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w wm8971l stereo codec for portable audio applications wolfson microelectronics plc www.wolfsonmicro.com advanced information, march 2004, rev 3.0 copyright ? 2004 wolfson microelectronics plc description the wm8971l is a low power, high quality stereo codec designed for portable digital audio applications. the device integrates complete interfaces to stereo or mono microphones and a stereo headphone. external component requirements are drastically reduced as no separate microphone or headphone amplifiers are required. advanced on-chip digital signal processing performs graphic equaliser, and automatic level control for the microphone or line input. the wm8971l can operate as a master or a slave, with various master clock frequencies including 12 or 24mhz for usb devices, or standard 256f s rates like 12.288mhz and 24.576mhz. different audio sample rates such as 96khz, 48khz, 44.1khz are generated directly from the master clock without the need for an external pll. the wm8971l operates at supply voltages down to 1.8v, although the digital core can operate at voltages down to 1.42v to save power, and the maximum for all supplies is 3.6 volts. different sections of the chip can also be powered down under software control. the wm8971l is supplied in a very small and thin 5x5mm qfn package, ideal for use in hand-held and portable systems. features ? dac snr 98db (?a? weighted), thd ?84db at 48khz, 3.3v ? adc snr 95db (?a? weighted), thd -82db at 48khz, 3.3v ? complete stereo / mono microphone interface - programmable alc / noise gate ? on-chip 400mw btl speaker driver (mono) ? on-chip headphone driver - >40mw output power on 16 ? / 3.3v - thd ?80db at 20mw, snr 90db with 16 ? load - no dc blocking capacitors required (capless mode) ? separately mixed mono output ? digital graphic equaliser ? low power - 7mw stereo playback (1.8v / 1.5v supplies) - 14mw record and playback (1.8v / 1.5v supplies) ? low supply voltages - analogue 1.8v to 3.6v - digital core: 1.42v to 3.6v - digital i/o: 1.8v to 3.6v ? 256fs / 384fs or usb master clock rates: 12mhz, 24mhz ? audio sample rates: 8, 11.025, 16, 22.05, 24, 32, 44.1, 48, 88.2, 96khz generated internally from master clock ? 5x5x0.9mm qfn package applications ? digital still cameras ? mp3 player / recorders ? minidisc player / recorders ? portable digital music systems block diagram csb sdin sclk mode vref avdd agnd vmid adcdat adclrc bclk mclk dacdat daclrc hpdetect
wm8971l advanced information w ai rev 3.0 march 2004 2 table of contents description .......................................................................................................1 features.............................................................................................................1 applications .....................................................................................................1 block diagram .................................................................................................1 table of contents .........................................................................................2 pin configuration...........................................................................................4 ordering information ..................................................................................4 pin description ................................................................................................5 absolute maximum ratings.........................................................................6 recommended operation conditions .....................................................6 electrical characteristics ......................................................................7 output pga?s linearity ......................................................................................... 9 headphone output thd versus power......................................................... 10 speaker output thd versus power .............................................................. 11 power consumption ....................................................................................12 signal timing requirements .....................................................................13 system clock timing............................................................................................ 13 audio interface timing ? master mode ......................................................... 13 audio interface timing ? slave mode ............................................................ 14 device description .......................................................................................17 introduction.......................................................................................................... 17 input signal path.................................................................................................. 17 automatic level control (alc) ....................................................................... 23 output signal path.............................................................................................. 26 analogue outputs ............................................................................................... 31 enabling the outputs ......................................................................................... 33 headphone switch ............................................................................................... 33 thermal shutdown .............................................................................................. 35 headphone output ............................................................................................... 35 digital audio interface...................................................................................... 36 audio interface control .................................................................................. 39 clocking and sample rates .............................................................................. 40 control interface .............................................................................................. 42 power supplies ..................................................................................................... 43 power management ............................................................................................. 44 register map...................................................................................................46 digital filter characteristics ...............................................................47 terminology........................................................................................................... 47 dac filter responses ......................................................................................... 48 adc filter responses ......................................................................................... 49 de-emphasis filter responses ........................................................................ 50 highpass filter ..................................................................................................... 51
advanced information wm8971l w ai rev 3.0 march 2004 3 applications information .........................................................................52 recommended external components........................................................... 52 line input configuration................................................................................... 53 microphone input configuration .................................................................. 53 minimising pop noise at the analogue outputs ........................................ 54 power management examples ......................................................................... 54 important notice ..........................................................................................56 address:................................................................................................................... 56
wm8971l advanced information w ai rev 3.0 march 2004 4 pin configuration ordering information order code temperature range package moisture sensitivity level peak soldering temperature wm8971lefl -25 c to +85 c 32-pin qfn (5x5x0.9mm) msl1 260c wm8971lefl/r -25 c to +85 c 32-pin qfn (5x5x0.9mm) (tape and reel) msl1 260c wm8971lgefl -25 c to +85 c 32-pin qfn (5x5x0.9mm) (lead free) msl1 260c wm8971lgefl/r -25 c to +85 c 32-pin qfn (5x5x0.9mm) (lead free, tape and reel) msl1 260c note: reel quantity = 3500
advanced information wm8971l w ai rev 3.0 march 2004 5 pin description pin no name type description 1 mclk digital input master clock 2 dcvdd supply digital core supply 3 dbvdd supply digital buffer (i/o) supply 4 dgnd supply digital ground (return path for both dcvdd and dbvdd) 5 bclk digital input / output audio interface bit clock 6 dacdat digital input dac digital audio data 7 daclrc digital input / output audio interface left / right clock/clock out 8 adcdat digital output adc digital audio data 9 adclrc digital input / output audio interface left / right clock 10 monoout analogue output mono output 11 nc no connect no connect 12 rout1 analogue output right output 1 (line or headphone) 13 lout1 analogue output left output 1 (line or headphone) 14 hpgnd supply supply for analogue output drivers (lout1/2, rout1/2) 15 rout2 analogue output right output 1 (line or headphone or speaker) 16 lout2 analogue output left output 1 (line or headphone or speaker) 17 hpvdd supply supply for analogue output drivers (lout1/2, rout1/2, monout) 18 avdd supply analogue supply 19 agnd supply analogue ground (return path for both avdd and mvdd) 20 vref analogue output reference voltage decoupling capacitor 21 vmid analogue output midrail voltage decoupling capacitor 22 micbias analogue output microphone bias 23 hpdetect analogue input headphone plug-in detection 24 nc no connect no connect 25 mic analogue input single ended microphone input 26 nc no connect no connect 27 rinput1 analogue input right channel input 1 28 linput1 analogue input left channel input 1 29 mode digital input control interface selection 30 csb digital input chip select / device address selection 31 sdin digital input/output control interface data input / 2-wire acknowledge output 32 sclk digital input control interface clock input
wm8971l advanced information w ai rev 3.0 march 2004 6 absolute maximum ratings absolute maximum ratings are stress ratings only. permanent damage to the device may be caused by continuously operating at or beyond these limits. device functional operating limits and guaranteed performance specifications are given under electrical characteristics at the test conditions specified. esd sensitive device. this device is manufactured on a cmos process. it is therefore generically susceptible to damage from excessive static voltages. proper esd precautions must be taken during handling and storage of this device. wolfson tests its package types according to ipc/jedec j-std-020b for moisture sensitivity to determine acceptable storage conditions prior to surface mount assembly. these levels are: msl1 = unlimited floor life at <30 c / 85% relative humidity. not normally stored in moisture barrier bag. msl2 = out of bag storage for 1 year at <30 c / 60% relative humidity. supplied in moisture barrier bag. msl3 = out of bag storage for 168 hours at <30 c / 60% relative humidity. supplied in moisture barrier bag. the moisture sensitivity level for each package type is specified in ordering information. condition min max supply voltages -0.3v +3.63v voltage range digital inputs dgnd -0.3v dbvdd +0.3v voltage range analogue inputs agnd -0.3v avdd +0.3v operating temperature range, t a -25 c +85 c storage temperature after soldering -65 c +150 c notes 1. analogue and digital grounds must always be within 0.3v of each other. 2. all digital and analogue supplies are completely independent from each other. 3. dcvdd must be less than or equal to avdd and dbvdd. recommended operation conditions parameter symbol min typ max unit digital supply range (core) dcvdd 1.42 2.0 3.6 v digital supply range (buffer) dbvdd 1.8 2.0 3.6 v analogue supplies range avdd, hpvdd 1.8 2.0 3.6 v ground dgnd,agnd, hpgnd 0 v
