checking 00 = One 1-bit measurement (default) 01 = Two 1-bit measurements 10 = Four 1-bit measurements 11 = Eight 1-bit measurements 0x0 1:0 R/W button_average Sets the number of repeated 8-bit ADC measurements used to generate an averaged result Using more measurements for averaging will increase button-checking noise immunity but also increases the detection latency by about 1 ms per measurement 00 = One 8-bit measurement (no averaging) 01 = Two 8-bit measurements used for averaging (default) 10 = Four 8-bit measurements used for averaging 11 = Eight 8-bit measurements used for averaging 0x1 Table 56: ACCDET_CONFIG_8 (Page 0: 0x000000CD) Bit Mode Symbol Description Reset

4 R hptest_comp HP TEST comparator result

1 = HP Impedance is < Threshold 0 = HP Impedance is > Threshold 0x0 2:1 R/W hptest_res_sel HP Impedance Test threshold control 00 - 1000 Ω 01 = 2500 Ω 10 = 5000 Ω 11 = 10000 Ω 0x1

0 R/W hptest_en Headphone Impedance test block control

0 = HP Impedance test block disabled 1 = HP Impedance test block enabled 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 68 of 123 © 2015 Dialog Semiconductor Table 57: Register map adc_filters_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000038 ADC_FILTER adc_hpf_en Reserved adc_audio_hpf_corner adc_voice_e n adc_voice_hpf_corner Table 58: ADC_FILTERS1 (Page 0: 0x00000038) Bit Mode Symbol Description Reset

7 R/W adc_hpf_en ADC high pass filter control

0 = ADC high pass filter disabled 1 = ADC high pass filter enabled 0x1 5:4 R/W adc_audio_hpf_corn er ADC high pass filter 3 dB cut-off point. At 48 kHz, the cutoff point is at: 00 = 2 Hz 01 = 4 Hz 10 = 8 Hz 11 = 16 Hz For other sample rates the 3 dB cuttoff point scales proportionately 0x0

3 R/W adc_voice_en ADC voice filter control

0 = Voice filter disabled 1 = Voice filter enabled 0x0 2:0 R/W adc_voice_hpf_corn er Voice (8 kHz) High pass 3 dB cutoff point 000 = 2.5 Hz 001 = 25 Hz 010 = 50 Hz 011 = 100 Hz 100 = 150 Hz 101 = 200 Hz 110 = 300 Hz 111 = 400 Hz For other sample rates the 3 dB cuttoff point scales proportionately 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 69 of 123 © 2015 Dialog Semiconductor Table 59: Register map alc_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x0000002F ALC_CTRL1 Reserved alc_calib_ove rflow alc_auto_cali b_en alc_en Reserved alc_sync_mo de alc_offset_en 0x0000009A ALC_CTRL2 alc_release alc_attack 0x0000009B ALC_CTRL3 alc_integ_release alc_integ_attack alc_hold 0x0000009C ALC_NOISE Reserved alc_noise 0x0000009D ALC_TARGET _MIN Reserved alc_threshold_min 0x0000009E ALC_TARGET _MAX Reserved alc_threshold_max 0x0000009F ALC_GAIN_LI MITS alc_gain_max alc_atten_max 0x000000A0 ALC_ANA_GA IN_LIMITS Reserved alc_ana_gain_max Reserved alc_ana_gain_min 0x000000A1 ALC_ANTICLI P_CTRL alc_antipclip_ en Reserved alc_anticlip_step 0x000000A2 ALC_ANTICLI P_LEVEL Reserved alc_anticlip_level 0x000000A3 ALC_OFFSET _AUTO_M_L alc_offset_auto_m_l 0x000000A4 ALC_OFFSET _AUTO_U_L Reserved alc_offset_auto_u_l

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 70 of 123 © 2015 Dialog Semiconductor Table 60: ALC_CTRL1 (Page 0: 0x0000002F) Bit Mode Symbol Description Reset

5 R alc_calib_overflow Indicates that an offset overflow occurred during

0 = No offset overflow 1 = Offset overflow occurred 0x0

4 R/W alc_auto_calib_en Automatic calibration control

0 = Automatic calibration not enabled 1 = Automatic calibration enabled This is a self-clearing bit 0x0

3 R/W alc_en Controls the ALC operation on the left ADC channel

0 = ALC is disabled 1 = ALC is enabled 0x0 1 R/W alc_sync_mode ALC hybrid mode control. Hybrid mode uses both analogue and digital gains. 0 = Hybrid mode is Off (digital gain only) 1 = Hybrid mode is On (digital and analogue gain) 0x0

0 R/W alc_offset_en DC Offset cancellation control

0 = DC Offset cancellation is disabled 1 = DC Offset cancellation is enabled 0x0 Table 61: ALC_CTRL2 (Page 0: 0x0000009A) Bit Mode Symbol Description Reset 7:4 R/W alc_release Sets the ALC release rate. This is the speed at which the ALC increases the gain. 0000 = 28.66/Fs (0.6 ms/dB @48 kHz) 0001 = 57.33/Fs (1.2 ms/dB @48 kHz) 0010 = 114.66/Fs (2.4 ms/dB @48 kHz) then doubling at every step to... 1001 = 14674/Fs (306 ms/dB @48 kHz) 1010 to 1111 = 29348/Fs (611 ms/dB @48 kHz) 0x0 3:0 R/W alc_attack ALC attack rate control. This is the speed at which the ALC reduces the gain. 0000 = 7.33/Fs (0.153 ms/dB @48 kHz) 0001 = 14.66/Fs (0.305 ms/dB @48 kHz) 0010 = 29.32/Fs (0.612 ms/dB @48 kHz) then doubling at every step to... 1011 = 15012/Fs (312 ms/dB @48 kHz) 1100 to 1111 = 30024/Fs (625 ms/dB @48 kHz) 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 71 of 123 © 2015 Dialog Semiconductor Table 62: ALC_CTRL3 (Page 0: 0x0000009B) Bit Mode Symbol Description Reset 7:6 R/W alc_integ_release Controls the rate at which the input signal envelope is tracked as the signal gets smaller 00 = 1/4 01 = 1/16 10 = 1/256 11 = 1/65537 0x0 5:4 R/W alc_integ_attack Comtrols the rate at which the input signal envelope is tracked as the signal gets larger 00 = 1/4 01 = 1/16 10 = 1/256 11 = 1/65537 0x0 3:0 R/W alc_hold ALC hold time control. This is the period the ALC waits before releasing. 0000 = 62/Fs (1.3 ms @48 kHz) 0001 = 124/Fs (2.6 ms @48 kHz) 0010 = 248/Fs (5.2 ms @48 kHz) then doubling at every step to… 1110 = 1015808/Fs (21 s @48 kHz) 1111 = 2031616/Fs (42 s @48 kHz) 0x0 Table 63: ALC_NOISE (Page 0: 0x0000009C) Bit Mode Symbol Description Reset 5:0 R/W alc_noise Sets the threshold below which input signals will not cause the ALC to change gain 000000 = 0 dBFS 000001 = -1.5 dBFS 000010 = -3.0 dBFS then continuing in -1.5 dBFS steps to… 111110 = -93.0 dBFS 111111 = -94.5 dBFS (default) 0x3F

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 72 of 123 © 2015 Dialog Semiconductor Table 64: ALC_TARGET_MIN (Page 0: 0x0000009D) Bit Mode Symbol Description Reset 5:0 R/W alc_threshold_min Sets the minimum amplitude of the ALC output signal at which the ALC increases the gain. If the minimum attenution level is reached, the ALC will not increase the gain even if this threshold is breached. 000000 = 0 dBFS 000001 = -1.5 dBFS 000010 = -3.0 dBFS then continuing in -1.5 dBFS steps to... 111110 = -93.0 dBFS 111111 = -94.5 dBFS (default) 0x3F Table 65: ALC_TARGET_MAX (Page 0: 0x0000009E) Bit Mode Symbol Description Reset 5:0 R/W alc_threshold_max Sets the maximum amplitude of the ALC output signal at which the ALC reduces the gain. If the maximum attenution level is reached, the ALC will not reduce the gain even if this threshold is exceeded. 000000 = 0 dBFS 000001 = -1.5 dBFS 000010 = -3.0 dBFS then continuing in -1.5 dBFS steps to... 111110 = -93.0 dBFS 111111 = -94.5 dBFS 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 73 of 123 © 2015 Dialog Semiconductor Table 66: ALC_GAIN_LIMITS (Page 0: 0x0000009F) Bit Mode Symbol Description Reset 7:4 R/W alc_gain_max Sets the maximum amount of gain that can be applied to the input signal by the ALC when the input signal is large relative to the maximum threshold 0000 = 0 dB 0001 = 6 dB 0010 = 12 dB then continuing in 6 dB steps to… 1110 = 84 dB 1111 = 90 dB 0xF 3:0 R/W alc_atten_max Sets the maximum amount of attenuation that can be applied to the input signal by the ALC when the input signal is large relative to the maximum threshold 0000 = 0 dB 0001 = 6 dB 0010 = 12 dB then continuing in 6 dB steps to… 1110 = 84 dB 1111 = 90 dB 0xF

