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| _______________________________________________________________ maxim integrated products 1 for pricing, delivery, and ordering information, please contact maxim direct at 1-888-629-4642, or visit maxims website at www.maxim-ic.com. stereo audio codec with flexsound technology MAX9888 19-5235; rev 0; 6/10 simplified block diagram general description the MAX9888 is a full-featured audio codec whose high performance and low power consumption make it ideal for portable applications. class d speaker amplifiers provide efficient amplification for two speakers. low radiated emissions enable com - pletely filterless operation. integrated bypass switches optionally connect an external amplifier to the transducer when the class d amplifiers are disabled. directdrive ? headphone amplifiers provide a true ground-referenced output, eliminating the need for large dc-blocking capacitors. 1.8v headphone opera - tion ensures low power consumption. the device also includes a differential receiver amplifier. three differential analog microphone inputs are available as well as support for two pdm digital microphones. integrated switches allow microphone signals to be routed out to external devices. two flexible single-ended or differential line inputs may be connected to an fm radio or other sources. integrated flexsound k technology improves loud- speaker performance by optimizing the signal level and frequency response while limiting the maximum distor - tion and power at the output to prevent speaker damage. automatic gain control (agc) and a noise gate optimize the signal level of microphone input signals to make best use of the adc dynamic range. the device is fully specified over the -40 n c to +85 nc extended temperature range. features s 100db dr stereo dac (8khz < f s < 96khz) s 91db dr stereo adc (8khz < f s < 96khz) s stereo low emi class d amplifiers 950mw/channel (8 i , v spkvdd_ = 4.2v) s stereo directdrive headphone amplifiers s differential receiver amplifier s 2 stereo single-ended/mono differential line inputs s 3 differential microphone inputs s flexsound technology 5-band parametric eq automatic level control (alc) excursion limiter speaker power limiter speaker distortion limiter microphone automatic gain control and noise gate s dual i 2 s/pcm/tdm digital audio interfaces s asynchronous digital mixing s supports master clock frequencies from 10mhz to 60mhz s rf immune analog inputs and outputs s extensive click-and-pop reduction circuitry s i 2 c control interface s 63 wlp package (3.80mm x 3.30mm, 0.4mm pitch) + denotes lead(pb)-free/rohs-compliant package. ordering information directdrive is a registered trademark and flexsound is a trademark of maxim integrated products, inc. part temp range pin-package MAX9888ewy+ -40nc to +85nc 63 wlp dac dac mix receiver amp speaker amp speaker amp headphone amp headphone amp adc adc mix flexsound technology ? 5-band parametric eq ? automatic level control ? loudspeaker processing ? excursion limiter ? thd limiter ? power limiter ? microphone processing ? automatic gain control ? noise gate ? asynchronous digital mixing digi tal mi crophone input + + i 2 c control i 2 s/pcm digital audio interface i 2 s/pcm digital audio interface linein a1 linein a2 linein b1 linein b2 MAX9888 datasheet.in
stereo audio codec with flexsound technology MAX9888 2 table of con tents general description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 simplified block diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 ordering information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 functional diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 absolute maximum ratings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 electrical characteristics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 digital input/output characteristics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19 input clock characteristics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21 audio interface timing characteristics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22 digital microphone timing characterstics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23 i 2 c timing characterstics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24 power consumption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25 typical operating characteristics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28 microphone to adc . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28 line to adc . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32 digital loopback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 analog loopback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34 dac to receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35 line to receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37 dac to speaker . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38 line to speaker . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42 dac to headphone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43 line to headphone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48 speaker bypass switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49 pin configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50 pin description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51 detailed description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53 i 2 c slave address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54 registers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54 power management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59 microphone inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60 line inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62 adc input mixers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63 record path signal processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64 microphone agc . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64 noise gate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64 datasheet.in stereo audio codec with flexsound technology MAX9888 3 table of contents (continued) adc record level control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67 sidetone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68 digital audio interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69 clock control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76 passband filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78 playback path signal processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81 automatic level control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81 parametric equalizer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82 playback level control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84 dac input mixers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85 preoutput signal path . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86 preoutput mixer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86 preoutput pga . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87 receiver amplifier . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87 receiver output volume . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88 receiver output mixer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88 speaker amplifiers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89 speaker output volume . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90 speaker output mixers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90 speaker amplifier signal processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91 excursion limiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91 power limiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93 distortion limiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94 headphone amplifier . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95 headphone output mixers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96 headphone output volume . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97 output bypass switches . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98 click-and-pop reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99 jack detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100 jack insertion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100 accessory button detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100 jack removal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101 datasheet.in stereo audio codec with flexsound technology MAX9888 4 table of contents (continued) battery measurement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102 device status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103 device revision . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104 i 2 c serial interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104 bit transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104 start and stop conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104 early stop conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104 slave address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105 acknowledge . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105 write data format . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105 read data format . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106 applications information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107 typical operating circuits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107 analog microphones and bypass switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107 digital microphones and receiver amplifier . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108 filterless class d operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109 rf susceptibility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109 startup/shutdown sequencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109 component selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110 optional ferrite bead filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110 input capacitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110 charge-pump capacitor selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110 charge-pump flying capacitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111 charge-pump holding capacitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111 unused pins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111 recommended pcb routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112 supply bypassing, layout, and grounding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112 wlp applications information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113 package information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114 revision history . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115 datasheet.in stereo audio codec with flexsound technology MAX9888 5 functional diagram sda scl i 2 c e5 f5 f4 e2 mclk clock control inbdiff mixout3 mixout2 pgainb: +20db to -6db c8 inb2 d8 inb1 d9 ina2/extmicn e9 ina1/extmicp g9 mic2n f9 mic2p f8 d6 jacksns jack detection jdeten mben g8 mi cb ia s reg + mix mix pgainb: +20db to -6db inadiff inabyp mic2byp pgaina: +20db to -6db pgaout1: 0db to -23db pgaout2: 0db to -23db pgaout3: 0db to -23db + pgaina: +20db to -6db pgam2: +20db to 0db pgam1: +20db to 0db mixout1 pr eo ut 1 mix mixadr adren adlen psclk mix mixadl mix pa2en: 0/20/30db extmic pa1en: 0/20/30db extmic mic1n/ digmicclk e8 mic1p/ digmicdata adcr adcl pr eo ut 2 pr eo ut 3 +9 db +9 db +9 db avlg: 0/6/12/18db avl: 3db to -12db mode1 avflt dsts mas1 dai1 sel1 sel2 bclk1 bclks1 d1 d2 e4 d4 e1 f2 f3 g1 g3 g2 lrclk1 sdout1 sdin1 sidetone dvst: 0db to -60db lten1 lben2 avrg: 0/6/12/18db avr: 3db to -2db automatic gain control eq1en eq2en dveq1: 0db to -15db dv1g: 0/6/12/18db dv2: 0db to -15db dcb2 5-band parametric eq 5-band parametric eq dveq2: 0db to -15db flexsound technology multi band alc excursion limiter dv1: 0db to -15db mode1 dvflt dalen mixdar dacl daren dacr + + bit clock frame clock data output data input mas1 hizoff1 mas2 dai2 bclk2 lrclk2 sdout2 sdin2 bit clock frame clock data output data input mas2 hizoff2 lrclks1 sdouts1 sdins1 dvdds1 bclks2 lrclks2 sdouts2 sdins2 dvdds2 g4 dvdd g5 avdd mux lben1 mixrec battery adc 0db recvol: +8db to -62db recp/ rxinp a6 recn/ rxinn spklp spkln b6 recbyp spkbyp recen mixspl +6db spvoll: +8db to -62db a4, b4 spklvdd a3, b3 a5, b5 spklgnd c4, c5 spkrvdd c3, d3 spkrp c1, c2 spkrn a1, b1 splen spren po we r/ di st or ti on li mi te r mixspr +6db spvolr: +8db to -62db hpvoll: +3db to -67db spkrgnd a2, b2 hpl b9 hpsns c6 hpr c9 hplen mixhpl hpvolr: +3db to -67db hpvdd a7 hpgnd a8 b7 b8 a9 g7 f1 hpren mixhpr hpvss agnd dgnd c1n c1p charge pump re f f 7 g6 f6 pr eg re g bias MAX9888 noise gate audio/ voice filters audio/ filters audio/ voice filters mix mix mix mixdal mix mix mix mix mix irq port s1 port s2 datasheet.in stereo audio codec with flexsound technology MAX9888 6 stresses beyond those listed under absolute maximum ratings may cause permanent damage to the device. these are stress ratings only, and functional operation of the device at these or any other conditions beyond those indicated in the operational sections of the specifications is not implied. exposure to absolute maximum rating conditions for extended periods may affect device reliability. (voltages with respect to agnd.) dvdd, avdd, hpvdd ......................................... -0.3v to +2.2v spklvdd, spkrvdd, dvdds1, dvdds2 .......... -0.3v to +6.0v dgnd, hpgnd, spklgnd, spkrgnd .............. -0.1v to +0.1v hpvss ............................... (hpgnd - 2.2v) to (hpgnd + 0.3v) c1n .................................... (hpvss - 0.3v) to (hpgnd + 0.3v) c1p ..................................... (hpgnd - 0.3v) to (hpvdd + 0.3v) preg ..................................................... -0.3v to (avdd + 0.3v) ref, micbias ................................. -0.3v to (spklvdd + 0.3v) mclk, sdins1, sdins2, jacksns, sda, scl, irq ................................................. -0.3v to +6.0v lrclks1, bclks1, sdouts1 ......... -0.3v to (dvdds1 + 0.3v) lrclks2, bclks2, sdouts2 ......... -0.3v to (dvdds2 + 0.3v) reg, ina1, ina2, inb1, inb2, mic1p/digmicdata, mic1n/digmicclk, mic2p, mic2n ............... -0.3v to +2.2v hpsns ............................... (hpgnd - 0.3v) to (hpgnd + 0.3v) hpl, hpr ............................ (hpvss - 0.3v) to (hpvdd + 0.3v) recp, recn .............. (spklgnd - 0.3v) to (spklvdd + 0.3v) spklp, spkln ........... (spklgnd - 0.3v) to (spklvdd + 0.3v) spkrp, spkrn ......... (spkrgnd - 0.3v) to (spkrvdd + 0.3v) continuous power dissipation (t a = +70 nc) 63-bump wlp (derate 25.6mw/ n c above +70nc) ........ 2.05w operating temperature range .......................... -40n c to +85nc storage temperature range ............................ -65n c to +150nc soldering temperature (reflow) ...................................... +260nc electrical characteristics (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) absolute maximum ratings parameter symbol conditions min typ max units power supply supply voltage range guaranteed by psrr v spklvdd , v spkrvdd 2.8 5.5 v v dvdd , v avdd , v hpvdd 1.65 1.8 2.0 v dvdds1 , v dvdds2 1.65 3.6 total supply current (note 2) i vdd full-duplex 8khz mono, receiver output (note 3) analog 6.37 10 ma speaker 1.98 3.5 digital 1.49 3 dac playback 48khz stereo, headphone outputs (note 3) analog 2.71 4 speaker 1.65 2.5 digital 2.93 4.5 dac playback 48khz stereo, speaker outputs (note 3) analog 1.85 3 speaker 8.22 16 digital 2.94 5 full-duplex 48khz stereo, microphone inputs, headphone outputs (note 3) analog 12.75 18 speaker 1.7 3 digital 3.75 5.5 stereo line playback, in_dif = 0, ina1 to hpl, ina2 to hpr, v mclk = 0v analog 5.11 7 speaker 0.58 1 digital 0.03 0.06 datasheet.in stereo audio codec with flexsound technology MAX9888 7 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units shutdown supply current (note 2) t a = +25nc analog 0.2 2 fa speaker 0.1 1 digital 1 5 ref voltage 2.5 v preg voltage 1.6 v reg voltage 0.7 v shutdown to full operation slew = 0 30 ms slew = 1 17 microphone to adc path dynamic range (note 4) dr f s = 8khz, mode = 0 (iir voice), av micpre_ = 0db 75 88 db total harmonic distortion + noise thd+n v in = 0.1v p-p , mclk = 12.288mhz, f s = 8khz, f = 1khz -77 -65 db av micpre_ = 0db, v in = 1v p-p , f = 1khz -82 av micpre_ = +30db, v in = 32mv p-p , f = 1khz -71 common-mode rejection ratio cmrr v in = 100mv p-p , f = 217hz 65 db power-supply rejection ratio psrr v avdd = 1.65v to 2.0v, input referred, mic inputs floating 60 100 db f = 217hz, v ripple = 100mv p-p , av adc = 0db, input referred 100 f = 1khz, v ripple = 100mv p-p , av adc = 0db, input referred 91 f = 10khz, v ripple = 100mv p-p , av adc = 0db, input referred 70 path phase delay 1khz, 0db input, highpass filter disabled measured from analog input to digital output mode = 0 (iir voice) 8khz 2.2 ms mode = 0 (iir voice) 16khz 1.1 mode = 1 (fir audio) 8khz 4.5 mode = 1 (fir audio) 48khz 0.76 datasheet.in stereo audio codec with flexsound technology MAX9888 8 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units microphone preamp full-scale input av micpre_ = 0db 1.05 v p-p preamplifier gain av micpre_ (note 5) pa1en/pa2en = 01 0 db pa1en/pa2en = 10 19.5 20 20.5 pa1en/pa2en = 11 29.4 30 30.5 pga gain av micpga_ (note 5) pgam1/pgam2 = 0x00 19.5 20 20.5 db pgam1/pgam2 = 0x14 0 mic input resistance r in_mic all gain settings, measured at mic1p/mic1n/ mic2p/mic2n 30 50 ki microphone bias micbias output voltage v micbias i load = 1ma 2.14 2.2 2.25 v load regulation i load = 1ma to 2ma 0.5 11 mv line regulation v spklvdd = 2.8v to 5.5v 100 fv ripple rejection f = 217hz, v ripple (spklvdd) = 100mv p-p 92 db f = 10khz, v ripple (spklvdd) = 100mv p-p 83 noise voltage a-weighted, f = 20hz to 20khz 3.8 fv rms p-weighted, f = 20hz to 4khz 2.1 f = 1khz 33 nv/hz microphone bypass switch on-resistance r on i mic1_ = 100ma, inabyp = mic2byp = 1, v mic2_ = v ina_ = (0v, v avdd ) 3.5 20 i total harmonic distortion + noise thd+n v in = 2v p-p , v cm = 0.9v, r l = 10ki, f = 1khz, inabyp = mic2byp = 1 -80 db off-isolation v in = 2v p-p , v cm = 0.9v, r l = 10ki, f = 1khz 60 db off-leakage current v mic1_ = (0v, v avdd ), v mic2_ /v ina_ = (v avdd , 0v) -2.5 +2.5 fa line input to adc path dynamic range (note 4) dr f s = 48khz, mclk = 12.288mhz, mode = 1 (fir audio) 91 db total harmonic distortion + noise thd+n v in = 1v p-p , f = 1khz -77 db gain error dc accuracy 1 5 % datasheet.in stereo audio codec with flexsound technology MAX9888 9 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units line input preamp full-scale input v in av pgain_ = 0db 1 v p-p av pgain_ = -6db 1.4 level adjust gain av pgain_ pgaina/pgainb = 0x0 19 20 21 db pgaina/pgainb = 0x1 13 14 15 pgaina/pgainb = 0x2 (note 5) 2 3 4 pgaina/pgainb = 0x3 t a = +25nc 0 pgaina/pgainb = 0x4 -4 -3 -2 pgaina/pgainb = 0x5, 0x6, 0x7 -7 -6 -5 input resistance r in av pgain_ = +20db 14.6 21 27.4 ki av pgain_ = +14db 20 av pgain_ = +3db 20 av pgain_ = 0db 7.3 10 13.7 av pgain_ = -3db 20 av pgain_ = -6db 20 feedback resistance r in_fb inaext/inbext = 1 t a = +25nc 18.5 20 21.5 ki t a = t min to t max 17.5 23 adc level control adc level adjust range av adclvl avl/avr = 0xf to 0x0 (note 5) -12 +3 db adc level adjust step size 1 db adc gain adjust range av adcgain avlg/avrg = 00 to 11 (note 5) 0 18 db adc gain adjust step size 6 db adc digital filters voice mode iir lowpass filter (mode1 = 0) passband cutoff f plp ripple limit cutoff 0.441 x f s hz -3db cutoff 0.449 x f s passband ripple f < f plp -0.1 +0.1 db stopband cutoff f slp 0.47 x f s hz stopband attenuation (note 6) f > f slp 74 db datasheet.in stereo audio codec with flexsound technology MAX9888 10 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units voice mode iir highpass filter (mode1 = 0) passband cutoff (-3db from peak) f ahppb avflt = 0x1 (elliptical tuned for f s = 16khz + 217hz notch) 0.0161 x f s hz avflt = 0x2 (500hz butterworth tuned for f s = 16khz) 0.0319 x f s avflt = 0x3 (elliptical tuned for f s = 8khz + 217hz notch) 0.0321 x f s avflt = 0x4 (500hz butterworth tuned for f s = 8khz) 0.0632 x f s avflt = 0x5 (f s /240 butterworth) 0.0043 x f s stopband cutoff (-30db from peak) f ahpsb avflt = 0x1 (elliptical tuned for f s = 16khz + 217hz notch) 0.0139 x f s hz avflt = 0x2 (500hz butterworth tuned for f s = 16khz) 0.0156 x f s avflt = 0x3 (elliptical tuned for f s = 8khz + 217hz notch) 0.0279 x f s avflt = 0x4 (500hz butterworth tuned for f s = 8khz) 0.0312 x f s avflt = 0x5 (f s /240 butterworth) 0.002 x f s dc attenuation dc atten avflt 000 90 db stereo audio mode fir lowpass filter (mode1 = 1, dhf1 = 0, lrclk < 50khz) passband cutoff f plp ripple limit cutoff 0.43 x f s hz -3db cutoff 0.48 x f s -6.02db cutoff 0.5 x f s passband ripple f < f plp -0.1 +0.1 db stopband cutoff f slp 0.58 x f s hz stopband attenuation (note 6) f < f slp 60 db adc stereo audio mode fir lowpass filter (mode1 = 1, dhf1 = 1, lrclk > 50khz) passband cutoff f plp ripple limit cutoff 0.208 x f s hz -3db cutoff 0.28 x f s datasheet.in stereo audio codec with flexsound technology MAX9888 11 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units passband ripple f < f plp -0.1 +0.1 db stopband cutoff f slp 0.417 x f s hz stopband attenuation f < f slp 60 db adc stereo audio mode dc-blocking highpass filter (mode1 = 1) passband cutoff (-3db from peak) f ahppb avflt 000 0.000125 x fs hz dc attenuation dc atten avflt 000 90 db microphone automatic gain control agc hold duration agchld = 01 50 ms agchld = 11 400 agc attack time agcatk = 00 2 ms agcatk = 11 123 agc release time agcrls = 000 0.078 s agcrls = 111 10 agc threshold level agcth = 0x0 to 0xf -3 +18 db agc threshold step size 1 db agc gain (note 5) 0 20 db adc noise gate ng threshold level anth = 0x3 to 0xf, referred to 0dbfs -64 -16 db ng attenuation (note 5) 0 12 db adc-to-dac digital sidetone (mode = 0) sidetone gain adjust range av stga dvst = 0x01 -0.5 db dvst = 0x1f -60.5 sidetone gain adjust step size 2 db sidetone path phase delay 1khz, 0db input, highpass filter disabled 8khz 2.2 ms 16khz 1.1 adc-to-dac digital loop-through path dynamic range (note 4) dr f s = 48khz, mclk = 12.288mhz, mode = 1 (fir audio) 89 db total harmonic distortion thd f = 1khz, f s = 48khz, mclk = 12.288mhz, mode = 1 (fir audio) -71 -66 db dac level control dac attenuation range av dacattn dv1dv2 = 0xf to 0x0 (note 5) -15 0 db dac attenuation step size 1 db dac gain adjust range av dacgain dv1g = 00 to 11 (note 5) 0 18 db dac gain adjust step size 6 db datasheet.in stereo audio codec with flexsound technology MAX9888 12 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units dac digital filters voice mode iir lowpass filter (mode1 = 0) passband cutoff f plp ripple limit cutoff 0.448 x f s hz -3db cutoff 0.451 x f s passband ripple f < f plp -0.1 +0.1 db stopband cutoff f slp 0.476 x f s hz stopband attenuation (note 6) f > f slp 75 db voice mode iir highpass filter (mode1 = 0) passband cutoff (-3db from peak) f dhppb dvflt = 0x1 (elliptical tuned for f s = 16khz + 217hz notch) 0.0161 x f s hz dvflt = 0x2 (500hz butterworth tuned for f s = 16khz) 0.0312 x f s dvflt = 0x3 (elliptical tuned for f s = 8khz + 217hz notch) 0.0321 x f s dvflt = 0x4 (500hz butterworth tuned for f s = 8khz) 0.0625 x f s dvflt = 0x5 (f s /240 butterworth) 0.0042 x f s stopband cutoff (-30db from peak) f dhpsb dvflt = 0x1 (elliptical tuned for f s = 16khz + 217hz notch) 0.0139 x f s hz dvflt = 0x2 (500hz butterworth tuned for f s = 16khz) 0.0156 x f s dvflt = 0x3 (elliptical tuned for f s = 8khz + 217hz notch) 0.0279 x f s dvflt = 0x4 (500hz butterworth tuned for f s = 8khz) 0.0312 x f s dvflt = 0x5 (f s /240 butterworth) 0.002 x f s dc attenuation dc atten dvflt 000 85 db datasheet.in stereo audio codec with flexsound technology MAX9888 13 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units stereo audio mode fir lowpass filter (mode1 = 1, dhf1/dhf2 = 0, lrclk < 50khz) passband cutoff f plp ripple limit cutoff 0.43 x f s hz -3db cutoff 0.47 x f s -6.02db cutoff 0.5 x f s passband ripple f < f plp -0.1 +0.1 db stopband cutoff f slp 0.58 x f s hz stopband attenuation (note 6) f > f slp 60 db stereo audio mode fir lowpass filter (mode1 = 1, dhf1/dhf2 = 1 for lrclk > 50khz) passband cutoff f plp ripple limit cutoff 0.24 x f s hz -3db cutoff 0.31 x f s passband ripple f < f plp -0.1 +0.1 db stopband cutoff f slp 0.477 x f s hz stopband attenuation (note 6) f < f slp 60 db stereo audio mode dc-blocking highpass filter passband cutoff (-3db from peak) f dhppb dvflt 000 (dai1), dcb2 = 1 (dai2) 0.000104 x f s hz dc attenuation dc atten dvflt 000 (dai1), dcb2 = 1 (dai2) 90 db automatic level control dual band lowpass corner frequency alcmb = 1 5 khz dual band highpass corner frequency alcmb = 1 5 khz gain range 0 12 db low signal threshold alcth = 111 to 001 -48 -12 dbfs release time alcrls = 101 0.25 s alcrls = 000 8 parametric equalizer number of bands 5 bands per band gain range -12 +12 db preattenuator gain range (note 5) -15 0 db preattenuator step size 1 db datasheet.in stereo audio codec with flexsound technology MAX9888 14 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units dac-to-receiver amplifier path dynamic range (note 4) dr f s = 48khz, mclk = 12.288mhz, f = 1khz 96 db total harmonic distortion + noise thd+n f = 1khz, p out = 25mw, r rec = 32i -70 -63 db click and pop level k cp peak voltage, a-weighted, 32 samples per second, av rec = 0db into shutdown -70 dbv out of shutdown -73 preoutput mixers level adjust gain av pgaout_ (note 5) pgaouta/pgaoutb/ pgaoutc = 0x0 0 