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ac 97 soundmax ? codec AD1985 rev. a information furnished by analog devices is believed to be accurate and reliable. however, no responsibility is assumed by analog devices for its use, nor for any infringements of patents or other rights of third parties that may result from its use. specifications subject to change without notice. no license is granted by implication or otherwise under any patent or patent ri ghts of analog devices. trademarks and registered trademarks are the prop erty of their respective owners. one technology way, p.o. box 9106, norwood, ma 02062-9106, u.s.a. tel: 781.329.4700 www.analog.com fax: 781.326.8703 ? 2004 analog devices, inc. all rights reserved. ac 97 2.3 compliant features 6 dac channels for 5.1 surround greater than 90 db dynamic range 20-bit resolution on all dacs s/pdif output integrated stereo headphone amplifiers variable rate audio double rate audio (f s = 96 khz) line-level mono phone input high quality cd mixer input selectable mic input with preamp aux and line in stereo inputs external amplifier power down (eapd) power management modes jack sensing and peripheral enumeration/identification 48-lead lqfp package enhanced features integrated parametric equalizer (eq) stereo microphone with preamplifiers integrated pll for system clocking variable sample rate 7 khz to 96 khz 7 khz to 48 khz in 1 hz increments 96 khz for double rate audio advanced jack sense with auto topology switching software enabled v refout for microphones and external power amp software enabled outputs for jack sharing auto down-mix and channel spreading microphone to mono output stereo microphone analog passthrough to outputs built-in stereo microphone and center/lfe pin sharing selectable center/lfe tip/ring swapping to support various speaker products functional block diagram pcm l/r adc rate pcm c/lfe dac rate pcm front dac rate pcm surr dac rate dac slot logic adc slot logic ac '97 interface v2.3 eq coef storage ac '97 control registers eapd analog mixing control jack sense pll spdif tx id0 id1 reset sync bitclk sdata_out sdata_in js0 js1 js2 js3 eapd 16-bit 16-bit - ? adc codec core 20-bit - ? dac 20-bit - ? dac 20-bit - ? dac 20-bit - ? dac bypass eq 20-bit - ? dac bypass cswp cswp eq 20-bit - ? dac - ? adc m g = gain a = attenuation m = mute z = high z m g m z g m ga mga m ga m ga m ga m ga g g m ga m ga m ga mmm ga m ga m ga m ga m ga m ga m m m mic1 mic2 phone_in cd_l cd_gnd cd_r aux_l aux_r line_in_l oms oms 2cmic ms cd diff amp line_in_r sprd mz a lfe_out sprd mz a center_out mz a line_out_l m a mono_out mz a line_out_r m a surr_out_l/ hp_out_l m a surr_out_r/ hp_out_r hp hp losel losel mix hpsel hpsel v ref g v refout voltage reference AD1985 xtl_out xtl_in spdif_out pc beep generator m a record selector 03610-a-001 figure 1.
AD1985 rev. a | page 2 of 48 table of contents detailed functional block diagram .............................................. 3 specifications..................................................................................... 4 analog input ................................................................................. 4 master volume .............................................................................. 4 programmable gain amplifieradc...................................... 4 analog mixerinput gain/amplifiers/attenuators............... 5 digital decimation and interpolation filters ........................... 5 analog-to-digital converters..................................................... 5 digital-to-analog converters..................................................... 5 analog output .............................................................................. 6 static digital specifications......................................................... 6 power supply................................................................................. 6 power-down states ...................................................................... 7 clock specifications ..................................................................... 7 timing parameters ....................................................................... 8 absolute maximum ratings.......................................................... 10 environmental conditions........................................................ 10 esd caution................................................................................ 10 pin configuration and functional descriptions........................ 11 pin descri ptions ......................................................................... 11 indexed control registers......................................................... 13 outline dimensions ....................................................................... 47 ordering guide........................................................................... 47 revision history 3/04data sheet changed from rev. 0 to rev. a updated format.................................................................universal changes to figure 1......................................................................... 1 changes to figure 2......................................................................... 3 changes to table 4........................................................................... 4 changes to tables 5, 7, and 8.......................................................... 5 changes to table 12 ........................................................................ 7 changes to pin configuration..................................................... 11 changes to circuit layout note .................................................. 11 changes to indexed control registers ....................................... 13 changes to master volume register (index 0x02).................... 15 changes to headphone volume register (index 0x04) ........... 16 changes to mono volume register (index 0x06) ..................... 17 changes to pc beep register (index 0x0a) .............................. 18 changes to microphone volume register (index 0x0e).......... 19 changes to aux in volume register (index 0x16) .................. 21 changes to record select control register (index 0x1a)....... 22 changes to record gain register (index 0x1c) ....................... 23 changes to extended audio id register (index 0x28) ............ 27 changes to center/lfe volume control register (index 0x36).................................................................... 30 changes to surround volume control register (index 0x38) ................................................................................... 31 changes to jack sense/audio interrupt/status register (index 0x72) ................................................................................... 36 changes to miscellaneous control bits (index 0x76) .............. 39 changes to advanced jack sense register (index 0x78).......... 41 updated outline dimensions...................................................... 47 updated ordering guide.............................................................. 47 3/03revision 0: initial version
AD1985 rev. a | page 3 of 48 detailed functional block diagram dac slot logic adc slot logic ac '97 interface v2.3 eq coef storage ac '97 control registers eapd analog mixing control jack sense pll spdif tx id0 id1 reset sync bitclk sdata_ou t sdata_in js0 js1 js2 js3 eapd 16-bit 16-bit - ? adc codec core 20-bit - ? dac 20-bit - ? dac 20-bit - ? dac 20-bit - ? dac bypass eq 20-bit - ? dac bypass cswp cswp eq 20-bit - ? dac - ? adc m g = gain a = attenuation m = mute z = high z m g m z g m ga m ga m ga m ga m ga m ga m a m ga m ga m ga mmm ga m ga m ga m ga m ga m ga m m m phone_in cd_l cd_gnd cd_r aux_l aux_r line_in_l cd diff amp line_in_r sprd mz a lfe_out sprd mz a center_out mz a line_out_l m a mono_out mz a line_out_r m a surr_out_l/ hp_out_l m a surr_out_r/ hp_out_r hp hp losel losel mix hpsel hpsel v ref g v refout voltage reference AD1985 xtl_out xtl_in spdif_out pc beep generator g g mic1 mic2 oms oms 2cmic ms 03610-a-002 oms: 0x74 d9 def=0(mic_1/2) m20: 0x0e d6 def=0 (0db gain) left center left lfe right center left lfe right right left left right right left right rec selector 0x1a d2-d0 right d10-d8 left def=0x000 000= mic 001= cd 010= --- 011= aux 100= line_in 101= str_mix 110= mon_mix 111= phone rec gain: 0x1c 0db to +22.5db def=0x8000 0db/muted pcm adc rate: 0x32 7khz-48khz in 1hz steps def 0xbb80 48khz pcbeep: 0x0a vol: d4-d0 mute: d15 frequency: d12-d5 mgb[1:0]: 0x76 d1-d0 def=0 (+20db) 00= +20db gain 01= +10db gain 10= +30db gain 11 (reserved) 00= +2.25v 01= high z 10= +3.7v 11= 0v +2.25v ac97 mode: 0x18 adi mode: pcm vol 0x36 +12db to -34.5db def 0x8808 0db/muted pcm c/lfe dac rate: 0x30 7khz-48khz in 1hz steps def 0xbb80 48khz ac97 mode: 0x18 adi mode: pcm vol 0x38 +12db to -34.5db def 0x8808 0db/muted pcm surr dac rate: 0x2e 7khz-48khz in 1hz steps def 0xbb80 48khz pcm vol: 0x18 +12db to -34.5db def 0x8808 0db/muted pcm front dac rate: 0x2c 7khz-48khz in 1hz steps def 0xbb80 48khz dsa: 0x28 d5-d4 def 00 eqb: 0x72 d9-d8 def=0 (eq enabled) jsense: 0x72 def=0x0000 eapd: 0x26 d15 def=0 (enabled) eqb 0 1 0 1 1 0 1 0 1 0 1 0 0 1 0 1 0 1 0 1 1 0 1 0 1 0 mono_out: mono vol 0x06 0db to -46.5db def=0x8000 0db/muted mix: 0x20 d9 def=0 (mixer) line_out: master vol 0x02 0db to -46.5db def=0x8000 0db/muted line_out: master vol 0x02 0db to -46.5db def=0x8000 0db/muted losel: 0x76 d5 def=0 (mixer) losel: 0x76 d5 def=0 (mixer) ac97 mode: surr vol 0x38 adi mode: headphone vol 0x04 0db to -46.5db def 0x8000 0db/muted hp stby: pr6 reg 0x26 d14 1=power down, def=0 hpsel: 0x76 d10 def=0 (surround) ac97 mode: c&lfe vol 0x36 adi mode: master vol 0x02 0db to -46.5db def 0x8000 0db/muted oms: 0x74 d9 def=0(mic_1/2) cldis: 0x76 d11 def=0(enabled) sprd: 0x76 d6 def=0 (nosprd) cswp: 0x74 d3 def=0 (nocswp) down mix: 0x76 dmix 1,0 (d9,d8) def=0x0 (muted) down mix: 0x76 dmix 1,0 (d9,d8) def=0x0 (muted) down mix: 0x76 dmix 1,0 (d9,d8) def=0x0 (muted) mic vol: 0x0e +12 to -34.5db def=0x8808 0db/muted phone vol: 0x0c +12 to -34.5db def=0x8008 0db/muted line_in vol: 0x10 +12 to -34.5db def=0x8808 0db/muted aux vol: 0x16 +12 to -34.5db def=0x8808 0db/muted aux vol: 0x16 +12 to -34.5db def=0x8808 0db/muted cd vol: 0x012 +12 to -34.5db def=0x8008 0db/muted 2cmic: 0x76 d7 def=0 (ms sel) ms: 0x20 d8 def=0 (mic_1) note: setting to enable surround 5.1 is losel=1 lodis: 0x76 d12 def=0 (enabled) lodis: 0x76 d12 def=0 (enabled) +0 to -45db def=0x8000 0db/muted vrefout: 0x76 d3-d2 def=00 (+2.25v) figure 2. detailed functional block diagram
AD1985 rev. a | page 4 of 48 specifications table 1. test conditions, unless otherwise noted parameter value/condition unit temperature 25 c digital supply (dv dd ) 3.3 v analog supply (av dd ) 5.0 v sample rate (f s ) 48 khz input sine wave signal 1,008 hz analog output pass band 20 to 20,000 hz calibrated output C3 db relative to full scale 10 k? output load: line 32 ? output load: headphone 2 k? output load: center and lfe dac test conditions 47.5 k? output load: mono calibrated 0 db pga gain adc test conditions input C3 db relative to full scale clock 24.576 mhz analog input table 2. parameter min typ max unit input voltage line_in, cd, aux, phone_in 1 v rms 2.83 v p-p mic_in with +30 db preamp 0.032 v rms 0.089 v p-p mic_in with +20 db preamp 0.1 v rms 0.283 v p-p mic_in with +10 db preamp 0.316 v rms 0.894 v p-p mic_in with 0 db preamp 1 v rms 2.83 v p-p input impedance 1 20 k? input capacitance 1 5 7.5 pf master volume table 3. parameter min typ max unit step size (line out, mono out, surround out, center, lfe) 1.5 db output attenuation range (0 db to ?46.5 db) 46.5 db mute attenuation of 0 db fundamental 1 80 db programmable gain amplifieradc table 4. parameter min typ max unit step size (0 db to +22.5 db) 1.5 db pga gain range 22.5 db 1 guaranteed, not tested.
AD1985 rev. a | page 5 of 48 analog mixerinput gain/amplifiers/attenuators table 5. parameter min typ max unit signal-to-noise ratio (snr) cd to line_out 90 db line, aux, phone, to line_out 85 db mic1 or mic2 to line_out 80 db step size: all mixer inputs, except pc beep 1.5 db input gain/attenuation range (+12 db to C34.5 db): all mixer inputs , except pc beep 46.5 db digital decimation and interpolation filters 1 table 6. parameter min typ max unit pass band 0 0.4 f s hz pass-band ripple 0.09 db transition band 0.4 f s 0.6 f s hz stop band 0.6 f s hz stop-band rejection C74 db group delay 16/f s s group delay variation over pass band 0 s analog-to-digital converters table 7. parameter min typ max unit resolution 16 bits total harmonic distortion (thd) ?85 db dynamic range (C60 db in; thd+n referenc ed to full-scale; a-weighted) 84 db signal-to-intermodulation distortion (ccif method) 1 85 db adc crosstalk 1 line inputs (input l, ground r, read r; input r, ground l, read l) ?85 db line_in to other ?95 db gain error (full-scale span relative to nominal input voltage) 10 % interchannel gain mismatch (difference of gain errors) 0.5 db adc offset error 1 5 mv digital-to-analog converters table 8. parameter min typ max unit resolution 20 bits total harmonic distortion (thd); line_out, c/lfe C90 db total harmonic distortion (thd); hp_out C75 db dynamic range (C60 db in; thd+n referenc ed to full-scale; a-weighted) 90 db signal-to-intermodulation distortion (ccif method) 1 100 db gain error (output full-scale voltage relat ive to nominal output full-scale voltage) 2 10 % interchannel gain mismatch (difference of gain errors) 0.7 db dac crosstalk (input l, zero r, read r_out; input r, zero l, read l_out) 1 C100 db total out-of-band energy (measured from 0.6 f s to 100 khz) 1 C85 db 1 guaranteed, not tested. 2 c/lfe specified with 10 k? load.