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 74 of 123 © 2015 Dialog Semiconductor Table 67: ALC_ANA_GAIN_LIMITS (Page 0: 0x000000A0) Bit Mode Symbol Description Reset 6:4 R/W alc_ana_gain_max Sets the maximum amount of analogue gain that can be applied to the input signal by the ALC when the input signal is large relative to the maximum threshold. This setting applies only to mixed analogue and digital gain mode (alc_sync_mode = 1). 000 = reserved 001 = 0 dB 010 = 6 dB 011 = 12 dB 100 = 18 dB 101 = 24 dB 110 = 30 dB 111 = 36 dB 0x7 2:0 R/W alc_ana_gain_min Sets the minimum amount of analogue gain that can be applied to the input signal by the ALC when the input signal is large relative to the maximum threshold. This setting applies only to mixed analogue and digital gain mode (alc_sync_mode = 1). 000 = reserved 001 = 0 dB 010 = 6 dB 011 = 12 dB 100 = 18 dB 101 = 24 dB 110 = 30 dB 111 = 36 dB 0x1 Table 68: ALC_ANTICLIP_CTRL (Page 0: 0x000000A1) Bit Mode Symbol Description Reset

7 R/W alc_antipclip_en Controls the ALC signal clip prevention mechanism

0 = Clip prevention is disabled 1 = Clip prevention is enabled 0x0 1:0 R/W alc_anticlip_step Sets the attack rate for the ALC when the output signal exceeds the anticlip threshold level 00 = 0.034 dB/Fs 01 = 0.068 dB/Fs 10 = 0.136 dB/Fs 11 = 0.272 dB/Fs 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 75 of 123 © 2015 Dialog Semiconductor Table 69: ALC_ANTICLIP_LEVEL (Page 0: 0x000000A2) Bit Mode Symbol Description Reset 6:0 R/W alc_anticlip_level ALC anticlip threshold control. The ALC anticlip operates when signals are above this threshold. The formula used to calculate the threshold value, using 'x' to denote the decimal value of this bit field, is: x = ((x+1)/128) Fs 0x00 = 0.0078 Fs 0x01 = 0.0156 Fs 0x02 = 0.0234 Fs then contnuing in aproximately 0.0078 steps to... 0x7E = 0.9922 Fs 0x7F = 1.0 Fs 0x0 Table 70: ALC_OFFSET_AUTO_M_L (Page 0: 0x000000A3) Bit Mode Symbol Description Reset 7:0 R alc_offset_auto_m_l This read-only bit field contains the middle eight bits (bits [15:8]) of the value used for automatic offset correction 0x0 Table 71: ALC_OFFSET_AUTO_U_L (Page 0: 0x000000A4) Bit Mode Symbol Description Reset 3:0 R alc_offset_auto_u_l This read-only bit field contains the upper four bits (bits [19:16]) of the value used for automatic offset correction 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 76 of 123 © 2015 Dialog Semiconductor Table 72: Register map analogue_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000006 MIC_1_GAIN_ STATUS Reserved mic_1_amp_gain_status 0x00000008 MIXIN_L_GAI N_STATUS Reserved mixin_l_amp_gain_status 0x0000000A ADC_L_GAIN _STATUS Reserved adc_l_digital_gain_status 0x0000000C DAC_L_GAIN _STATUS Reserved dac_l_digital_gain_status 0x0000000D DAC_R_GAIN _STATUS Reserved dac_r_digital_gain_status 0x0000000E HP_L_GAIN_ STATUS Reserved hp_l_amp_gain_status 0x0000000F HP_R_GAIN_ STATUS Reserved hp_r_amp_gain_status 0x00000010 MIC_1_SELE CT Reserved mic_1_amp_in_sel 0x00000032 REFERENCE S Reserved vmid_fast_ch arge bias_en Reserved 0x00000033 MIXIN_L_SEL ECT Reserved mixin_l_mix_ select 0x00000034 MIXIN_L_GAI N Reserved mixin_l_amp_gain 0x00000036 ADC_L_GAIN Reserved adc_l_digital_gain 0x00000039 MIC_1_GAIN Reserved mic_1_amp_gain 0x00000045 DAC_L_GAIN Reserved dac_l_digital_gain 0x00000046 DAC_R_GAIN Reserved dac_r_digital_gain 0x00000048 HP_L_GAIN Reserved hp_l_amp_gain 0x00000049 HP_R_GAIN Reserved hp_r_amp_gain 0x0000004B MIXOUT_L_S ELECT Reserved mixout_l_mix _select

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 77 of 123 © 2015 Dialog Semiconductor Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x0000004C MIXOUT_R_S ELECT Reserved mixout_r_mix _select 0x00000062 MICBIAS_CT RL Reserved micbias1_en micbias1_level 0x00000063 en mic_1_amp_ mute_en mic_1_amp_r amp_en Reserved 0x00000065 MIXIN_L_CTR L mixin_l_amp _en mixin_l_amp _mute_en mixin_l_amp _ramp_en mixin_l_amp _zc_en mixin_l_mix_ en Reserved 0x00000067 en adc_l_ramp_ en Reserved 0x00000069 en dac_l_ramp_ en Reserved 0x0000006A en dac_r_ramp_ en Reserved 0x0000006B ute_en hp_l_amp_ra mp_en hp_l_amp_zc n_gain_en Reserved 0x0000006C n hp_r_amp_m ute_en hp_r_amp_ra mp_en hp_r_amp_zc _en hp_r_amp_o e hp_r_amp_m in_gain_en Reserved 0x0000006E MIXOUT_L_C TRL mixout_l_am p_en Reserved 0x0000006F MIXOUT_R_C TRL mixout_r_am p_en Reserved 0x00000091 IO_CTRL Reserved io_voltage_le vel Table 73: MIC_1_GAIN_STATUS (Page 0: 0x00000006) Bit Mode Symbol Description Reset 2:0 R mic_1_amp_gain_st atus Contains the currently active mic_1_amp gain setting 000 = -6 dB 001 = 0 dB 010 = 6 dB 011 = 12 dB 100 = 18 dB 101 = 24 dB 110 = 30 dB 111 = 36 dB 0x1

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 78 of 123 © 2015 Dialog Semiconductor Table 74: MIXIN_L_GAIN_STATUS (Page 0: 0x00000008) Bit Mode Symbol Description Reset 3:0 R mixin_l_amp_gain_st atus Contains the currently active mixin_l_amp gain setting 0000 = -4.5 dB 0001 = -3.0 dB 0010 = -1.5 dB 0011 = 0.0 dB then continuing in 1.5 dB steps to… 1110 = 16.5 dB 1111 = 18.0 dB 0x0 Table 75: ADC_L_GAIN_STATUS (Page 0: 0x0000000A) Bit Mode Symbol Description Reset 6:0 R adc_l_digital_gain_st atus Contains the currently active ADC_L digital gain setting 0x00 = -83.25 dB 0x01 = -82.5 dB then continuing in 0.75 dB steps through 0x6F = 0 dB to... 0x7E = 11.25 dB 0x7F = 12 dB 0x0 Table 76: DAC_L_GAIN_STATUS (Page 0: 0x0000000C) Bit Mode Symbol Description Reset 6:0 R dac_l_digital_gain_st atus Contains the currently active DAC_L digital gain setting 0x00 to 0x07 = mute 0x08 = -77.25 dB 0x09 = -76.5 dB then continuing in 0.75 dB steps through 0x6F = 0 dB to... 0x7E = 11.25 dB 0x7F = 12 dB 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 79 of 123 © 2015 Dialog Semiconductor Table 77: DAC_R_GAIN_STATUS (Page 0: 0x0000000D) Bit Mode Symbol Description Reset 6:0 R dac_r_digital_gain_st atus Contains the currently active DAC_R digital gain setting 0x00 to 0x07 = mute 0x08 = -77.25 dB 0x09 = -76.5 dB then continuing in 0.75 dB steps through 0x6F = 0 dB to... 0x7E = 11.25 dB 0x7F = 12 dB 0x0 Table 78: HP_L_GAIN_STATUS (Page 0: 0x0000000E) Bit Mode Symbol Description Reset 5:0 R hp_l_amp_gain_stat us Contains the currently active HP_L_AMP gain setting 000000 = -57.0 dB 000001 = -56.0 dB 000010 = -55.0 dB then continuing in 1 dB steps to… 111001 = 0.0 dB 111111 = 6.0 dB 0x0 Table 79: HP_R_GAIN_STATUS (Page 0: 0x0000000F) Bit Mode Symbol Description Reset 5:0 R hp_r_amp_gain_stat us Contains the currently active HP_R_AMP gain setting 000000 = -57.0 dB 000001 = -56.0 dB 000010 = -55.0 dB then continuing in 1 dB steps to… 111001 = 0.0 dB 111111 = 6.0 dB 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 80 of 123 © 2015 Dialog Semiconductor Table 80: MIC_1_SELECT (Page 0: 0x00000010) Bit Mode Symbol Description Reset 1:0 R/W mic_1_amp_in_sel MIC_1 input source control 00 = Differential 01 = MIC_1_P single-ended 10 = MIC_1_N single-ended 11 = Reserved 0x0 Table 81: REFERENCES (Page 0: 0x00000032) Bit Mode Symbol Description Reset