db pgaouta/pgaoutb/ pgaoutc = 0xc -25 -23.4 -22 level adjust step size 2 db mute attenuation f = 1khz 85 db line input-to-receiver amplifier path dynamic range (note 4) dr referenced to full-scale output level 92 db total harmonic distortion + noise thd+n -70 db power-supply rejection ratio psrr v spklvdd = 2.8v to 5.5v 54 89 db f = 217hz, v ripple = 100mv p-p -63 f = 1khz, v ripple = 100mv p-p -63 f = 10khz, v ripple = 100mv p-p -65 click-and-pop level k cp peak voltage, a-weighted, 32 samples per second, av rec = 0db into shutdown -57 dbv out of shutdown -55 receiver amplifier output power p out r rec = 32i, f = 1khz, thd = 1% 100 mw full-scale output (note 7) 1 v rms volume control av rec (note 5) recvol = 0x00 -65 -62 -58 db recvol = 0x1f +7.5 +8 +8.5 volume control step size +8db to +6db 0.5 db +6db to +0db 1 0db to -14db 2 -14db to -38db 3 -38db to -62db 4 mute attenuation f = 1khz 95 db output offset voltage v os av rec = -62db t a = +25nc 0.13 1 mv capacitive drive capability no sustained oscillations r rec = 32i 500 pf r rec = j 100 datasheet.in stereo audio codec with flexsound technology MAX9888 15 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units dac-to-speaker amplifier path total harmonic distortion + noise thd+n f = 1khz, p out = 250mw, z spk = 8i + 68fh -71 db crosstalk spkl to spkr and spkr to spkl, p out = 640mw, f = 1khz -75 db output noise a-weighted 43 fv rms click-and-pop level k cp peak voltage, a-weighted, 32 samples per second, av spk_ = 0db into shutdown -65 dbv out of shutdown -65 line input-to-speaker amplifier path total harmonic distortion + noise thd+n f = 1khz, p out = 200mw, z spk = 8i + 68fh -66 db output noise a-weighted 56 fv rms power-supply rejection ratio psrr v spklvdd = v ripple = 2.8v to 5.5v 43 60 db f = 217hz, v ripple = 100mv 75 f = 1khz, v ripple = 100mv 73 f = 10khz, v ripple = 100mv 50 click-and-pop level k cp peak voltage, a-weighted, 32 samples per second, av spk_ = 0db into shutdown -48 dbv out of shutdown -50 speaker amplifier output power p out f = 1khz, thd = 1%, z spk = 8i + 68fh v spklvdd = v spkrvdd = 5.0v 1370 mw v spklvdd = v spkrvdd = 4.2v 954 v spklvdd = v spkrvdd = 3.7v 733 v spklvdd = v spkrvdd = 3.2v 544 full-scale output (note 7) 2 v rms volume control (note 5) av spk_ spvoll/spvolr = 0x00 -69 -64 -59 db spvoll/spvolr = 0x1f +7.5 +8 +8.5 volume control step size +8db to +6db 0.5 db +6db to +0db 1 0db to -14db 2 -14db to -38db 3 -38db to -64db 4 datasheet.in stereo audio codec with flexsound technology MAX9888 16 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units mute attenuation f = 1khz 86 db output offset voltage v os av spk_ = -64db, t a = +25nc 0.25 1.25 mv excursion limiter upper-corner frequency range dhpucf = 001 to 100 400 1000 hz lower-corner frequency dhplcf = 01 to 10 400 hz biquad minimum corner frequency dhpucf = 000 (fixed mode) 100 hz dhpucf = 001 200 dhpucf = 010 300 dhpucf = 011 400 dhpucf = 100 500 threshold voltage z spk = 8i + 68fh, v spklvdd = v spkrvdd = 5.5v, av spk_ = +8db dhpth = 000 0.34 v p dhpth = 111 4.95 release time alcrls = 101 0.25 s alcrls = 000 4 power limiter attenuation -64 db threshold z spk = 8i + 68fh, v spklvdd = v spkrvdd = 5.5v, av spk_ = +8db pwrth = 0x1 0.05 w pwrth = 0xf 1.80 time constant 1 t pwr1 pwrt1 = 0x1 0.5 s pwrt1 = 0xf 8.7 time constant 2 t pwr2 pwrt2 = 0x1 to 0xf 0.5 min pwrt2 = 0xf 8.7 weighting factor k pwr pwrk = 000 to 111 12.5 100 % distortion limiter distortion limit thdclp = 0x1 < 1 % thdclp = 0xf 24 release time constant thdt1 = 000 0.76 s thdt1 = 111 6.2 dac-to-headphone amplifier path dynamic range (note 4) dr f s = 48khz, mclk = 12.288mhz master or slave mode 100 db slave mode 94 total harmonic distortion + noise thd+n f s = 48khz, mclk = 12.288mhz, f = 1khz, p out = 20mw r hp = 16i -71 -64 db r hp = 32i -75 f s = 48khz, mclk = 12.288mhz, f = 1khz, v out = 1 vrms , r hp = 10ki -79 datasheet.in stereo audio codec with flexsound technology MAX9888 17 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units crosstalk f = 1khz, input = -1dbfs, r hp = 10ki -82 db hpl to hpr and hpr to hpl, p out = 5mw, f = 1khz, r hp = 32i -82 db power-supply rejection ratio psrr v avdd = v hpvdd = 1.65v to 2.0v 60 84 db f = 217hz, v ripple = 100mv, av vol = 0db 92 f = 1khz, v ripple = 100mv, av vol = 0db 91 f = 10khz, v ripple = 100mv, av vol = 0db 57 dac path phase delay 1khz, 0db input, highpass filter disabled measured from digital input to analog output mode = 0 (voice) 8khz 2.2 ms mode = 0 (voice) 16khz 1.1 mode = 1 (music) 8khz 4.5 mode = 1 (music) 48khz 0.76 gain error 1 % channel gain mismatch 0.5 % click-and-pop level k cp peak voltage, a-weighted, 32 samples per second, av hp_ = 0db into shutdown -66 dbv out of shutdown -67 line input-to-headphone amplifier path total harmonic distortion + noise thd+n v in = 1v p-p , f =1khz, r hp = 32i -70 db dynamic range (note 4) dr 91 db power-supply rejection ratio psrr v avdd = v hpvdd = 1.65v to 2.0v 42 66 db f = 217hz, v ripple = 100mv p-p 62 f = 1khz, v ripple = 100mv p-p 57 f = 10khz, v ripple = 100mv p-p 41 click and pop level k cp peak voltage, a-weighted, 32 samples per second, av hp_ = 0db into shutdown -62 dbv out of shutdown -60 headphone amplifier output power p out f = 1khz, thd = 1% r hp = 32i 32 mw r hp = 16i 40 full-scale output (note 7) 1 v rms volume control av hp_ t a = +25nc (note 5) hpvol_ = 0x00 -71 -67 -66 db hpvol_ = 0x1f 2.4 3 3.5 datasheet.in stereo audio codec with flexsound technology MAX9888 18 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units volume control step size +3db to +1db 0.5 db +1db to -5db 1 -5db to -19db 2 -19db to -43db 3 -43db to -67db 4 mute attenuation f = 1khz 82 db output offset voltage v os av hp_ = -67db t a = +25nc 0.2 1 mv t a = t min to t max 2 capacitive drive capability no sustained oscillations r hp = 32i 500 pf r hp = j 100 charge pump oscillator frequency f cp 300 667 900 khz slow mode 74 speaker bypass switch on-resistance r on i spkl_ = 100ma, spkbyp = 1, v rxin_ = [0v, v spklvdd] 2.8 4.5 i total harmonic distortion + noise thd+n v in = 2v p-p , v cm = v spklvdd /2, z spk = 8i + 68fh, f = 1khz, spkbyp = 1 r s = 10i -77 db r s = 0i -60 off-isolation v in = 2v p-p , v cm = v spklvdd /2, z l = 8i + 68fh, f = 1khz 96 db off-leakage current v rxin_ = [0v, v spklvdd ], v spkl_ = [v spklvdd , 0v] -1 +1 fa receiver bypass switch on-resistance r on i recp = 100ma, recbyp = 1, v recn = [0v, v spklvdd ] 1.2 2 i total harmonic distortion + noise thd+n v in = 2v p-p , v cm = v spklvdd /2, r l = 32i, f = 1khz, recbyp = 1 -66 % off-isolation v in = 2v p-p , v cm = v spklvdd /2, r l = 32i, f = 1khz 80 db off-leakage current v recp = [0v, v spklvdd ], v recn = [v spklvdd , 0v] -15 +15 fa datasheet.in stereo audio codec with flexsound technology MAX9888 19 electrical characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) digital input/output characteristics (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = 1.65v to 2.0v, v spklvdd = v spkrvdd = 3.7v, t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units jack detection jacksns high threshold v th1 micbias enabled 0.92 x v micbias 0.95 x v micbias 0.98 x v micbias v micbias disabled 0.92 x v spklvd d 0.95 x v spklvdd 0.98 x v spklvd d jacksns low threshold v th2 micbias enabled 0.06 x v micbias 0.10 x v micbias 0.17 x v micbias v micbias disabled 0.06 x v spklvd d 0.10 x v spklvdd 0.17 x v spklvd d jacksns sense voltage v sense micbias disabled v spklvdd v jacksns sense resistance r sense micbias disabled, jdwk = 0 1.7 2.4 2.9 ki jacksns weak pullup current i wpu micbias disabled, jdwk = 1 2 5 9.5 fa jacksns deglitch period t glitch jdeb = 00 25 ms jdeb = 11 200 battery adc input voltage range 2.8 5.5 v lsb size 0.1 v parameter symbol conditions min typ max units mclk input high voltage v ih 1.2 v input low voltage v il 0.6 v input leakage current i ih , i il v dvdd = 2.0v, v in = 0v, 5.5v, t a = +25c -1 +1 fa input capacitance 10 pf sdins1, bclks1, lrclks1input input high voltage v ih 0.7 x dvdds1 v input low voltage v il 0.29 x dvdds1 v input hysteresis 200 mv input leakage current i ih , i il v dvdds1 = 3.6v, v in = 0v, 3.6v; t a = +25c -1 +1 fa input capacitance 10 pf datasheet.in stereo audio codec with flexsound technology MAX9888 20 digital input/output characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = 1.65v to 2.0v, v spklvdd = v spkrvdd = 3.7v, t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units bclks1, lrclks1, sdouts1output output low voltage v ol v dvdds1 = 1.65v, i ol = 3ma 0.4 v output high voltage v oh v dvdds1 = 1.65v, i oh = 3ma dvdds1 - 0.4 v input leakage current i ih , i il v dvdd = 2.0v, v in = 0v, 5.5v; t a = +25c, high-impedance state -1 +1 fa sdins2, bclks2, lrclks2input input high voltage v ih 0.7 x dvdds2 v input low voltage v il 0.29 x dvdds2 v input hysteresis 200 mv input leakage current i ih , i il v dvdds2 = 3.6v, v in = 0v, 3.6v; t a = +25c -1 +1 fa input capacitance 10 pf bclks2, lrclks2, sdouts2output output low voltage v ol v dvdds2 = 1.65v, i ol = 3ma 0.4 v output high voltage v oh v dvdds2 = 1.65v, i oh = 3ma dvdds2 - 0.4 v input leakage current i ih , i il v dvdd = 2.0v, v in = 0v, 5.5v; t a = +25nc, high-impedance state -1 +1 fa sda, sclinput input high voltage v ih 0.7 x dvdd v input low voltage v il 0.3 x dvdd v input hysteresis 210 mv input leakage current i ih , i il v dvdd = 2.0v, v in = 0v, 5.5v, t a = +25nc -1 +1 fa input capacitance 10 pf sda, irqoutput output high current i oh v out = 5.5v, t a = +25c 1 ma output low voltage v ol v dvdd = 1.65v, i ol = 3ma 0.2 x dvdd v digmicdatainput input high voltage v ih 0.65 x dvdd v input low voltage v il 0.35 x dvdd v input hysteresis 125 mv input leakage current i ih , i il v dvdd = 2.0v, v in = 0v, 2.0v; t a = +25c -25 +25 fa input capacitance 10 pf datasheet.in stereo audio codec with flexsound technology MAX9888 21 input clock characteristics (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v, t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) digital input/output characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = 1.65v to 2.0v, v spklvdd = v spkrvdd = 3.7v, t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units mclk input frequency f mclk 10 60 mhz mclk input duty cycle psclk = 01 40 50 60 % psclk = 10 or 11 30 70 maximum mclk input jitter 100 ps rms lrclk sample rate (note 8) dhf_ = 0 8 48 khz dhf_ = 1 48 96 dai1 lrclk average frequency error (note 9) freq1 = 0x8 to 0xf 0 0 % freq1 = 0x0 -0.025 +0.025 dai2 lrclk average frequency error (note 9) -0.025 +0.025 % pll lock time rapid lock mode 2 7 ms nonrapid lock mode 12 25 maximum lrclk jitter to maintain pll lock 100 ns soft-start/stop time 10 ms parameter symbol conditions min typ max units digmicclkoutput output low voltage v ol v dvdd = 1.65v, i ol = 1ma 0.4 v output high voltage v oh v dvdd = 1.65v, i oh = 1ma dvdd - 0.4 v datasheet.in stereo audio codec with flexsound technology MAX9888 22 audio interface timing characteristics (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = 1.65v, v spklvdd = v spkrvdd = 2.8v, t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) figure 1. non-tdm audio interface timing diagrams (tdm_ = 0) parameter symbol conditions min typ max units bclk cycle time t bclk slave mode 90 ns bclk high time t bclkh slave mode 20 ns bclk low time t bclkl slave mode 20 ns bclk or lrclk rise and fall time t r , t f master mode, c l = 15pf ns sdin to bclk setup time t setup 20 ns lrclk to bclk setup time t syncset slave mode 20 ns sdin to bclk hold time t hold 20 ns lrclk to bclk hold time t synchold slave mode 20 ns minimum delay time from lsb bclk falling edge to high-impedance state t hizout master mode, tdm_ = 1 42 ns lrclk rising edge to sdout msb delay t synctx c l = 30pf, tdm_ = 1, fsw_ = 1 50 ns bclk to sdout delay t clktx c l = 30pf tdm_ = 1, bclk rising edge 50 ns tdm_ = 0 50 delay time from bclk to lrclk t clksync master mode tdm_ = 1 -15 +15 ns tdm_ = 0 0.8 x t bclkl delay time from lrclk to bclk after lsb t endsync master mode tdm_ = 1, fsw_ = 1 20 ns t r t clktx t setup t hold t bclk t bclkh t bclkl msb lsb lsb bclk (output) lrclk (output) sdout (output) sdin (input) bclk (input) lrclk (input) sdout (output) sdin (input) master mode t setup t hold msb msb lsb lsb hi-z slave mode hi-z msb t clksync t f t hizout t clktx t syncset t hizout datasheet.in stereo audio codec with flexsound technology MAX9888 23 digital microphone timing characterstics (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = 2.0v, v spklvdd = v spkrvdd = 2.8v, t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) figure 2. tdm audio interface timing diagram (tdm_ = 1, fsw_ = 0) figure 3. tdm audio interface timing diagram (tdm_ = 1, fsw_ = 1) parameter symbol conditions min typ max units digmicclk frequency f micclk micclk = 00 mclk/8 mhz micclk = 01 mclk/6 digmicdata to digmicclk setup time t su,mic either clock edge 20 ns digmicdata to