AD1985 rev. a | page 6 of 48 analog output table 9. parameter min typ max unit full-scale output voltage: line out, mono out, center, lfe 1 v rms 2.83 v p-p output impedance 1 300 500 ? external load impedance 1 2 10 k? output capacitance 1 15 pf external load capacitance 1000 pf full-scale output voltage: hp_out 1 v rms 2.83 v p-p external load capacitance 1 1000 pf external load impedance 1 32 ? v ref 2.1 2.25 2.4 v v refout (vrefh, vrefd = 00 in register 0x76) 2.25 v v refout (vrefh, vrefd = 10) 3.70 v v refout (vrefh, vrefd = 11) 0.0 v v refout current drive 5 ma mute click (muted output unmuted midscale dac output) 5 mv note that setting v refout to 0 v reduces crosstalk when center/lfe is sharing the mic jack. the center/lfe crosstalk should be better than ?60 db at 100 hz when sharing with a stereo microphone application circuit. static digital specifications table 10. parameter min typ max unit digital inputs/outputs high level input voltage (v ih ) 0.65 dv dd v low level input voltage (v il ) 0.35 dv dd v high level output voltage (v oh ), i oh = 2 ma 0.9 dv dd v low level output voltage (v ol ), i ol = 2 ma 0.1 dv dd v input leakage current C10 +10 a output leakage current C10 +10 a input/output pin capacitance 7.5 pf power supply table 11. parameter min typ max unit power supply rangeanalog (av dd ) 4.5 5.5 v power supply rangedigital (dv dd ) 2.97 3.63 v power dissipation5 v/3.3 v 465 mw power supply rejection (100 mv p-p signal @ 1 khz) 1 40 db 1 guaranteed, not tested.
AD1985 rev. a | page 7 of 48 power-down states 1 table 12. parameter pr[k:i] 2 pr[6:0] 2 i dv dd (3.3 v) typ i av dd (5 v) typ unit full power-up 000 000 0000 55.5 56.0 ma adc 000 000 0001 47.4 49.9 ma front dac 000 000 0010 49.5 47.9 ma center dac 001 000 0000 55.5 56.0 ma surround dac 010 000 0000 49.0 47.5 ma lfe dac 100 000 0000 55.1 56.0 ma adc + all dacs 111 000 0011 15.8 24.2 ma mixer 000 000 0100 55.5 34.3 ma adc + mixer 000 000 0101 47.4 27.4 ma all dacs + mixer 111 000 0110 34.1 10.0 ma adc + all dacs + mixer 111 000 0111 14.3 2.5 ma standby 111 011 1111 0.114 0.004 ma headphone standby 000 100 0000 55.5 48.3 ma clock specifications table 13. parameter min typ max unit input clock frequency (xtal mode or clock oscillator) 24.576 mhz input clock frequency (reference clock mode) 14.31818 mhz input clock frequency (usb clock mode) 48.000 mhz recommended clock duty cycle 40 50 60 % 1 currents measured with v refout unloaded. 2 pr bits are controlled in registers 0x2a and 0x26.
AD1985 rev. a | page 8 of 48 timing parameters guaranteed over operating temperature range. table 14. parameter symbol min typ max unit reset active low pulse width t rst_low 1.0 s reset inactive to sdata_in or bit_clk active delay t rst2clk 162.8 ns sync active high pulse width t sync_high 1.3 s sync low pulse width t sync_low 19.5 s sync inactive to bit_clk startup delay t sync2clk 162.8 ns bit_clk frequency 12.288 mhz bit_clk period t clk_period 81.4 ns bit_clk output jitter 1, 2 750 2000 ps bit_clk high pulse width t clk_high 33 42 48 ns bit_clk low pulse width t clk_low 33 38 48 ns sync frequency 48.0 khz sync period t sync_period 20.8 s setup to falling edge of bit_clk t setup 10 2.5 ns hold from falling edge of bit_clk t hold 5 ns bit_clk rise time t riseclk 2 4 6 ns bit_clk fall time t fallclk 2 4 6 ns sync rise time t risesync 2 4 6 ns sync fall time t fallsync 2 4 6 ns sdata_in rise time t risedin 2 4 6 ns sdata_in fall time t falldin 2 4 6 ns sdata_out rise time t risedout 2 4 6 ns sdata_out fall time t falldout 2 4 6 ns end of slot 2 to bit_clk, sdata_in low t s2_pdown 0 1.0 s setup to trailing edge of reset (applies to sync, sdata_out) t setup2rst 15.0 ns rising edge of reset to high-z delay t off 25.0 ns propagation delay 15 ns reset rise time 50 ns output valid delay from rising edge of bit_clk to sdi valid t co 15 ns reset inactive to bit_clk startup delay t tri2actv 25 ns 1 guaranteed, not tested. 2 output jitter directly dependent on crystal input jitter; maximum specified for noncrystal operation.
AD1985 rev. a | page 9 of 48 reset bit_clk sdata_in t rst_low t rst2clk t tri2actv t tri2actv 03610-0-003 figure 3. cold reset timing (codec is supplying the bit_clk signal) sync bit_clk t sync_high t sync2clk 03610-0-004 figure 4. warm reset timing bit_clk sync t clk_low t clk_high t clk_period t sync_low t sync_period t sync_high 03610-0-005 figure 5. clock timing bit_clk sync sdata_in sdata_out t riseclk t fallclk t risesync t fallsync t risedin t falldin t risedout t falldout 03610-0-006 figure 6. signal rise and fall times bit_clk sync sdata_in sdata_ou t bit_clk not to scale slot 1 slot 2 write to 03 26 data pr4 t s2_pdown 03610-0-007 figure 7. ac link low power mode timing bit_clk sdata_ou t sdata_in sync t co t setup v ih v il v oh v ol t hold 03610-0-008 figure 8. ac link low power mode timing reset sdata_out sdata_in, bit_clk , eapd, spdif_out and digital i/o hi-z t setup2rst t off 03610-0-009 figure 9. ate test mode
AD1985 rev. a | page 10 of 48 absolute maximum ratings table 15. parameter min max unit power supplies digital (dv dd ) C0.3 +3.6 v analog (av dd ) C0.3 +6.0 v input current (except supply pins) 10.0 ma analog input voltage (signal pins) C0.3 av dd + 0.3 v digital input voltage (signal pins) C0.3 dv dd + 0.3 v ambient temperature (operating) 0 70 c storage temperature C65 +150 c stresses above those listed under absolute maximum ratings may cause permanent damage to the device. this is a stress rating only; functional operation of the device at these or any other conditions above those indicated in the operational section of this specification is not implied. exposure to absolute maximum rating conditions for extended periods may affect device reliability. environmental conditions ambient temperature rating t case = case temperature in c pd = power dissipation in w ja = thermal resistance (junction to ambient) jc = thermal resistance (junction to case) table 16. thermal resistance package type ja jc lqfp 50.1c/w 17.8c/w all measurements per eia/jesd51 with 2s2p test board per eia/jesd51-7. esd caution esd (electrostatic discharge) sensitive device. electrosta tic charges as high as 4000 v readily accumulate on the human body and test equipment and can discharge with out detection. although this product features proprietary esd protection circuitry, permanent dama ge may occur on devices subjected to high energy electrostatic discharges. therefore, proper esd precautions are recommended to avoid performance degradation or loss of functionality.
AD1985 rev. a | page 11 of 48 pin configuration and fu nction descriptions spdif eapd id1 id0 av ss3 av dd3 nc surr/hp_out_r av ss2 surr/hp_out_l av dd2 mono_out nc = no connect phone_in aux_l aux_r js1 js0 cd_l cd_gnd_ref cd_r mic1 mic2 line_in_l line_in_r dv dd1 xtl_in xtl_out dv ss1 sdata_out bit_clk dv ss2 sdata_in dv dd2 sync reset js3 line_out_r (front)/surr_r line_out_l (front)/surr_l av dd4 js2 lfe_out center_out afilt2 afilt1 v refout v ref av ss1 av dd1 48 47 46 45 44 39 38 37 43 42 41 40 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 36 35 34 33 32 31 30 29 28 27 26 25 03610-a-011 pin 1 identifier top view (not to scale) AD1985 figure 10. 48-lead lqfp pin configuration circuit layout note : in normal operation, surround and line out are swapped to provide headphone drive on line outputs. therefore, pins 35 and 36 be come the surround l/r outputs and pins 39 and 41 become the line out (front) l/r outputs with he adphone drive. see bi ts losel and hpsel in register 0x76 for details. pin function descriptions table 17. digital i/o mnemonic pin no. i/o description xtl_in 2 i crystal input (24.576 mhz) or external clock in (24.576 mhz, 14.31818 mhz, or 48.000 mhz). xtl_out 3 o crystal output. sdata_out 5 i ac link serial da ta output. AD1985 input stream. bit_clk 6 o/i ac link bit clock. 12.288 mhz serial da ta clock. (input pin, for secondary mode only.) sdata_in 8 o ac link serial da ta input. AD1985 output stream. sync 10 i ac link frame sync. reset 11 i ac link reset. AD1985 master hardware reset. spdif 48 o spdif output. table 18. chip selects/clock strapping mnemonic pin no. i/o description id0 45 i chip select input 0 (active low). this pin can also be us ed as the chain input from a secondary codec. id1 46 i chip select input 1 (active low). table 19. jack sense/eapd mnemonic pin no. type description eapd 47 o eapd output. js0 17 i jack sense 0 input. js1 16 i jack sense 1 input. js2 33 i jack sense 2 input. js3 12 i jack sense 3 input.
AD1985 rev. a | page 12 of 48 table 20. analog i/o mnemonic pin no. i/o description phone_in 13 i monaural line-level input. aux_l 14 i auxiliary input, left channel. aux_r 15 i auxiliary input, right channel. cd_l 18 i cd audio left channel. cd_gnd_ref 19 i cd audio analog ground reference for differential cd input. cd_r 20 i cd audio right channel. mic1 21 i microphone 1 input. mic2 22 i microphone 2 input. line_in_l 23 i line in left channel. line_in_r 24 i line in right channel. center_out 31 i/o center channel output or input to recorder (depending on oms bit in reg 0x74). lfe_out 32 i/o low frequency enhanced output or inp ut to recorder (depending on oms bit in reg 0x74). line_out_l/surr_l 35 o line out (front) left channel or surround left channel (depending on losel bit in reg 0x76). line_out_r/surr_r 36 o line out (front) right channel or surround right channel (depending on losel bit in reg 0x76). mono_out 37 o monaural output to teleph ony subsystem speakerphone. surr_out_l/hp_out_l 39 o surround or front headphone left channel output (depending on hpsel bit in reg 0x76). surr_out_r/hp_out_r 41 o surround or front headphone righ t channel output (depending on hpsel bit in reg 0x76). table 21. filter/reference mnemonic pin no. i/o description v ref 27 o voltage reference filter. v refout 28 o voltage reference output (intended for mic bias). afilt1 29 o antialiasing filter capacitoradc right channel. aflit2 30 o antialiasing filter capacitoradc left channel. table 22. power and ground signals mnemonic pin no. type description dv dd1 1 i digital v dd : 3.3 v. dv ss1 4 i digital ground. dv ss2 7 i digital ground. dv dd2 9 i digital v dd : 3.3 v. av dd1 25 i analog v dd : 5.0 v. av ss1 26 i analog ground. av dd4 34 i analog v dd : 5.0 v. av dd2 38 i analog v dd : 5.0 v. av ss2 40 i analog ground. av dd3 43 i analog v dd : 5.0 v. av ss3 44 i analog ground. table 23. no connects mnemonic pin no. type description nc 42 n/a no connect.