advanced information wm8971l w ai rev 3.0 march 2004 7 electrical characteristics test conditions dcvdd = 1.5v, dbvdd = 3.3v, avdd = hpvdd = 3.3v, t a = +25 o c, 1khz signal, fs = 48khz, pga gain = 0db, 24-bit audio data unless otherwise stated. parameter symbol test conditions min typ max unit analogue inputs (linput1, rinput1, mic) to adc avdd = 3.3v 1.0 full scale input signal level (for adc 0db input at 0db gain) v infs avdd = 1.8v 0.545 v rms l/rinput1 to adc, pga gain = 0db 12 22 l/rinput1 to adc, pga gain = +30db 1 8.2 l/rinput1 to mixer 8.7 input resistance l/rinput1 unused 15 k ? input capacitance 10 pf avdd = 3.3v 80 95 signal to noise ratio (a-weighted) snr avdd = 1.8v 90 db dynamic range -60dbfs 90 95 db -1dbfs input, avdd = 3.3v -82 0.008 -77 0.014 total harmonic distortion thd -1dbfs input, avdd = 1.8v -74 0.02 db % adc channel separation 1khz signal 85 db channel matching 1khz signal 0.2 db analogue outputs (lout1/2, rout1/2, monoout) 0db full scale output voltage avdd/3.3 vrms 1khz, full scale signal 90 mute attenuation monoout pin 81 db channel separation analogue in to analogue out 85 db dac to line-out (l/rout2 with 10k ? ? ? ? / 50pf load) avdd=3.3v 90 98 signal to noise ratio (a-weighted) snr avdd=1.8v 93 db avdd=3.3v -84 total harmonic distortion thd avdd=1.8v -80 db channel separation 1khz signal 100 db
wm8971l advanced information w ai rev 3.0 march 2004 8 test conditions dcvdd = 1.5v, dbvdd = 3.3v, avdd = hpvdd = 3.3v, t a = +25 o c, 1khz signal, fs = 48khz, pga gain = 0db, 24-bit audio data unless otherwise stated. parameter symbol test conditions min typ max unit headphone output (lout1/rout1, using capacitors) output power per channel p o output power is very closely correlated with thd; see below. hpvdd=1.8v, r l =32 ? p o =5mw 0.016 -76 hpvdd=1.8v, r l =16 ? p o =5mw 0.022 -73 hpvdd=3.3v, r l =32 ? , p o =20mw 0.013 -78 total harmonic distortion thd hpvdd=3.3v, r l =16 ? , p o =20mw 0.018 -75 % db hpvdd = 3.3v 92 96 signal to noise ratio (a-weighted) snr hpvdd = 1.8v 96 db speaker output (lout2/rout2 with 8 ? ? ? ? bridge tied load, rout2inv=1) output power at 1% thd p o thd = 1% 400 mw (rms) abs. max power output p o max 500 mw (rms) total harmonic distortion thd po=180mw, r l =8 ? , hpvdd=3.3v -60 0.1 db % signal to noise ratio (a-weighted) snr hpvdd=3.3v, r l =8 ? 95 db analogue reference levels midrail reference voltage vmid ?3% avdd/2 +3% v buffered reference voltage vref ?3% avdd/2 +3% v microphone bias bias voltage v micbias 3ma load current ?5% 0.9 avdd + 5% v bias current source i micbias 3 ma output noise voltage vn 1k to 20khz 15 nv/ hz digital input / output input high level v ih 0.7 dbvdd v input low level v il 0.3 dbvdd v output high level v oh i ol =1ma 0.9 dbvdd v output low level v ol i oh -1ma 0.1 dbvdd v hpdetect (pin 23) input high level v ih 0.7 avdd v input low level v il 0.3 avdd v
advanced information wm8971l w ai rev 3.0 march 2004 9 output pga?s linearity output pga gains -70.000 -60.000 -50.000 -40.000 -30.000 -20.000 -10.000 0.000 10.000 40 50 60 70 80 90 100 110 120 130 xxxvol register setting (binary) measured gain [db] lout1 rout1 lout2 rout2 monoout output pga gain step size 0.000 0.250 0.500 0.750 1.000 1.250 1.500 1.750 2.000 40 50 60 70 80 90 100 110 120 130 xxxvol register setting (binary) step size [db] lout1 rout1 lout2 rout2 monoout
wm8971l advanced information w ai rev 3.0 march 2004 10 headphone output thd versus power headphone pow er vs thd+n (32ohm load) -100 -80 -60 -40 -20 0 0 5 10 15 20 25 30 pow er (m w) thd+n (db ) avdd=1.8v avdd=1.8v, capless avdd=3.3v avdd=3.3v, capless headphone pow er vs thd+n (16ohm load) -100 -80 -60 -40 -20 0 0 102030405060 pow er (m w) thd+n (db ) avdd=1.8v avdd=1.8v, capless avdd=3.3v avdd=3.3v, capless
advanced information wm8971l w ai rev 3.0 march 2004 11 speaker output thd versus power speaker power vs thd+n (8ohm btl load) -70 -60 -50 -40 -30 -20 -10 0 100 200 300 400 500 pow e r (m w) thd+n (db ) avdd=1.8v avdd=2.5v avdd=3.3v speaker power vs thd+n (8 ohm btl load) 0 2 4 6 8 10 0 100 200 300 400 500 power (mw) thd+n (%) avdd=1.8v avdd=2.5v avdd=3.3v
wm8971l advanced information w ai rev 3.0 march 2004 12 power consumption the power consumption of the wm8971l depends on the following factors. ? supply voltages: reducing the supply voltages also reduces supply currents, and therefore results in significant power savings, especially in the digital sections of the wm8971l. ? operating mode: significant power savings can be achieved by always disabling parts of the wm8971l that are not used (e.g. mic pre-amps, unused outputs, dac, adc, etc.) control register r23 other settings tot. power bit vmidsel vref ainl ainr adcl adcr micb dacl dacr lout1 rout1 lout2 rout2 mono adcosr dacosr vsel v i (ma) v i (ma) v i (ma) v i (ma) mw off 00 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 11 clocks stopped 3.3 0.0016 3.3 0.0190 3.3 0.0080 3.3 0.0002 0.0950 01 2.5 0.0008 2.5 0.0170 2.5 0.0050 2.5 0.0000 0.0570 00 1.8 0.0005 1.5 0.0120 1.5 0.0029 1.8 0.0000 0.0233 standby 10 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 11 interface stopped 3.3 0.3900 3.3 0.0390 3.3 0.0080 3.3 0.0000 1.4421 (500 kohm vmid string) 01 2.5 0.2880 2.5 0.0170 2.5 0.0050 2.5 0.0000 0.7750 00 1.8 0.1970 1.5 0.0120 1.5 0.0030 1.8 0.0000 0.3771 playback to line-out 01 1 0 0 0 0 0 1 1 0 0 1 1 0 0 0 11 3.3 3.7310 3.3 5.6600 3.3 0.3000 3.3 0.2370 32.7624 01 2.5 2.6940 2.5 3.8600 2.5 0.2200 2.5 0.2100 17.4600 00 1.8 1.8820 1.5 2.1400 1.5 0.1240 1.8 0.1500 7.0536 playback to line-out 01 1 0 0 0 0 0 1 1 1 1 0 0 0 0 1 11 3.3 3.5170 3.3 4.6470 3.3 0.3000 3.3 0.9500 31.0662 (64x oversampling mode) 01 2.5 2.5760 2.5 3.2030 2.5 0.2200 2.5 0.6480 16.6175 00 1.8 1.7760 1.5 1.7590 1.5 0.1240 1.8 0.4130 6.7647 playback to 16 ohm headphone 01 1 0 0 0 0 0 1 1 1 1 0 0 0 0 0 11 3.3 3.7260 3.3 5.6700 3.3 0.3000 3.3 0.9530 35.1417 01 2.5 2.7530 2.5 3.9250 2.5 0.2200 2.5 0.6570 18.8875 00 1.8 1.8900 1.5 2.1410 1.5 0.1240 1.8 0.4150 7.5465 pla y back to 8 ohm btl speake r 01 1 0 0 0 0 0 1 1 0 0 1 1 0 0 0 11 r24, rout2inv=1 3.3 3.8820 3.3 5.6470 3.3 0.3000 3.3 0.2830 33.3696 01 2.5 2.8780 2.5 3.9390 2.5 0.2200 2.5 0.2100 18.1175 00 1.8 1.9800 1.5 2.1630 1.5 0.1240 1.8 0.1510 7.2663 headphone amp 01 1 0 0 0 0 0 0 0 1 1 0 0 0 0 0 11 clocks stopped 3.3 1.8400 3.3 0.0200 3.3 0.0080 3.3 0.9540 9.3126 (line-in to 16 ohm headphone) 01 2.5 1.3300 2.5 0.0190 2.5 0.0050 2.5 0.6400 4.9850 00 1.8 0.9300 1.5 0.0130 1.5 0.0030 1.8 0.4100 2.4360 speaker amp 01 1 0 0 0 0 0 0 0 0 0 1 1 0 0 0 11 clocks stopped 3.3 1.9780 3.3 0.0200 3.3 0.0080 3.3 0.3310 7.7121 (line-in to 8 ohm speaker) 01 r24, rout2inv=1 2.5 1.4300 2.5 0.0190 2.5 0.0050 2.5 0.2430 4.2425 00 1.8 0.9860 1.5 0.0130 1.5 0.0030 1.8 0.1760 2.1156 phone call 01 1 0 0 0 0 1 0 0 0 0 1 1 1 0 0 11 clocks stopped 3.3 2.5230 3.3 0.0370 3.3 0.0080 3.3 0.4420 9.9330 (mono line-in to headphone, 01 2.5 1.8520 2.5 0.0190 2.5 0.0050 2.5 0.3200 5.4900 mic to monoout) 00 1.8 1.2900 1.5 0.0130 1.5 0.0030 1.8 0.2240 2.7492 record from line-in 01 1 1 1 1 1 0 0 0 0 0 0 0 0 0 0 11 3.3 8.6600 3.3 6.5700 3.3 0.3330 3.3 0.0000 51.3579 01 2.5 7.7100 2.5 4.2800 2.5 0.2320 2.5 0.0000 30.5550 00 1.8 6.8000 1.5 2.2100 1.5 0.1350 1.8 0.0000 15.7575 record from line-in 01 1 1 1 1 1 0 0 0 0 0 0 0 0 1 0 11 3.3 5.0720 3.3 5.9100 3.3 0.3390 3.3 0.0000 37.3593 (64x oversampling mode) 01 2.5 4.2550 2.5 3.7500 2.5 0.2320 2.5 0.0000 20.5925 00 1.8 3.5900 1.5 1.9100 1.5 0.1350 1.8 0.0000 9.5295 record from mono microphone 01 1 1 0 1 0 1 0 0 0 0 0 0 0 0 0 11 3.3 4.9330 3.3 6.5400 3.3 0.3390 3.3 0.0000 38.9796 01 r23, datsel=01 2.5 4.2970 2.5 4.2500 2.5 0.2400 2.5 0.0000 21.9675 00 1.8 3.7210 1.5 2.2200 1.5 0.1370 1.8 0.0000 10.2333 stereo record & playback 01 1 1 1 1 1 1 1 1 0 0 1 1 0 0 0 11 3.3 11.927 3.3 10.870 3.3 0.3320 3.3 0.2820 77.2563 01 2.5 10.112 2.5 7.3600 2.5 0.2340 2.5 0.2060 44.7800 00 1.8 7.3910 1.5 4.0610 1.5 0.1320 1.8 0.1480 19.8597 stereo record & playback 01 1 1 1 1 1 1 1 1 0 0 1 1 0 1 1 11 3.3 8.1090 3.3 9.3300 3.3 0.3330 3.3 0.2820 59.5782 (64x oversampling mode) 01 2.5 6.5500 2.5 6.3020 2.5 0.2340 2.5 0.2070 33.2325 00 1.8 4.7000 1.5 3.3800 1.5 0.1320 1.8 0.1490 13.9962 dbvdd hpvdd r24 r25 (19h) r26 (1ah) avdd dcvdd table 1 supply current consumption notes: 1. all figures are at t a = +25 o c, slave mode, fs = 48khz, mclk = 12.288 mhz (256fs), with zero signal (quiescent) 2. the power dissipated in the headphone or speaker is not included in the above table.
advanced information wm8971l w ai rev 3.0 march 2004 13 signal timing requirements system clock timing mclk t mclkl t mclkh t mclky figure 1 system clock timing requirements test conditions clkdiv2=0 , dcvdd = 1.42v, dbvdd = 3.3v, dgnd = 0v, t a = +25 o c, slave mode fs = 48khz, mclk = 384fs, 24-bit data, unless otherwise stated. parameter symbol min typ max unit system clock timing information mclk system clock pulse width high t mclkl 21 ns mclk system clock pulse width low t mclkh 21 ns mclk system clock cycle time t mclky 54 ns mclk duty cycle t mclkds 60:40 40:60 test conditions clkdiv2=1 , dcvdd = 1.42v, dbvdd = 3.3v, dgnd = 0v, t a = +25 o c, slave mode fs = 48khz, mclk = 384fs, 24-bit data, unless otherwise stated. parameter symbol min typ max unit system clock timing information mclk system clock pulse width high t mclkl 10 ns mclk system clock pulse width low t mclkh 10 ns mclk system clock cycle time t mclky 27 ns audio interface timing ? master mode bclk (output) adcdat adclrc/ daclrc (outputs) t dl dacdat t dda t dht t dst figure 2 digital audio data timing ? master mode (see control interface)
wm8971l advanced information w ai rev 3.0 march 2004 14 test conditions dcvdd = 1.42v, dbvdd = 3.3v, dgnd = 0v, t a = +25 o c, slave mode, fs = 48khz, mclk = 256fs, 24-bit data, unless otherwise stated. parameter symbol min typ max unit bit clock timing information bclk rise time (10pf load) t bclkr 3 ns bclk fall time (10pf load) t bclkf 3 ns bclk duty cycle (normal mode, bclk = mclk/n) t bclkds 50:50 bclk duty cycle (usb mode, bclk = mclk) t bclkds t mclkds audio data input timing information adclrc/daclrc propagation delay from bclk falling edge t dl 10 ns adcdat propagation delay from bclk falling edge t dda 10 ns dacdat setup time to bclk rising edge t dst 10 ns dacdat hold time from bclk rising edge t dht 10 ns audio interface timing ? slave mode bclk daclrc/ adclrc t bch t bcl t bcy dacdat adcdat t lrsu t ds t lrh t dh t dd figure 3 digital audio data timing ? slave mode test conditions dcvdd = 1.42v, dbvdd = 3.3v, dgnd = 0v, t a = +25 o c, slave mode, fs = 48khz, mclk = 256fs, 24-bit data, unless otherwise stated. parameter symbol min typ max unit audio data input timing information bclk cycle time t bcy 50 ns bclk pulse width high t bch 20 ns bclk pulse width low t bcl 20 ns adclrc/daclrc set-up time to bclk rising edge t lrsu 10 ns adclrc/daclrc hold time from bclk rising edge t lrh 10 ns dacdat hold time from bclk rising edge t dh 10 ns adcdat propagation delay from bclk falling edge t dd 10 ns note: bclk period should always be greater than or equal to mclk period.