4 R/W vmid_fast_charge VMID reference fast charge control

0 = low noise, slow charge mode 1 = high noise, fast charge mode 0x0 3 R/W bias_en Master Bias control. Master Bias is required for analogue circuitry. 0 = Master Bias disabled 1 = Master Bias enabled 0x1 Table 82: MIXIN_L_SELECT (Page 0: 0x00000033) Bit Mode Symbol Description Reset

0 R/W mixin_l_mix_select MIXIN_L mixer input control

0 = No input selected 1 = MIC_1 selected as input 0x0 Table 83: MIXIN_L_GAIN (Page 0: 0x00000034) Bit Mode Symbol Description Reset 3:0 R/W mixin_l_amp_gain mixin_l_amp gain control 0000 = -4.5 dB 0001 = -3.0 dB 0010 = -1.5 dB 0011 = 0.0 dB then continuing in 1.5 dB steps to… 1110 = 16.5 dB 1111 = 18.0 dB 0x3

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 81 of 123 © 2015 Dialog Semiconductor Table 84: ADC_L_GAIN (Page 0: 0x00000036) Bit Mode Symbol Description Reset 6:0 R/W adc_l_digital_gain ADC_L digital gain control 00x0 = -83.25 dB 0x01 = -82.5 dB then continuing in 0.75 dB steps through 0x6F = 0 dB to... 0x7E = 11.25 dB 0x7F = 12 dB 0x6F Table 85: MIC_1_GAIN (Page 0: 0x00000039) Bit Mode Symbol Description Reset 2:0 R/W mic_1_amp_gain mic_1_amp gain control 000 = -6 dB 001 = 0 dB 010 = 6 dB 011 = 12 dB 100 = 18 dB 101 = 24 dB 110 = 30 dB 111 = 36 dB 0x1 Table 86: DAC_L_GAIN (Page 0: 0x00000045) Bit Mode Symbol Description Reset 6:0 R/W dac_l_digital_gain DAC_L digital gain control 0x00 to 0x07 = mute 0x08 = -77.25 dB 0x09 = -76.5 dB then continuing in 0.75 dB steps through 0x6F = 0 dB to... 0x7E = 11.25 dB 0x7F = 12 dB 0x6F

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 82 of 123 © 2015 Dialog Semiconductor Table 87: DAC_R_GAIN (Page 0: 0x00000046) Bit Mode Symbol Description Reset 6:0 R/W dac_r_digital_gain DAC_R digital gain control 0x00 to 0x07 = mute 0x08 = -77.25 dB 0x09 = -76.5 dB then continuing in 0.75 dB steps through 0x6F = 0 dB to... 0x7E = 11.25 dB 0x7F = 12 dB 0x6F Table 88: HP_L_GAIN (Page 0: 0x00000048) Bit Mode Symbol Description Reset 5:0 R/W hp_l_amp_gain HP_L_AMP gain control 000000 = -57.0 dB 000001 = -56.0 dB 000010 = -55.0 dB then continuing in 1 dB steps through… 111001 = 0.0 dB to… 111111 = 6.0 dB 0x39 Table 89: HP_R_GAIN (Page 0: 0x00000049) Bit Mode Symbol Description Reset 5:0 R/W hp_r_amp_gain HP_R_AMP gain control 000000 = -57.0 dB 000001 = -56.0 dB 000010 = -55.0 dB then continuing in 1 dB steps through… 111001 = 0.0 dB to… 111111 = 6.0 dB 0x39 Table 90: MIXOUT_L_SELECT (Page 0: 0x0000004B) Bit Mode Symbol Description Reset

0 R/W mixout_l_mix_select Output left mixer channel selection

0 = No channel selected 1 = DAC_L selected as output 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 83 of 123 © 2015 Dialog Semiconductor Table 91: MIXOUT_R_SELECT (Page 0: 0x0000004C) Bit Mode Symbol Description Reset

0 R/W mixout_r_mix_select Ouput right mixer channel selection

0 = No channel selected 1 = DAC_R selected as output 0x0 Table 92: MICBIAS_CTRL (Page 0: 0x00000062) Bit Mode Symbol Description Reset

3 R/W micbias1_en Microphone Bias 1 control

0 = Micbias1 disabled 1 = Micbias1 enabled 0x0 2:0 R/W micbias1_level Microphone Bias 1 level control 000 = 1.6 V 001 = 1.8 V 010 = 2.0 V 011 = 2.2 V 100 = 2.4 V 101 = 2.6 V 110 = 2.8 V 111 = 2.9 V This must only be modified while micbias_1 is disabled (micbias1_en = 0) 0x3 Table 93: MIC_1_CTRL (Page 0: 0x00000063) Bit Mode Symbol Description Reset

7 R/W mic_1_amp_en MIC_1 amplifier control

0 = MIC_1 disabled 1 = MIC_1 enabled 0x0

6 R/W mic_1_amp_mute_e

n MIC_1 amplifier mute control 0 = MIC_1 unmuted 1 = MIC_1 muted 0x1 5 - mic_1_amp_ramp_e n MIC_1 amplifier gain ramping control 0 = Gain changes are instant 1 = Gain changes are ramped to the new level 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 84 of 123 © 2015 Dialog Semiconductor Table 94: MIXIN_L_CTRL (Page 0: 0x00000065) Bit Mode Symbol Description Reset

7 R/W mixin_l_amp_en MIXIN_L amplifier control

0 = MIXIN_L disabled 1 = MIXIN_L enabled 0x0

6 R/W mixin_l_amp_mute_

MIXIN_L amplifier mute control 0 = MIXIN_L unmuted 1 = MIXIN_L muted 0x1

5 R/W mixin_l_amp_ramp_

MIXIN_L amplifier gain ramping control 0 = Gain changes are instant 1 = Gain changes are ramped to the new level This setting overrides zero crossing 0x0

4 R/W mixin_l_amp_zc_en MIXIN_L amplifier zero cross control

0 = Gain changes are instant 1 = Gain changes are performed when the signal crosses zero If no zero-crossing is detected within the timeout period of approximately 100 ms, the update is applied unconditionally 0x0 3 R/W mixin_l_mix_en MIXIN_L mixer control. When this mixer is disabled, all inputs are deselected. 0 = Mixer disabled 1 = Mixer enabled 0x0 Table 95: ADC_L_CTRL (Page 0: 0x00000067) Bit Mode Symbol Description Reset

7 R/W adc_l_en ADC_L control

0 = ADC_L disabled 1 = ADC_L enabled 0x0

6 R/W adc_l_mute_en ADC_L mute control

0 = ADC_L unmuted 1 = ADC_L muted 0x1

5 R/W adc_l_ramp_en ADC_L digital gain ramping control

0 = Gain changes are instant 1 = Gain changes are ramped to the new level 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 85 of 123 © 2015 Dialog Semiconductor Table 96: DAC_L_CTRL (Page 0: 0x00000069) Bit Mode Symbol Description Reset

7 R/W dac_l_en DAC_L control

0 = DAC_L disabled 1 = DAC_L enabled 0x0

6 R/W dac_l_mute_en DAC_L mute control

0 = DAC_L unmuted 1 = DAC_L muted 0x1

5 R/W dac_l_ramp_en DAC_L digital gain ramping control

0 = Gain changes are instant 1 = Gain changes are ramped to the new level 0x0 Table 97: DAC_R_CTRL (Page 0: 0x0000006A) Bit Mode Symbol Description Reset

7 R/W dac_r_en DAC_R control

0 = DAC_R disabled 1 = DAC_R enabled 0x0

6 R/W dac_r_mute_en DAC_R mute control

0 = DAC_R unmuted 1 = DAC_R muted 0x1

5 R/W dac_r_ramp_en DAC_R digital gain ramping control

0 = Gain changes are instant 1 = Gain changes are ramped to the new level 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 86 of 123 © 2015 Dialog Semiconductor Table 98: HP_L_CTRL (Page 0: 0x0000006B) Bit Mode Symbol Description Reset