digmicclk hold time t hd,mic either clock edge 0 ns lrclk (output) bclk (output) sdout (output) sdin (input) t clktx t hizout hi-z lsb t hold t setup lsb msb msb t clksync t r t f lrclk (input) bclk (input) sdout (output) sdin (input) t clktx t hizout hi-z lsb t hold t setup lsb t synchold msb msb t bclkh t bclkl t bclk t syncset master mode slave mode t clksync t bclkh t bclkl t clksync t r t f l rc lk (out put) bc lk (out put) sd o ut (out put) s din (input) ma st er m ode hi- z lsb ms b t synctx t e nds yn c lsb ms b slav e mo de hi- z lsb lsb ms b ms b t clktx t hizout t hold t setup t hold t setup lrclk (input) bclk (input) sdout (output) sdin (input) t bclk t synctx t clktx t hizout datasheet.in stereo audio codec with flexsound technology MAX9888 24 figure 4. digital microphone timing diagram i 2 c t iming characterstics (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = 1.65v to 2.0v, v spklvdd = v spkrvdd = 3.7v, t a = t min to t max , unless otherwise noted. typical values are at t a = +25 n c.) (note 1) parameter symbol conditions min typ max units serial-clock frequency f scl guaranteed by scl pulse-width low and high 0 400 khz bus free time between stop and start conditions t buf 1.3 fs hold time (repeated) start condition t hd,sta 0.6 fs scl pulse-width low t low 1.3 fs scl pulse-width high t high 0.6 fs setup time for a repeated start condition t su,sta 0.6 fs data hold time t hd,dat r pu = 475i, cb = 100pf, 400pf 0 900 ns data setup time t su,dat 100 ns sda and scl receiving rise time t r (note 10) 20 + 0.1c b 300 ns sda and scl receiving fall time t f (note 10) 20 + 0.1c b 300 ns sda transmitting fall time t f r pu = 475i, c b = 100pf, 400pf (note 10) 20 + 0.05c b 250 ns setup time for stop condition t su,sto 0.6 fs bus capacitance c b guaranteed by sda transmitting fall time 400 pf pulse width of suppressed spike t sp 0 50 ns 1/ f mi cc lk t hd,mic t su,mic t hd,mic t su,mic le ft r igh t l ef t r ig ht datasheet.in stereo audio codec with flexsound technology MAX9888 25 note 1: the ic is 100% production tested at t a = +25 n c. specifications over temperature limits are guaranteed by design. note 2: analog supply current = i avdd + i hpvdd . speaker supply current = i spklvdd + i spkrvdd . digital supply current = i dvdd + i dvdds1 + i dvdds2 . note 3: clocking all zeros into the dac. slave mode. note 4: dynamic range measured using the eiaj method. -60dbfs, 1khz output signal, a-weighted and normalized to 0dbfs. f = 20hz to 20khz. note 5: gain measured relative to the 0db setting. note 6: the filter specification is accurate only for synchronous clocking modes, where ni is a multiple of 0x1000. note 7: 0dbfs for dac input. 1v p-p for ina/inb inputs. note 8: lrclk may be any rate in the indicated range. asynchronous or noninteger mclk/lrclk ratios may exhibit some full- scale performance degradation compared to synchronous integer related mclk/lrclk ratios. note 9: in master-mode operation, the accuracy of the mclk input proportionally determines the accuracy of the sample clock rate. note 10: cb is in pf. figure 5. i 2 c interface timing diagram power consumption (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v) mode i avdd (ma) i hpvdd (ma) i spklvdd + i spkrvdd (ma) i dvdd (ma) i dvdds1 + i dvdds2 (ma) power (mw) dac playback 48khz stereo hp dac hp 24-bit, music filters 1.35 1.37 1.65 2.91 0.02 16.25 dac playback 48khz stereo hp dac hp 24-bit, music filters, 0.1mw/channel, r hp = 32i 1.35 4.19 1.65 3.02 0.02 21.55 dac playback 48khz stereo hp dac hp 24-bit, music filters, alc enabled 1.35 1.37 1.65 2.96 0.02 16.36 scl sda t r t f t buf start condition stop condition repeated start condition start condition t su,sto t hd,sta t su,sta t hd,dat t su,dat t low t high t hd,sta t sp datasheet.in stereo audio codec with flexsound technology MAX9888 26 power consumption (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v) mode i avdd (ma) i hpvdd (ma) i spklvdd + i spkrvdd (ma) i dvdd (ma) i dvdds1 + i dvdds2 (ma) power (mw) dac playback 48khz stereo hp dac hp 24-bit, music filters, eq enabled 1.35 1.36 1.65 3.27 0.02 16.90 dac playback 48khz stereo hp dac hp 24-bit, music filters, digital mixing 1.34 1.36 1.65 2.91 0.02 16.27 dac playback 44.1khz stereo hp dac hp 24-bit, music filters 1.35 1.37 1.69 2.85 0.02 16.29 dac playback 8khz stereo hp dac hp 16-bit, voice filters 1.35 1.37 1.65 1.46 0.01 13.65 dac playback 8khz mono hp dac hp 16-bit, voice filters 1.00 0.71 1.01 1.36 0.01 9.27 dac playback 48khz stereo spk dac spk 24-bit, music filters 1.83 0.02 8.22 2.92 0.02 39.09 dac playback 48khz mono spk dac spk 24-bit, music filters 1.25 0.02 4.31 2.82 0.02 23.32 line stereo record 48khz ina adc 16-bit, music filters 9.91 0.02 0.39 1.62 0.11 22.48 line stereo record 48khz, stereo hp ina adc ina hp 16-bit, music filters 10.64 2.65 0.66 1.63 0.11 29.51 line stereo record 48khz, stereo spk ina adc ina spk 16-bit, music filters 10.97 0.03 7.15 1.63 0.12 49.50 differential line record 48khz ina adcl inb adcr differential input 10.49 0.02 0.39 1.63 0.16 23.58 microphone stereo record 48khz mic1/2 adc 16-bit, music filters 10.88 0.03 0.69 1.62 0.17 25.43 microphone stereo record 8khz mic1/2 adc 16-bit, voice filters 10.77 0.02 0.64 1.03 0.06 23.78 datasheet.in stereo audio codec with flexsound technology MAX9888 27 power consumption (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v) mode i avdd (ma) i hpvdd (ma) i spklvdd + i spkrvdd (ma) i dvdd (ma) i dvdds1 + i dvdds2 (ma) power (mw) microphone mono record 48khz mic1/2 adc 16-bit, music filters 6.01 0.02 0.66 1.37 0.10 15.97 microphone mono record 8khz mic1/2 adc 16-bit, voice filters 5.95 0.02 0.64 0.98 0.04 14.94 microphone mono record 8khz mic1/2 adc 16-bit, voice filters, agc 5.95 0.02 0.64 0.98 0.04 15.00 microphone mono record 8khz mic1/2 adc 16-bit, voice filters, agc, noise gate 5.96 0.02 0.64 0.98 0.04 14.98 full-duplex 48khz stereo hp mic1/2 adc dac hp 24-bit, music filters 11.38 1.37 1.70 3.56 0.19 36.06 full-duplex 8khz mono rcv mic1 adc dac rec 16-bit, voice filters 6.35 0.02 1.98 1.47 0.03 21.47 full-duplex 8khz mono hp mic1 adc dac hp 16-bit, voice filters 6.09 0.71 1.01 1.46 0.03 18.72 full-duplex 8khz stereo hp mic1/2 adc dac hp 16-bit, voice filters 10.92 1.37 1.09 1.51 0.05 28.95 line playback stereo hp ina hp single-ended inputs 1.89 2.65 0.58 0.03 0.01 10.41 line playback stereo spk ina spk single-ended inputs 2.21 0.02 7.05 0.04 0.02 30.19 line playback mono spk ina spk single-ended inputs 1.68 0.02 3.70 0.03 0.02 16.90 differential line playback stereo hp ina hpl inb hpr differential input 2.46 2.65 0.58 0.03 0.01 11.42 datasheet.in stereo audio codec with flexsound technology MAX9888 28 typical operating characteristics (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) microphone to adc total harmonic distortion plus noise vs. frequency (mic to adc) MAX9888 toc01 frequency (hz) thd+n ratio (db) 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 10,000 mclk = 13mhz lrclk = 8khz freq mode v in = 1v p-p av micpre_ = 0db total harmonic distortion plus noise vs. frequency (mic to adc) MAX9888 toc02 frequency (hz) thd+n ratio (db) 10,000 1000 100 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 10 100,000 mclk = 13mhz lrclk = 44.1khz pll mode v in = 1v p-p av micpre_ = 0db total harmonic distortion plus noise vs. frequency (mic to adc) MAX9888 toc03 frequency (hz) thd+n ratio (db) 10,000 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode v in = 1v p-p av micpre_ = 0db total harmonic distortion plus noise vs. frequency (mic to adc) MAX9888 toc04 frequency (hz) thd+n ratio (db) 10,000 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 100,000 mclk = 12.288mhz lrclk = 96khz ni mode v in = 1v p-p av micpre_ = 0db total harmonic distortion plus noise vs. frequency (mic to adc) MAX9888 toc05 frequency (hz) thd+n ratio (db) 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 10,000 mclk = 13mhz lrclk = 8khz freq mode v in = 0.1v p-p av micpre_ = +20db total harmonic distortion plus noise vs. frequency (mic to adc) MAX9888 toc06 frequency (hz) thd+n ratio (db) 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 10,000 mclk = 13mhz lrclk = 8khz freq mode v in = 0.032v p-p av micpre_ = +30db datasheet.in stereo audio codec with flexsound technology MAX9888 29 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) gain vs. frequency (mic to adc) MAX9888 toc07 frequency (hz) normalized gain (db) 1000 100 -75 -65 -55 -45 -35 -25 -15 -5 5 -85 10 10,000 mclk = 13mhz lrclk = 8khz freq mode v in = 1v p-p av micpre_ = 0db mode = 0 mode = 1 common-mode rejection ratio vs. frequency (mic to adc) MAX9888 toc08 frequency (hz) cmrr (db) 10,000 1000 100 10 20 30 40 50 60 70 80 90 0 10 100,000 av micpre_ = +30db av micpre_ = +20db av micpre_ = 0db v out, diff = 0dbfs power-supply rejection ratio vs. frequency (mic to adc) MAX9888 toc09 frequency (hz) psrr (db) 10,000 1000 100 -100 -80 -60 -40 -20 0 -120 10 100,000 v ripple = 200mv p-p inputs ac grounded ripple on spklvdd, spkrvdd ripple on avdd, dvdd, hpvdd fft, 0dbfs (mic to adc) MAX9888 toc10 frequency (hz) amplitude (dbv) 3500 3000 2500 2000 1500 1000 500 -120 -100 -80 -60 -40 -20 0 -140 0 4000 mclk = 13mhz lrclk = 8khz freq mode av micpre_ = 0db fft, -60dbfs (mic to adc) MAX9888 toc11 frequency (hz) amplitude (dbv) 3500 3000 2500 2000 1500 1000 500 -120 -100 -80 -60 -40 -20 0 -140 0 4000 mclk = 13mhz lrclk = 8khz freq mode av micpre_ = 0db fft, 0dbfs (mic to adc) MAX9888 toc12 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz lrclk = 44.1khz pll mode av micpre_ = 0db datasheet.in stereo audio codec with flexsound technology MAX9888 30 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) fft, -60dbfs (mic to adc) MAX9888 toc13 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz lrclk = 44.1khz pll mode av micpre_ = 0db fft, 0dbfs (mic to adc) MAX9888 toc14 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode av micpre_ = 0db fft, -60dbfs (mic to adc) MAX9888 toc15 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode av micpre_ = 0db fft, 0dbfs (mic to adc) MAX9888 toc16 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 96khz ni mode av micpre_ = 0db datasheet.in stereo audio codec with flexsound technology MAX9888 31 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) fft, -60dbfs (mic to adc) MAX9888 toc17 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 96khz ni mode av micpre_ = 0db software turn-on/off response (mic to adc) MAX9888 toc19 scl 2v/div adc output 0.5v/div 10ms/div adc enable/disable response (mic to adc) MAX9888 toc18 scl 2v/div adc output 0.5v/div 10ms/div datasheet.in stereo audio codec with flexsound technology MAX9888 32 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) line to adc total harmonic distortion plus noise vs. frequency (line to adc) MAX9888 toc20 frequency (hz) thd+n ratio (db) 10,000 1000 100 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode v in = 1.4v p-p av pgain_ = -6db c in = 1f total harmonic distortion plus noise vs. frequency (line to adc) MAX9888 toc21 frequency (hz) thd+n ratio (db) 10,000 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode v in = 1v p-p c in = 1f av pgain_ = 0db total harmonic distortion plus noise vs. frequency (line to adc) MAX9888 toc22 frequency (hz) thd+n ratio (db) 10,000 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode v in = 0.1v p-p av pgain_ = +20db total harmonic distortion plus noise vs. frequency (line to adc) MAX9888 toc23 frequency (hz) thd+n ratio (db) 10,000 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode v in = 1v rms external gain mode r in = 56k i , c in = 1f power-supply rejection ratio vs. frequency (line to adc) MAX9888 toc24 frequency (hz) psrr (db) 10,000 1000 100 -100 -80 -60 -40 -20 0 -120 10 100,000 v ripple = 200mv p-p inputs ac grounded ripple on avdd, dvdd, hpvdd ripple on spklvdd, spkrvdd datasheet.in stereo audio codec with flexsound technology MAX9888 33 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) digital loopback fft, 0dbfs (sdins1 to sdouts2 digital loopback) MAX9888 toc25 frequency (hz) amplitude (dbfs) 15,000 10,000 5000 -160 -140 -120 -100 -80 -60 -40 -20 0 -180 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode fft, -60dbfs (sdins1 to sdouts2 digital loopback) MAX9888 toc26 frequency (hz) amplitude (dbfs) 15,000 10,000 5000 -160 -140 -120 -100 -80 -60 -40 -20 0 -180 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode datasheet.in stereo audio codec with flexsound technology MAX9888 34 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) analog loopback total harmonic distortion plus noise vs. frequency (line to adc to dac to headphone) MAX9888 toc27 frequency (hz) thd+n ratio (db) 10,000 1000 100 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 10 100,000 mclk = 13mhz lrclk = 44.1khz pll mode r hp = 32 i , c in = 1f p out = 0.025w p out = 0.01w total harmonic distortion plus noise