advanced information wm8971l w ai rev 3.0 march 2004 15 control interface timing ? 3-wire mode csb sclk sdin t csl t dho t dsu t csh t scy t sch t scl t scs lsb t css figure 4 control interface timing ? 3-wire serial control mode test conditions dcvdd = 1.42v, dbvdd = 3.3v, dgnd = 0v, t a = +25 o c, slave mode, fs = 48khz, mclk = 256fs, 24-bit data, unless otherwise stated. parameter symbol min typ max unit program register input information sclk rising edge to csb rising edge t scs 80 ns sclk pulse cycle time t scy 200 ns sclk pulse width low t scl 80 ns sclk pulse width high t sch 80 ns sdin to sclk set-up time t dsu 40 ns sclk to sdin hold time t dho 40 ns csb pulse width low t csl 40 ns csb pulse width high t csh 40 ns csb rising to sclk rising t css 40 ns pulse width of spikes that will be suppressed t ps 5 ns
wm8971l advanced information w ai rev 3.0 march 2004 16 control interface timing ? 2-wire mode sdin sclk t 3 t 1 t 6 t 2 t 7 t 5 t 4 t 3 t 8 t 9 figure 5 control interface timing ? 2-wire serial control mode test conditions dcvdd = 1.42v, dbvdd = 3.3v, dgnd = 0v, t a = +25 o c, slave mode, fs = 48khz, mclk = 256fs, 24-bit data, unless otherwise stated. parameter symbol min typ max unit program register input information sclk frequency 0 400 khz sclk low pulse-width t 1 600 ns sclk high pulse-width t 2 1.3 us hold time (start condition) t 3 600 ns setup time (start condition) t 4 600 ns data setup time t 5 100 ns sdin, sclk rise time t 6 300 ns sdin, sclk fall time t 7 300 ns setup time (stop condition) t 8 600 ns data hold time t 9 900 ns pulse width of spikes that will be suppressed t ps 0 5 ns
advanced information wm8971l w ai rev 3.0 march 2004 17 device description introduction the wm8971l is a low power audio codec offering a combination of high quality audio, advanced features, low power and small size. these characteristics make it ideal for portable digital audio applications such as digital still cameras, mp3 and minidisk player / recorders. the device includes a stereo analogue input which can be switched internally. this input can be used as either a line level input, as a microphone input or be selected as a mono differential input. a mono input is also included which can be configured as a microphone or line input. when recording a programmable gain amplifier with automatic level control (alc) keeps the recording volume constant. the on-chip stereo adc and dac are of a high quality using a multi-bit, low-order oversampling architecture to deliver optimum performance with low power consumption. the dac output signal first enters an analogue mixer where an analogue input and/or the post-alc signal can be added to it. this mix is available on line and headphone outputs. the wm8971l has a configurable digital audio interface where adc data can be read and digital audio playback data fed to the dac. it supports a number of audio data formats including i 2 s, dsp mode (a burst mode in which frame sync plus 2 data packed words are transmitted), msb-first, left justified and msb-first, right justified, and can operate in master or slave modes. the wm8971l uses a unique clocking scheme that can generate many commonly used audio sample rates from either a 12.00mhz usb clock or an industry standard 256/384 f s clock. this feature eliminates the common requirement for an external phase-locked loop (pll) in applications where the master clock is not an integer multiple of the sample rate. sample rates of 8khz, 11.025khz, 12khz, 16khz, 22.05khz, 24khz, 32khz, 44.1khz, 48khz, 88.2khz and 96khz can be generated. the digital filters used for recording and playback are optimised for each sampling rate used. to allow full software control over all its features, the wm8971l offers a choice of 2 or 3 wire mpu control interface. it is fully compatible and an ideal partner for a wide range of industry standard microprocessors, controllers and dsps. the design of the wm8971l has given much attention to power consumption without compromising performance. it operates at very low voltages, and includes the ability to power off parts of the circuitry under software control, including standby and power off modes. input signal path the input signal path for each channel consists of a switch to select between three analogue inputs, followed by a pga (programmable gain amplifier) and an optional microphone gain boost. a differential input of (linput1 ? rinput1) may also be selected. the gain of the pga can be controlled either by the user or by the on-chip alc function (see automatic level control). the signal then enters an adc where it is digitised. alternatively, the two channels can also be mixed in the analogue domain and digitised in one adc while the other adc is switched off. the mono-mix signal appears on both digital output channels. signal inputs the wm8971l has three high impedance, low capacitance ac coupled analogue inputs, linput1, rinput1, and mic. the linsel and rinsel control bits select between them. these inputs can be configured as microphone or line inputs by enabling or disabling the microphone gain boost. a differential input, linput1-rinput1 may also be selected using l/rinsel. the signal inputs are biased internally to the reference voltage vref. whenever the line inputs are muted or the device placed into standby mode, the inputs are kept biased to vref using special anti-thump circuitry. this reduces any audible clicks that may otherwise be heard when changing inputs.
wm8971l advanced information w ai rev 3.0 march 2004 18 register address bit label default description 7:6 linsel 00 left channel input select 00 = linput1 11 = differential r32 (20h) adc signal path control (left) 5:4 lmicboost 00 left channel microphone gain boost 00 = boost off (bypassed) 01 = 13db boost 10 = 20db boost 11 = 29db boost 7:6 rinsel 00 right channel input select 00 = rinput1 01 = mic 11 = differential r33 (21h) adc signal path control (right) 5:4 rmicboost 00 right channel microphone gain boost 00 = boost off (bypassed) 01 = 13db boost 10 = 20db boost 11 = 29db boost table 2 input software control mono mixing the stereo adc can operate as a stereo or mono device, or the two channels can be mixed to mono, either in the analogue domain (i.e. before the adc) or in the digital domain (after the adc). monomix selects the mode of operation. the user also has the flexibility to select the data output from the audio interface using datsel. the default is for left and right channel adc data to be output, but the interface may also be configured so that e.g. left channel adc data is output as both left and right data for when an analogue mono mix is selected. register address bit label default description r31 (1fh) adc input mode 7:6 monomix [1:0] 00 00: stereo 01: analogue mono mix (using left adc) 10: analogue mono mix (using right adc) 11: digital mono mix table 3 mono mixing register address bit label default description r23 (17h) additional control (1) 3:2 datsel [1:0] 00 00: left data=left adc; right data =right adc 01: left data =left adc; right data = left adc 10: left data = right adc; right data =right adc 11: left data = right adc; right data = left adc table 4 adc data output configuration
advanced information wm8971l w ai rev 3.0 march 2004 19 the micbias output provides a low noise reference voltage suitable for biasing electret type microphones and the associated external resistor biasing network. refer to the applications information section for recommended external components. the output can be enabled or disables using the micb control bit (see also the ?power management? section). register address bit label default description r25 (19h) power management (1) 1 micb 0 microphone bias enable 0 = off (high impedance output) 1 = on table 5 microphone bias control the internal micbias circuitry is shown below. note that the is a maximum source current capability for micbias is 3ma. the external biasing resistors therefore must be large enough to limit the micbias current to 3ma. agnd micbias = 1.8 x vmid = 0.9 x avdd vmid internal resistor internal resistor micb figure 6 microphone bias schematic pga control the pga matches the input signal level to the adc input range. the pga gain is logarithmically adjustable from +30db to ?17.25db in 0.75db steps. each pga can be controlled either by the user or by the alc function (see automatic level control). when alc is enabled for one or both channels, then writing to the corresponding pga control register has no effect. the gain is independently adjustable on both right and left line inputs. however, by setting the livu or rivu bits whilst programming the pga gain, both channels are simultaneously updated. this reduces the required number of software writes required. setting the lzcen and rzcen bits enables a zero-cross detector which ensures that pga gain changes only occur when the signal is at zero, eliminating any zipper noise. if zero cross is enabled a timeout is also available to update the gain if a zero cross does not occur. this function may be enabled by setting toen in register r23 (17h).
wm8971l advanced information w ai rev 3.0 march 2004 20 the inputs can also be muted in the analogue domain under software control. the software control registers are shown in table 6 register address bit label default description 8 livu 0 left volume update 0 = store linvol in intermediate latch (no gain change) 1 = update left and right channel gains (left = linvol, right = intermediate latch) 7 linmute 1 left channel input analogue mute 1 = enable mute 0 = disable mute note: livu must be set to un-mute. 6 lzcen 0 left channel zero cross detector 1 = change gain on zero cross only 0 = change gain immediately r0 (00h) left channel pga 5:0 linvol [5:0] 010111 ( 0db ) left channel input volume control 111111 = +30db 111110 = +29.25db . . 0.75db steps down to 000000 = -17.25db 8 rivu 0 right volume update 0 = store rinvol in intermediate latch (no gain change) 1 = update left and right channel gains (right = rinvol, left = intermediate latch) 7 rinmute 1 right channel input analogue mute 1 = enable mute 0 = disable mute note: rivu must be set to un-mute. 6 rzcen 0 right channel zero cross detector 1 = change gain on zero cross only 0 = change gain immediately r1 (01h) right channel pga 5:0 rinvol [5:0] 010111 ( 0db ) right channel input volume control 111111 = +30db 111110 = +29.25db . . 0.75db steps down to 000000 = -17.25db r23 (17h) additional control (1) 0 toen 0 timeout enable 0 : timeout disabled 1 : timeout enabled table 6 input pga software control
advanced information wm8971l w ai rev 3.0 march 2004 21 analogue to digital converter (adc) the wm8971l uses a multi-bit, oversampled sigma-delta adc for each channel. the use of multi-bit feedback and high oversampling rates reduces the effects of jitter and high frequency noise. the adc full scale input level is proportional to avdd. with a 3.3v supply voltage, the full scale level is 1.0 volts r.m.s. any voltage greater than full scale may overload the adc and cause distortion. adc digital filter the adc filters perform true 24 bit signal processing to convert the raw multi-bit oversampled data from the adc to the correct sampling frequency to be output on the digital audio interface. the digital filter path is illustrated in figure 7. from adc digital hpf digital filter to digital audio interface digital decimator adchpd figure 7 adc digital filter the adc digital filters contain a digital high pass filter, selectable via software control. the high-pass filter response is detailed in the digital filter characteristics section. when the high-pass filter is enabled the dc offset is continuously calculated and subtracted from the input signal. by setting hpor, the last calculated dc offset value is stored when the high-pass filter is disabled and will continue to be subtracted from the input signal. if the dc offset is changed, the stored and subtracted value will not change unless the high-pass filter is enabled. this feature can be used for calibration purposes. the output data format can be programmed by the user to accommodate stereo or monophonic recording on both inputs. the polarity of the output signal can also be changed under software control. the software control is shown in table 7. register address bit label default description 6:5 adcpol [1:0] 00 00 = polarity not inverted 01 = l polarity invert 10 = r polarity invert 11 = l and r polarity invert 4 hpor 0 store dc offset when high pass filter disabled 1 = store offset 0 = clear offset r5 (05h) adc and dac control 0 adchpd 0 adc high pass filter enable (digital) 1 = disable high pass filter 0 = enable high pass filter table 7 adc signal path control
wm8971l advanced information w ai rev 3.0 march 2004 22 digital adc volume control the output of the adcs can be digitally amplified or attenuated over a range from ?97db to +30db in 0.5db steps. the volume of each channel can be controlled separately. the gain for a given eight-bit code x is given by: 0.5 (x-195) db for 1 x 255; mute for x = 0 the lavu and ravu control bits control the loading of digital volume control data. when lavu or ravu are set to 0, the ladcvol or radcvol control data will be loaded into the respective control register, but will not actually change the digital gain setting. both left and right gain settings are updated when either lavu or ravu are set to 1. this makes it possible to update the gain of both channels simultaneously. register address bit label default description 7:0 ladcvol [7:0] 11000011 ( 0db ) left adc digital volume control 0000 0000 = digital mute 0000 0001 = -97db 0000 0010 = -96.5db ... 0.5db steps up to 1111 1111 = +30db r21 (15h) left adc digital volume 8 lavu 0 left adc volume update 0 = store ladcvol in intermediate latch (no gain change) 1 = update left and right channel gains (left = ladcvol, right = intermediate latch) 7:0 radcvol [7:0] 11000011 ( 0db ) right adc digital volume control 0000 0000 = digital mute 0000 0001 = -97db 0000 0010 = -96.5db ... 0.5db steps up to 1111 1111 = +30db r22 (16h) right adc digital volume 8 ravu 0 right adc volume update 0 = store radcvol in intermediate latch (no gain change) 1 = update left and right channel gains (left = intermediate latch, right = radcvol) table 8 adc digital volume control
advanced information wm8971l w ai rev 3.0 march 2004 23 automatic level control (alc) the wm8971l has an automatic level control that aims to keep a constant recording volume irrespective of the input signal level. this is achieved by continuously adjusting the pga gain so that the signal level at the adc input remains constant. a digital peak detector monitors the adc output and changes the pga gain if necessary. note that when the alc function is enabled, the settings of registers 0 and 1 (linvol, livu, lizc, rinvol, rivu and rizc, but not linmute and rinmute) are ignored. hold time decay time attack time input signal signal after alc pga gain alc target level figure 8 alc operation the alc function is enabled using the alcsel control bits. when enabled, the recording volume can be programmed between ?6db and ?28.5db (relative to adc full scale) using the alcl register bits. an upper limit for the pga gain can be imposed by setting the maxgain control bits. hld, dcy and atk control the hold, decay and attack times, respectively: hold time is the time delay between the peak level detected being below target and the pga gain beginning to ramp up. it can be programmed in power-of-two (2 n ) steps, e.g. 2.67ms, 5.33ms, 10.67ms etc. up to 43.7s. alternatively, the hold time can also be set to zero. the hold time only applies to gain ramp-up, there is no delay before ramping the gain down when the signal level is above target. decay (gain ramp-up) time is the time that it takes for the pga gain to ramp up across 90% of its range (e.g. from ?15b up to 27.75db). the time it takes for the recording level to return to its target value therefore depends on both the decay time and on the gain adjustment required. if the gain adjustment is small, it will be shorter than the decay time. the decay time can be programmed in power-of-two (2 n ) steps, from 24ms, 48ms, 96ms, etc. to 24.58s. attack (gain ramp-down) time is the time that it takes for the pga gain to ramp down across 90% of its range (e.g. from 27.75db down to -15b gain). the time it takes for the recording level to return to its target value therefore depends on both the attack time and on the gain adjustment required. if the gain adjustment is small, it will be shorter than the attack time. the attack time can be programmed in power-of-two (2 n ) steps, from 6ms, 12ms, 24ms, etc. to 6.14s. when operating in stereo, the peak detector takes the maximum of left and right channel peak values, and any new gain setting is applied to both left and right pgas, so that the stereo image is preserved. however, the alc function can also be enabled on one channel only. in this case, only one pga is controlled by the alc mechanism, while the other channel runs independently with its pga gain set through the control register. when one adc channel is unused, the peak detector disregards that channel. the alc function can also operate when the two adc outputs are mixed to mono in the digital domain, but not if they are mixed to mono in the analogue domain, before entering the adcs.