7 R/W hp_l_amp_en HP_L_AMP amplifier control

0 = HP_L_AMP disabled 1 = HP_L_AMP enabled 0x0

6 R/W hp_l_amp_mute_en HP_L_AMP amplifier mute control

0 = HP_L_AMP unmuted 1 = HP_L_AMP muted 0x1

5 R/W hp_l_amp_ramp_en HP_L_AMP amplifier gain ramping control

0 = Gain changes are instant 1 = Gain changes are ramped to the new level This setting overrides zero crossing 0x0

4 R/W hp_l_amp_zc_en HP_L_AMP amplifier zero cross control

3 R/W hp_l_amp_oe HP_L_AMP amplifier output control

0 = Output is high-impedance 1 = Output is driven 0x0

2 R/W hp_l_amp_min_gain

_en HP_L_AMP amplifier minimum gain control. 0 = Normal gain operation 1 = Minimum gain only. HP_L amplifier is held at minimum gain regardless of other gain settings 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 87 of 123 © 2015 Dialog Semiconductor Table 99: HP_R_CTRL (Page 0: 0x0000006C) Bit Mode Symbol Description Reset

7 R/W hp_r_amp_en HP_R_AMP amplifier control

0 = HP_R_AMP disabled 1 = HP_R_AMP enabled 0x0

6 R/W hp_r_amp_mute_en HP_R_AMP amplifier mute control

0 = HP_R_AMP unmuted 1 = HP_R_AMP muted 0x1

5 R/W hp_r_amp_ramp_en HP_R_AMP amplifier gain ramping control

0 = Gain changes are instant 1 = Gain changes are ramped to the new level This setting overrides zero crossing 0x0

4 R/W hp_r_amp_zc_en HP_R_AMP amplifier zero cross control

3 R/W hp_r_amp_oe HP_R_AMP amplifier output control

0 = Output is high-impedance 1 = Output is driven 0x0

2 R/W hp_r_amp_min_gain

_en HP_R_AMP amplifier minimum gain control. 0 = Normal gain operation 1 = Minimum gain only. HP_R_AMP is held at minimum gain regardless of other gain settings 0x0 Table 100: MIXOUT_L_CTRL (Page 0: 0x0000006E) Bit Mode Symbol Description Reset

7 R/W mixout_l_amp_en MIXIN_L amplifier control

0 = Mixer disabled 1 = Mixer enabled 0x0 Table 101: MIXOUT_R_CTRL (Page 0: 0x0000006F) Bit Mode Symbol Description Reset

7 R/W mixout_r_amp_en MIXIN_R amplifier control

0 = Mixer disabled 1 = Mixer enabled 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 88 of 123 © 2015 Dialog Semiconductor Table 102: IO_CTRL (Page 0: 0x00000091) Bit Mode Symbol Description Reset

0 R/W io_voltage_level Digital I/O voltage range control

0 = 2.5 to 3.6 V 1 = 1.2 to 2.8 V 0x0 Table 103: Register map charge_pump_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000047 CP_CTRL cp_en Reserved cp_mchange Reserved 0x00000095 CP_VOL_THR ESHOLD1 Reserved cp_thresh_vdd2 Table 104: CP_CTRL (Page 0: 0x00000047) Bit Mode Symbol Description Reset

7 R/W cp_en Chargepump control

0 = Chargepump disabled 1 = Chargepump enabled 0x0 5:4 R/W cp_mchange Charge pump tracking mode control 00 = Reserved 01 = Voltage level is controlled by the largest output volume level 10 = Voltage level is controlled by the DAC volume level 11 = Voltage level is controlled by the signal magnitude 0x2 Table 105: CP_VOL_THRESHOLD1 (Page 0: 0x00000095) Bit Mode Symbol Description Reset 5:0 R/W cp_thresh_vdd2 Threshold at and below which the charge pump can use the CPVDD/2 rail. This setting is only effective when cp_mchange = 10 or cp_mchange = 11. It is ignored for cp_mchange settings of 00 and 01 0xE

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 89 of 123 © 2015 Dialog Semiconductor Table 106: Register map cif_i2c_addr_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x0000001B CIF_I2C_ADD R_CFG Reserved cif_i2c_addr_cfg Table 107: CIF_I2C_ADDR_CFG (Page 0: 0x0000001B) Bit Mode Symbol Description Reset 1:0 R/W cif_i2c_addr_cfg I2C address [1:0] configuration This allows multiple DA7219 devices to reside on the same bus by allowing the least significant two bits to be written to a specific value. The I2C clock must be externally controlled while writing this register to ensure that only the target DA7219 device's I2C address is modified. 0x2 Table 108: Register map common1_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000012 CIF_TIMEOUT _CTRL Reserved i2c_timeout_ en 0x00000013 CIF_CTRL cif_reg_soft_r eset Reserved cif_i2c_write_ mode 0x00000016 SR_24_48 Reserved sr_24_48 0x00000017 SR Reserved sr 0x00000092 GAIN_RAMP_ CTRL Reserved gain_ramp_rate 0x00000094 PC_COUNT Reserved pc_resync_a uto pc_freerun

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 90 of 123 © 2015 Dialog Semiconductor Table 109: CIF_TIMEOUT_CTRL (Page 0: 0x00000012) Bit Mode Symbol Description Reset 0 R/W i2c_timeout_en I2C (2-wire) timeout control. The timeout period is approximately 43.9 ms. 0 = Timeout disabled 1 = Timeout enabled 0x0 Table 110: CIF_CTRL (Page 0: 0x00000013) Bit Mode Symbol Description Reset

7 R/W cif_reg_soft_reset Software reset which returns all the registers back

to their default values. Writing to this bit causes all the registers to reset. 0 = No reset 1 = Reset all registers to their default values 0x0

0 R/W cif_i2c_write_mode I2C (2-wire) interface write mode control

0 = Page mode. The register address is autoincremented after the first write. 1 = Repeat mode. The register address and data are sent for each write. 0x0 Table 111: SR_24_48 (Page 0: 0x00000016) Bit Mode Symbol Description Reset 0 R/W sr_24_48 24_48_mode control. Setting this bit runs the ADC and the DAC paths at different speeds. 0 = The ADC path and the DAC path both run at the same speed. This speed is determined by the setting of the sr bit in this register 1 = The ADC path runs at 24 kHz, and the DAC path and the rest of the system run at 48 KHz To use this mode, the system sample rate sr must be set to 48 kHz. Therefore the I2S will also run at 48 kHz and the 24 kHz ADC output will be double sampled. 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 91 of 123 © 2015 Dialog Semiconductor Table 112: SR (Page 0: 0x00000017) Bit Mode Symbol Description Reset 3:0 R/W sr Sample rate control: 0001 = 8.000 kHz 0010 = 11.025 kHz 0011 = 12.000 kHz 0101 = 16.000 kHz 0110 = 22.050 kHz 0111 = 24.000 kHz 1001 = 32.000 kHz 1010 = 44.100 kHz 1011 = 48.000 kHz 1110 = 88.200 kHz 1111 = 96.000 kHz 0xA Table 113: GAIN_RAMP_CTRL (Page 0: 0x00000092) Bit Mode Symbol Description Reset 1:0 R/W gain_ramp_rate Controls the speed of the gain ramping when ramping is activated 0 = nominal rate * 8 1 = nominal rate 2 = nominal rate / 8 3 = nominal rate / 16 (slowest) The nominal rate (excluding headphone circuits) = 0.88 ms/dB. The nominal rate for the headphone circuits is = 1.3 ms/dB. 0x0 Table 114: PC_COUNT (Page 0: 0x00000094) Bit Mode Symbol Description Reset

1 R/W pc_resync_auto Program Counter resynchronisation control

0 = No resynchronisation. If the DAI drifts with respect to the system clocks, either a sample is skipped or it is double-sampled 1 = Automatic resynchronisation if the DAI drifts with respect to the system clock 0x1

0 R/W pc_freerun Controls the filter operation when the DAI is not

enabled or when no DAI clocks are available on the ADC to DAC processing path 0 = Filters are synchronised to the DAI 1 = Filters are free running 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 92 of 123 © 2015 Dialog Semiconductor Table 115: Register map common2_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000081 CHIP_ID1 chip_id1 0x00000082 CHIP_ID2 chip_id2 0x00000083 CHIP_REVISI ON chip_major chip_minor Table 116: CHIP_ID1 (Page 0: 0x00000081) Bit Mode Symbol Description Reset 7:0 R chip_id1 First two digits of the four-digit Chip ID The last two numbers of the Chip ID are held in chip_id2 0x23 Table 117: CHIP_ID2 (Page 0: 0x00000082) Bit Mode Symbol Description Reset 7:0 R chip_id2 Last two digits of the four-digit Chip ID The first two numbers of the Chip ID are held in chip_id1 0x93 Table 118: CHIP_REVISION (Page 0: 0x00000083) Bit Mode Symbol Description Reset 7:4 R chip_major Chip major revision 0x0 3:0 R chip_minor Chip minor revision 0x1