vs. frequency (line to adc to dac to headphone ) MAX9888 toc28 frequency (hz) thd+n ratio (db) 10,000 1000 100 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32i, c in = 1f p out = 0.025w p out = 0.01w fft, 0dbfs (line to adc to dac to headphone) MAX9888 toc29 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz lrclk = 44.1khz pll mode r hp = 32 i fft, -60dbfs (line to adc to dac to headphone) MAX9888 toc30 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz lrclk = 44.1khz pll mode r hp = 32 i fft, 0dbfs (line to adc to dac to headphone) MAX9888 toc31 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32i fft, -60dbfs (line to adc to dac to headphone) MAX9888 toc32 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32 i datasheet.in stereo audio codec with flexsound technology MAX9888 35 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) dac to receiver total harmonic distortion vs. output power (dac to receiver) MAX9888 toc33 output power (w) thd+n ratio (db) 0.10 0.08 0.06 0.04 0.02 -80 -70 -60 f = 100hz -50 -40 -30 -20 -10 0 -90 0 0.12 mclk = 13mhz lrclk = 8khz freq mode r rec = 32 i av rec = +8db f = 3000hz f = 1000hz total harmonic distortion vs. frequency (dac to receiver) MAX9888 toc34 frequency (hz) thd+n ratio (db) 1000 100 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 10 10,000 mclk = 13mhz lrclk = 8khz freq mode r rec = 32 i p out = 0.05w p out = 0.025w output power vs. supply voltage (dac to receiver) MAX9888 toc35 supply voltage (v) output power per channel (mw) 5.0 4.5 3.0 3.5 4.0 70 80 90 100 110 120 130 140 60 2.5 5.5 thd+n = 10% thd+n = 1% mclk = 13mhz lrclk = 8khz freq mode r rec = 32 i av rec = +8db gain vs. frequency (dac to receiver) MAX9888 toc36 frequency (hz) normalized gain (db) 1000 100 -4 -3 -2 -1 0 1 2 3 4 5 -5 10 10,000 mclk = 13mhz lrclk = 8khz freq mode r rec = 32 i power consumption vs. output power (dac to receiver) MAX9888 toc37 output power (w) power consumption (w) 0.10 0.08 0.06 0.04 0.02 0.05 0.10 0.15 0.20 0.25 0 0 0.12 mclk = 13mhz lrclk = 8khz freq mode r rec = 32 i av rec = +8db power-supply rejection ratio vs. frequency (dac to receiver) MAX9888 toc38 frequency (hz) psrr (db) 10,000 1000 100 -80 -60 -40 -20 0 -100 10 100,000 v ripple = 200mv p-p all zeros at input ripple on spklvdd, spkrvdd ripple on avdd, dvdd, hpvdd datasheet.in stereo audio codec with flexsound technology MAX9888 36 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) MAX9888 toc39 receiver output 1v/div scl 2v/div 10ms/div software turn-on/ off response (dac to receiver, vsen = 0) MAX9888 toc40 receiver output 1v/div scl 2v/div 10ms/div software turn-on/ off response (dac to receiver, vsen = 1) fft, 0dbfs (dac to receiver) MAX9888 toc41 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz lrclk = 8khz freq mode r rec = 32 i fft, -60dbfs (dac to receiver) MAX9888 toc42 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz lrclk = 8khz freq mode r rec = 32 i frequency (khz) amplitude (dbm) 1000 100 10 1 -100 -80 -60 -40 -20 0 -120 0 10,000 wideband fft, 0dbfs (dac to receiver) MAX9888 toc43 mclk = 13mhz lrclk = 8khz pll mode r rec = 32 i frequency (khz) amplitude (dbm) 1000 100 10 1 -100 -80 -60 -40 -20 0 -120 0 10,000 wideband fft, -60dbfs (dac to receiver) MAX9888 toc44 mclk = 13mhz lrclk = 8khz pll mode r rec = 32 i datasheet.in stereo audio codec with flexsound technology MAX9888 37 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) line to receiver total harmonic distortion plus noise vs. output power (line to receiver) MAX9888 toc45 output power (w) thd+n ratio (db) 0.08 0.06 0.02 0.04 -70 -60 -50 -40 -20 -30 -10 0 -80 0 0.10 r rec = 32 i av rec = +8db f = 6000hz f = 1000hz f = 100hz total harmonic distortion plus noise vs. frequency (line to receiver) MAX9888 toc46 frequency (hz) thd+n ratio (db) 10,000 1000 100 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 10 100,000 r rec = 32 i c in = 1f av rec = +8db p out = 0.025w p out = 0.05w gain vs. frequency (line to receiver) MAX9888 toc47 frequency (hz) normalized gain (db) 10,000 1000 100 -4 -3 -2 -1 0 1 2 3 4 5 -5 10 100,000 r rec = 32 i c in = 1f power-supply rejection ratio vs. frequency (line to receiver) MAX9888 toc48 frequency (hz) psrr (db) 10,000 1000 100 -80 -60 -40 -20 0 -100 10 100,000 v ripple = 200mv p-p input ac grounded ripple on spklvdd, spkrvdd ripple on avdd, dvdd, hpvdd datasheet.in stereo audio codec with flexsound technology MAX9888 38 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) dac to speaker total harmonic distortion plus noise vs. output power (dac to speaker) MAX9888 toc49 output power (w) thd+n ratio (db) 1.0 0.8 0.6 0.4 0.2 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 0 1.2 v spk_vdd = 4.2v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h av spk_ = +8db f = 6000hz f = 1000hz f = 100hz thd+n ratio (db) -70 -60 -50 -40 -30 -20 -10 0 -80 MAX9888 toc50 output power (w) 0.8 0.6 0.4 0.2 0 1.0 total harmonic distortion plus noise vs. output power (dac to speaker) f = 100hz v spk_vdd = 3.7v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h av spk_ = +8db f = 1000hz f = 6000hz total harmonic distortion plus noise vs. output power (dac to speaker) MAX9888 toc51 thd+n ratio (db) -70 -60 -50 -40 -20 -30 -10 -80 0 output power (w) 0.5 0.4 0.3 0.2 0.1 0 0.6 v spk_vdd = 3.v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h av spk_ = +8db f = 6000hz f = 1000hz f = 100hz v spk_vdd = 4.2v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 4i + 33h av spk_ = +8db total harmonic distortion plus noise vs. output power (dac to speaker) MAX9888 toc52 output power (w) thd+n ratio (db) 2.0 1.5 1.0 0.5 0 2.5 -80 -70 -60 -50 -40 -30 -20 -10 0 -90 f = 6000hz f = 1000hz f = 100hz total harmonic distortion plus noise vs. output power (dac to speaker) v spk_vdd = 3.7v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 4 i + 33h av spk_ = +8db MAX9888 toc53 thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 0 -90 output power (w) 1.5 1.0 0.5 0 2.0 f = 6000hz f = 1000hz f = 100hz total harmonic distortion plus noise vs. output power (dac to speaker) MAX9888 toc54 output power (w) 1.2 1.0 0.6 0.8 0.4 0.2 0 1.4 thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 v spk_vdd = 3v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 4i + 33h av spk_ = +8db f = 6000hz f = 1000hz f = 100hz datasheet.in stereo audio codec with flexsound technology MAX9888 39 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (dac to speaker) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc55 v spk_vdd = 4.2v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h av spk_ = +8db p out = 0.25w p out = 0.55w frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (dac to speaker) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc56 v spk_vdd = 3.7v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h av spk_ = +8db p out = 0.25w p out = 0.55w frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (dac to speaker) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc57 v spk_vdd = 4.2v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 4i + 33h p out = 0.5w p out = 1.0w frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (dac to speaker) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc58 v spk_vdd = 3.7v mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 4i + 33h p out = 0.5w p out = 1.0w output power vs. supply voltage (dac to speaker) MAX9888 toc59 supply voltage (v) output power per channel (mw) 5.0 4.5 4.0 3.5 3.0 600 800 1000 1200 1400 1600 1800 2000 2200 400 2.5 5.5 mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h thd+n = 10% thd+n = 10% thd+n = 10% thd+n = 10% thd+n = 1% output power vs. supply voltage (dac to speaker) MAX9888 toc60 output power per channel (mw) 500 1000 1500 2000 2500 3000 3500 0 supply voltage (v) 5.0 4.5 4.0 3.5 3.0 2.5 5.5 mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 4i + 33h thd+n = 10% thd+n = 1% datasheet.in stereo audio codec with flexsound technology MAX9888 40 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) gain vs. frequency (dac to speaker) MAX9888 toc61 normalized gain (db) -4 -3 -2 -1 0 1 2 3 4 5 -5 mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h frequency (hz) 10,000 1000 100 10 100,000 efficiency (%) 10 20 30 40 50 60 70 80 90 100 0 efficiency vs. output power (dac to speaker) MAX9888 toc62 output power per channel (w) 0.5 1.0 1.5 2.0 0 2.5 v spk_vdd = 4.2v mclk = 12.288mhz, lrclk = 48khz ni mode av spk_ = +8db z spk = 4i + 33h z spk = 4i + 33h z spk = 8i + 68h efficiency vs. output power (dac to speaker) MAX9888 toc63 output power per channel (w) 1.4 1.2 0.8 1.0 0.4 0.6 0.2 0 1.6 efficiency (%) 10 20 30 40 50 60 70 80 90 100 0 v spk_vdd = 3.7v mclk = 12.288mhz, lrclk = 48khz ni mode av spk_ = +8db z spk = 4i + 33h z spk = 8i + 68h supply current vs. supply voltage (dac to speaker) MAX9888 toc64 spk_vdd supply voltage (v) spk_vdd supply current (ma) 5.0 4.5 4.0 3.5 3.0 5 10 15 20 25 30 0 2.5 5.5 mclk = 12.288mhz, lrclk = 48khz z spk = 8i + 68h ni mode av spk_ = +8db all zeros at input power-supply rejection ratio vs. frequency (dac to speaker) MAX9888 toc65 psrr (db) -80 -60 -40 -20 0 -100 frequency (hz) 10,000 1000 100 10 100,000 v ripple = 200mv p-p ripple on spklvdd, spkrvdd ripple on avdd, dvdd, hpvdd power-supply rejection ratio vs. supply voltage (dac to speaker) MAX9888 toc66 supply voltage (v) 5.0 4.5 4.0 3.5 3.0 2.5 5.5 psrr (db) -80 -60 -40 -20 -100 0 ripple on spklvdd, spkrvdd v ripple = 200mv p-p f = 1khz datasheet.in stereo audio codec with flexsound technology MAX9888 41 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) frequency (hz) 10,000 1000 100 10 100,000 crosstalk vs. frequency (dac to speaker) crosstalk (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc67 mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h software turn-o n/ off response (dac to speaker, vsen = 0) MAX9888 toc68 scl 2v/div speaker output 1v/div 10ms/div software turn-o n/ off response (dac to speaker, vsen = 1) MAX9888 toc69 scl 2v/div speaker output 1v/div 10ms/div fft, -60dbfs (dac to speaker) MAX9888 toc70 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz, lrclk = 48khz ni mode z spk = 8i + 68h fft, -60dbfs (dac to speaker) MAX9888 toc71 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz, lrclk = 44.1khz pll mode z spk = 8i + 68h wideband fft (dac to speaker) MAX9888 toc72 frequency (mhz) amplitude (dbm) 10 -50 -40 -30 -20 -10 0 10 20 -60 1 100 mclk = 13mhz, lrclk = 44.1khz pll mode z spk = 8i + 68h datasheet.in stereo audio codec with flexsound technology MAX9888 42 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) line to speaker total harmonic distortion plus noise vs. output power (line to speaker) output power (w) 0.8 0.6 0.2 0.4 0 MAX9888 toc73 thd+n ratio (db) -70 -60 -50 -40 -20 -30 -10 0 -80 z spk = 8i + 68h av spk_ = +8db f = 6000hz f = 1000hz f = 100hz frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (line to speaker) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc74 z spk = 8i + 68h c in = 1f av spk_ = +8db p out = 0.55w p out = 0.25w gain vs. frequency (line to speaker) MAX9888 toc75 normalized gain (db) -4 -3 -2 -1 0 1 2 3 4 5 -5 frequency (hz) 10,000 1000 100 10 100,000 z spk = 8i + 68h c in = 1f power-supply rejection ratio vs. frequency (line to speaker) MAX9888 toc76 psrr (db) -80 -60 -40 -20 0 -100 frequency (hz) 10,000 1000 100 10 100,000 v ripple = 200mv rms input ac grounded ripple on avdd, dvdd, hpvdd ripple on spklvdd, spkrvdd frequency (hz) 10,000 1000 100 10 100,000 crosstalk vs. frequency (line to speaker) crosstalk (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc77 z spk = 8i + 68h datasheet.in stereo audio codec with flexsound technology MAX9888 43 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) dac to headphone total harmonic distortion plus noise vs. output power (dac to headphone) MAX9888 toc78 output power (w) thd+n ratio (db) 0.04 0.03 0.02 0.01 0 0.05 -80 -70 -60 -50 -40 -30 -20 -10 -90 0 mclk = 13mhz lrclk = 8khz freq mode r hp = 32i av hp_ = +3db f = 3000hz f = 100hz f = 1000hz total harmonic distortion plus noise vs. output power (dac to headphone) MAX9888 toc79 output power (w) thd+n ratio (db) 0.04 0.03 0.02 0.01 0 0.05 -80 -70 -60 -50 -40 -30 -20 -10 -90 0 mclk = 13mhz lrclk = 44.1khz pll mode r hp = 32i av hp_ = +3db f = 6000hz f = 100hz f = 1000hz total harmonic distortion plus noise vs. output power (dac to headphone) MAX9888 toc80 output power (w) thd+n ratio (db) 0.04 0.03 0.02 0.01 0 0.05 -80 -70 -60 -50 -40 -30 -20 -10 -90 0 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32i av hp_ = +3db f = 6000hz f = 100hz f = 1000hz total harmonic distortion plus noise vs. output power (dac to headphone) MAX9888 toc81 output power (w) thd+n ratio (db) 0.04 0.03 0.02 0.01 0 0.05 -80 -70 -60 -50 -40 -30 -20 -10 -90 0 mclk = 12.288mhz lrclk = 96khz ni mode r hp = 32i av hp_ = +3db f = 6000hz f = 100hz f = 1000hz MAX9888 toc82 output power (w) 