wm8971l advanced information w ai rev 3.0 march 2004 24 register address bit label default description 8:7 alcsel [1:0] 00 (off) alc function select 00 = alc off (pga gain set by register) 01 = right channel only 10 = left channel only 11 = stereo (pga registers unused) note: ensure that linvol and rinvol settings (reg. 0 and 1) are the same before entering this mode. 6:4 maxgain [2:0] 111 (+30db) set maximum gain of pga 111 : +30db 110 : +24db ?.(-6db steps) 001 : -6db 000 : -12db r17 (11h) alc control 1 3:0 alcl [3:0] 1011 (-12db) alc target ? sets signal level at adc input 0000 = -28.5db fs 0001 = -27.0db fs ? (1.5db steps) 1110 = -7.5db fs 1111 = -6db fs 7 alczc 0 (zero cross off) alc uses zero cross detection circuit. r18 (12h) alc control 2 3:0 hld [3:0] 0000 (0ms) alc hold time before gain is increased. 0000 = 0ms 0001 = 2.67ms 0010 = 5.33ms ? (time doubles with every step) 1111 = 43.691s 7:4 dcy [3:0] 0011 (192ms) alc decay (gain ramp-up) time 0000 = 24ms 0001 = 48ms 0010 = 96ms ? (time doubles with every step) 1010 or higher = 24.58s r19 (13h) alc control 3 3:0 atk [3:0] 0010 (24ms) alc attack (gain ramp-down) time 0000 = 6ms 0001 = 12ms 0010 = 24ms ? (time doubles with every step) 1010 or higher = 6.14s table 9 alc control peak limiter to prevent clipping when a large signal occurs just after a period of quiet, the alc circuit includes a limiter function. if the adc input signal exceeds 87.5% of full scale (?1.16db), the pga gain is ramped down at the maximum attack rate (as when atk = 0000), until the signal level falls below 87.5% of full scale. this function is automatically enabled whenever the alc is enabled. note: if atk = 0000, then the limiter makes no difference to the operation of the alc. it is designed to prevent clipping when long attack times are used.
advanced information wm8971l w ai rev 3.0 march 2004 25 noise gate when the signal is very quiet and consists mainly of noise, the alc function may cause ?noise pumping?, i.e. loud hissing noise during silence periods. the wm8971l has a noise gate function that prevents noise pumping by comparing the signal level at the linput1/2/3 and/or rinput1/2/3 pins against a noise gate threshold, ngth. the noise gate cuts in when: ? signal level at adc [db] < ngth [db] + pga gain [db] + mic boost gain [db] this is equivalent to: ? signal level at input pin [db] < ngth [db] the adc output can then either be muted or digitally attenuated by 18db. alternatively, the pga gain can be held constant (preventing it from ramping up as it normally would when the signal is quiet). the table below summarises the noise gate control register. the ngth control bits set the noise gate threshold with respect to the adc full-scale range. the threshold is adjusted in 1.5db steps. levels at the extremes of the range may cause inappropriate operation, so care should be taken with set?up of the function. note that the noise gate only works in conjunction with the alc function, and always operates on the same channel(s) as the alc (left, right, both, or none). register address bit label default description 7:3 ngth [4:0] 00000 noise gate threshold 00000 -76.5dbfs 00001 -75dbfs ? 1.5 db steps 11110 -31.5dbfs 11111 -30dbfs 2:1 ngg [1:0] 00 noise gate type x0 = pga gain held constant 01 = mute adc output 10 = reserved (do not use this setting) 11 = reserved (do not use this setting) r20 (14h) noise gate control 0 ngat 0 noise gate function enable 1 = enable 0 = disable table 10 noise gate control register address bit label default description 8 adcdiv2 0 adc 6db attenuate enable 0 = disabled (0db) 1 = -6db enabled r5 (05h) adc and dac control 7 dacdiv2 0 dac 6db attenuate enable 0 = disabled (0db) 1 = -6db enabled table 11 adc and dac 6db attenuation select
wm8971l advanced information w ai rev 3.0 march 2004 26 output signal path the wm8971l output signal paths consist of digital filters, dacs, analogue mixers and output drivers. the digital filters and dacs are enabled when the wm8971l is in ?playback only? or ?record and playback? mode. the mixers and output drivers can be separately enabled by individual control bits (see analogue outputs). thus it is possible to utilise the analogue mixing and amplification provided by the wm8971l, irrespective of whether the dacs are running or not. the wm8971l receives digital input data on the dacdat pin. the digital filter block processes the data to provide the following functions: ? digital volume control ? graphic equaliser and dynamic bass boost ? sigma-delta modulation two high performance sigma-delta audio dacs convert the digital data into two analogue signals (left and right). these can then be mixed with analogue signals from the linput1/2/3 and rinput1/2/3 pins, and the mix is fed to the output drivers, lout1/rout1, lout2/rout2 and monoout. ? lout1/rout1: can drive a 16 ? or 32 ? stereo headphone or stereo line output. ? lout2/rout2: can drive a 16 ? or 32 ? stereo headphone or stereo line output, or an 8 ? mono speaker. ? monoout: can drive a mono line output or other load down to 10k ? digital dac volume control the signal volume from each dac can be controlled digitally, in the same way as the adc volume (see digital adc volume control). the gain and attenuation range is ?127db to 0db in 0.5db steps. the level of attenuation for an eight-bit code x is given by: 0.5 (x-255) db for 1 x 255; mute for x = 0 the ldvu and rdvu control bits control the loading of digital volume control data. when ldvu or rdvu are set to 0, the ldacvol or rdacvol control data is loaded into an intermediate register, but the actual gain does not change. both left and right gain settings are updated simultaneously when either ldvu or rdvu are set to 1. register address bit label default description 8 ldvu 0 left dac volume update 0 = store ldacvol in intermediate latch (no gain change) 1 = update left and right channel gains (left = ldacvol, right = intermediate latch) r10 (0ah) left channel digital volume 7:0 ldacvol [7:0] 11111111 ( 0db ) left dac digital volume control 0000 0000 = digital mute 0000 0001 = -127db 0000 0010 = -126.5db ... 0.5db steps up to 1111 1111 = 0db 8 rdvu 0 right dac volume update 0 = store rdacvol in intermediate latch (no gain change) 1 = update left and right channel gains (left = intermediate latch, right = rdacvol) r11 (0bh) right channel digital volume 7:0 rdacvol [7:0] 11111111 ( 0db ) right dac digital volume control similar to ldacvol table 12 digital volume control
advanced information wm8971l w ai rev 3.0 march 2004 27 graphic equaliser the wm8971l has a digital graphic equaliser and adaptive bass boost function. this function operates on digital audio data before it is passed to the audio dacs. bass enhancement can take two different forms: ? linear bass control: bass signals are amplified or attenuated by a user programmable gain. this is independent of signal volume, and very high bass gains on loud signals may lead to signal clipping. ? adaptive bass boost: the bass volume is amplified by a variable gain. when the bass volume is low, it is boosted more than when the bass volume is high. this method is recommended because it prevents clipping, and usually sounds more pleasant to the human ear. treble control applies a user programmable gain, without any adaptive boost function. bass and treble control are completely independent with separately programmable gains and filter characteristics. register address bit label default description 7 bb 0 bass boost 0 = linear bass control 1 = adaptive bass boost 6 bc 0 bass filter characteristic 0 = low cutoff (130hz at 48khz sampling) 1 = high cutoff (200hz at 48khz sampling) bass intensity code bb=0 bb=1 0000 +9db 15 (max) 0001 +9db 14 0010 +7.5db 13 0011 +6db 12 0100 +4.5db 11 0101 +3db 10 0110 +1.5db 9 0111 0db 8 1000 -1.5db 7 1001 -3db 6 1010 -4.5db 5 1011 -6db 4 1100 -6db 3 1101 -6db 2 1110 -6db 1 r12 (0ch) bass control 3:0 bass [3:0] 1111 (disabled) 1111 bypass (off) 6 tc 0 treble filter characteristic 0 = high cutoff (8khz at 48khz sampling) 1 = low cutoff (4khz at 48khz sampling) r13 (0dh) treble control 3:0 trbl [3:0] 1111 (disabled) treble intensity 0000 or 0001 = +9db 0010 = +7.5db ? (1.5db steps) 1011 to 1110 = -6db 1111 = disable table 13 graphic equaliser
wm8971l advanced information w ai rev 3.0 march 2004 28 digital to analogue converter (dac) after passing through the graphic equaliser filters, digital ?de-emphasis? can be applied to the audio data if necessary (e.g. when the data comes from a cd with pre-emphasis used in the recording). de-emphasis filtering is available for sample rates of 48khz, 44.1khz and 32khz. the wm8971l also has a soft mute function, which gradually attenuates the volume of the digital signal to zero. when removed, the gain will ramp back up to the digital gain setting. this function is enabled by default. to play back an audio signal, it must first be disabled by setting the dacmu bit to zero. register address bit label default description 2:1 deemp [1:0] 00 de-emphasis control 11 = 48khz sample rate 10 = 44.1khz sample rate 01 = 32khz sample rate 00 = no de-emphasis r5 (05h) adc and dac control 3 dacmu 1 digital soft mute 1 = mute 0 = no mute (signal active) table 14 dac control the digital audio data is converted to oversampled bit streams in the on-chip, true 24-bit digital interpolation filters. the bitstream data enters two multi-bit, sigma-delta dacs, which convert them to high quality analogue audio signals. the multi-bit dac architecture reduces high frequency noise and sensitivity to clock jitter. it also uses a dynamic element matching technique for high linearity and low distortion. in normal operation, the left and right channel digital audio data is converted to analogue in two separate dacs. however, it is also possible to disable one channel, so that the same signal (left or right) appears on both analogue output channels. additionally, there is a mono-mix mode where the two audio channels are mixed together digitally and then converted to analogue using only one dac, while the other dac is switched off. the mono-mix signal can be selected to appear on both analogue output channels. the dac output defaults to non-inverted. setting dacinv will invert the dac output phase on both left and right channels. register address bit label default description 5:4 dmonomix [1:0] 00 dac mono mix 00: stereo 01: mono ((l+r)/2) into dacl, ?0? into dacr 10: mono ((l+r)/2) into dacr, ?0? into dacl 11: mono ((l+r)/2) into dacl and dacr r23 (17h) additional control (1) 1 dacinv 0 dac phase invert 0 : non-inverted 1 : inverted table 15 dac mono mix and phase invert select
advanced information wm8971l w ai rev 3.0 march 2004 29 output mixers the wm8971l provides the option to mix the dac output signal with analogue line-in signals from the linput1, rinput1, mic pins or a mono differential input (linput1 ? rinput1). the level of the mixed-in signals can be controlled with pgas (programmable gain amplifiers). register address bit label default description r34 (22h) left mixer (1) 2:0 lmixsel 000 left input selection for output mix 000 = linput1 011 = left adc input (after pga / micboost) 100 = differential input r36 (24h) right mixer (1) 2:0 rmixsel 000 right input selection for output mix 000 = rinput1 001 = mic 010 = reserved (do not use) 011 = right adc input (after pga / micboost) 100 = differential input table 16 output mixer signal selection register address bit label default description 8 ld2lo 0 left dac to left mixer 0 = disable (mute) 1 = enable path 7 li2lo 0 lmixsel signal to left mixer 0 = disable (mute) 1 = enable path r34 (22h) left mixer control (1) 6:4 li2lovol [2:0] 101 (-9db) lmixsel signal to left mixer volume 000 = +6db ? (3db steps) 111 = -15db 8 rd2lo 0 right dac to left mixer 0 = disable (mute) 1 = enable path 7 ri2lo 0 rmixsel signal to left mixer 0 = disable (mute) 1 = enable path r35 (23h) left mixer control (2) 6:4 ri2lovol [2:0] 101 (-9db) rmixsel signal to left mixer volume 000 = +6db ? (3db steps) 111 = -15db table 17 left output mixer control
wm8971l advanced information w ai rev 3.0 march 2004 30 register address bit label default description 8 ld2ro 0 left dac to right mixer 0 = disable (mute) 1 = enable path 7 li2ro 0 lmixsel signal to right mixer 0 = disable (mute) 1 = enable path r36 (24h) right mixer control (1) 6:4 li2rovol [2:0] 101 (-9db) lmixsel signal to right mixer volume 000 = +6db ? (3db steps) 111 = -15db 8 rd2ro 0 right dac to right mixer 0 = disable (mute) 1 = enable path 7 ri2ro 0 rmixsel signal to right mixer 0 = disable (mute) 1 = enable path r37 (25h) right mixer control (2) 6:4 ri2rovol [2:0] 101 (-9db) rmixsel signal to right mixer volume 000 = +6db ? (3db steps) 111 = -15db table 18 right output mixer control register address bit label default description 8 ld2mo 0 left dac to mono mixer 0 = disable (mute) 1 = enable path 7 li2mo 0 lmixsel signal to mono mixer 0 = disable (mute) 1 = enable path r38 (26h) mono mixer control (1) 6:4 li2movol [2:0] 101 (-9db) lmixsel signal to mono mixer volume 000 = +6db ? (3db steps) 111 = -15db 8 rd2mo 0 right dac to mono mixer 0 = disable (mute) 1 = enable path 7 ri2mo 0 rmixsel signal to mono mixer 0 = disable (mute) 1 = enable path r39 (27h) mono mixer control (2) 6:4 ri2movol [2:0] 101 (-9db) rmixsel signal to mono mixer volume 000 = +6db ? (3db steps) 111 = -15db table 19 mono output mixer control