AD1985 rev. a | page 13 of 48 indexed control registers notes odd register addresses are aliased to the next lower even address. registers not shown and bits containing an x are assumed to be reserved. reserved registers should not be written. zeros should be written to reserved bits. 1 for ac 97 compatibility, these rm bits must be enabled before they can have any effect. reg name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x00 reset x se4 se3 se2 se1 se0 id9 id8 id7 id6 id5 id4 id3 id2 id1 id0 0x0090 0x02 master volume mm x x lmv4 lmv3 lmv2 lmv1 lmv0 mmrm 1 x x rmv4 rmv3 rm v2 rmv1 rmv0 0x8000 0x04 headphones volume hpm x x lhv4 lhv3 lhv2 lhv1 lhv0 hprm 1 x x rhv4 rhv3 rhv2 rhv1 rhv0 0x8000 0x06 mono volume mvm x x x x x x x x x x mv4 mv2 mv2 mv1 mv0 0x8000 0x0a pc beep pcbm x x pcbf7 pcbf6 pcbf5 pcbf4 pcbf3 pcbf 2 pcbf1 pcbf0 pcbv3 pc bv2 pcbv1 pcbv0 x 0x8000 0x0c phone_in volume phm x x x x x x x x x x phv4 phv3 phv2 phv1 phv0 0x8008 0x0e mic volume mcm x x x x x x x x m20 x mcv4 mcv3 mcv2 mcv1 mcv0 0x8008 0x10 line in volume lm x x llv4 llv3 llv2 llv1 llv0 lvrm 1 x x rlv4 rlv3 rlv2 rlv1 rlv0 0x8808 0x12 cd volume cm x x lcv4 lcv3 lcv2 lcv1 lcv0 cdrm 1 x x rcv4 rcv3 rcv2 rcv1 rcv0 0x8808 0x16 aux in volume am x x lav4 lav3 lav2 lav1 lav0 avrm 1 x x rav4 rav3 rav2 rav1 rav0 0x8808 0x18 pcm out vol om x x lov4 lov3 lov2 lov1 lov0 omrm 1 x x rov4 rov3 rov2 rov1 rov0 0x8808 0x1a record select x x x x x ls2 ls1 ls0 x x x x x rs2 rs1 rs0 0x0000 0x1c record gain im x x x liv3 liv2 liv1 liv0 imrm 1 x x x riv3 riv2 riv1 riv0 0x8000 0x20 general purpose x x x x drss1 drss0 mix ms lpbk x x x x x x x 0x0000 0x24 audio int. and paging i4 i3 i2 i1 i0 x x x x x x x pg3 pg2 pg1 pg0 0xxxxx 0x26 power-down ctrl/stat eapd pr6 pr5 pr4 pr3 pr2 pr1 pr0 x x x x ref anl dac adc n/a 0x28 extended audio id aid1 aid0 x x rev1 rev0 amap aidldac aidsdac aidcdac dsa1 dsa0 x aidspdif dra aidvra 0xxbc7 0x2a extd audio stat/ctrl vforce x prk prj pri spcv x ascldac ascsdac asccdac spsa1 spsa 0 x ascspdf ascdra ascvra 0xxxxx 0x2c pcm front dac rate srf15 srf14 srf13 srf12 sr f11 srf10 srf9 srf8 srf7 srf6 srf5 srf4 srf3 srf2 srf1 srf0 0xbb80 0x2e pcm surr dac rate srs15 srs14 srs13 srs12 sr s11 srs10 srs9 srs8 srs7 srs6 srs5 srs4 srs3 sr s2 srs1 srs0 0xbb80 0x30 pcm lfe/c dac rate srcl15 srcl14 srcl13 srcl12 sr cl11 srcl10 srcl9 srcl8 srcl 7 srcl6 srcl5 srcl4 srcl3 srcl2 srcl1 srcl0 0xbb80 0x32 pcm l/r adc rate sra15 sra14 sra13 sra12 sr a11 sra10 sra9 sra8 sra7 sra6 sra5 sra4 sra3 sra2 sra1 sra0 0xbb80 0x36 center/lfe volume lfem x x lfe4 lfe3 lfe2 lfe1 lfe0 cn tm x x cnt4 cnt3 cnt2 cnt1 cnt0 0x8080 0x38 surround volume lsm x lsr5 lsr4 lsr3 lsr2 lsr1 lsr0 rs m x rsr5 rsr4 rsr3 rsr2 rsr1 rsr0 0x8080 0x3a spdif control v x spsr1 spsr0 l cc6 cc5 cc4 cc3 cc2 cc1 cc0 pre copy aud pro 0x2000 0x60 eq control eqm x x x x x x x sym chs bca5 bca4 bca3 bca2 bca1 bca0 0x8080 0x62 eq data cfd15 cfd14 cfd13 cfd12 cfd11 cfd10 cfd9 cfd8 cfd7 cfd6 cfd5 cfd4 cfd3 cfd2 cfd1 cfd0 0x0000 0x70 j sense/general x x x x x x x x mmdis js2sel x x x x x x n/a 0x72 j sense/audio/status js sprd js1 dmx js0 dmx js mt2 js mt1 js mt0 js1 eqb js0 eqb js1 tmr js0 tmr js1 md js0 md js1 st js0 st js1 int js0 int n/a 0x74 serial configuration slot16 regm2 regm1 regm0 regm3 drf oms chen spovr lbks1 lbks0 ints cswp spal spdz splnk 0x1001 0x76 misc control bits dacz ac97nc msplt lodis cldis hpsel dmix1 dmix0 sp rd 2cmic losel sru vrefh vrefd mbg1 mbg0 0x0000 0x78 advanced jack sense x x x x x x x x js3tmr js2tmr js3md js2md js3st js2st js3int js2int n/a 0x7c vendor id1 vidf7 vidf6 vidf5 vidf4 vidf3 vidf2 vidf1 vidf0 vi ds7 vids6 vids5 vids4 vids3 vids2 vids1 vids0 0x4144 0x7e vendor id2 vidt7 vidt6 vidt5 vidt4 vidt3 vidt2 vidt1 vidt0 vidrev7 vi drev6 vidrev5 vidrev4 vidrev3 vidrev2 vidrev1 vidrev0 0x5375 0x60 pg. 1 codec class/rev x x x cl4 cl3 cl2 cl1 cl0 rv7 rv6 rv5 rv4 rv3 rv2 rv1 rv0 n/a 0x62 pg. 1 pci svid pvi15 pvi14 pvi13 pv i12 pvi11 pvi10 pvi9 pvi8 pvi7 pvi6 pvi5 pvi4 pv i3 pvi2 pvi1 pvi0 n/a 0x64 pg. 1 pci sid pi15 pi14 pi13 pi12 pi11 pi10 pi9 pi8 pi7 pi6 pi5 pi 4 pi3 pi2 pi1 pi0 n/a 0x66 pg. 1 function select x x x x x x x x x x x fc3 fc2 fc1 fc0 t/r 0x0000 0x68 pg. 1 function information g4 g3 g2 g1 g0 inv dl4 dl3 dl2 dl1 dl0 iv x x x fip n/a 0x6a pg. 1 sense register st2 st1 st0 s4 s3 s2 s1 s0 or1 or0 sr5 sr4 sr3 sr 2 sr1 sr0 n/a
AD1985 rev. a | page 14 of 48 reset (index 0x00) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x00 reset x se4 se3 se2 se1 se0 id9 id8 id7 id6 id5 id4 id3 id2 id1 id0 0x0090 note: writing any value to this register perf orms a register reset, which causes all registers (except register 0x74) to revert to their default values. register 0x74 will only reset bits cswp (d3), lbks[1:0] (d[6:5]), an d oms (d9). the regm and ser ial configuration bits are re set only by an external ha rdware reset. reading this register returns the id code of the part and a code for the type of 3d stereo enhancement. id[9:0] identify capability: the id decodes the capabilities of AD1985 based on the following: bit = 1 function AD1985 id0 dedicated mic pcm in channel 0 id1 reserved (per ac 97, 2.3) 0 id2 bass and treble control 0 id3 simulated stereo (mono to stereo) 0 id4 headphone out support 1 id5 loudness (bass boost) support 0 id6 18-bit dac resolution 0 id7 20-bit dac resolution 1 id8 18-bit adc resolution 0 id9 20-bit adc resolution 0 se[4:0] stereo enhancement. the AD1985 does not provide hardware 3d stereo enhancement (all bits are 0).
AD1985 rev. a | page 15 of 48 master volume register (index 0x02) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x02 master volume mm x x lmv4 lmv3 lmv2 lmv1 lmv0 mmrm 1 x x rmv4 rmv3 rmv2 rmv1 rmv0 0x8000 1 for ac 97 compatibility, bit d7 (mmrm) is available only by setting the msplt bit in register 0x76. the msplt bit enables sep arate mute bits for the left and right channels. this register controls the line_out volume and mute bits. each volume subregister contains five bits, generating 32 volume levels with increments of 1.5 db each. ac 97 defines the 6-bit volume registers, therefore, to maintain compatibility whenever the d5 or d13 bit is set to 1, its respective lower five volume bits are automatically set to 1 by the codec logic. on readback, all lower five bits will read 1s whenever these bits are set to 1. note that depending on the state of the ac97nc bit in register 0x76, this register has the following additional functionality: ? for ac97nc = 0, the register controls the line_out output attenuators only. ? for ac97nc = 1, the register controls the line_out, center, and lfe output attenuators. rmv[4:0] right master volume control. the least significant bit repr esents 1.5 db. this register controls the output from 0 db to a maximum attenuation of 46.5 db. mmrm right channel mute. once enabled by the msplt bit in register 0x76, this bit mutes the right channel separately from the mm bit. otherwise, this bit will a lways read 0 and will have no effect when set to 1. lmv[4:0] left master volume control. the least significant bit repr esents 1.5 db. this register controls the output from 0 db to a maximum attenuation of 46.5 db. mm master volume mute. when this bit is set to 1, all channels are muted, un less the msplt bit in register 0x76 is set to 1, in which case, this mute bit will only affect the left channels. volume settings for master and headphone reg. 0x76 control bits master volume (0x02) and headphone volume (0x04) left channel volume d[13:8] right channel volume d[5:0] msplt 1 d15 write readback function d7 1 write readback function 0 0 00 0000 00 0000 0 db gain x 00 0000 00 0000 0 db gain 0 0 00 1111 00 1111 C22.5 db gain x 00 1111 00 1111 C22.5 db gain 0 0 01 1111 01 1111 C46.5 db gain x 01 1111 01 1111 C46.5 db gain 0 0 1x xxxx 01 1111 C46.5 db gain x 1x xxxx 01 1111 C46.5 db gain 0 1 xx xxxx xx xxxx C db gain, muted x xx xxxx xx xxxx C db gain, muted 1 0 1x xxxx 01 1111 C46.5 db gain 1 xx xxxx xx xxxx C db gain, only right muted 1 1 xx xxxx xx xxxx C db gain, only left muted 0 1x xxxx 01 1111 C46.5 db gain 1 1 xx xxxx xx xxxx C db gain, left muted 1 xx xxxx xx xxxx C db gain, right muted note: x in the above table is a wild ca rd, meaning the value has no effect. 1 for ac 97 compatibility, bit d7 is available only by setting the msplt bit, register 0x76. th e msplt bit enables separate mut e bits for the left and right channels. if msplt is not set, bit d7 has no effect.
AD1985 rev. a | page 16 of 48 headphone volume regi ster (index 0x04) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x04 headphones volume hpm x x lhv4 lhv3 lhv2 lhv1 lhv0 hprm 1 x x rhv4 rhv3 rhv2 rhv1 rhv0 0x8000 1 for ac 97 compatibility, bit d7 (hprm) is available only by setting the msplt bit in register 0x76. the msplt bit enables sep arate mute bits for the left and right channels. this register controls the headphone volume for both stereo channels and mute bits. each volume subregister contains five bits, generating 32 volume levels with increments of 1.5 db each. ac 97 defines the 6-bit volume registers, therefore, to maintain compatibility whenever the d5 or d13 bit is set to 1, its respective lower five volume bits are automatically set to 1 by the codec logic. on readback, all lower five bits will read 1s whenever these bits are set to 1 (see the volume settings for master and headphone table on the previous page). note that depending on the state of the ac97nc bit in register 0x76, this register has the following additional functionality: ? for ac97nc = 0, the register has no control over the surr_out/hp_out outputs (see register 0x38). ? for ac97nc = 1, the register controls the surr_out/hp_out output attenuators. rhv[4:0] right headphone volume control. the least significant bit represents 1. 5 db. this register controls the output from 0 db to a maximum attenuation of 46.5 db. hprm right channel mute. once enabled by the msplt bit in register 0x76, this bit mutes the right channel separately from the hpm bit. otherwise, this bit will a lways read 0 and will have no effect when set to 1. lhv[4:0] left headphone volume control. the least significant bit represents 1.5 db. this register controls the output from 0 db to a maximum attenuation of 46.5 db. hpm headphones volume mute. when this bit is set to 1, both the left and right channels are muted, unless the msplt bit in register 0x76 is set to 1, in which case, this mute bit will only affect the left channel.
AD1985 rev. a | page 17 of 48 mono volume register (index 0x06) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x06 mono volume mvm x x x x x x x x x x mv4 mv3 mv2 mv1 mv0 0x8000 this register controls the mono output volume and mute bit. the volume register contains five bits, generating 32 volume levels with increments of 1.5 db each. ac 97 defines the 6-bit volume registers, therefore, to maintain compatibility whenever the d5 bit is set to 1, its respective lower five volume bits are automatically set to 1 by the codec logic. on readback, all lower five bits will read 1s whenever this bit is set to 1. mv[4:0] mono volume control. the least significant bit represents 1.5 db. this register cont rols the output from 0 db to a maximum attenuation of 46.5 db. mvm mono volume mute. when this bit is set to 1, the channel is muted. volume settings for mono control bits d[4:0] for mono (0x06) d15 write readback function 0 0 0000 0 0000 0 db gain 0 0 1111 0 1111 C22.5 db gain 0 1 1111 1 1111 C46.5 db gain 1 x xxxx x xxxx C db gain, muted note that the x in the above table is a wild card, meaning the value has no effect.