0.07 0.06 0.04 0.05 0.02 0.03 0.01 0 0.08 total harmonic distortion plus noise vs. output power (dac to headphone) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 16i av hp_ = +3db f = 6000hz f = 100hz f = 1000hz frequency (hz) 1000 100 10 10,000 total harmonic distortion plus noise vs. frequency (line to speaker) MAX9888 toc83 mclk = 13mhz lrclk = 8khz freq mode r hp = 32i av hp_ = +3db thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 p out = 0.01w p out = 0.02w datasheet.in stereo audio codec with flexsound technology MAX9888 44 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (dac to headphone) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc84 p out = 0.025w p out = 0.1w mclk = 13mhz lrclk = 44.1khz pll mode r hp = 32i av hp_ = +3db total harmonic distortion plus noise vs. frequency (dac to headphone) MAX9888 toc85 frequency (hz) thd+n ratio (db) 10,000 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32i av hp_ = +3db p out = 0.025w p out = 0.01w total harmonic distortion plus noise vs. frequency (dac to headphone) MAX9888 toc86 frequency (hz) thd+n ratio (db) 10,000 1000 100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 -100 10 100,000 mclk = 12.288mhz lrclk = 96khz ni mode r hp = 32i av hp_ = +3db p out = 0.025w p out = 0.01w frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (dac to headphone) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc87 p out = 0.01w p out = 0.0.25w mclk = 12.288mhz lrclk = 48khz ni mode r hp = 16i gain vs. frequency (dac to headphone) MAX9888 toc88 normalized gain (db) -60 -50 -40 -30 -20 -10 0 10 -70 frequency (hz) 10,000 1000 100 10 100,000 mclk = 13mhz lrclk = 8khz ni mode r hp = 32i mode = 1 mode = 0 hpvdd input current vs. output power (dac to headphone) MAX9888 toc89 output power per channel (mw) hpvdd input current (ma) 10 1 0.1 20 40 60 80 100 120 0 0.01 100 r hp = 16i r hp = 32i mclk = 12.288mhz lrclk = 48khz ni mode datasheet.in stereo audio codec with flexsound technology MAX9888 45 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) power-supply rejection ratio vs. frequency (dac to headphone) MAX9888 toc90 psrr (db) -80 -60 -40 -20 0 -100 frequency (hz) 10,000 1000 100 10 100,000 v ripple = 200mv p-p input all zeros ripple on spklvdd, spkrvdd ripple on avdd, dvdd, hpvdd crosstalk vs. frequency (dac to headphone) MAX9888 toc91 crosstalk (db) -90 -80 -70 -60 -50 -40 -100 frequency (hz) 10,000 1000 100 10 100,000 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32i software turn-o n/ off response (dac to headphone, vsen = 0) MAX9888 toc92 scl 2v/div headphone output 0.5v/div 10ms /div software turn-o n/ off response (dac to headphone, vsen = 1) MAX9888 toc93 scl 2v/div headphone output 0.5v/div 10ms /div fft, 0dbfs (dac to headphone) MAX9888 toc94 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz, lrclk = 8khz freq mode r hp = 32i fft, -60dbfs (dac to headphone) MAX9888 toc95 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -140 -120 -100 -80 -60 -40 -20 0 -160 0 20,000 mclk = 13mhz, lrclk = 8khz freq mode r hp = 32i datasheet.in stereo audio codec with flexsound technology MAX9888 46 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) fft, -60dbfs (dac to headphone) MAX9888 toc97 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -140 -120 -100 -80 -60 -40 -20 0 -160 0 20,000 mclk = 13mhz, lrclk = 44.1khz pll mode r hp = 32i fft, 0dbfs (dac to headphone) MAX9888 toc98 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32i fft, 0dbfs (dac to headphone) MAX9888 toc96 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 13mhz, lrclk = 44.1khz pll mode r hp = 32i datasheet.in stereo audio codec with flexsound technology MAX9888 47 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) fft, 0dbfs (dac to headphone) MAX9888 toc100 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -120 -100 -80 -60 -40 -20 0 -140 0 20,000 mclk = 12.288mhz lrclk = 96khz ni mode r hp = 32i fft, -60dbfs (dac to headphone) MAX9888 toc101 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -140 -120 -100 -80 -60 -40 -20 0 -160 0 20,000 mclk = 2.288mhz lrclk = 96khz ni mode r hp = 32i wideband fft, 0dbfs (dac to headphone) frequency (khz) amplitude (dbm) 1000 100 10 1 -80 -60 -40 -20 0 20 -100 0 10,000 MAX9888 toc102 mclk = 13mhz lrclk = 44.1khz pll mode a vhp_ = -3db r hp = 32i fft, -60dbfs (dac to headphone) MAX9888 toc99 frequency (hz) amplitude (dbv) 15,000 10,000 5000 -140 -120 -100 -80 -60 -40 -20 0 -160 0 20,000 mclk = 12.288mhz lrclk = 48khz ni mode r hp = 32i wideband fft, -60dbfs (dac to headphone) frequency (khz) amplitude (dbm) 1000 100 10 1 -80 -60 -40 -20 0 20 -100 0 10,000 MAX9888 toc103 mclk = 13mhz lrclk = 44.1khz pll mode a vhp_ = -3db r hp = 32i datasheet.in stereo audio codec with flexsound technology MAX9888 48 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) line to headphone total harmonic distortion plus noise vs. output power (line to headphone) output power (w) thd+n ratio (db) 0.04 0.03 0.01 0.02 -70 -60 -50 -40 -20 -30 -10 0 -80 0 0.05 MAX9888 toc104 r hp = 32i a vhp_ = +3db f = 6000hz f = 100hz f = 1000hz frequency (hz) 10,000 1000 100 10 100,000 total harmonic distortion plus noise vs. frequency (line to headphone) thd+n ratio (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc105 r hp = 32i c in = 1f p out = 0.01w p out = 0.025w gain vs. frequency (line to headphone) MAX9888 toc106 normalized gain (db) -4 -3 -2 -1 0 1 2 3 4 5 -5 frequency (hz) 10,000 1000 100 10 100,000 r hp = 32i c in = 1f power-supply rejection ratio vs. frequency (line to headphone) MAX9888 toc107 psrr (db) -80 -60 -40 -20 0 -100 frequency (hz) 10,000 1000 100 10 100,000 v ripple = 200mv p-p ripple on spklvdd, spkrvdd ripple on avdd, dvdd, hpvdd frequency (hz) 10,000 1000 100 10 100,000 crosstalk vs. frequency (line to headphone) crosstalk (db) -80 -70 -60 -50 -40 -30 -20 -10 -90 0 MAX9888 toc108 r hp = 32i common-mode rejection ratio vs. frequency (line to headphone) MAX9888 toc109 cmrr (db) 10 20 30 40 50 60 70 0 frequency (hz) 10,000 1000 100 10 100,000 v out = -6dbv c in = 1f r hp = 32i av pgain_ = 20db av pgain_ = 0db datasheet.in stereo audio codec with flexsound technology MAX9888 49 typical operating characteristics (continued) (v avdd = v hpvdd = v dvdd = v dvdds1 = v dvdds2 = +1.8v, v spklvdd = v spkrvdd = 3.7v. speaker loads (z spk ) connected between spk_p and spk_n. receiver load (r rec ) connected between recp and recn. headphone loads (r hp ) connected from hpl or hpr to gnd. r hp = j , r rec = j , z spk = j , c ref = 2.2 f f, c micbias = c preg = c reg = 1 f f, c c1n-c1p = 1 f f, c hpvss = 1 f f. av micpre_ = +20db, av micpga_ = 0db, av dacattn = 0db, av dacgain = 0db, av adclvl = 0db, av adcgain = 0db, av pgain_ = 0db, av pgaout_ = 0db, av hp_ = 0db, av rec = 0db, av spk_ = 0db, mclk = 12.288mhz, lrclk = 48khz, mas = 1. t a = +25 n c, unless otherwise noted.) speaker bypass switch total harmonic distortion plus noise vs. output power (speaker bypass switch) output power (w) thd+n ratio (db) 0.15 0.20 0.10 0.05 -70 -60 -50 -40 -30 -20 -10 0 -80 0 MAX9888 toc110 receiver amplifier driving loudspeaker z spk = 8i + 68h f = 1000hz f = 100hz f = 6000hz on resistance vs. v com (speaker bypass switch) MAX9888 toc111 v com (v) r on (i) 5 4 1 2 3 0.5 1.0 1.5 2.0 2.5 3.0 3.5 i sw = 20ma v spk_vdd = 3.0v v spk_vdd = 3.7v v spk_vdd = 4.2v v spk_vdd = 5.0v 4.0 0 0 6 off-isolation vs. frequency (speaker bypass switch) MAX9888 toc112 frequency (hz) off-isolation (db) 10,000 1000 100 -100 -80 -60 -40 -20 0 -120 10 100,000 speaker amp driving loudspeaker speaker bypass switch open measured at rxin_ 50 i load on rxin_ receiver amp driving rxin_ datasheet.in stereo audio codec with flexsound technology MAX9888 50 pin configuration top view (bump side down) 1 2 3 4 5 6 7 8 9 b c d e f g MAX9888 a mclk irq spkrvdd spkrvdd sdins1 spklgnd spkrp spklgnd hpsns jacksns scl sda reg avdd dvdd preg ref agnd micbias spklvdd spklp spkrgnd spkln recn/ rxinn spklvdd spklp spkrgnd spkln recp/ rxinp hpvdd hpgnd spkrp dgnd spkrn spkrn hpr hpl hpvss c1p c1n n.c. inb2 bclks1 lrclks1 n.c. n.c. n.c. n.c. n.c. inb1 ina2/ extmicn dvdds1 sdouts1 mic1p/ digmicdata ina1/ extmicp bclks2 lrclks2 mic1n/ digmicclk mic2p sdouts2 dvdds2 sdins2 mic2n datasheet.in stereo audio codec with flexsound technology MAX9888 51 pin description pin name function a1, b1 spkrn negative right-channel class d speaker output a2, b2 spkrgnd right-speaker ground a3, b3 spklvdd left-speaker, ref, receiver amplifier power supply. bypass to spklgnd with a 1ff and a 10ff capacitor. a4, b4 spklp positive left-channel class d speaker output a5, b5 spkln negative left-channel class d speaker output a6 recp/rxinp positive receiver amplifier output. can be positive bypass switch input when receiver amp is shut down. a7 hpvdd headphone power supply. bypass to hpgnd with a 1ff capacitor. a8 hpgnd headphone ground a9 hpvss inverting charge-pump output. bypass to hpgnd with a 1ff ceramic capacitor. b6 recn/rxinn negative receiver amplifier output. can be negative bypass switch input when receiver amp is shut down. b7 c1p charge-pump flying capacitor positive terminal. connect a 1ff ceramic capacitor between c1n and c1p. b8 c1n charge-pump flying capacitor negative terminal. connect a 1ff ceramic capacitor between c1n and c1p. b9 hpl left-channel headphone output c1, c2 spkrp positive right-channel class d speaker output c3, d3 spkrvdd right-speaker power supply. bypass to spkrgnd with a 1ff capacitor. c4, c5 spklgnd left-speaker ground c6 hpsns headphone amplifier ground sense. connect to the headphone jack ground terminal or connect to ground. c7, d5, d7, e3, e6, e7 n.c. no connection c8 inb2 single-ended line input b2. also positive differential line input b. c9 hpr right-channel headphone output d1 bclks1 s1 digital audio bit clock input/output. bclks1 is an input when the MAX9888 is in slave mode and an output when in master mode. the input/output voltage is referenced to dvdds1. d2 lrclks1 s1 digital audio left-right clock input/output. lrclks1 is the audio sample rate clock and determines whether s1 audio data is routed to the left or right channel. in tdm mode, lrclks1 is a frame sync pulse. lrclks1 is an input when the MAX9888 is in slave mode and an output when in master mode. the input/output voltage is referenced to dvdds1. d4 sdins1 s1 digital audio serial-data dac input. the input voltage is referenced to dvdds1. d6 jacksns jack sense. detects the insertion of a jack. see the headset detection section. d8 inb1 single-ended line input b1. also negative differential line input b. d9 ina2/ extmicn single-ended line input a2. also positive differential line input a or negative differential external microphone input. datasheet.in stereo audio codec with flexsound technology MAX9888 52 pin description (continued) pin name function e1 dvdds1 s1 digital audio interface power-supply input. bypass to dgnd with a 1ff capacitor. e2 mclk master clock input. acceptable input frequency range is 10mhz to 60mhz. e4 sdouts1 s1 digital audio serial-data adc output. the output voltage is referenced to dvdds1. e5 irq hardware interrupt output. irq can be programmed to pull low when bits in status register 0x00 change state. read status register 0x00 to clear irq once set. repeat faults have no effect on irq until it is cleared by reading the i 2 c status register 0x00. connect a 10ki pullup resistor to dvdd for full output swing. e8 mic1p/ digmicdata positive differential microphone 1 input. ac-couple a microphone with a series 1ff capacitor. can be retasked as a digital microphone data input. e9 ina1/ extmicp single-ended line input a1. also negative differential line input a or positive differential external microphone input. f1 dgnd digital ground f2 bclks2 s2 digital audio bit clock input/output. bclks2 is an input when the ic is in slave mode and an output when in master mode. the input/output voltage is referenced to dvdds2. f3 lrclks2 s2 digital audio left-right clock input/output. lrclks2 is the audio sample rate clock and determines whether audio data on s2 is routed to the left or right channel. in tdm mode, lrclks2 is a frame sync pulse. lrclks2 is an input when the ic is in slave mode and an output when in master mode. the input/output voltage is referenced to dvdds2. f4 sda i 2 c serial-data input/output. connect a pullup resistor to dvdd for full output swing. f5 scl i 2 c serial-clock input f6 reg common-mode voltage reference. bypass to agnd with a 1ff capacitor. f7 ref converter reference. bypass to agnd with a 2.2ff capacitor. f8 mic1n/ digmicclk negative differential microphone 1 input. ac-couple a microphone with a series 1ff capacitor. can be retasked as a digital microphone clock output. f9 mic2p positive differential microphone 2 input. ac-couple a microphone with a series 1ff capacitor. g1 sdouts2 s2 digital audio serial-data adc output. the output voltage is referenced to dvdds2. g2 dvdds2 