advanced information wm8971l w ai rev 3.0 march 2004 31 analogue outputs lout1/rout1 outputs the lout1 and rout1 pins can drive a 16 ? or 32 ? headphone or a line output (see headphone output and line output sections, respectively). the signal volume on lout1 and rout1 can be independently adjusted under software control by writing to lout1vol and rout1vol, respectively. note that gains over 0db may cause clipping if the signal is large. any gain setting below 0101111 (minimum) mutes the output driver. the corresponding output pin remains at the same dc level (the reference voltage on the vref pin), so that no click noise is produced when muting or un-muting. a zero cross detect on the analogue output may also be enabled when changing the gain setting to minimize audible clicks and zipper noise as the gain updates. if zero cross is enabled a timeout is also available to update the gain if a zero cross does not occur. this function may be enabled by setting toen in register r23 (17h). register address bit label default description 8 lo1vu 0 left volume update 0 = store lout1vol in intermediate latch (no gain change) 1 = update left and right channel gains (left = lout1vol, right = intermediate latch) 7 lo1zc 0 left zero cross enable 1 = change gain on zero cross only 0 = change gain immediately r2 (02h) lout1 volume 6:0 lout1vol [6:0] 1111001 (0db) lout1 volume 1111111 = +6db ? (80 steps) 0110000 = -67db 0101111 to 0000000 = analogue mute 8 ro1vu 0 right volume update 0 = store rout1vol in intermediate latch (no gain change) 1 = update left and right channel gains (left = intermediate latch, right = rout1vol) 7 ro1zc 0 right zero cross enable 1 = change gain on zero cross only 0 = change gain immediately r3 (03h) rout1 volume 6:0 rout1vol [6:0] 1111001 rout1 volume similar to lout1vol table 20 lout1/rout1 volume control
wm8971l advanced information w ai rev 3.0 march 2004 32 lout2/rout2 outputs the lout2 and rout2 output pins are essentially similar to lout1 and rout1, but they are independently controlled and can also drive an 8 ? mono speaker (see speaker output section). for speaker drive, the rout2 signal must be inverted (rout2inv = 1), so that the left and right channel are mixed to mono in the speaker [l?(-r) = l+r]. register address bit label default description 6:0 lout2vol [6:0] 1111001 (0db) similar to lout1vol 7 lo2zc 0 left zero cross enable 1 = change gain on zero cross only 0 = change gain immediately r40 (28h) lout2 volume 8 lo2vu 0 same as lo1vu 6:0 rout2vol [6:0] 1111001 (0db) similar rout1vol 7 ro2zc 0 right zero cross enable 1 = change gain on zero cross only 0 = change gain immediately r41 (29h) rout2 volume 8 ro2vu 0 same as ro1vu r24 (18h) additional control (2) 4 rout2inv 0 rout2 invert 0 = no inversion (0 phase shift) 1 = signal inverted (180 phase shift) table 21 lout2/rout2 volume control mono output the monoout pin can drive a mono line output. the signal volume on monoout can be adjusted under software control by writing to monooutvol. register address bit label default description 6:0 monoout vol [6:0] 1111001 (0db) monoout volume 1111111 = +6db ? (80 steps) 0110000 = -67db 0101111 to 0000000 = analogue mute r42 (2ah) monoout volume 7 mozc 0 monoout zero cross enable 1 = change gain on zero cross only 0 = change gain immediately table 22 monoout volume control
advanced information wm8971l w ai rev 3.0 march 2004 33 enabling the outputs each analogue output of the wm8971l can be separately enabled or disabled. the analogue mixer associated with each output is powered on or off along with the output pin. all outputs are disabled by default. to save power, unused outputs should remain disabled. outputs can be enabled at any time, except when vref is disabled (vr=0), as this may cause pop noise (see ?power management? and ?applications information? sections) register address bit label default description 6 lout1 0 lout1 enable 5 rout1 0 rout1 enable 4 lout2 0 lout2 enable 3 rout2 0 rout2 enable r26 (1ah) power management (2) 2 mono 0 monoout enable note: all ?enable? bits are 1 = on, 0 = off table 23 analogue output control whenever an analogue output is disabled, it remains connected to vref (pin 20) through a resistor. this helps to prevent pop noise when the output is re-enabled. the resistance between vref and each output can be controlled using the vroi bit in register 27. the default is low (1.5k ? ), so that any capacitors on the outputs can charge up quickly at start-up. if a high impedance is desired for disabled outputs, vroi can then be set to 1, increasing the resistance to about 40k ? . register address bit label default description r27 (1bh) additional (1) 6 vroi 0 vref to analogue output resistance 0: 1.5 k ? 1: 40 k ? table 24 disabled outputs to vref resistance headphone switch the hpdetect pin can be used as a headphone switch control input to automatically disable the speaker output and enable the headphone output e.g. when a headphone is plugged into a jack socket. in this mode, enabled by setting hpswen, hpdetect switches between headphone and speaker outputs (e.g. when the pin is connected to a mechanical switch in the headphone socket to detect plug-in). the hpswpol bit reverses the pin?s polarity. note that the lout1, rout1, lout2 and rout2 bits in register 26 must also be set for headphone and speaker output (see table 25 and table 26). hpswen hpswpol hpdetect (pin23) l/rout1 (reg. 26) l/rout2 (reg. 26) headphone enabled speaker enabled 0 x x 0 0 no no 0 x x 0 1 no yes 0 x x 1 0 yes no 0 x x 1 1 yes yes 1 0 0 x 0 no no 1 0 0 x 1 no yes 1 0 1 0 x no no 1 0 1 1 x yes no 1 1 0 0 x no no 1 1 0 1 x yes no 1 1 1 x 0 no no 1 1 1 x 1 no yes table 25 headphone switch operation
wm8971l advanced information w ai rev 3.0 march 2004 34 register address bit label default description 6 hpswen 0 headphone switch enable 0 : headphone switch disabled 1 : headphone switch enabled r24 (18h) additional control (2) 5 hpswpol 0 headphone switch polarity 0 : hpdetect high = headphone 1 : hpdetect high = speaker table 26 headphone switch figure 9 example headset detection circuit using normally-open switch figure 10 example headset detection circuit using normally-closed switch
advanced information wm8971l w ai rev 3.0 march 2004 35 thermal shutdown the speaker and headphone outputs can drive very large currents. to protect the wm8971l from overheating a thermal shutdown circuit is included. if the device temperature reaches approximately 150 0 c and the thermal shutdown circuit is enabled (tsden = 1) then the analogue outputs (out1l/r, out2l/r) will be disabled. register address bit label default description r23 (17h) additional control (1) 8 tsden 0 thermal shutdown enable 0 : thermal shutdown disabled 1 : thermal shutdown enabled table 27 thermal shutdown headphone output analogue outputs lout1/rout1 and lout2/rout2, can drive a 16 ? or 32 ? headphone load, through dc blocking capacitors. figure 11 recommended headphone output configuration the dc blocking capacitors and the load resistance together determine the lower cut-off frequency, f c . increasing the capacitance lowers f c , improving the bass response. smaller capacitance values will diminish the bass response. assuming a 16 ohm load and c1, c2 = 220 f: f c = 1 / 2 r l c 1 = 1 / (2 x 16 ? x 220 f) = 45 hz speaker output lout2 and rout2 can differentially drive a mono 8 ? speaker as shown below. figure 12 speaker output connection the right channel is inverted by setting the rout2inv bit, so that the signal across the loudspeaker is the sum of left and right channels. line output the analogue outputs, lout1/rout1 and lout2/rout2, can be used as line outputs. recommended external components are shown below. figure 13 recommended circuit for line output
wm8971l advanced information w ai rev 3.0 march 2004 36 the dc blocking capacitors and the load resistance together determine the lower cut-off frequency, f c . assuming a 10 kohm load and c1, c2 = 1 f: f c = 1 / 2 (r l +r 1 ) c 1 = 1 / (2 x 10.1k ? x 1 f) = 16 hz increasing the capacitance lowers f c , improving the bass response. smaller values of c1 and c2 will diminish the bass response. the function of r1 and r2 is to protect the line outputs from damage when used improperly. digital audio interface the digital audio interface is used for inputting dac data into the wm8971l and outputting adc data from it. it uses five pins: ? adcdat: adc data output ? adclrc: adc data alignment clock ? dacdat: dac data input ? adclrc: dac data alignment clock ? bclk: bit clock, for synchronisation the clock signals bclk, adclrc and daclrc can be outputs when the wm8971l operates as a master, or inputs when it is a slave (see master and slave mode operation, below). four different audio data formats are supported: ? left justified ? right justified ? i 2 s ? dsp mode all four of these modes are msb first. they are described in audio data formats, below. refer to the electrical characteristic section for timing information. master and slave mode operation the wm8971l can be configured as either a master or slave mode device. as a master device the wm8971l generates bclk, adclrc and daclrc and thus controls sequencing of the data transfer on adcdat and dacdat. in slave mode, the wm8971l responds with data to clocks it receives over the digital audio interface. the mode can be selected by writing to the ms bit (see table 23). master and slave modes are illustrated below. figure 14 master mode figure 15 slave mode
advanced information wm8971l w ai rev 3.0 march 2004 37 audio data formats in left justified mode, the msb is available on the first rising edge of bclk following a lrclk transition. the other bits up to the lsb are then transmitted in order. depending on word length, bclk frequency and sample rate, there may be unused bclk cycles before each lrclk transition. left channel right channel daclrc/ adclrc bclk dacdat/ adcdat 1/fs n 3 2 1 n-2 n-1 lsb msb n 3 2 1 n-2 n-1 lsb msb input word length (wl) note: input word length is defined by the wl register. timing is shown with lrp = 1 figure 16 left justified audio interface (assuming n-bit word length) in right justified mode, the lsb is available on the last rising edge of bclk before a lrclk transition. all other bits are transmitted before (msb first). depending on word length, bclk frequency and sample rate, there may be unused bclk cycles after each lrclk transition. left channel right channel daclrc/ adclrc bclk dacdat/ adcdat 1/fs n 3 2 1 n-2 n-1 lsb msb n 3 2 1 n-2 n-1 lsb msb input word length (wl) note: input word length is defined by the wl register. timing is shown with lrp = 1 figure 17 right justified audio interface (assuming n-bit word length) in i 2 s mode, the msb is available on the second rising edge of bclk following a lrclk transition. the other bits up to the lsb are then transmitted in order. depending on word length, bclk frequency and sample rate, there may be unused bclk cycles between the lsb of one sample and the msb of the next. left channel right channel daclrc/ adclrc bclk dacdat/ adcdat 1/fs n 3 2 1 n-2 n-1 lsb msb n 3 2 1 n-2 n-1 lsb msb 1 bclk 1 bclk input word length (wl) note: input word length is defined by the wl register. timing is shown with lrp = 1 figure 18 i 2 s justified audio interface (assuming n-bit word length)
wm8971l advanced information w ai rev 3.0 march 2004 38 in dsp mode, the left channel msb is available on either the 1 st (mode b) or 2 nd (mode a) rising edge of bclk (selectable by lrp) following a rising edge of lrclk. right channel data immediately follows left channel data. depending on word length, bclk frequency and sample rate, there may be unused bclk cycles between the lsb of the right channel data and the next sample. left channel right channel daclrc/ adclrc bclk dacdat/ adcdat n 3 2 1 n-2 n-1 lsb msb n 3 2 1 n-2 n-1 1 bclk input word length (wl) 1/fs figure 19 dsp mode audio interface (mode a, lrp=0) left channel right channel daclrc/ adclrc bclk dacdat/ adcdat n 3 2 1 n-2 n-1 lsb msb n 3 2 1 n-2 n-1 1 bclk input word length (wl) 1/fs figure 20 dsp mode audio interface (mode b, lrp=1)