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 93 of 123 © 2015 Dialog Semiconductor Table 119: Register map dac_filters_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000040 DAC_FILTER dac_softmute _en dac_softmute_rate Reserved 0x00000041 DAC_FILTER dac_eq_band2 dac_eq_band1 0x00000042 DAC_FILTER dac_eq_band4 dac_eq_band3 0x00000043 DAC_FILTER dac_eq_en Reserved dac_eq_band5 0x00000044 DAC_FILTER dac_hpf_en Reserved dac_audio_hpf_corner dac_voice_e n dac_voice_hpf_corner Table 120: DAC_FILTERS5 (Page 0: 0x00000040) Bit Mode Symbol Description Reset 7 R/W dac_softmute_en DAC softmute control. When this bit is set, both channels are soft-muted. 0 = Soft-mute disabled 1 = Soft-mute enabled 0x0 6:4 R/W dac_softmute_rate Softmute gain update control 000 = 1 sample per 0.1875 dB 001 = 2 samples per 0.1875 dB 010 = 4 samples per 0.1875 dB 011 = 8 samples per 0.1875 dB 100 = 16 samples per 0.1875 dB 101 = 32 samples per 0.1875 dB 110 = 64 samples per 0.1875 dB 111 = Reserved 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 94 of 123 © 2015 Dialog Semiconductor Table 121: DAC_FILTERS2 (Page 0: 0x00000041) Bit Mode Symbol Description Reset 7:4 R/W dac_eq_band2 Gain control of Band 2 in the 5-band EQ 0000 = -10.5 dB 0001 = -9.0 dB 0010 = -7.5 dB Continuing in 1.5 dB steps through 0111 = 0 dB to… 1110 = 10.5 dB 1111 = 12 dB 0x8 3:0 R/W dac_eq_band1 Gain control of Band 1 in the 5-band EQ 0000 = -10.5 dB 0001 = -9.0 dB 0010 = -7.5 dB Continuing in 1.5 dB steps through 0111 = 0 dB to… 1110 = 10.5 dB 1111 = 12 dB 0x8 Table 122: DAC_FILTERS3 (Page 0: 0x00000042) Bit Mode Symbol Description Reset 7:4 R/W dac_eq_band4 Gain control of Band 4 in the 5-band EQ 0000 = -10.5 dB in 1.5 dB steps 0001 = -9.0 dB 0010 = -7.5 dB Continuing in 1.5 dB steps through 0111 = 0 dB to… 1110 = 10.5 dB 1111 = 12 dB 0x8 3:0 R/W dac_eq_band3 Gain control of Band 3 in the 5-band EQ 0000 = -10.5 dB 0001 = -9.0 dB 0010 = -7.5 dB Continuing in 1.5 dB steps through 0111 = 0 dB to… 1110 = 10.5 dB 1111 = 12 dB 0x8

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 95 of 123 © 2015 Dialog Semiconductor Table 123: DAC_FILTERS4 (Page 0: 0x00000043) Bit Mode Symbol Description Reset

7 R/W dac_eq_en DAC 5-band EQ control

0 = Equaliser disabled 1 = Equaliser enabled 0x0 3:0 R/W dac_eq_band5 Gain control of Band 5 in the 5-band EQ 0000 = -10.5 dB 0001 = -9.0 dB 0010 = -7.5 dB Continuing in 1.5 dB steps through 0111 = 0 dB to… 1110 = 10.5 dB 1111 = 12 dB 0x8 Table 124: DAC_FILTERS1 (Page 0: 0x00000044) Bit Mode Symbol Description Reset

7 R/W dac_hpf_en DAC High Pass Filter control

0 = High Pass Filter disabled 1 = High Pass Filter enabled 0x1 5:4 R/W dac_audio_hpf_corn er High Pass Filter 3 dB cutoff control. At 48 kHz, the 3 dB cutoff point is at: 00 = 2 Hz 01 = 4 Hz 10 = 8 Hz 11 = 16 Hz For other sample rates, the corner cutoff point scales proportionately. 0x0 3 R/W dac_voice_en DAC Voice Filter control : For 8/11.025/12/16 kHz sample rates and for best performance should always be enabled when running at one these rates. 0 = DAC Voice Filter disabled 1 = DAC Voice Filter enabled This DAC Voice Filter control overrides the 5-band EQ setting in dac_eq_en 0x0 2:0 R/W dac_voice_hpf_corn er Voice Filter 3 dB cutoff control. At 8 kHz, the 3 dB cutoff point is at: 000 = 2.5 Hz 001 = 25 Hz, 010 = 50 Hz 011 = 100 Hz 100 = 150 Hz 101 = 200 Hz 110 = 275 Hz 111 = 363 Hz 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 96 of 123 © 2015 Dialog Semiconductor Table 125: Register map dac_ng_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x000000AF DAC_NG_SET UP_TIME Reserved dac_ng_ram pdn_rate dac_ng_ram pup_rate dac_ng_setup_time 0x000000B0 DAC_NG_OF F_THRESH Reserved dac_ng_off_threshold 0x000000B1 DAC_NG_ON _THRESH Reserved dac_ng_on_threshold 0x000000B2 DAC_NG_CT RL dac_ng_en Reserved Table 126: DAC_NG_SETUP_TIME (Page 0: 0x000000AF) Bit Mode Symbol Description Reset

3 R/W dac_ng_rampdn_rat

e DAC Noise Gate ramp down control 0 = 0.88 ms/dB 1 = 14.08 ms/dB 0x0

2 R/W dac_ng_rampup_rat

e DAC Noise Gate ramp up control 0 = 0.22 ms/dB 1 = 0.0138 ms/dB 0x0 1:0 R/W dac_ng_setup_time Noise Gate timing control This specifies the number of samples for which the largest signal through the DACs must be above (or below) dac_ng_off _threshold (or dac_ng_on_threshold) for the Noise Gate to unmute (or mute) the data 00 = 256 samples 01 = 512 samples 10 = 1024 samples 11 = 2048 samples 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 97 of 123 © 2015 Dialog Semiconductor Table 127: DAC_NG_OFF_THRESH (Page 0: 0x000000B0) Bit Mode Symbol Description Reset 2:0 R/W dac_ng_off_threshol d Threshold above which the Noise Gate is deactivated. If the signal rises above this level, the Noise Gate is deactivated. 000 = -102 dB 001 = -96 dB 010 = -90 dB 011 = -84 dB 100 = -78 dB 101 = -72 dB 110 = -66 dB 111 = -60 dB 0x0 Table 128: DAC_NG_ON_THRESH (Page 0: 0x000000B1) Bit Mode Symbol Description Reset 2:0 R/W dac_ng_on_threshol d Threshold below which the Noise Gate is dactivated. If the signal drops below this level for dac_ng_setup_time samples, the Noise Gate is activated. 000 = -102 dB 001 = -96 dB 010 = -90 dB 011 = -84 dB 100 = -78 dB 101 = -72 dB 110 = -66 dB 111 = -60 dB 0x0 Table 129: DAC_NG_CTRL (Page 0: 0x000000B2) Bit Mode Symbol Description Reset

7 R/W dac_ng_en DAC Noise Gate control

0 = Noise Gate is disabled 1 = Noise Gate is enabled 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 98 of 123 © 2015 Dialog Semiconductor Table 130: Register map dai_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x0000002B DAI_CLK_MO DE dai_clk_en Reserved dai_wclk_tri_ state dai_wclk_pol dai_clk_pol dai_bclks_per_wclk 0x0000002C DAI_CTRL dai_en Reserved dai_ch_num dai_word_length dai_format 0x0000002D DAI_TDM_CT RL dai_tdm_mod e_en dai_oe Reserved dai_tdm_ch_en 0x00000030 DAI_OFFSET _LOWER dai_offset_lower 0x00000031 DAI_OFFSET _UPPER Reserved dai_offset_upper Table 131: DAI_CLK_MODE (Page 0: 0x0000002B) Bit Mode Symbol Description Reset

7 R/W dai_clk_en DAI Master mode control

0 = Slave mode (BCLK/WCLK inputs) 1 = Master mode (BCLK/WCLK outputs) 0x0

4 R/W dai_wclk_tri_state WLCK tri-state control

0 = WCLK state is set by dai_clk_en. WCLK is set as output in master mode, and as input in slave mode 1 = WCLK forced as an input 0x0

3 R/W dai_wclk_pol DAI word clock polarity control

0 = Normal polarity 1 = Inverted polarity 0x0

2 R/W dai_clk_pol DAI bit clock polarity control

0 = Normal polarity 1 = Inverted polarity 0x0 1:0 R/W dai_bclks_per_wclk Number of BCLKs per WCLK period when in DAI Master mode 00 = 32 BCLKS per WCLK 01 = 64 BCLKS per WCLK 10 = 128 BCLKS per WCLK 11 = 256 BCLKS per WCLK 0x1

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 99 of 123 © 2015 Dialog Semiconductor Table 132: DAI_CTRL (Page 0: 0x0000002C) Bit Mode Symbol Description Reset