AD1985 rev. a | page 18 of 48 pc beep register (index 0x0a) reg name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x0a pc beep volume pcbm x x pcbf7 pcbf6 pcbf5 pcb f4 pcbf3 pcbf2 pcbf1 pcbf0 pcbv3 pcbv2 pcbv1 pcbv0 x 0x8000 this register controls the level and frequency for the digital pc beep generated by the codec. please note that pc beep should be muted when not in use. pcbv[3:0] controls the volume of the generated signal. each step corresponds to approximately 3 db of attenuation. the msb of the register is the mute bit. when this bit is set to 1, the level for th e signal is set at C db. pcbf[7:0] these bits are writeable, and the code c-digital pc beep generation is supp orted. the beep frequency generated is the result of dividing the 48 khz clock by 4x the numb er specified in pcbf[7:0], allowing tones from 47 hz to 12 khz. a value of 0x00 in bits pcbf[7 :0] disables internal pc beep genera tion. the pv bits control the volume level of the generated signal. the register default value is 0x8000, which corresponds to 0 db attenuation with mute on. pcbm pv3 to pv0 function 0 0000 0 db attenuation 0 1111 45 db attenuation 1 xxxx db attenuation pcbf frequency of pc beep 0x00 pc beep disabled 0x01 12 khz 0x0c 1 khz 0xff 47 hz
AD1985 rev. a | page 19 of 48 phone_in volume register (index 0x0c) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x0c phone_in volume phm x x x x x x x x x x phv4 phv3 phv2 phv1 phv0 0x8008 phv[4:0] phone volume. allows setting the phone volume gain/attenu ator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. the default value is 0 db, with mute bit enabled. phm phone mute. when this bit is set to 1, the phone channel is muted. volume settings for phone and mic control bits d[4:0] phone (0x0c) and mic (0x0e) d15 write readback function 0 0 0000 0 0000 12 db gain 0 0 1000 0 1000 0 db gain 0 1 1111 1 1111 C34.5 db gain 1 x xxxx x xxxx C db gain, muted note that the x in the above table is a wild card, meaning the value has no effect. microphone volume re gister (index 0x0e) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x0e mic volume mcm x x x x x x x x m 20 x mcv4 mcv3 mcv2 mcv1 mcv0 0x8008 this register controls the volume, gain boost, and mute for the gain/attenuators on both the mic1 and mic2 paths to the mixer. there is no separate control for left and right on this path. the signal paths must be identical, hence the single control for both. mcv[4:0] mic volume. allows setting the mic volume gain/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. the de fault value is 0 db, with mute enabled. m20 mic gain boost. this bit allows setting additional mic gain to increase the mic rophone sensitivity. gain is applied to the microphone path to both the analog mixer and the adc(s). the nominal gain boost by default is + 20 db; however, bits d0 and d1 (mbg[1:0]) on the miscellaneous control bits register (0x76), allow changing the gain boost to +10 db or +30 db, if necessary. 0 = disabled; gain = 0 db. 1 = enabled; default gain = +20 db (s ee register 0x76, bits d0 and d1). mcm mic mute. when this bit is se t to 1, both channels are muted.
AD1985 rev. a | page 20 of 48 line in volume (index 0x10) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x10 line in volume lm x x llv4 llv3 llv2 llv1 llv0 lvrm 1 x x rlv4 rlv3 rlv2 rlv1 rlv0 0x8808 1 for ac 97 compatibility, bit d7 (lvrm) is available only by setting the msplt bit in register 0x76. the msplt bit enables sep arate mute bits for the left and right channels. rlv[4:0] right line in volume. allows setting the line in righ t channel gain/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain rang e is +12 db to C34.5 db. the defa ult value is 0 db, mute enabled. lvrm right channel mute. once enabled by the msplt bit in register 0x76, this bit mutes the right channel separately from the lm bit. otherwise, this bit will always read 0 and will have no effect when set to 1. llv[4:0] line in volume left. allows setting the line in left channel gain/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain rang e is +12 db to C34.5 db. the defa ult value is 0 db, mute enabled. lm line in mute. when this bit is set to 1, both the left and right channels are muted, unless the msplt bit in register 0x76 is set to 1, in which case, this mute bit will affect only the left channel. volume settings for line in, cd volume, aux, and pcm out reg. 0x76 control bits line in (0x10), cd (0x12), aux (0x16), and pcm out (0x18) left channel volume d[12:8] right channel volume d[4:0] msplt 1 d15 write readback function d7 1 write readback function 0 0 0 0000 0 0000 12 db gain x 0 0000 0 0000 12 db gain 0 0 0 1000 0 1000 0 db gain x 0 1000 0 1000 0 db gain 0 0 1 1111 1 1111 C34.5 db gain x 1 1111 1 1111 C34.5 db gain 0 1 x xxxx x xxxx C db gain, muted x x xxxx x xxxx C db gain, muted 1 0 1 1111 1 1111 C34.5 db gain 1 x xxxx x xxxx C db gain, only right muted 1 1 x xxxx x xxxx C db gain, only left muted 0 1 1111 1 1111 C34.5 db gain 1 1 x xxxx x xxxx C db gain, left muted 1 x xxxx x xxxx C db gain, right muted 1 for ac 97 compatibility, bit d7 is available only by setting the msplt bit, register 0x76. th e msplt bit enables separate mut e bits for the left and right channels. if msplt is not set, rm bit has no effect. note that the x in the above table is a wild card, meaning the value has no effect. cd volume register (index 0x12) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x12 cd volume cm x x lcv4 lcv3 lcv2 lcv1 lcv0 cdrm 1 x x rcv4 rcv3 rcv2 rcv1 rcv0 0x8808 1 for ac97 compatibility, bit d7 (cdrm) is av ailable only by setting the msplt bit in register 0x76. the msplt bit enables separ ate mute bits for the left and right channels. see the volume settings for line in, cd volume, aux, and pcm out table. rcv[4:0] right cd volume. allows setting the cd right channel ga in/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. the default value is 0 db, mute enabled. cdrm right channel mute. once enabled by the msplt bit in register 0x76, this bit mutes the right channel separately from the cm bit. otherwise, this bit will a lways read 0 and will have no effect when set to 1. lcv[4:0] left cd volume. allows setting the cd left channel ga in/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. the default value is 0 db, mute enabled. cm cd volume mute. when this bit is set to 1, both the le ft and right channels are m uted, unless the msplt bit in register 0x76 is set to 1, in which case, this mute bit will only affect the left channel.
AD1985 rev. a | page 21 of 48 aux in volume register (index 0x16) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x16 aux in volume am x x lav4 lav3 lav2 lav1 lav0 avrm 1 x x rav4 rav3 rav2 rav1 rav0 0x8808 1 for ac 97 compatibility, bit d7 (avrm) is available only by setting the msplt bit in register 0x76. the msplt bit enables sep arate mute bits for the left and right channels. see the volume settings for line in, cd volume, aux, and pcm out table. rav[4:0] right aux volume. allows setting the aux righ t channel gain/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. the default va lue is 0 db, mute enabled. avrm right channel mute: once enabled by th e msplt bit in register 0x76, this bi t mutes the right channel separately from the am bit. otherwise, this bit will always read 0 and will have no effect when set to 1. lav[4:0] left aux volume. allows setting the aux left channel gain/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. the default value is 0 db, mute enabled. am aux volume mute. when this bit is set to 1, both the le ft and right channels are m uted, unless the msplt bit in register 0x76 is set to 1, in which case, this mute bit will only affect the left channel. pcm out volume (index 0x18) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x18 pcm out vol om x x lov4 lov3 lov2 lov1 lov0 omrm 1 x x rov4 rov3 rov2 rov1 rov0 0x8808 1 for ac 97 compatibility, bit d7 (omrm) is available only by setting the msplt bit in register 0x76. the msplt bit enables sep arate mute bits for the left and right channels. see the volume settings for line in, cd volume, aux, and pcm out table. note that depending on the state of the ac97nc bit in register 0x76, this register has the following additional functionality: ? for ac97nc = 0, the register also controls the surround, center, and lfe dac gain/attenuators. ? for ac97nc = 1, the register controls the pcm out volume only. rov[4:0] right pcm out volume. allows setting the pc m right channel gain/attenuator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. th e default value is 0 db, mute enabled. omrm right channel mute. once enabled by the ms plt bit in register 0x76, this bit mutes the right channel separately from the om bit. otherwise, this bit will always read 0 and will have no effect when set to 1. lov[4:0] left pcm out volume. allows setting the pcm left channel gain/a ttenuator to one of 32 levels. the lsb represents 1.5 db, and the gain range is +12 db to C34.5 db. th e default value is 0 db, mute enabled. om pcm out volume mute. when this bit is se t to 1, both the left and right channels ar e muted, unless the msplt bit in register 0x76 is set to 1, in which case, this mute bit will only affect the left channel.
AD1985 rev. a | page 22 of 48 record select control register (index 0x1a) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x1a record select x x x x x ls2 ls1 ls0 x x x x x rs2 rs1 rs0 0x0000 this register is used to select the record source, independently for the right and left channels. for single mic recording, see the ms bit (register 0x20) for mic1 and mic2 input selection. for dual mic recording, see the 2cmic bit (register 0x76) to enable simultaneous recording into l/r channels. for output line sharing for the microphones, see the oms bit (register 0x74) to swap between the mic1/mic2 and c/lfe pins. the default value is 0x0000, which corresponds to the mic input for both channels. rs[2:0] right record select: see table below. ls[2:0] left record select: see table below. ls[2:0] left record select oms 2cmic ms 0 0 0 mic1 (mono) 0 0 1 mic2 (mono) 0 1 0 mic1 (stereo) 0 1 1 mic2 (stereo) 1 0 0 center (mono) 1 0 1 lfe (mono) 1 1 0 center (stereo) 0 1 1 1 lfe (stereo) 1 cd_in (left) 2 muted 3 aux_in (left) 4 line_in (left) 5 stereo mix (left) 6 mono mix (mono) 7 phone_in (mono) rs[2:0] right record select oms 2cmic ms 0 0 0 mic1 (mono) 0 0 1 mic2 (mono) 0 1 0 mic2 (stereo) 0 1 1 mic1 (stereo) 1 0 0 center (mono) 1 0 1 lfe (mono) 1 1 0 lfe (stereo) 0 1 1 1 center (stereo) 1 cd_in (right) 2 muted 3 aux_in (right) 4 line_in (right) 5 stereo mix (right) 6 mono mix (mono) 7 phone_in (mono)
AD1985 rev. a | page 23 of 48 record gain register (index 0x1c) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x1c record gain im x x x liv3 liv2 liv1 liv0 imrm 1 x x x riv3 riv2 riv1 riv0 0x8000 1 for ac 97 compatibility, bit d7 (imrm) is available only by setting the msplt bit in register 0x76. the msplt bit enables sep arate mute bits for the left and right channels. riv[3:0] adc right channel input volume ga in control. each lsb represents 1.5 db and the gain range is 0 db to +22.5 db. imrm right channel mute. once enabled by th e msplt bit in register 0x76, this bi t mutes the right channel separately from the im bit. otherwise, this bit will always read 0 and will have no effect when set to 1. liv[3:0] adc left channel input volume ga in control. each lsb represents 1.5 db and the gain range is 0 db to +22.5 db. im input mute. when this bit is set to 1, both the left and right channels ar e muted, unless the msplt bit in register 0x76 is set to 1, in which case, this mute bit will only affect the left channel. im xim[3:0] function 0 1111 +22.5 db gain 0 0000 0 db gain 1 xxxxx muted
AD1985 rev. a | page 24 of 48 general purpose register (index 0x20) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x20 general purpose x x x x drss1 drss0 mix ms lpbk x x x x x x x 0x0000 lpbk loopback control. this bit enables the digital internal l oopback from the adc to the front dac. this feature is normally used for testing and troubleshooting. see lbks bits in register 0x74 for changing the loopback path to use the surround or center/lfe dacs. mic select. selects mic input into the record selector input and also selects the input to the analog mixer. (see the 2cmic [bit d7 register 0x76] to enable simult aneous dual microphone record ing, and the oms [bit d9 register 0x74] to enable output line sharing for the microphone inputs.) the following chart shows which of the codec i/o pins are used for the microphone source from the various settings of the microphone control bits. oms 2cmic ms left channel right channel 0 0 0 mic1 (mono) mic1 (mono) 0 0 1 mic2 (mono) mic2 (mono) 0 1 0 mic1 (stereo) mic2 (stereo) 0 1 1 mic2 (stereo) mic1 (stereo) 1 0 0 center (mono) center (mono) 1 0 1 lfe (mono) lfe (mono) 1 1 0 center (stereo) lfe (stereo) ms 1 1 1 lfe (stereo) center (stereo) mono output select. selects mono output audio source. 0 = mixer output. default. 1 = selected mic channel (selected cha nnel is unaffected by 2cmic setting). mix oms ms mono output 0 x x mixer_mono (default) 1 0 0 mic1 (mono) 1 0 1 mic2 (mono) 1 1 0 center (mono) mix 1 1 1 lfe (mono) drss[1:0] double rate slot select. the drss bi ts specify the slots for the n + 1 sa mple outputs. pcm l (n + 1) and pcm r (n + 1) data are by default provided in output slots 10 and 11. 00: pcm l, r n + 1 data is on slots 10, 11 (reset default). 01: pcm l, r n + 1 data is on slots 7, 8. 10: reserved. 11: reserved.