advanced information wm8971l w ai rev 3.0 march 2004 39 audio interface control the register bits controlling audio format, word length and master / slave mode are summarised in table 28. ms selects audio interface operation in master or slave mode. in master mode bclk, adclrc and daclrc are outputs. the frequency of adclrc and daclrc is set by the sample rate control bits sr[4:0] and usb. in slave mode bclk, adclrc and daclrc are inputs. register address bit label default description 7 bclkinv 0 bclk invert bit (for master and slave modes) 0 = bclk not inverted 1 = bclk inverted 6 ms 0 master / slave mode control 1 = enable master mode 0 = enable slave mode 5 lrswap 0 left/right channel swap 1 = swap left and right dac data in audio interface 0 = output left and right data as normal right, left and i2s modes ? lrclk polarity 1 = invert lrclk polarity 0 = normal lrclk polarity 4 lrp 0 dsp mode ? mode a/b select 1 = msb is available on 1st bclk rising edge after lrc rising edge (mode b) 0 = msb is available on 2nd bclk rising edge after lrc rising edge (mode a) 3:2 wl[1:0] 10 audio data word length 11 = 32 bits (see note) 10 = 24 bits 01 = 20 bits 00 = 16 bits r7 (07h) digital audio interface format 1:0 format[1:0] 10 audio data format select 11 = dsp mode 10 = i 2 s format 01 = left justified 00 = right justified table 28 audio data format control note: right justified mode does not support 32-bit data. audio interface output tristate register bit tri, register 24(18h) bit[3] can be used to tristate the adcdat pin and switch adclrc, daclrc and bclk to inputs. in slave mode (master=0) adclrc, daclrc and bclk are by default configured as inputs and only adcdat will be tri-stated, (see table 29). register address bit label default description r24(18h) additional control (2) 3 tri 0 tristates adcdat and switches adclrc, daclrc and bclk to inputs. 0 = adcdat is an output, adclrc, daclrc and bclk are inputs (slave mode) or outputs (master mode) 1 = adcdate is tristated, adclrc, daclrc and bclk are inputs table 29 tri-stating the audio interface
wm8971l advanced information w ai rev 3.0 march 2004 40 master mode adclrc and daclrc enable in master mode, by default adclrc is disabled when the adc is disabled and daclrc is disabled when the dac is disabled. register bit lrcm, register 24(18h) bit[2] changes the control so that the adclrc and daclrc are disabled only when adc and dac are disabled. this enables the user to use e.g. adclrc for both adc and dac lrclk and disable the adc when dac only operation is required, (see table 30). register address bit label default description r24(18h) additional control (2) 2 lrcm 0 selects disable mode for adclrc and daclrc 0 = adclrc disabled when adc (left and right) disabled, daclrc disabled when dac (left and right) disabled. 1 = adclrc and daclrc disabled only when adc (left and right) and dac (left and right) are disabled. table 30 adclrc/daclrc enable clock output by default adclrc (pin 9) is the adc word clock input/output. under the control of adclrm[1:0], register 27(1bh) bits [8:7] the adclrc pin may be configured as a clock output. if adclrm is 01, 10 or 11 then adclrc pin is always an output even in slave mode or when tri = ?1?, (see table 31). register address bit label default description r27(18h) additional control (3) [8:7] adclrm [1:0] 00 configures adclrc pin 00 = adclrc is adc word clock input (slave mode) or adclrc output (master mode) 01 = adclrc pin is mclk output 10 = adclrc pin is mclk / 5.5 output 11 = adclrc pin is mclk / 6 output table 31 adclrc clock output clocking and sample rates the wm8971l supports a wide range of master clock frequencies on the mclk pin, and can generate many commonly used audio sample rates directly from the master clock. the adc and dac do not need to run at the same sample rate; several different combinations are possible. there are two clocking modes: ? ?normal? mode supports master clocks of 128f s , 192f s , 256f s , 384f s , and their multiples (note: f s refers to the adc or dac sample rate, whichever is faster) ? usb mode supports 12mhz or 24mhz master clocks. this mode is intended for use in systems with a usb interface, and eliminates the need for an external pll to generate another clock frequency for the audio codec. register address bit label default description 6 clkdiv2 0 master clock divide by 2 1 = mclk is divided by 2 0 = mclk is not divided 5:1 sr [4:0] 00000 sample rate control r8 (08h) clocking and sample rate control 0 usb 0 clocking mode select 1 = usb mode 0 = ?normal? mode table 32 clocking and sample rate control the clocking of the wm8971l is controlled using the clkdiv2, usb, and sr control bits. setting the clkdiv2 bit divides mclk by two internally. the usb bit selects between ?normal? and usb mode. each value of sr[4:0] selects one combination of mclk division ratios and hence one combination of sample rates (see next page). since all sample rates are generated by dividing mclk, their
advanced information wm8971l w ai rev 3.0 march 2004 41 accuracy depends on the accuracy of mclk. if mclk changes, the sample rates change proportionately. note that some sample rates (e.g. 44.1khz in usb mode) are approximated, i.e. they differ from their target value by a very small amount. this is not audible, as the maximum deviation is only 0.27% (8.0214khz instead of 8khz in usb mode). by comparison, a half-tone step corresponds to a 5.9% change in pitch. mclk clkdiv2=0 mclk clkdiv2=1 adc sample rate (adclrc) dac sample rate (daclrc) usb sr [4:0] filter type bclk (ms=1) ?normal? clock mode (?*? indicates backward compatibility with wm8731) 8 khz (mclk/1536) 8 khz (mclk/1536) 0 00110 * 1 mclk/4 8 khz (mclk/1536) 48 khz (mclk/256) 0 00100 * 1 mclk/4 12 khz (mclk/1024) 12 khz (mclk/1024) 0 01000 1 mclk/4 16 khz (mclk/768) 16 khz (mclk/768) 0 01010 1 mclk/4 24 khz (mclk/512) 24 khz (mclk/512) 0 11100 1 mclk/4 32 khz (mclk/384) 32 khz (mclk/384) 0 01100 * 1 mclk/4 48 khz (mclk/256) 8 khz (mclk/1536) 0 00010 * 1 mclk/4 48 khz (mclk/256) 48 khz (mclk/256) 0 00000 * 1 mclk/4 12.288 mhz 24.576 mhz 96 khz (mclk/128) 96 khz (mclk/128) 0 01110 * 3 mclk/2 8.0182 khz (mclk/1408) 8.0182 khz (mclk/1408) 0 10110 * 1 mclk/4 8.0182 khz (mclk/1408) 44.1 khz (mclk/256) 0 10100 * 1 mclk/4 11.025 khz (mclk/1024) 11.025 khz (mclk/1024) 0 11000 1 mclk/4 22.05 khz (mclk/512) 22.05 khz (mclk/512) 0 11010 1 mclk/4 44.1 khz (mclk/256) 8.0182 khz (mclk/1408) 0 10010 * 1 mclk/4 44.1 khz (mclk/256) 44.1 khz (mclk/256) 0 10000 * 1 mclk/4 11.2896mhz 22.5792mhz 88.2 khz (mclk/128) 88.2 khz (mclk/128) 0 11110 * 3 mclk/2 8 khz (mclk/2304) 8 khz (mclk/2304) 0 00111 * 1 mclk/6 8 khz (mclk/2304) 48 khz (mclk/384) 0 00101 * 1 mclk/6 12 khz (mclk/1536) 12 khz (mclk/1536) 0 01001 1 mclk/6 16khz (mclk/1152) 16 khz (mclk/1152) 0 01011 1 mclk/6 24khz (mclk/768) 24 khz (mclk/768) 0 11101 1 mclk/6 32 khz (mclk/576) 32 khz (mclk/576) 0 01101 * 1 mclk/6 48 khz (mclk/384) 48 khz (mclk/384) 0 00001 * 1 mclk/6 48 khz (mclk/384) 8 khz (mclk/2304) 0 00011 * 1 mclk/6 18.432mhz 36.864mhz 96 khz (mclk/192) 96 khz (mclk/192) 0 01111 * 3 mclk/3 8.0182 khz (mclk/2112) 8.0182 khz (mclk/2112) 0 10111 * 1 mclk/6 8.0182 khz (mclk/2112) 44.1 khz (mclk/384) 0 10101 * 1 mclk/6 11.025 khz (mclk/1536) 11.025 khz (mclk/1536) 0 11001 1 mclk/6 22.05 khz (mclk/768) 22.05 khz (mclk/768) 0 11011 1 mclk/6 44.1 khz (mclk/384) 8.0182 khz (mclk/2112) 0 10011 * 1 mclk/6 44.1 khz (mclk/384) 44.1 khz (mclk/384) 0 10001 * 1 mclk/6 16.9344mhz 33.8688mhz 88.2 khz (mclk/192) 88.2 khz (mclk/192) 0 11111 * 3 mclk/3 usb mode (?*? indicates backward compatibility with wm8731) 8 khz (mclk/1500) 8 khz (mclk/1500) 1 00110 * 0 mclk 8 khz (mclk/1500) 48 khz (mclk/250) 1 00100 * 0 mclk 8.0214 khz (mclk/1496) 8.0214khz (mclk/1496) 1 10111 * 1 mclk 8.0214 khz (mclk/1496) 44.118 khz (mclk/272) 1 10101 * 1 mclk 11.0259 khz (mclk/1088) 11.0259khz (mclk/1088) 1 11001 1 mclk 12 khz (mclk/1000) 12 khz (mclk/1000) 1 01000 0 mclk 16khz (mclk/750) 16khz (mclk/750) 1 01010 0 mclk 22.0588khz (mclk/544) 22.0588khz (mclk/544) 1 11011 1 mclk 24khz (mclk/500) 24khz (mclk/500) 1 11100 0 mclk 32 khz (mclk/375) 32 khz (mclk/375) 1 01100 * 0 mclk 44.118 khz (mclk/272) 8.0214khz (mclk/1496) 1 10011 * 1 mclk 44.118 khz (mclk/272) 44.118 khz (mclk/272) 1 10001 * 1 mclk 48 khz (mclk/250) 8 khz (mclk/1500) 1 00010 * 0 mclk 48 khz (mclk/250) 48 khz (mclk/250) 1 00000 * 0 mclk 88.235khz (mclk/136) 88.235khz (mclk/136) 1 11111 * 3 mclk 12.000mhz 24.000mhz 96 khz (mclk/125) 96 khz (mclk/125) 1 01110 * 2 mclk table 33 master clock and sample rates
wm8971l advanced information w ai rev 3.0 march 2004 42 control interface selection of control mode the wm8971l is controlled by writing to registers through a serial control interface. a control word consists of 16 bits. the first 7 bits (b15 to b9) are address bits that select which control register is accessed. the remaining 9 bits (b8 to b0) are data bits, corresponding to the 9 bits in each control register. the control interface can operate as either a 3-wire or 2-wire mpu interface. the mode pin selects the interface format. mode interface format low 2 wire high 3 wire table 34 control interface mode selection 3-wire serial control mode in 3-wire mode, every rising edge of sclk clocks in one data bit from the sdin pin. a rising edge on csb latches in a complete control word consisting of the last 16 bits. b15 b14 b13 b12 b11 b10 b9 b8 b7 b6 b5 b4 b3 b2 b1 b0 sdin sclk csb control register address control register data bits latch figure 21 3-wire serial control interface 2-wire serial control mode the wm8971l supports software control via a 2-wire serial bus. many devices can be controlled by the same bus, and each device has a unique 7-bit address (this is not the same as the 7-bit address of each register in the wm8971l). the wm8971l operates as a slave device only. the controller indicates the start of data transfer with a high to low transition on sdin while sclk remains high. this indicates that a device address and data will follow. all devices on the 2-wire bus respond to the start condition and shift in the next eight bits on sdin (7-bit address + read/write bit, msb first). if the device address received matches the address of the wm8971l and the r/w bit is ?0?, indicating a write, then the wm8971l responds by pulling sdin low on the next clock pulse (ack). if the address is not recognised or the r/w bit is ?1?, the wm8971l returns to the idle condition and wait for a new start condition and valid address. once the wm8971l has acknowledged a correct address, the controller sends the first byte of control data (b15 to b8, i.e. the wm8971l register address plus the first bit of register data). the wm8971l then acknowledges the first data byte by pulling sdin low for one clock pulse. the controller then sends the second byte of control data (b7 to b0, i.e. the remaining 8 bits of register data), and the wm8971l acknowledges again by pulling sdin low. the transfer of data is complete when there is a low to high transition on sdin while sclk is high. after receiving a complete address and data sequence the wm8971l returns to the idle state and waits for another start condition. if a start or stop condition is detected out of sequence at any point during data transfer (i.e. sdin changes while sclk is high), the device jumps to the idle condition. sdin sclk register address and 1st register data bit device address (7 bits) rd / wr bit ack (low) control byte 1 (bits 15 to 8) control byte 2 (bits 7 to 0) remaining 8 bits of register data stop start ack (low) ack (low) figure 22 2-wire serial control interface
advanced information wm8971l w ai rev 3.0 march 2004 43 the wm8971l has two possible device addresses, which can be selected using the csb pin. csb state device address low 0011010 (0 x 34h) high 0011011 (0 x 36h) table 35 2-wire mpu interface address selection power supplies the wm8971l can use up to four separate power supplies: ? avdd / agnd: analogue supply, powers all analogue functions except the headphone drivers. avdd can range from 1.8v to 3.6v and has the most significant impact on overall power consumption (except for power consumed in the headphone). a large avdd slightly improves audio quality. ? hpvdd / hpgnd: headphone supply, powers analogue outputs l/rout1, l/rout2 and monoout. hpvdd is normally tied to avdd, but requires separate layout and decoupling capacitors to curb harmonic distortion. if hpvdd is lower than avdd, the output signal may be clipped. ? dcvdd: digital core supply, powers all digital functions except the audio and control interfaces. dcvdd can range from 1.42v to 3.6v, and has no effect on audio quality. the return path for dcvdd is dgnd, which is shared with dbvdd. ? dbvdd: digital buffer supply, powers the audio and control interface buffers. this makes it possible to run the digital core at very low voltages, saving power, while interfacing to other digital devices using a higher voltage. dbvdd draws much less power than dcvdd, and has no effect on audio quality. dbvdd can range from 1.8v to 3.6v. the return path for dbvdd is dgnd, which is shared with dcvdd. it is possible to use the same supply voltage on all four. however, digital and analogue supplies should be routed and decoupled separately to keep digital switching noise out of the analogue signal paths.