7 R/W dai_en DAI control

0 = DAI disabled. No data is transferred. 1 = DAI enabled. Input and output data streams are transferred 0x0 5:4 R/W dai_ch_num Channel control 00 = No channels are enabled 01 = Left channel is enabled 10 = Left and right channels are enabled 11 = Reserved 0x2 3:2 R/W dai_word_length DAI data word length control 0 = 16 bits per channel 1 = 20 bits per channel 2 = 24 bits per channel 3 = 32 bits per channel 0x2 1:0 R/W dai_format DAI data format 00 = I2S mode 01 = Left justified mode 10 = Right justified mode 11 = DSP mode 0x0 Table 133: DAI_TDM_CTRL (Page 0: 0x0000002D) Bit Mode Symbol Description Reset 7 R/W dai_tdm_mode_en DAI TDM mode control. In TDM mode, the output is high impedence when not actively driving data as this allows other devices to share the DATOUT line. 0 = DAI normal mode 1 = DAI TDM mode 0x0

6 R/W dai_oe DAI output control

0 = DAI DATOUT pin is high impedence 1 = DAI DATOUT pin is driven when required 0x1 1:0 R/W dai_tdm_ch_en DAI TDM channel control. Bit 0 = Left channel; Bit 1: Riight channel. For each bit, 0 = Disabled; 1 = Enabled. 00 = Left channel and right channel both disabled 01 = Left channel enabled, right channel disabled 10 = Left channel disabled, right channel enabled 11 = Left channel and right channel both enabled 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 100 of 123 © 2015 Dialog Semiconductor Table 134: DAI_OFFSET_LOWER (Page 0: 0x00000030) Bit Mode Symbol Description Reset 7:0 R/W dai_offset_lower DAI data offset with respect to WCLK measured in BCLK periods. The total offset is determined by an 11-bit binary number formed by a combination of this register (dai_offset_lower) and dai_offset_upper. With the maximum BCLK frequency of 6 MHz, the maximum number of BCLK periods is 768. The maximum DAI offset value is therefore 767 (0x2FF), represented by dai_offset_lower = 1111 1111, and dai_offset_upper = 010. 0x000 = No offset relative to the normal formatting 0x001 = One BCLK period offset relative to the normal formatting 0x002 = Two BCLK periods offset relative to the normal formatting 0xn = n BCLK periods offset relative to the normal formatting (max = 0x2FF) 0x0 Table 135: DAI_OFFSET_UPPER (Page 0: 0x00000031) Bit Mode Symbol Description Reset 2:0 R/W dai_offset_upper DAI data offset with respect to WCLK measured in BCLK periods. The total offset is determined by an 11-bit binary number formed by a combination of dai_offset_lower and this register (dai_offset_upper). With the maximum BCLK frequency of 6 MHz, the maximum number of BCLK periods is 768. The maximum DAI offset value is therefore 767 (0x2FF), represented by dai_offset_lower = 1111 1111, and dai_offset_upper = 010. 0x000 = No offset relative to the normal formatting 0x001 = One BCLK period offset relative to the normal formatting 0x002 = Two BCLK periods offset relative to the normal formatting 0xn = n BCLK periods offset relative to the normal formatting (max = 0x2FF) 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 101 of 123 © 2015 Dialog Semiconductor Table 136: Register map pll_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000020 PLL_CTRL pll_mode pll_mclk_sqr _en pll_indiv Reserved 0x00000022 PLL_FRAC_T OP Reserved pll_fbdiv_frac_top 0x00000023 PLL_FRAC_B OT pll_fbdiv_frac_bot 0x00000024 PLL_INTEGE R Reserved pll_fbdiv_integer 0x00000025 PLL_SRM_ST S pll_srm_status Reserved Table 137: PLL_CTRL (Page 0: 0x00000020) Bit Mode Symbol Description Reset 7:6 R/W pll_mode PLL mode control 00 = Bypass mode. The PLL is disabled, and the system clock is MCLK (after input divider) 01 = Normal mode.The PLL is enabled, and the system clock is a fixed multiple of MCLK 10 = SRM. The PLL is enabled, and the system clock tracks WCLK 11 = Reserved 0x0 5 R/W pll_mclk_sqr_en PLL clock squarer control. 0 = Clock squarer is disabled 1 = Clock squarer is enabled 0x0 4:2 R/W pll_indiv PLL reference input clock (MCLK) control 0 = 2 to 4.5 MHz 1 = 4.5 to 9 MHz 2 = 9 to 18 MHz 3 = 18 to 36 MHz 4 = 36+ MHz 0x4

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 102 of 123 © 2015 Dialog Semiconductor Table 138: PLL_FRAC_TOP (Page 0: 0x00000022) Bit Mode Symbol Description Reset 4:0 R/W pll_fbdiv_frac_top PLL fractional division value (top bits). The full PLL fractional division value is a concatenation of these bits (MSB) and PLL_FBDIV_FRAC_BOT (LSB). The value in this register does not take effect until pll_fbdiv_integer is written. 0x0 Table 139: PLL_FRAC_BOT (Page 0: 0x00000023) Bit Mode Symbol Description Reset 7:0 R/W pll_fbdiv_frac_bot PLL fractional division value (bottom bits). The full PLL fractional division value is a concatenation of PLL_FBDIV_FRAC_TOP (MSB) and these bits (LSB). The value in this register does not take effect until pll_fbdiv_integer is written. 0x0 Table 140: PLL_INTEGER (Page 0: 0x00000024) Bit Mode Symbol Description Reset 6:0 R/W pll_fbdiv_integer PLL integer division value. Writing this register causes the entire pll_fbdiv value (PLL_INTEGER, PLL_FRAC_TOP, PLL_FRAC_BOT) to be updated. 0x20 Table 141: PLL_SRM_STS (Page 0: 0x00000025) Bit Mode Symbol Description Reset 7:4 R pll_srm_status PLL/SRM status (user mode). Within this four-bit register field, Bit position [3] = SRM lock Bit position [2] = PLL/SRM active Bit position [1] = PLL lock Bit position [0] = MCLK status (1=valid MCLK detected, subject to minimum detection frequency of approximately 1 MHz) For each bit position, 0 = Inactive or invalid 1 = Active or valid 0x1

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 103 of 123 © 2015 Dialog Semiconductor Table 142: Register map router_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x0000002A DIG_ROUTIN G_DAI Reserved dai_r_src Reserved dai_l_src 0x0000002E DIG_ROUTIN G_DAC dac_r_mono Reserved dac_r_src dac_l_mono Reserved dac_l_src 0x00000099 DIG_CTRL dac_r_inv Reserved dac_l_inv Reserved Table 143: DIG_ROUTING_DAI (Page 0: 0x0000002A) Bit Mode Symbol Description Reset 5:4 R/W dai_r_src Data selection for the DAI right output stream 00 = ADC left 01 = Tone generator 10 = DAI input left data / DAI mono mix 11 = DAI input right data / DAI mono mix 0x1 1:0 R/W dai_l_src Data selection for the DAI left output stream 00 = ADC left 01 = Tone generator 10 = DAI input left data / DAI mono mix 11 = DAI input right data / DAI mono mix 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 104 of 123 © 2015 Dialog Semiconductor Table 144: DIG_ROUTING_DAC (Page 0: 0x0000002E) Bit Mode Symbol Description Reset

7 R/W dac_r_mono Mono-mix control for the DAI right input stream

0 = No mono-mix 1 = The DAI right input stream is replaced with a mono mix of left and right 0x0 5:4 R/W dac_r_src Data selection to the DAC_R path 00 = ADC left output 01 = Tone generator 10 = DAI input left / dai mono mix 11 = DAI input right / dai mono mix 0x3

3 R/W dac_l_mono Mono-mix control for the DAI left input stream

0 = No mono-mix 1 = The DAI left input stream is replaced with a mono mix of left and right 0x0 1:0 R/W dac_l_src Data selection to the DAC_L path 00 = ADC left output 01 = Tone generator 10 = DAI input left / dai mono mix 11 = DAI input right / dai mono mix 0x2 Table 145: DIG_CTRL (Page 0: 0x00000099) Bit Mode Symbol Description Reset

7 R/W dac_r_inv DAC right input stream invertion control

0 = No inversion of the right input strem 1 = The right input stream is inverted 0x0

3 R/W dac_l_inv DAC left input stream invertion control

0 = No inversion of the left input stream 1 = The left input stream is inverted 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 105 of 123 © 2015 Dialog Semiconductor Table 146: Register map sidetone_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x0000003A SIDETONE_C TRL sidetone_en sidetone_mut e_en Reserved 0x0000003B SIDETONE_G AIN Reserved sidetone_gain 0x0000003C DROUTING_S T_OUTFILT_1 L Reserved outfilt_st_1l_src 0x0000003D DROUTING_S T_OUTFILT_1 R Reserved outfilt_st_1r_src Table 147: SIDETONE_CTRL (Page 0: 0x0000003A) Bit Mode Symbol Description Reset