AD1985 rev. a | page 25 of 48 audio interrupt and paging mech anism register (index 0x24) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x24 audio int. and paging i4 i3 i2 i1 i0 x x x x x x x pg3 pg2 pg1 pg0 0xxxxx note that this register controls the audio interrupt and register paging mechanisms. i4 interrupt status (rea d/write), default is 0. 0: interrupt is clear. 1: interrupt was generated. interrupt event is cleared by writing a 1 to this bit. the interrupt status bit will change regard less of the setting of interrupt enable (i0). an interrupt in the gpi in slot 12 in the ac link will follow this bit change when interru pt enable (i0) is unmasked. if this bit is set, one or both of i3 or i2 must be set to indicate the interrupt cause. i[3:2] interrupt cause (rea d-only), default is 0. i[2] = 0: sense status has not changed (did not cause interrupt). i[2] = 1: sense cycle completed or new sense information is available. i[3] = 0: gpio status change did not cause inte rrupt (default) i[3] = 1: gpio status change caused interrupt. these bits will indicate the cause(s) of an interrupt. this information should be used to service the correct interrupting even t(s). if the interrupt status bit (i4) is set, one or both of thes e bits must be set to indi cate the interrupt cause. hardware resets these bits back to 0 wh en the interrupt status bit is cleared. i1 sense cycle (read/write), default is 0. 0: sense cycle not in progress. 1: sense cycle start. writing a 1 to this bit causes a sense cycle start if supported. if a sense cycle is in progress, writing a 0 to this bit will a bort the sense cycle. the data in the sense result register (0x6 a, page 1) may or may not be valid, as determin ed by the iv bit in that register. i0 interrupt enable (rea d/write), default is 0. 0: interrupt generation is masked. 1: interrupt generation is unmasked. software should not unmask the interrupt unless the ac 97 controller ensu res that no conflict is possible with modem slot 12 gpi functionality. ac 97 2.2 compliant controllers will not likely support audio codec interrupt infrastructure. in that case, software can poll the interrupt status afte r initiating a sense cycle and waiting for the sense cycle maximum delay (which is defined by the software) to determine if an interrupting event has occurred. x reserved. pg[3:0] page selector (rea d/write), default is 0x0. 0: vendor specific. 0x1: page id 01, registers defined in ac97, rev. 2.3. 0x2 to 0xf: reserved pages. this register is used to select a descriptor of 16-word pages between registers 0x60 and 0x6f. a value of 0x0 is used to select vendor specific space to maintain compatibility with ac 97 2.2 vendor specific registers. system software can determine implemented pages by writing the page number and reading the value back. if the value read back does not match the value writte n, the page is not implemented. all implemented pages must be in consecutive order, i.e ., page 0x2 cannot be implem ented without page 0x1.
AD1985 rev. a | page 26 of 48 power-down control/status register (index 0x26) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x26 power-down ctrl/stat eapd pr6 pr5 pr4 pr3 pr2 pr1 pr0 x x x x ref anl dac adc n/a note that the ready bits are read-only; writing to ref, anl, dac, and adc will have no effect. these bits indicate the status f or the AD1985 subsections. if the bit is a 1, then that subsection is ready. ready is defined as the subsection able to perform in its nominal state. adc adc sections ready to transmit data. dac dac sections ready to accept data. anl analog amplifiers, attenu ators, and mixers ready. ref voltage references, v ref and v refout , up to nominal level. pr[6:0] codec power-down modes. some bits can be used individually, while others are used in combination. pr0 adcs and input mux power-d own. default setting is 0. clearing this bit will enable v ref , regardless of the state of pr3. pr1 dacs power-down. also powers down th e eq circuitry. default setting is 0. clearing this bit will enable v ref , regardless of the state of pr3. pr2 mixer power-down. default setting is 0. pr3 power down v ref and v refout . default setting is 0. may be used in combination with pr2 or by itself. if all the adcs and dacs are not powered do wn, setting this bit will have no effect on the v ref , and it will only power down v refout . pr4 ac link interface power-down. default setti ng is 0. the reference and the mixer can be either up or down, but all power-up sequences must be allowed to run to comp letion before pr5 and pr4 are both set. in multiple-codec systems, the master codecs pr4 bit controls the slave codec. in the slave codec, the pr4 bit has no effect, except to enable or disable pr5. pr5 internal clocks disabl ed. default setting is 0. pr5 has no effect, unless all adcs, dacs, and the ac link are powered down, e.g., pr0, pr1, pr4. the reference and the mixer can be either up or down, but all power-up sequences must be allo wed to run to completion before pr5 and pr4 are both set. in multiple codec systems, the master codecs pr5 controls the slave codec. pr5 is effective in the slave codec if the master's pr5 bit is clear. pr6 powers down the headphone am plifiers. default setting is 0. eapd external audio power-down control. controls the state of the eapd pin. eapd = 0 sets the eapd pin low, enabling an external power amplif ier. (reset default.) eapd = 1 sets the eapd pin high, shutting the external power amplifier off. normal adcs off pr0 dacs off pr1 analog off pr2 or pr3 digital i/f off pr4 shut off ac-link pr0 = 1 pr1 = 1 pr2 = 1 pr4 = 1 default pr0 = 0 and adc = 1 pr1 = 0 and dac = 1 pr2 = 0 and anl = 1 warm reset cold reset ready = 1 figure 11. example of ac 97 power-down/power-up flow
AD1985 rev. a | page 27 of 48 extended audio id register (index 0x28) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x28 extended audio id aid1 aid0 x x rev1 rev0 amap aidldac aidsdac aidcdac dsa1 dsa0 x aidspdif dra aidvra 0xxbc7 the extended audio id register identifies which extended audio features are supported. a nonzero extended audio id value indica tes that one or more of the extended audio features are supported. aidvra variable rate pcm audio support (read-only). this bit returns a 1 when read to indi cate that variable rate pcm audio is supported. aidspdif spdif support (read-only). this bit returns a 1 when re ad to indicate that spdif tr ansmitter is supported (iec958). dra double rate audio (read-only). this bit returns a 1 when read to indicate that the optional double rate pcm audio is supported for pcm l and pcm r. dac slot assignments. (read/write; reset default = 00.) 00: dacs 1, 2 = 3 and 4 dacs 3, 4 = 7 and 8 dacs 5, 6 = 6 and 9 01: dacs 1, 2 = 7 and 8 dacs 3, 4 = 6 and 9 dacs 5, 6 = 10 and 11 10: dacs 1, 2 = 6 and 9 dacs 3, 4 = 10 and 11 dacs 5, 6 = 3 and 4 dsa[1:0] 11: dacs 1, 2 = 10 and 11 dacs 3, 4 = 3 and 4 dacs 5, 6 = 7 and 8 aidcdac pcm center dac support (read-only). this bit returns a 1 when read to indicate that pcm center dac is supported. aidsdac pcm surround dac support (read-only). this bi t returns a 1 when read to indicate th at pcm surround left and right dacs are supported. aidldac pcm lfe dac support (read-only). this bit returns a 1 when read to indicate th at pcm lfe dac is supported. amap slot dac mappings based on codec id (read-only). this bit retur ns a 1 when read to indicate that slot/dac mappings based on codec id is supported. rev[1:0] rev[1:0] = 10 indicates that codec is ac 97 revision 2.3 compliant (read-only). aid[1:0] indicates codec configuration (read-only). 00 = primary. 01, 10, 11 = secondary.
AD1985 rev. a | page 28 of 48 extended audio status and control register (index 0x2a) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x2a extended audio stat/ctrl vforce x prk prj pri spcv x ascldac ascsdac asccdac spsa1 spsa0 x ascspdf ascdra ascvra 0xxxxx the extended audio status and control register is a read/write register that provides status and control of the extended audio features. ascvra variable rate audio (read/write). ascvra = 0 sets fixed sample rate audio at 48 khz (reset default). ascvra = 1 enables variable rate audio mode (enables sample rate registers and ac 97 slotreq signaling). ascdra double rate audio. ascdra = 1 enables double-rate audio mode in which data fr om pcm l and pcm r in output slots 3 and 4 is used in conjunction with pcm l (n + 1) and pcm r (n + 1) data to provide dac streams at twic e the sample rate designated by the pcm front sample rate control register. when using the double rate audio only th e front dacs are supported, and all othe r dacs (surround, center and lfe) are automatically powered down. note that ascdra can be used without ascvra; in that case, the converter rates are forced to 96 khz if ascdra = 1. ascspdf spdif transmitter subsystem enable/disable bit (read/write). ascspdf = 1 enables the spdif transmitter. ascspdf = 0 disables the spdif transmitter (default). this bit is also used to va lidate that the spdif transmi tter output is actually enabled. the ascspdf bit is only allowed to be set high if the pin 48 (s pdif) is pulled down at power-up, enabling the codec transmitter logic. if the spdif pin is floating or pulled high at power-up, the transmitter logic is disabled, and therefore, the ascspdf b it returns a low, indicating that the spdif tran smitter is not available. this bit must a lways be read back to verify that the spd if transmitter is actually enabled. spsa[1,0] spdif slot assignment bits (read/write). these bits control the spdif slot assignment and respective defa ults, depending on the codec id configuration (see the the ac 97 2.3 amap compliant default spdif slot assignments table for more information.) asccdac center dac status (read-only). asccdac = 1 indicates the pcm center dac is ready. ascsdac surround dac status. (read-only.) ascsdac = 1 indicates the pcm surround dacs are ready. ascldac lfe dac status (read-only). ascldac = 1 indicates the pcm lfe dac is ready. spcv spdif configuration valid (read-only). in dicates the status of the spdif transmi tter subsystem, enabling the driver to determine if the currently programmed spdif configuration is supported. spcv is always valid, independent of the spdif enable bit status. spcv = 0 indicates current spdif configuraton (spsa, spsr, dac slot rate, drs) is not valid (not supported). spcv = 1 indicates current spdif configuration (spsa, spsr, dac slot rate, drs) is valid (is supported). pri center dac power-down (read/write). pri = 1 mutes the pcm center dac. essentially , pri + prk = powered off center/lfe dacs. prj surround dacs power-down (read/write). prj = 1 turns off the pcm surround dacs. prk lfe dac power-down (read/write). prk = 1 mutes the pcm lfe dac. essentially, pri + prk = powered off center/lfe dacs. vforce validity force bit. (reset default = 0). when asserted, this bit forces the spdif st ream validity flag (bit <28> within each spdif l/r subframe) to be controlled by the v bit (d15) in register 0x3a (spdif control register). vforce = 0 and v = 0: the validity bit is ma naged by the codec error detection logic. vforce = 0 and v = 1: the validity bit is forced high, indicating subframe data is invalid. vforce = 1 and v = 0: the validity bit is forced low, indicating subframe data is valid. vforce = 1 and v = 1: the validity bit is forced high, indicating subframe data is invalid.
AD1985 rev. a | page 29 of 48 ac 97 2.3 amap compliant de fault spdif slot assignments codec id function spsa = 00 spsa = 01 spsa = 10 spsa = 11 00 2-ch primary with spdif 3 and 4 7 and 8 [default] 6 and 9 10 and 11 00 4-ch primary with spdif 3 and 4 7 and 8 6 and 9 (default) 10 and 11 00 6-ch primary with spdif 3 and 4 7 and 8 6 and 9 10 and 11 (default) 01 +2-ch secondary with spdif 3 an d 4 7 and 8 6 and 9 (default) 01 +4-ch secondary with spdif 3 and 4 7 and 8 6 and 9 10 and 11 (default) 10 +2-ch secondary with spdif 3 an d 4 7 and 8 6 and 9 (default) 10 +4-ch secondary with spdif 3 and 4 7 and 8 6 and 9 10 and 11 (default) 11 +2-ch secondary with spdif 3 and 4 7 and 8 6 and 9 10 and 11 (default) pcm front dac rate register (index 0x2c) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x2c pcm front dac rate srf15 srf14 srf13 srf12 srf11 srf 10 srf9 srf8 srf7 srf6 srf5 srf4 srf3 srf2 srf1 srf0 0xbb80 this read/write sample rate control register contains 16-bit unsigned values, representing the rate of operation in hz. sr[15:0] sample rate. the sampling frequency range is from 7 khz (0x1 b58) to 48 khz (0xbb80) in 1 hz increments. the codec may not function correctly if frequencies are wr itten outside this range. if 0 is written to ascvra (reg 0x2a), then the sample rate is reset to 48 khz. pcm surround dac rate register (index 0x2e) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x2e pcm surr dac rate srs15 srs14 srs13 srs12 srs11 sr s10 srs9 srs8 srs7 srs6 srs5 srs4 srs3 srs2 srs1 srs0 0xbb80 this read/write sample rate control register contains 16-bit unsigned values, representing the rate of operation in hz. this register sets the sample rate for the surround dac. this registers reset default is to be locked to the pcm front dac sam ple rate register (0x2c). to unlock this register, bit ssru in register 0x76 must be asserted. srs[15:0] sample rate. the sampling frequency rang e is from 7 khz (0x1b58) to 48 khz (0 xbb80) in 1 hz increments. the codec may not function correctly if frequencies are written outside this range. if 0 is written to ascvra (reg 0x2a), then the sample rate is reset to 48 khz. pcm lfe/center dac rate register (index 0x30) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x30 pcm lfe/c dac rate srcl15 srcl14 srcl13 srcl12 srcl11 sr cl10 srcl9 srcl8 srcl7 srcl6 srcl5 src l4 srcl3 srcl2 srcl1 srcl0 0xbb80 this read/write sample rate control register contains 16-bit unsigned values, representing the rate of operation in hz. this register sets the sample rate for the lfe dac and center dac. this registers reset default is to be locked to the pcm fro nt dac sample rate register (0x2c). to unlock this register, bit ssru in register 0x76 must be asserted. srcl[15:0] sample rate. the sampling frequency rang e is from 7 khz (0x1b58) to 48 khz (0 xbb80) in 1 hz increments. the codec may not function correctly if frequencies are written outside this range. if 0 is written to ascvra (register 0x2a), then the sample rate is reset to 48 khz.