wm8971l advanced information w ai rev 3.0 march 2004 44 power management the wm8971l has two control registers that allow users to select which functions are active. for minimum power consumption, unused functions should be disabled. to avoid any pop or click noise, it is important to enable or disable functions in the correct order (see applications information). vmidsel is the enable for the vmid reference, which defaults to disabled and can be enabled as a 50kohm potential divider or, for low power maintenance of vref when all other blocks are disabled, as a 500kohm potential divider. register address bit label defaul t description 8:7 vmidsel 00 vmid divider enable and select 00 ? vmid disabled (for off mode) 01 ? 50kohm divider enabled (for playback/record) 10 ? 500kohm divider enabled (for low-power standby) 11 ? 5kohm divider enabled (for fast start-up) 6 vref 0 vref (necessary for all other functions) 5 ainl 0 analogue in pga left 4 ainr 0 analogue in pga right 3 adcl 0 adc left 2 adcr 0 adc right r25 (19h) power management (1) 1 micb 0 micbias 8 dacl 0 dac left 7 dacr 0 dac right 6 lout1 0 lout1 output buffer* 5 rout1 0 rout1 output buffer* 4 lout2 0 lout2 output buffer* 3 rout2 0 rout2 output buffer* r26 (1ah) power management (2) 2 mono 0 monoout output buffer and mono mixer note: all control bits are 0=off, 1=on * the left mixer is enabled when lout1=1 or lout2=1. the right mixer is enabled when rout1=1 or rout2=1. table 36 power management stopping the master clock in order to minimise power consumed in the digital core of the wm8971l, the master clock should be stopped in standby and off modes. if this is cannot be done externally at the clock source, the digenb bit (r25, bit 0) can be set to stop the mclk signal from propagating into the device core. in standby mode with all supplies at 3.3v, setting digenb saves approximately 0.27ma on dcvdd and 0.2ma on dbvdd. however, since setting digenb has no effect on the power consumption of other system components external to the wm8971l, it is preferable to disable the master clock at its source wherever possible. register address bit label default description r25 (19h) additional control (1) 1 digenb 0 master clock disable 0: master clock enabled 1: master clock disabled table 37 adc and dac oversampling rate selection note: before digenb can be set, the control bits adcl, adcr, dacl and dacr must be set to zero and a waiting time of 1ms must be observed. any failure to follow this procedure may prevent dacs and adcs from re-starting correctly.
advanced information wm8971l w ai rev 3.0 march 2004 45 saving power by reducing oversampling rate the default mode of operation of the adc and dac digital filters is in 128x oversampling mode. under the control of adcosr and dacosr the oversampling rate may be halved. this will result in a slight decrease in noise performance but will also reduce the power consumption of the device. register address bit label default description 1 adcosr 0 adc oversample rate select 1 = 64x (lowest power) 0 = 128x (best snr) r24 (18h) additional control (2) 0 dacosr 0 dac oversample rate select 1 = 64x (lowest power) 0 = 128x (best snr) table 38 adc and dac oversampling rate selection saving power at low supply voltages the analogue supplies to the wm8971l can run from 1.8v to 3.6v. by default, all analogue circuitry on the device is optimized to run at 3.3v. this set-up is also good for all other supply voltages down to 1.8v. however, at lower voltages, it is possible to save power by reducing the internal bias currents used in the analogue circuitry. this is controlled as shown below. register address bit label default description r23 (17h) additional control(1) 7:6 vsel [1:0] 11 analogue bias optimization 00: lowest bias current, optimized for avdd=1.8v 01: low bias current, optimized for avdd=2.5v 1x: default bias current, optimized for avdd=3.3v
wm8971l advanced information w ai rev 3.0 march 2004 46 register map register address (bit 15 ? 9) remarks bit[8] bit[7] bit[6] bit[5] bit[4] bit[3] bit[2] bit[1] bit[0] default page ref r0 (00h) 0000000 left input volume livu linmute lizc linvol 010010111 20 r1 (01h) 0000001 right input volume rivu rinmute rizc rinvol 010010111 20 r2 (02h) 0000010 lout1 volume lo1vu lo1zc lout1vol[6:0] 001111001 31 r3 (03h) 0000011 rout1 volume ro1vu ro1zc rout1vol[6:0] 001111001 31 r4 (04h) 0000100 reserved 0 0 0 0 0 0 0 0 0 000000000 - r5 (05h) 0000101 adc and dac control adcdiv2 dacdiv2 adcpol[1:0] hpor dacmu deemph[1:0] adchpd 000001000 20,25,28 r6 (06h) 0000110 reserved 0 0 0 0 0 0 0 0 0 000000000 - r7 (07h) 0000111 audio interface 0 bclkinv ms lrswap lrp wl[1:0] format[1:0] 000001010 39 r8 (08h) 0001000 sample rate 0 0 clkdiv2 sr[4:0] usb 000000000 40 r9 (09h) 0001001 reserved 0 0 0 0 0 0 0 0 0 000000000 - r10 (0ah) 0001010 left dac volume ldvu ldacvol[7:0] 011111111 26 r11 (0bh) 0001011 right dac volume rdvu rdacvol[7:0] 011111111 26 r12 (0ch) 0001100 bass control 0 bb bc 0 0 bass[3:0] 000001111 27 r13 (0dh) 0001101 treble control 0 0 tc 0 0 trbl[3:0] 000001111 27 r15 (0fh) 0001111 reset writing to this register resets all registers to their default state not reset - r16 (10h) 0010000 reserved 0 0 0 0 0 0 0 0 0 000000000 - r17 (11h) 0010001 alc1 alcsel[1:0] maxgain[2:0] alcl[3:0] 001111011 24 r18 (12h) 0010010 alc2 0 alczc 0 0 0 hld[3:0] 000000000 24 r19 (13h) 0010011 alc3 0 dcy[3:0] atk[3:0] 000110010 24 r20 (14h) 0010100 noise gate 0 ngth[4:0] ngg[1:0] ngat 000000000 25 r21 (15h) 0010101 left adc volume lavu ladcvol[7:0] 011000011 22 r22 (16h) 0010110 right adc volume ravu radcvol[7:0] 011000011 22 r23 (17h) 0010111 additional control(1) tsden vsel[1:0] dmonomix[1:0] datsel[1:0] dacinv toen 011000000 18,20,28,35 r24 (18h) 0011000 additional control(2) 0 0 hpswen hpswpol rout2inv tri lrcm adcosr dacosr 000000000 32, 34,46 r25 (19h) 0011001 pwr mgmt (1) vmidsel[1:0] vref ainl ainr adcl adcr micb digenb 000000000 44 r26 (1ah) 0011010 pwr mgmt (2) dacl dacr lout1 rout1 lout2 rout2 mono 0 0 000000000 44 r27 (1bh) 0011011 additional control (3) adclrm[1:0] vroi 0 0 0 0 0 0 000000000 35 r31 (1fh) 0011111 adc input mode ds monomix[1:0] 0 0 0 0 0 0 000000000 18 r32 (20h) 0100000 adcl signal path 0 linsel[1:0] lmicboost[1:0] 0 0 0 0 000000000 18 r33 (21h) 0100001 adcr signal path 0 rinsel[1:0] rmicboost[1:0] 0 0 0 0 000000000 18 r34 (22h) 0100010 left out mix (1) ld2lo li2lo li2lovol[2:0] 0 lmixsel[2:0] 001010000 29 r35 (23h) 0100011 left out mix (2) rd2lo ri2lo ri2lovol[2:0] 0 0 0 0 001010000 29 r36 (24h) 0100100 right out mix (1) ld2ro li2ro li2rovol[2:0] 0 rmixsel[2:0] 001010000 30 r37 (25h) 0100101 right out mix (2) rd2ro ri2ro ri2rovol[2:0] 0 0 0 0 001010000 30 r38 (26h) 0100110 mono out mix (1) ld2mo li2mo li2movol[2:0] 0 0 0 0 001010000 30 r39 (27h) 0100111 mono out mix (2) rd2mo ri2mo ri2movol[2:0] 0 0 0 0 001010000 30 r40 (28h) 0101000 lout2 volume lo2vu lo2zc lout2vol[6:0] 001111001 32 r41 (29h) 0101001 rout2 volume ro2vu ro2zc rout2vol[6:0] 001111001 32 r42 (2ah) 0101010 monoout volume 0 mozc moutvol[6:0] 001111001 32 note: 1. all unused register bits must be set to ?0? when writing to wm8971l.