s2 digital audio interface power-supply input. bypass to dgnd with a 1ff capacitor. g3 sdins2 s2 digital audio serial-data dac input. the input voltage is referenced to dvdds2. g4 dvdd digital power supply. supply for the digital core and i 2 c interface. bypass to dgnd with a 1ff capacitor. g5 avdd analog power supply. bypass to agnd with a 1ff capacitor. g6 preg positive internal regulated supply. bypass to agnd with a 1ff capacitor. g7 agnd analog ground g8 micbias low-noise bias voltage. outputs a 2.2v microphone bias. an external resistor in the 2.2ki to 1ki range should be used to set the microphone current. g9 mic2n negative differential microphone 2 input. ac-couple a microphone with a series 1ff capacitor. datasheet.in stereo audio codec with flexsound technology MAX9888 53 detailed description the MAX9888 is a fully integrated stereo audio codec with flexsound technology and integrated amplifiers. two differential microphone amplifiers can accept sig - nals from three analog inputs. one input can be retasked to support two digital microphones. any combination of two microphones (analog or digital) can be recorded simultaneously. the analog signals are amplified up to 50db and recorded by the stereo adc. the digital record path supports voice filtering with selectable preset highpass filters and high stopband attenuation at f s /2. an automatic gain control (agc) circuit moni - tors the digitized signal and automatically adjusts the analog microphone gain to make best use of the adcs dynamic range. a noise gate attenuates signals below the user-defined threshold to minimize the noise output by the adc. the ic includes two analog line inputs. one of the line inputs can be optionally retasked as a third analog micro - phone input. both line inputs support either stereo single- ended input signals or mono differential signals. the line inputs are preamplified and then routed either to the adc for recording or to the output amplifiers for playback. integrated analog switches allow two differential micro - phone signals to be routed out the third microphone input to an external device. this eliminates the need for an external analog switch in systems that have two devices recording signals from the same microphone. through two digital audio interfaces, the device can transmit one stereo audio signal and receive two stereo audio signals in a wide range of formats including i 2 s, pcm, and up to four mono slots in tdm. each interface can be connected to either of two audio ports (s1 and s2) for communication with external devices. both audio interfaces support 8khz to 96khz sample rates. each input signal is independently equalized using 5-band parametric equalizers. a multiband automatic level control (alc) boosts signals by up to 12db. one signal path additionally supports the same voiceband filtering as the adc path. the ic includes a differential receiver amplifier, stereo class d speaker amplifiers, and directdrive true ground stereo headphone amplifiers. when the receiver amplifier is disabled, analog switches allow recp/rxinp and recn/rxinn to be reused for signal routing. in systems where a single transducer is used for both the loudspeaker and receiver, an exter - nal receiver amplifier can be routed to the left speaker through recp/rxinp and recn/rxinn, bypassing the class d amplifier, to connect to the loudspeaker. if the internal receiver amplifier is used, then leave recp/ rxinp and recn/rxinn unconnected. in systems where an external amplifier drives both the receiver and the MAX9888s input, one of the differential signals can be disconnected from the receiver when not needed by passing it through the analog switch that connects recp/rxinp to recn/rxinn. the stereo class d amplifier provides efficient amplifi - cation for two speakers. the amplifier includes active emissions limiting to minimize the radiated emissions (emi) traditionally associated with class d. in most systems, no output filtering is required to meet standard emi limits. to optimize speaker sound quality, the ic includes an excursion limiter, a distortion limiter, and a power limiter. the excursion limiter is a dynamic highpass filter with variable corner frequency that increases in response to high signal levels. low-frequency energy typically causes more distortion than useful sound at high sig - nal levels, so attenuating low frequencies allows the speaker to play louder without distortion or damage. at lower signal levels, the filter corner frequency reduces to pass more low frequency energy when the speaker can handle it. the distortion limiter reduces the volume when the output signal exceeds a preset distortion level. this ensures that regardless of input signal and battery voltage, excessive distortion is never heard by the user. the power limiter monitors the continuous power into the loudspeaker and lowers the signal level if the speaker is at risk of overheating. the stereo directdrive headphone amplifier uses an inverting charge pump to generate a ground-referenced output signal. this eliminates the need for dc-blocking capacitors or a midrail bias for the headphone jack ground return. ground sense reduces output noise caused by ground return current. the ic integrates jack detection allowing the detection of insertion and removal of accessories as well as button presses. datasheet.in stereo audio codec with flexsound technology MAX9888 54 table 1. register map i 2 c slave address configure the MAX9888 using the i 2 c control bus. the ic uses a slave address of 0x20 or 00100000 for write operations and 0x21 or 00100001 for read operations. see the i 2 c serial interface section for a complete inter - face description. registers table 1 lists all of the registers, their addresses, and power-on-reset states. registers 0x00 to 0x03 and 0xff are read-only while all of the other registers are read/ write. write zeros to all unused bits in the register table when updating the register, unless otherwise noted. register b7 b6 b5 b4 b3 b2 b1 b0 address default r/w page status status cld sld ulk jdet 0x00 r 103 microphone agc/ng ng agc 0x01 r 65 jack status jksns 0x02 r 101 battery voltage vbat 0x03 r/w 102 interrupt enable icld isld iulk 0 0 0 ijdet 0 0x0f 0x00 r/w 103 master clock control master clock 0 0 psclk 0 0 0 0 0x10 0x00 r/w 76 dai1 clock control clock mode sr1 freq1 0x11 0x00 r/w 76 any clock control pll1 ni1[14:8] 0x12 0x00 r/w 77 ni1[7:1] ni1[0] 0x13 0x00 r/w 77 dai1 configuration format mas1 wci1 bci1 dly1 0 tdm1 fsw1 ws1 0x14 0x00 r/w 71 clock osr1 0 0 0 bsel1 0x15 0x00 r/w 72 i/o configuration sel1 lten1 lben1 dmono1 hizoff1 sdoen1 sdien1 0x16 0x00 r/w 72 time-division multiplex slotl1 slotr1 slotdly1 0x17 0x00 r/w 73 filters mode1 avflt1 dhf1 dvflt1 0x18 0x00 r/w 79 dai2 clock control clock mode sr2 0 0 0 0 0x19 0x00 r/w 76 any clock control pll2 ni2[14:8] 0x1a 0x00 r/w 77 ni2[7:1] ni2[0] 0x1b 0x00 r/w 77 dai2 configuration format mas2 wci2 bci2 dly2 0 tdm2 fsw2 ws2 0x1c 0x00 r/w 71 clock 0 0 0 0 0 bsel2 0x1d 0x00 r/w 72 i/o configuration sel2 0 lben2 dmono2 hizoff2 sdoen2 sdien2 0x1e 0x00 r/w 72 datasheet.in stereo audio codec with flexsound technology MAX9888 55 table 1. register map (continued) register b7 b6 b5 b4 b3 b2 b1 b0 address default r/w page time-division multiplex slotl2 slotr2 slotdly2 0x1f 0x00 r/w 73 filters 0 0 0 0 dhf2 0 0 dcb2 0x20 0x00 r/w 79 mixers dac mixer mixdal mixdar 0x21 0x00 r/w 85 left adc mixer mixadl 0x22 0x00 r/w 64 right adc mixer mixadr 0x23 0x00 r/w 64 preoutput 1 mixer 0 0 0 0 mixout1 0x24 0x00 r/w 86 preoutput 2 mixer 0 0 0 0 mixout2 0x25 0x00 r/w 86 preoutput 3 mixer 0 0 0 0 mixout3 0x26 0x00 r/w 86 headphone amplifier mixer mixhpl mixhpr 0x27 0x00 r/w 97 receiver amplifier mixer 0 0 0 0 mixrec 0x28 0x00 r/w 88 speaker amplifier mixer mixspl mixspr 0x29 0x00 r/w 90 level control sidetone dsts 0 dvst 0x2a 0x00 r/w 69 dai1 playback level dv1m 0 dv1g dv1 0x2b 0x00 r/w 84 dai1 playback level 0 0 0 eqclp1 dveq1 0x2c 0x00 r/w 83 dai2 playback level dv2m 0 0 0 dv2 0x2d 0x00 r/w 84 dai2 playback level 0 0 0 eqclp2 dveq2 0x2e 0x00 r/w 83 left adc level 0 0 avlg avl 0x2f 0x00 r/w 68 right adc level 0 0 avrg avr 0x30 0x00 r/w 68 microphone 1 input level 0 pa1en pgam1 0x31 0x00 r/w 61 datasheet.in stereo audio codec with flexsound technology MAX9888 56 table 1. register map (continued) register b7 b6 b5 b4 b3 b2 b1 b0 address default r/w page microphone 2 input level 0 pa2en pgam2 0x32 0x00 r/w 61 ina input level 0 inaext 0 0 0 pgaina 0x33 0x00 r/w 63 inb input level 0 inbext 0 0 0 pgainb 0x34 0x00 r/w 63 preoutput 1 level 0 0 0 0 pgaout1 0x35 0x00 r/w 87 preoutput 2 level 0 0 0 0 pgaout2 0x36 0x00 r/w 87 preoutput 3 level 0 0 0 0 pgaout3 0x37 0x00 r/w 87 left headphone amplifier volume control hplm 0 0 hpvoll 0x38 0x00 r/w 97 right headphone amplifier volume control hprm 0 0 hpvolr 0x39 0x00 r/w 97 receiver amplifier volume control recm 0 0 recvol 0x3a 0x00 r/w 88 left speaker amplifier volume control splm 0 0 spvoll 0x3b 0x00 r/w 90 right speaker amplifier volume control sprm 0 0 spvolr 0x3c 0x00 r/w 90 microphone agc configuration agcsrc agcrls agcatk agchld 0x3d 0x00 r/w 65 threshold anth agcth 0x3e 0x00 r/w 66 speaker signal processing excursion limiter filter 0 dhpucf 0 0 dhplcf 0x3f 0x00 r/w 92 excursion limiter threshold 0 0 0 0 0 dhpth 0x40 0x00 r/w 92 alc alcen alcrls alcmb alcth 0x41 0x00 r/w 82 datasheet.in stereo audio codec with flexsound technology MAX9888 57 table 1. register map (continued) register b7 b6 b5 b4 b3 b2 b1 b0 address default r/w page power limiter pwrth 0 pwrk 0x42 0x00 r/w 93 power limiter pwrt2 pwrt1 0x43 0x00 r/w 94 distortion limiter thdclp 0 thdt1 0x44 0x00 r/w 95 configuration audio input inadiff inbdiff 0 0 0 0 0 0 0x45 0x00 r/w 63 microphone micclk digmicl digmicr 0 0 extmic 0x46 0x00 r/w 61 level control vs2en vsen zden 0 0 0 eq2en eq1en 0x47 0x00 r/w 99, 83 bypass switches inabyp 0 0 mic2byp 0 0 recbyp spkbyp 0x48 0x00 r/w 62, 98 jack detection jdeten 0 0 0 0 0 jdeb 0x49 0x00 r/w 101 power management input enable inaen inben 0 0 mben 0 adlen adren 0x4a 0x00 r/w 59 output enable hplen hpren splen spren recen 0 dalen daren 0x4b 0x00 r/w 59 system enable shdn vbaten 0 0 0 0 jdwk 0 0x4c 0x00 r/w 59 dsp coefficients eq band 1 (dai1/dai2) k_1[15:8] 0x50/0x82 0xxx r/w 82 k_1[7:0] 0x51/0x83 0xxx r/w 82 k1_1[15:8] 0x52/0x84 0xxx r/w 82 k1_1[7:0] 0x53/0x85 0xxx r/w 82 k2_1[15:8] 0x54/0x86 0xxx r/w 82 k2_1[7:0] 0x55/0x87 0xxx r/w 82 c1_1[15:8] 0x56/0x88 0xxx r/w 82 c1_1[7:0] 0x57/0x89 0xxx r/w 82 c2_1[15:8] 0x58/0x8a 0xxx r/w 82 c2_1[7:0] 0x59/0x8b 0xxx r/w 82 eq band 2 (dai1/dai2) k_2[15:8] 0x5a/0x8c 0xxx r/w 82 k_2[7:0] 0x5b/0x8d 0xxx r/w 82 k1_2[15:8] 0x5c/0x8e 0xxx r/w 82 k1_2[7:0] 0x5d/0x8f 0xxx r/w 82 k2_2[15:8] 0x5e/0x90 0xxx r/w 82 k2_2[7:0] 0x5f/0x91 0xxx r/w 82 c1_2[15:8] 0x60/0x92 0xxx r/w 82 c1_2[7:0] 0x61/0x93 0xxx r/w 82 c2_2[15:8] 0x62/0x94 0xxx r/w 82 c2_2[7:0] 0x63/0x95 0xxx r/w 82 datasheet.in stereo audio codec with flexsound technology MAX9888 58 table 1. register map (continued) register b7 b6 b5 b4 b3 b2 b1 b0 address default r/w page eq band 3 (dai1/dai2) k_3[15:8] 0x64/0x96 0xxx r/w 82 k_3[7:0] 0x65/0x97 0xxx r/w 82 k1_3[15:8] 0x66/0x98 0xxx r/w 82 k1_3[7:0] 0x67/0x99 0xxx r/w 82 k2_3[15:8] 0x68/0x9a 0xxx r/w 82 k2_3[7:0] 0x69/0x9b 0xxx r/w 82 c1_3[15:8] 0x6a/0x9c 0xxx r/w 82 c1_3[7:0] 0x6b/0x9d 0xxx r/w 82 c2_3[15:8] 0x6c/0x9e 0xxx r/w 82 c2_3[7:0] 0x6d/0x9f 0xxx r/w 82 eq band 4 (dai1/dai2) k_4[15:8] 0x6e/0xa0 0xxx r/w 82 k_4[7:0] 0x6f/0xa1 0xxx r/w 82 k1_4[15:8] 0x70/0xa2 0xxx r/w 82 k1_4[7:0] 0x71/0xa3 0xxx r/w 82 k2_4[15:8] 0x72/0xa4 0xxx r/w 82 k2_4[7:0] 0x73/0xa5 0xxx r/w 82 c1_4[15:8] 0x74/0xa6 0xxx r/w 82 c1_4[7:0] 0x75/0xa7 0xxx r/w 82 c2_4[15:8] 0x76/0xa8 0xxx r/w 82 c2_4[7:0] 0x77/0xa9 0xxx r/w 82 eq band 5 (dai1/dai2) k_5[15:8] 0x78/0xaa 0xxx r/w 82 k_5[7:0] 0x79/0xab 0xxx r/w 82 k1_5[15:8] 0x7a/0xac 0xxx r/w 82 k1_5[7:0] 0x7b/0xad 0xxx r/w 82 k2_5[15:8] 0x7c/0xae 0xxx r/w 82 k2_5[7:0] 0x7d/0xaf 0xxx r/w 82 c1_5[15:8] 0x7e/0xb0 0xxx r/w 82 c1_5[7:0] 0x7f/0xb1 0xxx r/w 82 c2_5[15:8] 0x80/0xb2 0xxx r/w 82 c2_5[7:0] 0x81/0xb3 0xxx r/w 82 excursion limiter biquad (dai1/dai2) a1[15:8] 0xb4/0xbe 0xxx r/w 91 a1[7:0] 0xb5/0xbf 0xxx r/w 91 a2[15:8] 0xb6/0xc0 0xxx r/w 91 a2[7:0] 0xb7/0xc1 0xxx r/w 91 b0[15:8] 0xb8/0xc2 0xxx r/w 91 b0[7:0] 0xb9/0xc3 0xxx r/w 91 b1[15:8] 0xba/0xc4 0xxx r/w 91 b1[7:0] 0xbb/0xc5 0xxx r/w 91 b2[15:8] 0xbc/0xc6 0xxx r/w 91 b2[7:0] 0xbd/0xc7 0xxx r/w 91 revision id rev id rev 0xff 0x43 r 104 datasheet.in stereo audio codec with flexsound technology MAX9888 59 power management the ic includes comprehensive power management to allow the disabling of all unused circuits, minimizing supply current. table 2. power management registers register bit name description 0x4c 7 shdn global shutdown disables everything except the headset detection circuitry, which is controlled separately. 