AD1985 rev. a | page 30 of 48 pcm l/r adc rate register (index 0x32) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x32 pcm l/r adc rate sra15 sra14 sra13 sra12 sra11 sra10 sra9 sra8 sra7 sra6 sra5 sra4 sra3 sra2 sra1 sra0 0xbb80 this read/write sample rate control register contains 16-bit unsigned values, representing the rate of operation in hz. sr[15:0] sample rate. the sampling frequency range is from 7 khz (0x1 b58) to 48 khz (0xbb80) in 1 hz increments. the codec may not function correctly if frequencies are written outside this range. if 0 is wri tten to ascvra (register 0x2a), then the sample rate is reset to 48 khz. center/lfe volume regi ster (index 0x36) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x36 center/lfe volume lfem x x lfe4 lfe3 lfe2 lfe1 lfe0 cntm x x cnt4 cnt3 cnt2 cnt1 cnt0 0x8080 this register controls the center and lfe output volumes and mute bits. the volume register contains five bits, generating 32 volume levels with increments of 1.5 db each. ac 97 defines the 6-bit volume registers, therefore, to maintain compatibility whenever the d5 or d13 bit is set to 1, its respective lower five volume bits are automatically set to 1 by codec logic. on read-back, all lower five bits will read 1s whenever this bit is set to 1. note that depending on the state of the ac 97nc bit in register 0x76, this register operates as follows: ? for ac'97nc = 0, the register controls the center and lfe output pin attenuators. the range is 0 db to ?46.5 db. ? for ac97nc = 1, the register controls the center and lfe dac gain/attenuators. the range is +12 db to ?34.5 db. cnt[4:0] center volume control. cntm center volume mute. when this bi t is set to 1, the channel is muted. lfe[4:0] lfe volume control. lfem lfe volume mute. when this bit is set to 1, the channel is muted. note that when the cswp bit (register 0x74) is set to 1, the definition of these volume controls for left and right are swapped , i.e., lfe[4:0] and lfem control center volume/mute, and cnt[4:0] and cntm control lfe volume/mute. cnt/lfe[5:0] cntm (lfem) write readback function with ac97nc = 0 function with ac97nc = 1 0 00 0000 00 0000 0 db gain +12 db gain 0 00 1111 00 1111 ?22.5 db gain ?10.5 db gain 0 01 1111 01 1111 ?46.5 db gain ?34.5 db gain 0 1x xxxx 01 1111 ?46.5 db gain indeterminate 1 1 xx xxxx xx xxxx muted, ? db gain muted, ? db gain note: x in the above table is a wild card, meaning the value has no effect. 1 when ac97nc is set to 1, there is only a 5-bit gain/attenuator to control, so the sixth bit, i.e., the msb, does not control a nything. therefore, the 5-bit gain setting is indeterminate, since the five lsbs are listed as wild cards.
AD1985 rev. a | page 31 of 48 surround volume control register (index 0x38) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x38 surround volume lsm x lsr5 lsr4 lsr3 lsr2 lsr1 lsr0 rsm x rsr5 rsr4 rsr3 rsr2 rsr1 rsr0 0x8080 this register controls the surround volume controls for both stereo channels and mute bits. each volume subregister contains five bits, generating 32 volume levels with increments of 1.5 db each. ac 97 defines the 6-bit volume registers, therefore, to maintain compatibility whenever the d5 or d13 bit is set to 1, its respective lower five volume bits are automatically set to 1 by the codec logic. on read-back, all lower five bits will read 1s whenever these bits are set to 1. note that depending on the state of the ac97nc bit in register 0x76, this register operates as follows: ? for ac97nc = 0, the register controls the surround output pin attenuators. the range is 0 db to C46.5 db. ? for ac97nc = 1, the register controls the surround dac gain/attenuators. the range is +12 db to C34.5 db. rsr[5:0] right surround volume control. rsm right surround volume mute. when this bit is set to 1, the right channel is muted. lsr[5:0] left surround volume control. lsm left surround volume mute. when this bit is set to 1, the left channel is muted. xsr[5:0] lsm (rsm) write readback function with ac97nc = 0 function with ac97nc = 1 0 00 0000 00 0000 0 db gain +12 db gain 0 00 1111 00 1111 ?22.5 db gain ?10.5 db gain 0 01 1111 01 1111 ?46.5 db gain ?34.5 gain 0 1x xxxx 01 1111 ?46.5 db gain indeterminate 1 1 xx xxxx xx xxxx muted, ? db gain muted, ? db gain note: x in the above table is a wild card, meaning the value has no effect. 1 when ac97nc is set to 1, there is only a 5-bit gain/attenuator to control, so the sixth bit, i.e., the msb, does not control a nything. therefore, the 5-bit gain setting is indeterminate, since the five lsbs are listed as wild cards.
AD1985 rev. a | page 32 of 48 spdif control register (index 0x3a) reg.num. name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x3a spdif control v x spsr1 spsr0 l cc6 cc5 cc4 cc3 cc2 cc1 cc0 pre copy aud pro 0x2000 register 0x3a is a read/write register that controls spdif functionality and manages bit fields propagated as channel status (or subframe in the v case). with the exception of v, this register should only be written to when the spdif transmitter is disabled (spdif bit in register 0x2a is 0). this ensures that control and status information initialize correctly at the beginning of spdif transmission. pro professional. 1 indicates professional use of channel status, 0 indicates consumer. aud nonaudio. 1 indicates data is nonpcm format, 0 indicates data is pcm. copy copyright. 1 indicates copyright is asse rted, 0 indicates copyright is not asserted. pre pre-emphasis. 1 indicates filter pre-emphasis is 50/15 s, 0 indicates no pre-emphasis. cc[6:0] category code. programmed accordin g to iec standards, or as appropriate. l generation level. programmed according to iec standards, or as appropriate. spsr[1:0] spdif transmit sample rate. spsr[1:0] = 00: transmit sample rate = 44.1 khz. spsr[1:0] = 01: reserved. spsr[1:0] = 10: transmit samp le rate = 48 khz (default). spsr[1:0] = 11: not supported. v validity. this bit affects the validity flag (bit <28> transmitted in each spdif l/r subframe) and enables the spdif transmitter to maintain connectio n during error or mute conditions. v = 1: each spdif subframe (l+r) has bit <28> set to 1. this tags both samples as invalid. v = 0: each spdif subframe (l+r) has bit <28> set to 0 for valid data and 1 for inva lid data (error condition). note that when v = 0, asserting the vforce bit (d15) in register 0x2a (extended audio stat/ctrl) will force the validity flag low, marking both samples as valid.
AD1985 rev. a | page 33 of 48 eq control register (index 0x60) reg.num. name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x60 eq control eqm x x x x x x x sym chs bca5 bca4 bca3 bca2 bca1 bca0 0x8080 register 0x60 is a read/write register that controls equalizer function and data setup. the register also contains the biquad and coefficient address pointer, which is used in conjunction with the eq data register (0x78) to set up the equalizer coefficients. the reset default disables the equalizer function until the coefficients can be properly set up by the software and sets the symmetry bit to allow equal coefficients for left and right channels. bca[5:0] biquad and coefficient address pointer: biquad 0 coef a0 bca[5:0] = 011011 biquad 0 coef a1 bca[5:0] = 011010 biquad 0 coef a2 bca[5:0] = 011001 biquad 0 coef b1 bca[5:0] = 011101 biquad 0 coef b2 bca[5:0] = 011100 biquad 1 coef a0 bca[5:0] = 100000 biquad 1 coef a1 bca[5:0] = 011111 biquad 1 coef a2 bca[5:0] = 011110 biquad 1 coef b1 bca[5:0] = 100010 biquad 1 coef b2 bca[5:0] = 100001 biquad 2 coef a0 bca[5:0] = 100101 biquad 2 coef a1 bca[5:0] = 100100 biquad 2 coef a2 bca[5:0] = 100011 biquad 2 coef b1 bca[5:0] = 100111 biquad 2 coef b2 bca[5:0] = 100110 biquad 3 coef a0 bca[5:0] = 101010 biquad 3 coef a1 bca[5:0] = 101001 biquad 3 coef a2 bca[5:0] = 101000 biquad 3 coef b1 bca[5:0] = 101100 biquad 3 coef b2 bca[5:0] = 101011 biquad 4 coef a0 bca[5:0] = 101111 biquad 4 coef a1 bca[5:0] = 101110 biquad 4 coef a2 bca[5:0] = 101101 biquad 4 coef b1 bca[5:0] = 110001 biquad 4 coef b2 bca[5:0] = 110000 biquad 5 coef a0 bca[5:0] = 110100 biquad 5 coef a1 bca[5:0] = 110011 biquad 5 coef a2 bca[5:0] = 110010 biquad 5 coef b1 bca[5:0] = 110110 biquad 5 coef b2 bca[5:0] = 110101 biquad 6 coef a0 bca[5:0] = 111001 biquad 6 coef a1 bca[5:0] = 111000 biquad 6 coef a2 bca[5:0] = 110111 biquad 6 coef b1 bca[5:0] = 111011 biquad 6 coef b2 bca[5:0] = 111010 chs channel select. chs = 0: selects left channel coefficients da ta block. chs = 1: selects right channel coefficients data blo ck. sym symmetry. when set to 1 this bit indicates that the left and right channel coefficients are equal. this shortens the coefficien t setup sequence since only the left channel coefficients need to be addressed and set up. (the right channel coefficients are simultaneously copied into memory.) eqm equalizer mute. when set to 1, this bit disables the equalizer function (allows all data to pass through). the reset default se ts this bit to 1, disabling the equalizer function until the biquad coefficients can be properly set.
AD1985 rev. a | page 34 of 48 eq data register (index 0x62) reg.num. name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x62 eq data cfd15 cfd14 cfd13 cfd12 cfd11 cfd10 cfd9 cfd8 cfd7 cfd6 cfd5 cfd4 cfd3 cfd2 cfd1 cfd0 0x0000 this read/write register is used to transfer eq biquad coefficients into memory. the register data is transferred to, or retrieved from, the address pointed to by the bca bits in the eq control register (0x60 ). data will be written to memory only if the eqm bit (register 0x60, bit 15) is asserted. cfd[15:0] coefficient data. the biquad coefficients are fixed-point format values with 16 bits of resolution. the cfd15 bit is the msb and the cfd0 bit is the lsb.
AD1985 rev. a | page 35 of 48 jack sense/general register (index 0x70) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x70 j sense/general x x x x x x x x mmdis js2sel x x x x x x n/a js2sel selects js2 input behavior. 0: standard operation for js2 jack sensing (default). 1: enable microphone input sensing with js2. mmdis mono mute disable. 0: use jsmt[2:0] (register 0x72) to de termine mono out ope ration (default). 1: independent of the settings in the jsmt table, this ensures mono out remains active all the time. x reserved, do not read/write these bits.
AD1985 rev. a | page 36 of 48 jack sense/audio interrupt/status register (index 0x72) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x72 jack sense/ audio/status js sprd js1 dmx js0 dmx js mt2 js mt1 js mt0 js1 eqb js0 eqb js1 tmr js0 tmr js1 md js0 md js1 st js0 st js1 int js0 int n/a note: all register bits are read/write exce pt for js0st and js1st, which are read-only. js0int indicates pin js0 has generated an interrupt. remains set until the software services the js0 interrupt; i.e., js0 isr (interru pt service routine) should clear th is bit by writing a 0 to it. notes: 1) interrupts are generated by valid state changes of js pins. 2) interrupt to the system is actually an or combination of this bit and js3int to js0int. 3) the interrupt implementation path is se lected by the ints bit (register 0x74). 4) it is also possible to generate a software system interrupt by writing a 1 to this bit. js1int indicates pin js1 has generated an interrupt. remains set until the software services the js1 interrupt; i.e., js1 isr (interru pt service routine) should clear this bit by writing a 0 to it. see js0int desc ription above for additional details. js0st js0 state. this bit always repo rts the logic state of the js0 pin. cannot be used for mic sensing. js1st js1 state. this bit always repo rts the logic state of the js1 pin. cannot be used for mic sensing. js0md js0 mode. this bit selects the operation mode for the js0 pin. 0: jack sense mode (reset default). 1: interrupt mode. js1md js1 mode: this bit selects the operation mode for the js1 pin. 0: jack sense mode (reset default). 1: interrupt mode. js0tmr js0 timer enable. if this bit is set to a 1, js0 must be high for greater than 298 ms to be recognized. js1tmr js1 timer enable. if this bit is set to a 1, js1 must be high for greater than 298 ms to be recognized. js0eqb js0 eq bypass enable. this bit enables js0 to control the eq bypass. when this bit is set to 1, js0 = 1 will ca use the eq to be bypa ssed. default is 0. js1eqb js1 eq bypass enable. this bit enables js1 to control the eq bypass. when this bit is set to 1, js1 = 1 will ca use the eq to be bypa ssed. default is 0. jsmt[2:0] js mute enable selector. these three bits select and enable th e jack sense muting action (see thejack sense mute select (jsmt [2:0]) table). js0dmx js0 down-mix control enable. this bit enables js0 to control th e down-mix function. this functi on allows a digital mix of 6-channel audio into 2-channel audio. the mix can then be routed to the stereo line_out or hp_out jacks. when this bit is set to 1, js0 = 1 will activate the down-mix conversion. see the dmix description in register 0x76. the dmix bits se lect the down-mix implementati on type and can also force the function to be activated. js1dmx js1 down-mix control enable. this bit enables js1 to cont rol the down-mix function (see the js0dmx description). when this bit is set to 1, js1 = 1 will activate the down-mix conversion. jssprd js spread control enable. this bit enables 2-channel to 6-channel audio spread function when both jack senses are active (logic state 1). note that the sprd bit can also force the spread function without being gated by the jack senses. see this bit s description in register 0x76 for more details on the spread function.