advanced information wm8971l w ai rev 3.0 march 2004 47 digital filter characteristics the adc and dac employ different digital filters. there are 4 types of digital filter, called type 0, 1, 2 and 3. the performance of types 0 and 1 is listed in the table below, the responses of all filters is shown in the proceeding pages. parameter test conditions min typ max unit adc filter type 0 (usb mode, 250fs operation) +/- 0.05db 0 0.416fs passband -6db 0.5fs passband ripple +/- 0.05 db stopband 0.584fs stopband attenuation f > 0.584fs -60 db adc filter type 1 (usb mode, 272fs or normal mode operation) +/- 0.05db 0 0.4535fs passband -6db 0.5fs passband ripple +/- 0.05 db stopband 0.5465fs stopband attenuation f > 0.5465fs -60 db -3db 3.7 -0.5db 10.4 high pass filter corner frequency -0.1db 21.6 hz dac filter type 0 (usb mode, 250fs operation) +/- 0.03db 0 0.416fs passband -6db 0.5fs passband ripple +/-0.03 db stopband 0.584fs stopband attenuation f > 0.584fs -50 db dac filter type 1 (usb mode, 272fs or normal mode operation) +/- 0.03db 0 0.4535fs passband -6db 0.5fs passband ripple +/- 0.03 db stopband 0.5465fs stopband attenuation f > 0.5465fs -50 db table 39 digital filter characteristics terminology 1. stop band attenuation (db) ? the degree to which the frequency spectrum is attenuated (outside audio band) 2. pass-band ripple ? any variation of the frequency response in the pass-band region
wm8971l advanced information w ai rev 3.0 march 2004 48 dac filter responses -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.06 -0.05 -0.04 -0.03 -0.02 -0.01 0 0.01 0.02 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5 response (db) frequency (fs) figure 23 dac digital filter frequency response ? type 0 figure 24 dac digital filter ripple ? type 0 -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.06 -0.05 -0.04 -0.03 -0.02 -0.01 0 0.01 0.02 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5 response (db) frequency (fs) figure 25 dac digital filter frequency response ? type 1 figure 26 dac digital filter ripple ? type 1 -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.06 -0.05 -0.04 -0.03 -0.02 -0.01 0 0.01 0.02 0 0.05 0.1 0.15 0.2 0.25 response (db) frequency (fs) figure 27 dac digital filter frequency response ? type 2 figure 28 dac digital filter ripple ? type 2
advanced information wm8971l w ai rev 3.0 march 2004 49 -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.25 -0.2 -0.15 -0.1 -0.05 0 0.05 0.1 0.15 0.2 0.25 0 0.05 0.1 0.15 0.2 0.25 response (db) frequency (fs) figure 29 dac digital filter frequency response ? type 3 figure 30 dac digital filter ripple ? type 3 adc filter responses -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.04 -0.03 -0.02 -0.01 0 0.01 0.02 0.03 0.04 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5 response (db) frequency (fs) figure 31 adc digital filter frequency response ? type 0 figure 32 adc digital filter ripple ? type 0 -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.06 -0.05 -0.04 -0.03 -0.02 -0.01 0 0.01 0.02 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5 response (db) frequency (fs) figure 33 adc digital filter frequency response ? type 1 figure 34 adc digital filter ripple ? type 1
wm8971l advanced information w ai rev 3.0 march 2004 50 -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.25 -0.2 -0.15 -0.1 -0.05 0 0.05 0.1 0.15 0.2 0.25 0 0.05 0.1 0.15 0.2 0.25 response (db) frequency (fs) figure 35 adc digital filter frequency response ? type 2 figure 36 adc digital filter ripple ? type 2 -100 -80 -60 -40 -20 0 0 0.5 1 1.5 2 2.5 3 response (db) frequency (fs) -0.25 -0.2 -0.15 -0.1 -0.05 0 0.05 0.1 0.15 0.2 0.25 0 0.05 0.1 0.15 0.2 0.25 response (db) frequency (fs) figure 37 adc digital filter frequency response ? type 2 figure 38 adc digital filter ripple ? type 3 de-emphasis filter responses -10 -8 -6 -4 -2 0 0 2000 4000 6000 8000 10000 12000 14000 16000 response (db) frequency (fs) -0.4 -0.3 -0.2 -0.1 0 0.1 0.2 0.3 0.4 0 2000 4000 6000 8000 10000 12000 14000 16000 response (db) frequency (fs) figure 39 de-emphasis frequency response (32khz) figure 40 de-emphasis error (32khz)
advanced information wm8971l w ai rev 3.0 march 2004 51 -10 -8 -6 -4 -2 0 0 5000 10000 15000 20000 response (db) frequency (fs) -0.4 -0.3 -0.2 -0.1 0 0.1 0.2 0.3 0.4 0 5000 10000 15000 20000 response (db) frequency (fs) figure 41 de-emphasis frequency response (44.1khz) figure 42 de-emphasis error (44.1khz) -10 -8 -6 -4 -2 0 0 5000 10000 15000 20000 response (db) frequency (fs) -0.4 -0.3 -0.2 -0.1 0 0.1 0.2 0.3 0.4 0 5000 10000 15000 20000 response (db) frequency (fs) figure 43 de-emphasis frequency response (48khz) figure 44 de-emphasis error (48khz) highpass filter the wm8971l has a selectable digital highpass filter in the adc filter path to remove dc offsets. the filter response is characterised by the following polynomial: figure 45 adc highpass filter response 1 - z -1 1 - 0.9995z -1 h(z) = -15 -10 -5 0 0 0.0005 0.001 0.0015 0.002 response (db) frequency (fs)
wm8971l advanced information w ai rev 3.0 march 2004 52 applications information recommended external components figure 46 recommended external components diagram
advanced information wm8971l w ai rev 3.0 march 2004 53 component reference suggested value description c1 ? c4 100nf de-coupling for dbvdd, dcvdd, avdd, hpvdd c5 ? c6 10uf reservoir capacitor for dvdd, avdd. should the supplies use separate sources then additional capacitors will be required of each additional source. c7 ? c9 1uf ac input coupling capacitors c13 2.2uf output ac coupling capacitors to remove dc level from monoout c14 and c15 220uf output ac coupling capacitors to remove dc level from headphone output (if used as a line-out only, use 2.2uf caps) c16 100nf de-coupling for vmid. c17 10uf reservoir capacitor for vmid c18 100nf de-coupling for vref c19 10uf reservoir capacitor for vref c20 100nf de-coupling for micbias ? not required if micbias output is not used c21 10uf reservoir capacitor for micbias ? not required if micbias output is not used table 40 external components descriptions note: 1. for capacitors c5, c6, c17, c19 and c21 it is recommended that very low esr components are used. line input configuration when linput1/rinput1 or mic are used as line inputs, the microphone boost and alc functions should normally be disabled. in order to avoid clipping, the user must ensure that the input signal does not exceed avdd. this may require a potential divider circuit in some applications. it is also recommended to remove rf interference picked up on any cables using a simple first-order rc filter, as high-frequency components in the input signal may otherwise cause aliasing distortion in the audio band. ac signals with no dc bias should be fed to the wm8971l through a dc blocking capacitor, e.g. 1 f. microphone input configuration figure 47 recommended circuit for line input for interfacing to a microphone, the alc function should be enabled and the microphone boost switched on. microphones held close to a speaker?s mouth would normally use the 13db gain setting, while tabletop or room microphones would need a 29db boost. the recommended application circuit is shown above. r1 and r2 form part of the biasing network (refer to microphone bias section). r1 connected to micbias is necessary only for electret type microphones that require a voltage bias. r2 should always be present to prevent the microphone r2 47kohm c1 220pf c2 1uf agnd agnd agnd l/rinput1 mic from microphone r1 680 ohm to 2.2kohm check microphone's specification micbias
wm8971l advanced information w ai rev 3.0 march 2004 54 input from charging to a high voltage which may damage the microphone on connection. r1 and r2 should be large so as not to attenuate the signal from the microphone, which can have source impedance greater than 2kohm. c1 together with the source impedance of the microphone and the wm8971l input impedance forms an rf filter. c2 is a dc blocking capacitor to allow the microphone to be biased at a different dc voltage to the micin signal. minimising pop noise at the analogue outputs to minimise any pop or click noise when the system is powered up or down, the following procedures are recommended. power up ? switch on power supplies. by default the wm8971l is in standby mode, the dac is digitally muted and the audio interface, line outputs and headphone outputs are all off (dacmu = 1 power management registers 1 and 2 are all zeros). ? enable vmid and vref, then wait for time tbd ? enable dacs as required ? enable line and / or headphone output buffers as required. ? set dacmu = 0 to soft-un-mute the audio dacs. power down ? set dacmu = 1 to soft-mute the audio dacs. ? disable all output buffers, then wait for time tbd. ? switch off the power supplies. power management examples power management (1) power management (2) pgas adcs dacs output buffers operation mode vref ainl/r pgl pgr adl adr mbi dal dar lo1 ro1 lo2 ro2 mo stereo headphone playback 1 1 0 0 0 0 0 1 1 1 1 0 0 0 stereo line-in record 1 1 1 1 1 1 0 0 0 0 0 0 0 0 stereo microphone record 1 1 1 1 1 1 1 0 0 0 0 0 0 0 mono microphone record 1 1 1 0 1 0 1 0 0 0 0 0 0 0 stereo line-in to headphone out 1 1 0 0 0 0 0 0 0 1 1 0 0 0 phone call 1 1 1 0 0 0 1 0 0 1 1 0 0 1 speaker phone call [rout2inv = 1] 1 1 1 0 0 0 1 0 0 0 0 1 1 1 record phone call [l channel = mic with boost, r channel = rx, enable mono mix] 1 1 1 1 1 1 1 0 0 1 1 0 0 1 table 41 register settings for power management
advanced information wm8971l w ai rev 3.0 march 2004 55 package dimensions dm030.c fl: 32 pin qfn plastic package 5 x 5 x 0.9 mm body, 0.50 mm lead pitch notes: 1. dimension b applied to metallized terminal and is measured between 0.25 mm and 0.30 mm from terminal tip. dimension l1 repr esents terminal pull back from package side wall. maximum of 0.1mm is acceptable. where terminal pull back exists, only upper half of lead is visible on package side wall due to half etching of leadframe. 2. falls within jedec, mo-220 with the exception of d2, e2: d2,e2: larger pad size chosen which is just outside jedec specification 3. all dimensions are in millimetres 4. this drawing is subject to change without notice. 5. shape and size of corner tie bar may vary with package terminal count. corner tie bar is connected to exposed pad internall y see detail b e2 e2/2 b b 16 15 a 8 9 e 5 corner tie bar c 0.08 c ccc a a1 c (a3) seating plane 1 b d2 l d2/2 see detail a index area (d/2 x e/2) top view d c aaa 2 x c aaa 2 x e detail b terminal tip r datum e e/2 l1 1 detail a b c bbb m a 32x b l 32x k l1 r 1 1 0. 5 66 mm 0.43 mm 5 corner tie bar symbols dimensions (mm) min nom max note a a1 a3 b d d2 e e2 e l l1 r 0.85 0.90 1.00 0.05 0.02 0 0.2 ref 0.30 0.23 0.18 5.00 3.4 3.3 3.2 0.5 bsc 0.35 0.4 0.45 0.1 b(min)/2 1 2 2 1 k 0.20 aaa bbb ccc ref: 0.15 0.10 0.10 jedec, mo-220, variation vkkd-2 tolerances of form and position 4.90 5.10 5.00 4.90 5.10 3.4 3.3 3.2 exposed centre pad 1 17 24 25 32
wm8971l advanced information w ai rev 3.0 march 2004 56 important notice wolfson microelectronics plc (wm) reserve the right to make changes to their products or to discontinue any product or service without notice, and advise customers to obtain the latest version of relevant information to verify, before placing orders, that information being relied on is current. all products are sold subject to the wm terms and conditions of sale supplied at the time of order acknowledgement, including those pertaining to warranty, patent infringement, and limitation of liability. wm warrants performance of its products to the specifications applicable at the time of sale in accordance with wm?s standard warranty. testing and other quality control techniques are utilised to the extent wm deems necessary to support this warranty. specific testing of all parameters of each device is not necessarily performed, except those mandated by government requirements. in order to minimise risks associated with customer applications, adequate design and operating safeguards must be used by the customer to minimise inherent or procedural hazards. wolfson products are not authorised for use as critical components in life support devices or systems without the express written approval of an officer of the company. life support devices or systems are devices or systems that are intended for surgical implant into the body, or support or sustain life, and whose failure to perform when properly used in accordance with instructions for use provided, can be reasonably expected to result in a significant injury to the user. a critical component is any component of a life support device or system whose failure to perform can be reasonably expected to cause the failure of the life support device or system, or to affect its safety or effectiveness. wm assumes no liability for applications assistance or customer product design. wm does not warrant or represent that any license, either express or implied, is granted under any patent right, copyright, mask work right, or other intellectual property right of wm covering or relating to any combination, machine, or process in which such products or services might be or are used. wm?s publication of information regarding any third party?s products or services does not constitute wm?s approval, license, warranty or endorsement thereof. reproduction of information from the wm web site or datasheets is permissible only if reproduction is without alteration and is accompanied by all associated warranties, conditions, limitations and notices. representation or reproduction of this information with alteration voids all warranties provided for an associated wm product or service, is an unfair and deceptive business practice, and wm is not responsible nor liable for any such use. resale of wm?s products or services with statements different from or beyond the parameters stated by wm for that product or service voids all express and any implied warranties for the associated wm product or service, is an unfair and deceptive business practice, and wm is not responsible nor liable for any such use. address: wolfson microelectronics plc 26 westfield road edinburgh eh11 2qb united kingdom tel :: +44 (0)131 272 7000 fax :: +44 (0)131 272 7001 email :: sales@wolfsonmicro.com

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