7 R/W sidetone_en Sidetone path control

0 = Sidetone path disabled 1 = Sidetone path enabled 0x0

6 R/W sidetone_mute_en SideTone mute control

0 = Sidetone mute disabled 1 = Sidetone mute enabled 0x1 Table 148: SIDETONE_GAIN (Page 0: 0x0000003B) Bit Mode Symbol Description Reset 3:0 R/W sidetone_gain Sidetone gain control 0000 = -42 dB 0001 = -39 dB 0010 = -36 dB Continuing in 3 dB steps to… 1101 = -3dB 1110 = 0dB 1111 = Reserved 0xE

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 106 of 123 © 2015 Dialog Semiconductor Table 149: DROUTING_ST_OUTFILT_1L (Page 0: 0x0000003C) Bit Mode Symbol Description Reset 2:0 R/W outfilt_st_1l_src Data selection for the output filter 1 left output stream bit 0 = Output filter 1L bit 1 = Output filter 1R bit 2 = Sidetone For each bit position/output stream, 0 = disabled and 1 = enabled 0x1 Table 150: DROUTING_ST_OUTFILT_1R (Page 0: 0x0000003D) Bit Mode Symbol Description Reset 2:0 R/W outfilt_st_1r_src Data selection for the output filter 1 right output stream bit 0 = Output filter 1L bit 1 = Output filter 1R bit 2 = Sidetone For each bit position/output stream, 0 = disabled and 1 = enabled 0x2 Table 151: Register map system_active_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x000000FD SYSTEM_ACT IVE Reserved system_activ e Table 152: SYSTEM_ACTIVE (Page 0: 0x000000FD) Bit Mode Symbol Description Reset

0 R/W system_active System Standby mode

0 = Standby mode 1 = Active mode 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 107 of 123 © 2015 Dialog Semiconductor Table 153: Register map system_controller_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x00000050 SYSTEM_MO DES_INPUT adc_mode mode_submit 0x00000051 SYSTEM_MO DES_OUTPU T dac_mode mode_submit 0x000000E0 SYSTEM_STA TUS Reserved sc2_busy sc1_busy Table 154: SYSTEM_MODES_INPUT (Page 0: 0x00000050) Bit Mode Symbol Description Reset 7:1 R/W adc_mode preconfigured system modes (input side): [1] = reserved [2] = MIC [3] = reserved [4] = MIXIN [5] = reserved [6] = ADC [7] = reserved 0x0

0 R/W mode_submit Causes both the adc_mode and dac_mode to

Table 155: SYSTEM_MODES_OUTPUT (Page 0: 0x00000051) Bit Mode Symbol Description Reset 7:1 R/W dac_mode preconfigured system modes (output side): [1] = reserved [2] = reserved [3] = reserved [4] = HP_L [5] = HP_R [6] = DAC_L [7] = DAC_R 0x0 0 - mode_submit Causes both the adc_mode and dac_mode to become active 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 108 of 123 © 2015 Dialog Semiconductor Table 156: SYSTEM_STATUS (Page 0: 0x000000E0) Bit Mode Symbol Description Reset

1 R sc2_busy Indicates the current status of the system mode

0 = complete 1 = busy 0x0

0 R sc1_busy Indicates the current status of the system controller

0 = complete 1 = busy 0x0 Table 157: Register map tone_gen_cad_00 page 0 Address Name # 7 6 5 4 3 2 1 0 Register Page 0 0x000000B4 TONE_GEN_ CFG1 start_stopn Reserved dtmf_en dtmf_reg 0x000000B5 TONE_GEN_ CFG2 tone_gen_gain Reserved swg_sel 0x000000B6 TONE_GEN_ CYCLES Reserved beep_cycles 0x000000B7 TONE_GEN_F REQ1_L freq1_l 0x000000B8 TONE_GEN_F REQ1_U freq1_u 0x000000B9 TONE_GEN_F REQ2_L freq2_l 0x000000BA TONE_GEN_F REQ2_U freq2_u 0x000000BB TONE_GEN_ ON_PER Reserved beep_on_per 0x000000BC TONE_GEN_ OFF_PER Reserved beep_off_per

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 109 of 123 © 2015 Dialog Semiconductor Table 158: TONE_GEN_CFG1 (Page 0: 0x000000B4) Bit Mode Symbol Description Reset 7 R/W start_stopn Tone Generator stop-start control. Setting this bit = 1 starts the Tone Generator for the number of beeps defined by beep_cycles. Once complete, the bit is automatically cleared. If beep_cycles = 111 (continuous), then this bit must be cleared manually 0 = Tone Generator disabled 1 = Tone Generator enabled 0x0

4 R/W dtmf_en DTMF control

0 = DTMF is disabled. The Tone Generator uses values in the registers freq1 and freq2 to generate sine wave(s) 1 = DTMF is enabled. The Tone Generator uses values from the register dtmf_reg to generate sine- waves 0x0 3:0 R/W dtmf_reg The DTMF key pad values 0 to 15 0000 = 0 0001 = 1 0010 = 2 0011 = 3 0100 = 4 0101 = 5 0110 = 6 0111 = 7 1000 = 8 1001 = 9 1010 = A 1011 = B 1100 = C 1101 = D 1110 = * 1111 = # 0x0

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 110 of 123 © 2015 Dialog Semiconductor Table 159: TONE_GEN_CFG2 (Page 0: 0x000000B5) Bit Mode Symbol Description Reset 7:4 R/W tone_gen_gain Tone Generator gain control 0000 = 0 dB 0001 = -2.5 dB 0010 = -6 dB Continuing in 2.5/3.5 dB steps to… 1110 = -42 dB 1111 = -44.5 dB 0x0 1:0 R/W swg_sel Sine wave selection control 00 = Sum of both Sine Wave Generator (SWG) values is mixed into the audio stream 01 = Only the first SWG value is output 10 = Only the second SWG value is output 11 = 1-Cos(SWG1) or S_ramp function for headphone detection. The high period is determined by the beep_on_per setting 0x0 Table 160: TONE_GEN_CYCLES (Page 0: 0x000000B6) Bit Mode Symbol Description Reset 2:0 R/W beep_cycles Beep control. This specified the number of beep cycles required. 000 = 1 cycle 001 = 2 cycles 010 = 3 cycles 011 = 4 cycles 100 = 8 cycles 101 = 16 cycles 110 = 32 cycles 111 = continuous (until start_stopn is set to 0) 0x0 Table 161: TONE_GEN_FREQ1_L (Page 0: 0x000000B7) Bit Mode Symbol Description Reset 7:0 R/W freq1_l Lower byte of the output frequency for the first Sine Wave Generator (SWG) If sample rate (SR) = 8/12/16/24/32/48/96 kHz If sample rate (SR) =11.025/22.05/44.4/88.2 kHz, 0x55

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 111 of 123 © 2015 Dialog Semiconductor Table 162: TONE_GEN_FREQ1_U (Page 0: 0x000000B8) Bit Mode Symbol Description Reset 7:0 R/W freq1_u Upper byte of the output frequency for the first Sine Wave Generator (SWG) If sample rate (SR) = 8/12/16/24/32/48/96 kHz If sample rate (SR) =11.025/22.05/44.4/88.2 kHz, 0x15 Table 163: TONE_GEN_FREQ2_L (Page 0: 0x000000B9) Bit Mode Symbol Description Reset 7:0 R/W freq2_l Lower byte of the output frequency for the second Sine Wave Generator (SWG) If sample rate (SR) = 8/12/16/24/32/48/96 kHz If sample rate (SR) =11.025/22.05/44.4/88.2 kHz, 0x0 Table 164: TONE_GEN_FREQ2_U (Page 0: 0x000000BA) Bit Mode Symbol Description Reset 7:0 R/W freq2_u Upper byte of the output frequency for the second Sine Wave Generator (SWG) If sample rate (SR) = 8/12/16/24/32/48/96 kHz If sample rate (SR) =11.025/22.05/44.4/88.2 kHz, 0x40

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 112 of 123 © 2015 Dialog Semiconductor Table 165: TONE_GEN_ON_PER (Page 0: 0x000000BB) Bit Mode Symbol Description Reset 5:0 R/W beep_on_per Beep ON period control 0x0 = 10 ms 0x1 = 20 ms 0x2 = 30 ms Continuing in 10 ms steps to… 0x14 = 200 ms then… 0x15 = Reserved 0x16 = Reserved 0x17 = Reserved 0x18 = Reserved then... 0x19 = 250 ms 0x1A = 300 ms 0x1B = 350 ms Continuing in 50 ms steps to 0x3B = 1950 ms 0x3C = 2000 ms 0x3D = Reserved 0x3E = Reserved 0x3F = Continuous 0x2 Table 166: TONE_GEN_OFF_PER (Page 0: 0x000000BC) Bit Mode Symbol Description Reset 5:0 R/W beep_off_per Beep OFF period control 0x0 = 10 ms 0x1 = 20 ms 0x2 = 30 ms Continuing in 10 ms steps to… 0x14 = 200 ms then… 0x15 = Reserved 0x16 = Reserved 0x17 = Reserved 0x18 = Reserved then... 0x19 = 250 ms 0x1A = 300 ms 0x1B = 350 ms Continuing in 50 ms steps to… 0x3B = 1950 ms 0x3C = 2000 ms 0x3D = Reserved 0x3E = Reserved 0x3F = Reserved 0x1