AD1985 rev. a | page 37 of 48 jack sense mute select (jsmt [2:0]) ref js1 js0 jsmt2 jsmt1 jsmt0 hp out line out c/lfe out mono out notes 0 out (0) out (0) 0 0 0 active active active active 1 out (0) in (1) 0 0 0 active active active active 2 in (1) out (0) 0 0 0 active active active active 3 in (1) in (1) 0 0 0 active active active active js0 and js1 ignored. 4 out (0) out (0) 0 0 1 active fmute fmute active 5 out (0) in (1) 0 0 1 active fmute fmute active 6 in (1) out (0) 0 0 1 active active active fmute 7 in (1) in (1) 0 0 1 active active active fmute js0 no mute action, js1 mutes mono and enables line_out + c/lfe. standard 6-channel configuration. swapped hp_out and line_out. 8 out (0) out (0) 0 1 0 fmute active fmute active 9 out (0) in (1) 0 1 0 fmute active fmute active 10 in (1) out (0) 0 1 0 active active active fmute 11 in (1) in (1) 0 1 0 active active active fmute js0 no mute action, js1 mutes mono and enables hp_out and c/lfe. standard 6-channel configuration. no swap. 12 out (0) out (0) 0 1 1 ** ** ** ** 13 out (0) in (1) 0 1 1 ** ** ** ** 14 in (1) out (0) 0 1 1 ** ** ** ** 15 in (1) in (1) 0 1 1 ** ** ** ** ** reserved. 16 out (0) out (0) 1 0 0 active fmute fmute active 17 out (0) in (1) 1 0 0 active active active fmute 18 in (1) out (0) 1 0 0 active fmute fmute fmute 19 in (1) in (1) 1 0 0 active fmute fmute fmute js0 = 0 and js1 = 0 enables mono. js1 = 1 enables front only. js0 = 1 and js1 = 0 enables all rear. 6-channel configuration with front jack wrap-back. 20 out (0) out (0) 1 0 1 fmute fmute fmute active 21 out (0) in (1) 1 0 1 fmute fmute fmute active 22 in (1) out (0) 1 0 1 active active active fmute 23 in (1) in (1) 1 0 1 active active active fmute js0 no mute action, js1 mutes mono and enable s line_out and hp_out and c/lfe. standard 6-channel configuration. swapped hp_out and line_out. 24 out (0) out (0) 1 1 0 ** ** ** ** 25 out (0) in (1) 1 1 0 ** ** ** ** 26 in (1) out (0) 1 1 0 ** ** ** ** 27 in (1) in (1) 1 1 0 ** ** ** ** ** reserved. 28 out (0) out (0) 1 1 1 ** ** ** ** 29 out (0) in (1) 1 1 1 ** ** ** ** 30 in (1) out (0) 1 1 1 ** ** ** ** 31 in (1) in (1) 1 1 1 ** ** ** ** ** reserved. fmute = output is forced to mute independent of the respective volume register setting. active = output is not muted and its status is dependent on the respective volume register setting. out = nothing is plugged into the jack, and therefore, the js status is 0 (via the load resistor pull-down action). in = jack has plug inserted, and therefore, the js status is 1 (via the codec js pin internal pull-up). note: mmdis (register 0x70, bit d7) set to 1 will keep mono out enabled for all the settings above.
AD1985 rev. a | page 38 of 48 serial configuration (index 0x74) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x74 serial config. slot16 regm2 regm1 regm0 regm3 drf oms chen spovr lbks1 lbks0 ints cswp spal spdz splnk 0x1001 note: this register will only reset bits cswp (d3), lbks[1:0] (d[6:5]), and oms (d9) when register 0x00 is written to (soft res et). all bits are reset on a hard or hardware reset. splnk spdif link. this bit enables the spdif to link with the front dacs for data requesting. 0: spdif and front dacs are not linked. 1: spdif and front dacs are linked and rece ive same data reques ts. (reset default.) spdz spdif dacz. 0: repeat last sample out the spdif stre am if fifo underruns . (reset default.) 1: forces midscale sample out the spdif stream if fifo underruns. spal spdif adc loop-around. 0: spdif transmitter is connected to the ac link stream. (reset default.) 1: spdif transmitter is connected to th e digital adc stream, not the ac link. cswp swap the center/lfe channels. some systems have a swappe d external connection for the center and lfe channels. setting this bit will swap these channels inte rnal to the codec. setting this bit also swaps the definitions of the center/lfe volume controls in register 0x36. ints interrupt mode select. this bit sele cts the audio interrupt implementation path. 0: slot 12, bit 0 (modem interrupt) (reset default). 1: slot 6, valid bit (mic adc interrupt). note: this bit does not generate an interrupt. rather, it steer s the path of the generated interrupt. lbks[1:0] loop-back selection. these bits select the internal digital loop-back path when the lpbk bit is active (see register 0x20). 00: loop back through the front dacs. (reset default.) 01: loop back through the surround dacs. 11: loop back through the center and lfe dacs. (center dac l oops back from the adc left channel, the lfe dac from the adc right channel.) 10: reserved. spovr spdif override. 0: spdif transmitter is enabled only if the spdif pin is pulled low on reset. (reset default.) 1: spdif transmitter is enabled regardless of the spdif pin configuration. chen chain enable. this bit enables chaining of a slave codec sdata_in stream into the id0 pin (pin 45). 0: disable chaining. (reset default.) 1: enable chaining into id0 pin. oms output microphone select. this bit will switch the microphon e inputs between the mic1/mic2 pins and the center/lfe pins. this feature is used for those syst ems that have input/output jack sharing. note: see the charts describing the microphone inputsthe de scription of ms (bit d8, register 0x20) and the record select control register (0x1a). 0: microphone inputs come from mic1 and mic2 pins . center/lfe outputs beha ve as expected (default). 1: microphone inputs now come from the center/lfe pins. th e codec will place the center /lfe outputs into a high-z stateequivalent to setting cldis (bit d11, register 0x76) . setting the oms bit, however, will not overwrite the cldis; center/lfe outputs respond equally to both bits. drf dac request force: this allo ws the AD1985 to synchronize dac requests with the ad1981a/ ad1981b. 0: normal dac requesting sequence. (reset default.) 1: synchronize to ad1981a/b dac requests. regm3 slave 3 codec register mask. regm0 master codec register mask. (reset default.) regm1 slave 1 codec register mask. regm2 slave 2 codec register mask. slot16 enable 16-bit slot mode. slot16 makes all ac link slots 16 bits in length, formatte d into 16 slots. this is a preferred mode for dsp serial port interfacing.
AD1985 rev. a | page 39 of 48 miscellaneous control bits (index 0x76) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x76 misc control bits dacz ac97nc msplt lodis cldis hpsel dmix1 dmix0 sprd 2cmic losel sru vrefh vrefd mbg1 mbg0 0x0000 mbg[1:0] mic boost gain select register. these two bits allow changi ng both mic preamp gain blocks from the nominal +20 db gain boost. both mic1 and mic2 preamps will be set to the same selected gain. note: this gain setting only takes effect wh ile bit d6 (m20) on the mic volume register (0x0e) is set to 1, otherwise, the mic boost blocks have a gain of 0 db. 00: +20 db gain. (reset default.) 01: +10 db gain. 10: +30 db gain. 11: reserved. controls the level of the v refout pin as follows: vrefh vrefd 0 0 2.25 v (default) 0 1 high-z 1 0 3.7 v vrefd, vrefh 1 1 0 v sru sample rate unlock. controls all dac sample rate locking. 0: all dac sample rates are locked to the front sample rate . (reset default.) 1: dac sample rates can be set indepe ndently for front, surround, and lfe. losel line_out amplifiers input select. this bit allows the line_out output amplifiers to be driven by the mixer or the surround dacs. the main purpose for this is to allow swapping of the front and surround channels to make better use of the surr/hp_out output amplifiers. this bit should normally be used in tandem with the hpsel bit (see below). 0: line_out amplifiers are driven by the mixer outputs. (reset default.) 1: line_out amplifiers are dri ven by the surround dac outputs. 2cmic 2-channel mic select. this bit enables simultaneous record ing from mic1 and mic2 inputs, using a stereo microphone array. note that this register works in conjunction with the ms and oms bits. 0: mic1 or mic2 (determined by ms bit) is routed to the mixer /record selectors left and righ t mic channels. (reset default.) 1: mic1 is routed to the mixer/record selectors left mic cha nnel, and mic2 is routed to th e mixer/record selectors right mic channel. the ms bit will swap the mic1/mic2 left and right assignments. the oms bit will select from the mic1/mic2 or center/lfe pi ns as the input source for th e microphone. see the record selector (register 0x1a), ms bit (bit d8, register 0x20), or om s bit (bit d9, register 0x74) de finitions for more information. sprd spread enable. this bit enables spreading of 2-channel media to all six output channels. this function is implemented in the analog section by using the output se lector control lines for the center/lfe, surround, and line_out output channels. note that the jack sense pins can also be set up to control (gate) this func tion depending on the js1sprd bit (see register 0x72). 0: no spreading occurs, unless activated by the jack senses an d js1sprd bit. (reset default.) 1: the sprd selector drives the cent er and lfe outputs from th e mono_out, the hpsel selector drives the surr/hp_out outputs from the mixer outp uts, and the losel selector drives the li ne_out outputs from the mixer outputs. note that the sprd bit overrides the current output selector control lines, set up by bi ts losel and hpsel as follows: losel = 0 and hpsel = 1.
AD1985 rev. a | page 40 of 48 dmix[1:0] down-mix mode select. provides analog down-mixing of the center, lfe, and/or surround channels into the mixer channels. this allows the full content of 5.1 or quad medi a to be played through stereo headphones or speakers. note that the jack sense pins can also be set up to control (gate) this function depending on the js0dmx and js1dmx bits (see register 0x72). the upper bit allows forcing the down-mix function: dmix[1] = 0: no down-mix unless activated by the jack senses and jsxdmx bits. (default.) dmix[1] = 1: forces down-mix function. the lower bit selects the down-mix type: dmix[0] = 0: selects 6-to-4 down-mix. the center and lf e channels are summed equally into the mixer l/r channels. (default.) dmix[0] = 1: selects 6-to-2 down-mix. the surround l/r channe ls are summed into the mixer l/r channels. the center and lfe are summed equally into the mixer left and right channels. default for dmix[1:0] is 00. hpsel headphone amplifier input select. this bit allows the headph one power amps to be driven from the surround dacs or from the mixer outputs. there are two reasons for this, on e is to allow 2-channel media to use the higher power headphone amplifiers available on the surr/hp_out outputs, an d the other is to allow spreading of 2-channel media to the surround outputs. together with the losel bit (see above), this bit also provides for analog swapp ing of the mixer (front) and surround outputs. 0: surr_out/hp_out outputs are driven by the surround dacs. (reset default.) 1: surr_out/hp_out outputs are driven by the mixer outputs. cldis center and lfe disable: disables the center and lfe output pins, placing th em into high-z (approximately 30 k? impedance) mode so that the assigned output audio jacks can be shared fo r mic inputs or other functions. 0: center and lfe output pins have normal audio drive capability. (reset default.) 1: center and lfe output pins are placed into high-z mode. lodis line_out disable: disables the line_out pins (l/r), placing them into high-z (approximately 30 k? impedance) mode so that the assigned output audio jack ca n be shared for line input function. 0: line_out pins have normal audio drive capability. (reset default.) 1: line_out pins are placed into high-z mode. msplt mute split. allows separate mute control bits for left and ri ght channels in master, hp, line_in, cd, pcm out, and record volume/gain control registers. 0: both left and right channel mutes are controlled by bit d15 in their re spective registers. (reset default.) 1: bit d15 affects only the left channel mute and bit d7 affects only the right channel mute. ac97nc ac 97 no compatibility mode. changing this bit allows the surround, center, and lfe volume control registers and output attenuators to operate in a more functional mode than whats defined by the ac 97 spec. this is call ed adi compatibility mode. in ac 97 compatibility mode, the dac ga in/attenuators for the surround, center, and lfe are controlled by register 0x18 (pcm volume). the output pin attenuators for the surround are controlled by register 0x38 and the output pin attenuators for the center and lfe are controlled by register 0x36. in adi compatibility mode, the surround dac gain/attenuators are controlled by register 0x38 and the center/lfe dac gain/attenuators are controlled by register 0x36. the outp ut pin attenuators for the surround, center, and lfe are controlled by register 0x02 (master volume). 0: ac 97 compatibility mode. (reset default.) 1: adi compatibility mode. dacz dac zero-fill. determines dac data fill under starved condition. 0: dac data is repeated when dacs are starved for data. (reset default.) 1: dac data is zero-filled when dacs are starved for data.