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 114 of 123 © 2015 Dialog Semiconductor The ordering number consists of the part number followed by a suffix indicating the packing method. For details and availability, please contact Dialog Semiconductor’s local sales representative. Table 167: Ordering information Part number Package Size (mm) Shipment form Pack quantity DA7219-02VBA 32 WL-CSP 4.5 x 1.64 mm Tape & reel 4,500 DA7219-02VB6 32 WL-CSP 4.5 x 1.64 mm Tray/Waffle Pack (engineering samples only - not for mass production

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 115 of 123 © 2015 Dialog Semiconductor Applications information Appendix A A.1 Codec initialisation Depending on the specific application, some general settings need to be set. Examples of these settings include the sample rate, the PLL, and the Digital Audio Interface. Then the amplifiers, the mixers and channels of the ADC/DAC have to be configured and enabled using their respective control registers. An example sequence is shown below: 1. Configure clock mode as required for operation, (for example PLL bypass, PLL or SRM mode) 2. Configure the digital audio interface 3. Configure the charge pump if the headphone path is in use 4. Set input and output mixer paths and gains 5. Enable input and output paths using the System Controller A.2 Automatic ALC calibration When using the automatic level control (ALC) in sync-mode the DC offset between the digital and analogue PGAs must be cancelled. This is performed automatically if the following procedure is performed: 6. Enable microphone amplifiers unmuted 7. Mute microphones 8. Enable input mixer and ADC unmuted 9. Enable AIF interface 10. Set alc_auto_calib_en in ALC_CTRL1 to ‘1’ (ALC_CTRL1 = 0x2F). This bit will auto clear when calibration is complete. 11. When calibration is complete, enable the ALC with alc_sync_mode and alc_offset_en (ALC_CTRL1 = 0x2F) 12. Unmute microphones

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 116 of 123 © 2015 Dialog Semiconductor Components Appendix B The following recommended components are examples selected from requirements of a typical application. The electrical characteristics (that is, the supported voltage/current ranges) have to be cross-checked and component types may need to be adapted from the individual needs of the target circuitry. B.1 Audio inputs Table 168: Audio inputs Pin name Bump/ pin Power domain Description Type MIC_N A15 VDD Differential mic. input (negative) / Single-ended mic. Input Analogue input MIC_P B16 VDD Differential mic. input (positive) / Single-ended mic. Input Analogue input MIC C15 N/A Supply input for headset microphone power Analogue input The DA7219 microphone inputs can be configured to accommodate single-ended or differential analogue microphones, line inputs or digital microphones. When using the inputs in an analogue configuration, a DC blocking capacitor is required for each used input bump used in the target application. The choice of capacitor is determined by the filter that is formed between that capacitor and the input impedance of the input pin, which can be found in the ‘Input Mixing Units’ section of the datasheet. 𝐶 = 1 2𝜋. 𝑅. 𝐹𝑐 Where Fc is the 3 dB cut off frequency of the low pass filter (typically 20 Hz for audio applications). A 1 µF capacitor is suitable for most applications. Due to their high stability, tantalum capacitors are particularly suitable for this application. Ceramic equivalents with an X5R dielectric are recommended as a cost-effective alternative. Care should be taken to ensure that the desired capacitance is maintained over operating temperature and voltage. Z5U dielectric ceramics should be avoided due to their susceptibility to microphonic effects. Unused input bumps can be left floating or connected via a capacitor to ground . The MIC pin would normally be connected to MICBIAS using a 2k2 resistor. This pin is an input to supply the microphone power when a headset is connected to the headset socket. The polarity of the microphone pin is determined by the accessory detect circuitry and the power is switched internally in the device to allow the microphone bias to be provided from this pin.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 120 of 123 © 2015 Dialog Semiconductor B.7 Supplies Table 175: Power supplies Pin name Bump/pin Power domain Description Type VDD C5 Min: 1.71 V Max: 2.65 V Supply for analogue & digital circuits / Supply for headphone charge pump Power supply VDD_IO D4 Min: 1.71 V Max: 3.6 V Supply for digital interfaces Power supply VDD_MIC A13 Min: 1.8 V * Max: 3.6 V Supply for microphone bias circuits * Minimum level must be 300 mV higher than VDD level Power supply Decoupling capacitors are recommended between all supplies and GND. These capacitors should be located as near to the device as possible. M11MICBIAS 1 µF A13VDD_MIC 1 µF VDD_IO 1 µF 1.71 V to 3.6 V 1.8 V to 3.6 V 1.71 V to 2.65 VVDD Figure 42: Power supply decoupling B.8 Ground Table 176: Ground Pin name Bump/pin Power domain Description Type GND B10 Analogue ground Power ground GND_HP B12 Headphone Ground Power ground GND_CP B2 Charge pump/digital ground Power ground GND, GND_HP and GND_CP should be connected directly to the system ground.

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 121 of 123 © 2015 Dialog Semiconductor PCB layout guidelines Appendix C DA7219 uses Dialog Semiconductor’s RouteEasy™ technology allowing the device to be routed using conventional, low cost, PCB technology. All device balls are routable on the top level and conventional plated through hole vias can be used throughout. This design is fully realisable using a 2-layer PCB. For optimum performance it is recommended that a 4-layer PCB is used with layers 2 and 3 as solid ground planes. Decoupling and reference capacitors should be located as close to the device as possible and appropriately sized tracks should be used for all power connections. Figure 43: DA7219 example layout C.1 Layout and schematic support Copies of the evaluation board schematics and layout are available on request to aid in PCB development. Dialog Semiconductor also offer a schematic and layout review service for all designs using Dialog’s devices. Please contact your local Dialog Semiconductor Office if you wish to use this service. C.2 General recommendations

Appropriate trace width and number of vias should be used for all power supply paths
A common ground plane should be used, which allows proper electrical and thermal performance
Noise-sensitive analogue signals such as feedback lines or clock connections should be kept away from traces carrying pulsed analogue or digital signals. This can be achieved by separation (distance) or by shielding with quiet signals or ground traces
Decoupling capacitors should be X5R ceramics and should be placed as near to the device as possible
Charge pump capacitors should be X5R ceramics and should be placed as near to the device as possible

Audio codec with Advanced Accessory Detect Company confidential Datasheet Revision 2.1 17-Dec -2015 122 of 123 © 2015 Dialog Semiconductor C.3 Capacitor selection Ceramic capacitors are manufactured with a variety of dielectrics, each with a different behaviour over temperature and applied voltage. Capacitors must have a dielectric adequate to ensure the minimum capacitance over the necessary temperature range, dc bias conditions and low Equ ivalent Series Resistance (ESR). X5R or X7R dielectrics with a voltage rating of 6.3 V or 10 V are recommended for best performance. Y5V and Z5U dielectrics are not recommended for use because of their poor temperature and dc bias characteristics. The worst-case capacitance accounting for capacitor variation over temperature, component tolerance, and voltage is calculated using the following equation: 𝐶𝐸𝐹𝐹 = 𝐶𝑂𝑈𝑇 𝑥 (1 − 𝑇𝐸𝑀𝑃𝐶𝑂) 𝑥 (1 − 𝑇𝑂𝐿) where: CEFF is the effective capacitance at the operating voltage. TEMPCO is the worst-case capacitor temperature coefficient. TOL is the worst-case component tolerance. These figures can be found in the manufacturer’s datasheet. In the example below, the worst-case temperature coefficient (TEMPCO) over −55°C to +85°C is assumed to be 15%. The tolerance of the capacitor (TOL) is assumed to be 10%, and COUT is 0.65 μF at 1.8 V. Substituting these values in the equation yields 𝐶𝐸𝐹𝐹 = 0.65𝜇𝐹 𝑥 (1 − 0.15) 𝑥 (1 − 0.1) = 0.497 𝜇𝐹 Below is a table with recommended capacitor types: Application Value Size Temp. char. Tolerance Rated voltage Type VDD,VDD_IO, VDD_MIC, DACREF, VMID,VREF, HPCFP/HPCFN, HPCSP, HPCSN, MICBIAS 10 x 1 µF 0201 X5R +/- 15% +/-10% 6.3 V Murata GRM033R60J105M

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Audio codec with Advanced Accessory Detect

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Audio codec with Advanced Accessory Detect

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