AD1985 rev. a | page 41 of 48 advanced jack sense register (index 0x78) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x78 advanced jack sense x x x x x x x x js3tmr js2tmr js3m d js2md js3st js2st js3int js2int n/a note: all register bits are read/write except for js2st and js 3st, which are read-only. js2int indicates pin js2 has generated an interrupt . remains set until the software services js2 interrupt, i.e., js2 isr (interrupt service routine) should clear th is bit by writing a 0 to it. notes: 1) interrupts are generated by valid state changes of js pins. 2) interrupt to the system is actually an or combination of this bit and js3int to js0int. 3) the interrupt implementation path is se lected by the ints bit (register 0x74). 4) it is also possible to generate a software system interrupt by writing a 1 to this bit. js3int indicates pin js3 has generated an interrupt . remains set until the software services the js3 interrupt, i.e., js3 isr (interru pt service routine) should clear this bit by writing a 0 to it. see the js2int desc ription above for additional details. js2st js2 state. this bit always re ports the logic state of js2 pin. cannot be used for mic sensing. js3st js3 state. this bit always re ports the logic state of js3 pin. when the voltage reference is set to 3.7 v, the behavior of th is pin is inverted to allow for mic sensing. in this mode, an inserted jack causes the pin to go low. all other voltage refere nce levels enable this pin to be used as a standard sense pin. only js3 can be used for mic sensing. js2md js2 mode. this bit selects the operation mode for the js2 pin. 0: jack sense mode. (reset default.) 1: interrupt mode. js3md js3 mode. this bit selects the operation mode for the js3 pin. 0: jack sense mode. (reset default.) 1: interrupt mode. js2tmr js2 timer enable. if this bit is set to a 1, js2 must be high for greater than 298 ms to be recognized. js3tmr js3 timer enable. if this bit is se t to a 1, js3 must be asserted (see js3st ) for greater than 298 ms to be recognized. x reserved. do not write. vendor id registers (index 0x7c to 0x7e) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x7c vendor id1 vidf7 vidf6 vidf5 vidf4 vidf3 vidf2 vidf1 vidf0 vids7 vids6 vids5 vids4 vids3 vids2 vids1 vids0 0x4144 vids[7:0]: this register is ascii encoded to a. vidf[7:0]: this register is ascii encoded to d. reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x7e vendor id2 vidt7 vidt6 vidt5 vidt4 vidt3 vidt2 vidt1 vidt0 vidrev7 vid rev6 vidrev5 vidrev4 vidrev3 vi drev2 vidrev1 vidrev0 0x5375 vidt[7:0]: this register is ascii encoded to s. vidrev[7:0]: this register is se t to 0x75, identifying the AD1985.
AD1985 rev. a | page 42 of 48 codec class/revision regist er (index 0x60, page 01) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x60 codec class/rev x x x cl4 cl3 cl2 cl1 cl0 rv7 rv6 rv5 rv4 rv3 rv2 rv1 rv0 n/a new register added for ac 97 2.3. rv[7:0] revision id. (read-only.) the initial production version of the AD1985 reports three. this will increment with each stepping/revision of the codec chip. cl[4:0] codec compatibility class. (read-only .) the AD1985 will return 0x00 from these bits. x reserved. pci subsystem vendor id register (index 0x62, page 01) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x62 pci svid pvi15 pvi14 pvi13 pvi12 pvi11 pvi10 pvi9 pvi8 pvi7 pvi6 pvi5 pvi4 pvi3 pvi2 pvi1 pvi0 n/a new register added for ac 97 2.3. pvi[15:0] pci subsystem vendor id. (read/write.) this field provides the pci subsystem vendor id of the audio or modem suba ssembly vendor (i.e., cnr manufacturer, motherboard vendor). this is not the codec vendor pci vendor id, nor the ac 97 controller pci vendor id. if data is not written by bios or other applications, it will return 0xffff. pci subsystem device id register (index 0x64, page 01) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x64 pci sid pi15 pi14 pi13 pi12 pi11 pi10 pi9 pi8 pi7 pi6 pi5 pi4 pi3 pi2 pi1 pi0 n/a new register added for ac 97 2.3. pi[15:0] pci vendor id. (read/write.) this field provides the pci subsystem id of the audio or mo dem subassembly (i.e., cnr model, motherboard sku). this is not the codec vendor pci id, nor the ac 97 controller pci id. information in this field mu st be available for ac 97 controller reads when codec ready is assert ed in ac link. if data is not written by bios or other applications, it will return 0xffff.
AD1985 rev. a | page 43 of 48 function select register (index 0x66, page 01) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x66 function select x x x x x x x x x x x fc3 fc2 fc1 fc0 t/r 0x0000 new register added for ac 97 2.3. function code bits. (default 0.) th ese bits specify the type of audio function described by this page. 0x00: dac 1 (master out). maps to li ne out l/r, codec pins 35 and 36. 0x01: dac 2 (aux out). maps to surro und/hp out l/r, code c pins 39 and 41. 0x02: dac 3 (center/lfe). maps to center/lfe, codec pins 31 and 32. 0x03: s/p-dif out. 0x04: phone in. 0x05: mic 1 (mic select = 0). 0x06: mic 2 (mic select = 1). 0x07: line in. 0x08: cd in. 0x09: video in. 0x0a: aux in. 0x0b: mono out 0x0c to 0x0f: reserved. fc[3:0] these bits are read/write and represent current ac 97 2.3 defi ned i/o capabilities. software will program the corresponding i/o number in this field together with the tip/ring selector bi t t/r. once software programs the value and properly reads it back to confirm selection and implementatio n, it will access the rest of the bit fields in the descriptor. a read-only value of 0 in this register, along with a read-only value of 0 in the iv (register 0x68 page 01) bit, indi cates the codec does not support the information and i/o register. tip or ring selection bit. (default is 0.) this bit sets which jack cond uctor the sense value is measured from. soft ware will program the corresponding ring/tip selector bit together with the i/o number in bits fc[3:0]. on ce software programs the value and properly reads it back to confirm selection and implementatio n, it will access the rest of the bit fields in the descriptor. 0: tip 1: ring t/r mono inputs and outputs report the relevant function and sense in formation when t/r is set to 0 (tip).
AD1985 rev. a | page 44 of 48 information and i/o register (index 0x68, page 01) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x68 function information g4 g3 g21 g1 g0 inv dl4 dl3 dl2 dl1 dl0 iv x x x fip n/a new register added for ac 97 2.3. data read back from this register is invalid unless a function is first specified in register 0x66, page 01. gain bits. (read/write.) the codec updates these bits with the gain value (db relative to level out) in 1.5 dbv increments, not including the volume control gains. for example, if the volume gain is 0 db, then the output pin should be 0 db l evel. any difference is the gain reflected here. when relevant, the bios fu rther updates these bits to take into co nsideration external amplifiers or other external logic that it knows about. g[3:0] indicates the magni tude of the gain. g[4] indicates whether the value is a gain or attenuationessentially a sign bit. g[4:0] gain or attenuation (db relative to level out) 0 0000 0 dbv 0 0001 1.5 dbv 0 1111 24 dbv 1 0001 C1.5 dbv g[4:0] 1 1111 C24 dbv inversion bit. (read/write.) indicates that the co dec presents a 180 phase shift to the signal. 0: no inversion reported. inv 1: inverted. buffer delays (read/write) the default valu e is the delay internal to the codec. th e bios may add to this value the known delays external to the code c, such as for an external amplifier, logic, etc. the codec provides a number representing a delay measuremen t for the input and output cha nnels. software will use this value to accurately calculate the audio st ream position with respect to what has b een reproduced or recorded. these values are in 20.83 s (1/48000 second) units. for output channels, this timing is from th e end of the ac link frame in which the sample is provided, until the time the analog signal appears at the output pin. for input streams, this is from when the analog signal is presented at the pin until t he representative sample is provided on the ac link. analog-to- analog paths are not considered in this measurement. the measurement is a typical measurement, at a 48 kh z sample rate, with minimal in-codec processing. 0x00: information not provided. 0x01 to 0x1e: buffer delay in 20.83 s units. dl[4:0] 0x1f: reserved. iv information valid bit. indicates whether a sensing method is prov ided by the codec and if inform ation field is valid. this fiel d is updated by the codec. 0: (a) after codec reset de-assert ion, it indicates the codec does not provide sensing logic and this bit will be read-only . (b) after a sense cycle is completed, indicates that no information is provided on the sensing method. 1: (a) after codec reset de-assertion, it indicates the co dec provides sensing logic for this i/o and this bit is read/write . (b) after clearing this bit by writing 1, when a sense cycle is completed, indicates that ther e is valid information in the remaining descriptor bits. writin g 0 to this bit has no effect. fip function information present. (read-only.) 1: the g[4:0], inv, and dl[4:0] (in regist er 0x68) bits and the st[2:0] (in register 0x6a) bits are supported and are read/writ e capable. 0: the g[4:0], inv, dl[4:0], and st[2:0] bits are not supported and are read-only with a value of 0.
AD1985 rev. a | page 45 of 48 sense register (index 0x6a, page 01) reg num name d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 default 0x6a sense register st2 st1 st0 s4 s3 s2 s1 s0 or1 or0 sr5 sr4 sr3 sr2 sr1 sr0 n/a new register added for ac 97 2.3. st[2:0] connector/jack location bits. (read/write.) th is field describes the location of the ja ck in the system. this field is updated by the bios. 0x0: rear i/o panel. 0x1: front panel. 0x2: motherboard. 0x3: dock/external. 0x4 to 0x6: reserved. 0x7: no connection/unused i/o. s[4:0] sensed bits relate to the i/o bein g sensed as input or output. (read-only.) sensed bits (when output sense cycle initiated). this field allows for the reporting of the type of output peripheral/device plugged in the jack. values specified below should be interrogated with the sr[5:0] and or[1:0] for accurate reporting. 0x00: data not valid. indicates that the reported value(s) is invalid. 0x01: no connection. indicates that th ere are no connected devices. default. 0x02: fingerprint. indicates a specific fingerprint va lue for devices that are not specified or are unknown. 0x03: speakers (8 ?). 0x04: speakers (4 ?). 0x05: powered speakers. 0x06: stereo headphone. 0x07: spdif out (electrical). 0x08: spdif out (tos). 0x09: mono headset. (mono speaker left channel and mic. re ad functions 5 and 6 for matching microphone.) (not supported.) 0x0a: other. allows a vendor to report sensing other type of de vices/peripherals. sr[5:0] together with or[1:0] provide information regarding the type of device sensed. 0x0b to 0x0e: reserved. 0x0f: unknown (use fingerprint). sensed bits (when input sense cycle initiated) this field allows for the reporting of the type of input peripheral/device plugged in the jack. values specified below should be interrogated with the sr[5:0] and or [1:0] bits for accurate reporting. 0x10: data not valid. indicates that the reported value(s) is invalid. 0x11: no connection. indicates that th ere are no connected devices. default. 0x12: fingerprint. indicates a specific fingerprint value for devices that are not specified. 0x13: microphone (mono). 0x14: stereo microphone. 0x15: stereo line in (ce device attached). 0x16: mono line in (ce device attached) . 0x17: spdif in (electrical). 0x18: spdif in (tos). 0x19: headset. (mono speaker left channel and mic. read function s 0 to 3 for matching dac out.) 0x1a: other. allows a vendor to report sensing other types of devices/peripherals. sr[5:0] together with or[1:0] provide information regarding the type of device sensed. 0x1b to 0x1e: reserved. 0x1f: unknown (use fingerprint).
AD1985 rev. a | page 46 of 48 or[1:0] order bits. these bits indicate the order of magnitude that the sense result bi ts sr[5:0] are using. default is 0x0. 00: 10 0 01: 10 1 10: 10 2 11: 10 3 for example, sr = 1, or = 11; if measuring resistance, result is 1 k?. sr[5:0] sense result bits. (read-only, default 0.) these bits are used to report a vendor spec ific fingerprint or value. (resistance, impedance, reactance, etc.) codec id and clock selection xtl_in id1 id0 codec id codec clocking source gnd 0 0 secondary, id = 3 12.288 mhz (bit_clk from primary codec) gnd 0 1 secondary, id = 2 12.288 mhz (bit_clk from primary codec) gnd 1 0 secondary, id = 1 12.288 mhz (bit_clk from primary codec) xtal 1 1 primary, id = 0 24.576 mhz (local xtal or external clk into xtl_in) clk input 0 0 primary, id = 0 14.31818 mhz (external into xtl_in) clk input 0 1 primary, id = 0 48.000 mhz (external into xtl_in) clk input 1 x reserved reserved note that internally, the id pins have weak pull -ups and are inverted. note also that the clock detection is done per ac 97 rev. 2.3 specification.
AD1985 rev. a | page 47 of 48 outline dimensions top view (pins down) 1 12 13 25 24 36 37 48 0.27 0.22 0.17 0.50 bsc 7.00 bsc sq seating plane 1.60 max 0.75 0.60 0.45 view a 9.00 bsc sq pin 1 0.20 0.09 1.45 1.40 1.35 0.08 max coplanarity view a rotated 90 ccw seating plane 10 6 2 7 3.5 0 0.15 0.05 compliant to jedec standards ms-026bbc figure 12. 48-lead low profile quad flat package [lqfp] (st-48) dimensions shown in millimeters ordering guide model temperature range package description package option AD1985jst 0c to +70c 48-lead lqfp st-48 AD1985jst-reel 0c to +70c 48-lead lqfp st-48 AD1985jstz 1 0c to +70c 48-lead lqfp st-48 AD1985jstz-reel 1 0c to +70c 48-lead lqfp st-48 1 z = pb-free part.
AD1985 rev. a | page 48 of 48 notes ? 2004 analog devices, inc. all rights reserved. trademarks and registered trademarks are the prop erty of their respective owners. c03610C0C3/04(a)

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