0 = device shutdown 1 = device enabled 6 vbaten see the battery measurement section. 1 jdwk see the headset detection section. 0x4a 7 inaen line input a enable 0 = disabled 1 = enabled 6 inben line input b enable 0 = disabled 1 = enabled 3 mben microphone bias enable 0 = disabled 1 = enabled 1 adlen left adc enable 0 = disabled 1 = enabled 0 adren right adc enable 0 = disabled 1 = enabled 0x4b 7 hplen left headphone enable 0 = disabled 1 = enabled 6 hpren right headphone enable 0 = disabled 1 = enabled 5 splen left speaker enable 0 = disabled 1 = enabled 4 spren right speaker enable 0 = disabled 1 = enabled 3 recen receiver enable 0 = disabled 1 = enabled 1 dalen left dac enable 0 = disabled 1 = enabled 0 daren right dac enable 0 = disabled 1 = enabled datasheet.in stereo audio codec with flexsound technology MAX9888 60 microphone inputs the device includes three differential microphone inputs and a low-noise microphone bias for powering the micro - phones (figure 6). one microphone input can also be con - figured as a digital microphone input accepting signals from up to two digital microphones. two microphones, analog or digital, can be recorded simultaneously. in the typical application, one microphone input is used for the handset microphone and the other is used as an accessory microphone. in systems using a background noise microphone, ina can be retasked as another microphone input. in systems where the codec is not the only device recording microphone signals, connect microphones to mic2p/mic2n and extmicp/extmicn. mic1p/mic1n then become outputs that route the microphone signals to an external device as needed. two devices can then record microphone signals without needing external analog switches. analog microphone signals are amplified by two stages of gain and then routed to the adcs. the first stage offers selectable 0db, 20db, or 30db settings. the second stage is a programmable-gain amplifier (pga) adjustable from 0db to 20db in 1db steps. to maximize the signal- to-noise ratio, use the gain in the first stage whenever possible. zero-crossing detection is included on the pga to minimize zipper noise while making gain changes. figure 6. microphone input block diagram mi c1p/ d igm i cda ta micbias mben mclk psclk reg mi c1n/ d igm i ccl k mi c2byp inabyp extmic pa1en: 0/20/30db pgam1: +20db to 0db mix mix mixadl mixadr adcl extmic pa2en: 0/20/30db pgaina: +20db to -6db pgaina: +20db to -6db mi c2p mi c2n in a1 /ex tm ic p pgam1: +20db to 0db inadiff adlen adren in a2/ex tm ic n adcr clock control datasheet.in stereo audio codec with flexsound technology MAX9888 61 table 3. microphone input registers register bit name description 0x31/0x32 6 pa1en/pa2en mic1/mic2 preamplifier gain course microphone gain adjustment. 00 = preamplifier disabled 01 = 0db 10 = 20db 11 = 30db 5 4 pgam1/pgam2 mic1/mic2 pga fine microphone gain adjustment. 3 value gain (db) value gain (db) 0x00 +20 0x0b +9 0x01 +19 0x0c +8 2 0x02 +18 0x0d +7 0x03 +17 0x0e +6 0x04 +16 0x0f +5 1 0x05 +15 0x10 +4 0x06 +14 0x11 +3 0x07 +13 0x12 +2 0 0x08 +12 0x13 +1 0x09 +11 0x14 to 0x1f 0 0x0a +10 0x46 7 micclk digital microphone clock frequency select a frequency that is within the digital microphones clock frequency range. set osr1 = 1 when using a digital microphone. 00 = pclk/8 01 = pclk/6 10 = 64 x lrclk 11 = reserved 6 5 digmicl left digital microphone enable set pal1en = 00 for proper operation. 0 = disabled 1 = enabled 4 digmicr right digital microphone enable set par1en = 00 for proper operation. 0 = disabled 1 = enabled 1 extmic external microphone connection routes ina_/extmic_ to the microphone preamplifiers. set inaen = 0 when using ina_/extmic_ as a microphone input. 00 = disabled 01 = mic1 input 10 = mic2 input 11 = reserved 0 datasheet.in stereo audio codec with flexsound technology MAX9888 62 line inputs the device includes two sets of line inputs (figure 7). each set can be configured as a stereo single-ended input or as a mono differential input. each input includes adjustable gain to match a wide range of input signal levels. if a custom gain is needed, the external gain mode provides a trimmed feedback resistor. set the gain by choosing the appropriate input resistor and using the following formula: av pgain = 20 x log (20k/rin) the external gain mode also allows summing multiple signals into a single input, by connecting multiple input resistors as show in figure 8, and inputting signals larger than 1v p-p . table 3. microphone input registers (continued) figure 7. line input block diagram figure 8. summing multiple input signals into ina/inb register bit name description 0x48 7 inabyp ina_/extmic_ to mic1_ bypass switch 0 = disabled 1 = enabled 4 mic2byp mic1_ to mic2_ bypass switch 0 = disabled 1 = enabled 1 recbyp see the output bypass switches section. 0 spkbyp mix mixout1 inadiff pgaina: +20db to -6db inabyp inbdiff ina1/ extmicp ina2/ extmicn inb1 inb2 mix mixout2 mix mixout3 pgaina: +20db to -6db pgainb: +20db to -6db pgainb: +20db to -6db left input 1 left input 2 ina1/extmicp vcm ina2/extmicn 20ki right input 1 right input 2 vcm 20ki datasheet.in stereo audio codec with flexsound technology MAX9888 63 adc input mixers the devices stereo adc accepts input from the micro - phone amplifiers and line inputs. the adc mixer routes any combination of the six audio inputs to the left and right adcs (figure 9). table 4. line input registers figure 9. adc input mixer block diagram register bit name description 0x33/0x34 6 inaext/inbext line input a/b external gain switches out the internal input resistor and selects a trimmed 20ki feedback resistor. use an external input resistor to set the gain of the line input. 0 = disabled 1 = enabled 2 pgaina/pgainb line input a/b internal gain settings 000 = +20db 001 = +14db 010 = +3db 011 = 0db 100 = -3db 101 = -6db 110 = -6db 111 = -6db 1 0 0x45 7 inadiff line input a differential enable 0 = stereo single-ended input 1 = mono differential input 6 inbdiff line input b differential enable 0 = stereo single-ended input 1 = mono differential input inbdiff pgainb: +20db to -6db + pgainb: +20db to -6db inadiff pgaina: +20db to -6db + pgaina: +20db to -6db pgam2: +20db to 0db pgam1: +20db to 0db mixadr adren adlen mix mixadl mix pa2en: 0/20/30db pa1en: 0/20/30db adcr adcl datasheet.in stereo audio codec with flexsound technology MAX9888 64 table 5. adc input mixer register figure 10. record path signal processing block diagram figure 11. agc and noise gate input vs. output gain record path signal processing the devices record signal path includes both automatic gain control (agc) for the microphone inputs and a digi - tal noise gate at the output of the adc (figure 10). microphone agc the ics agc monitors the signal level at the output of the adc and then adjusts the mic1 and mic2 analog pga settings automatically. when the signal level is below the predefined threshold, the gain is increased up to its maximum (20db). if the signal exceeds the thresh - old, the gain is reduced to prevent the output signal level exceeding the threshold. when agc is enabled, the microphone pga is not user programmable. the agc provides a more constant signal level and improves the available adc dynamic range. noise gate since the agc increases the levels of all signals below a user-defined threshold, the noise floor is effectively increased by 20db. to counteract this, the noise gate reduces the gain at low signal levels. unlike typical noise gates that completely silence the output below a defined level, the noise gate in the ic applies downward expan - sion. the noise gate attenuates the output at a rate of 1db for each 2db the signal is below the threshold. the noise gate can be used in conjunction with the agc or on its own. when the agc is enabled, the noise gate reduces the output level only when the agc has set the gain to the maximum setting. figure 11 shows the gain response resulting from using the agc and noise gate. register bit name description 0x22/0x23 7 mixadl/mixadr left/right adc input mixer selects which analog inputs are recorded by the left/right adc. 1xxxxxxx = mic1 x1xxxxxx = mic2 xx1xxxxx = reserved xxx1xxxx = reserved xxxx1xxx = ina1 xxxxx1xx = ina2 (inadiff = 0) or ina2 - ina1 (inadiff = 1) xxxxxx1x = inb1 xxxxxxx1 = inb2 (inbdiff = 0) or inb2 - inb1 (inbdiff = 1) 6 5 4 3 2 1 0 pgam2: +20db to 0db pgam1: +20db to -6db mixadr adren adlen mix mixadl mix pa2en: 0/20/30db pa1en: 0/20/30db adcr adcl avlg: 0/6/12/18db avl: 3db to -12db mode1 avflt avrg: 0/6/12/18db avr: 3db to -2db automatic gain control noise gate audio/ voice filters agc and noise gate amplitude response input amplitude (dbfs) agc only agc and noise gate noise gate only agc and noise gate disabled output amplitude (dbfs) -20 -40 -60 -80 -100 -120 -100 -80 -60 -40 -20 0 -120 0 datasheet.in stereo audio codec with flexsound technology MAX9888 65 table 6. record path signal processing registers register bit name description 0x01 7 ng noise gate attenuation reports the current noise gate attenuation. 000 = 0db 001 = 1db 010 = 2db 011 = 3db to 5db 100 = 6db to 7db 101 = 8db to 9db 110 = 10db to 11db 111 = 12db 6 5 4 agc agc gain reports the current agc gain setting. value gain (db) value gain (db) 3 0x00 +20 0x0b +9 0x01 +19 0x0c +8 0x02 +18 0x0d +7 2 0x03 +17 0x0e +6 0x04 +16 0x0f +5 0x05 +15 0x10 +4 1 0x06 +14 0x11 +3 0x07 +13 0x12 +2 0x08 +12 0x13 +1 0 0x09 +11 0x14 to 0x1f 0 0x0a +10 0x3d 7 agcsrc agc/noise gate signal source determines which adc channel the agc and noise gates analyze. gain is adjusted on both channels regardless of the agcsrc setting. 0 = left adc output 1 = maximum of either the left or right adc output 6 agcrls agc release time defined as the duration from start to finish of gain increase in the region shown in figure 12. release times are longer for low agc threshold levels. 000 = 78ms 001 = 156ms 010 = 312ms 011 = 625ms 100 = 1.25s 101 = 2.5s 110 = 5s 111 = 10s 5 4 datasheet.in stereo audio codec with flexsound technology MAX9888 66 table 6. record path signal processing registers (continued) register bit name description 0x3d 3 agcatk agc attack time defined as the time required to reduce gain by 63% of the total gain reduction (one time constant of the exponential response). attack times are longer for low agc threshold levels. see figure 12 for details. 00 = 2ms 01 = 7.2ms 10 = 31ms 11 = 123ms 2 1 agchld agc hold time the delay before the agc release begins. the hold time counter starts whenever the signal drops below the agc threshold and is reset by any signal that exceeds the threshold. set agchld to enable the agc circuit. see figure 12 for details. 00 = agc disabled 01 = 50ms 10 = 100ms 11 = 400ms 0 0x3e 7 anth noise gate threshold gain is reduced for signals below the threshold to quiet noise. the thresholds are relative to the adcs full-scale output voltage. 6 value threshold (dbfs) value threshold (dbfs) 0x0 noise gate disabled 0x8 -45 0x1 reserved 0x9 -41 5 0x2 reserved 0xa -38 0x3 -64 0xb -34 0x4 -62 0xc -30 4 0x5 -58 0xd -27 0x6 -53 0xe -22 0x7 -50 0xf -16 3 agcth agc threshold gain is reduced when signals exceed the threshold to prevent clipping. the thresholds are relative to the adcs full-scale voltage. 2 value threshold (dbfs) value threshold (dbfs) 0x0 -3 0x8 -11 0x1 -4 0x9 -12 1 0x2 -5 0xa -13 0x3 -6 0xb -14 0x4 -7 0xc -15 0 0x5 -8 0xd -16 0x6 -9 0xe -17 0x7 -10 0xf -18 datasheet.in stereo audio codec with flexsound technology MAX9888 67 adc record level control the ic includes separate digital level control for the left and right adc outputs (figure 13). to optimize dynamic range, use analog gain to adjust the signal level and set the digital level control to 0db whenever possible. digital level control is primarily used when adjusting the record level for digital microphones. figure 12. agc timing figure 13. adc record level control block diagram attack time hold time release time automatic gain control noise gate mode1 avflt1 avlg: 0/6/12/18db avl: 3db to -12db avrg: 0/6/12/18db avr: 3db to -2db adren adlen audio/ voice filters adcr adcl datasheet.in stereo audio codec with flexsound technology MAX9888 68 table 7. adc record level control register figure 14. sidetone block diagram sidetone enable sidetone during full-duplex operation to add a low-level copy of the recorded audio signal to the play - back audio signal (figure 14). sidetone is commonly used in telephony to allow the speaker to hear himself speak, providing a more natural user experience. the ic implements sidetone digitally. doing so helps prevent unwanted feedback into the playback signal path and better matches the playback audio signal. register bit name description 0x2f/0x30 5 avlg/avrg left/right adc gain 00 = 0db 01 = 6db 10 = 12db 11 = 18db 4 3 avl/avr left/right adc level value gain (db) value gain (db) 2 0x0 +3 0x8 -5 0x1 +2 0x9 -6 1 0x2 +1 0xa -7 0x3 0 0xb -8 0x4 -1 0xc -9 0 0x5 -2 0xd -10 0x6 -3 0xe -11 0x7 -4 0xf -12 adren adlen adcr adcl avlg: 0/6/12/18db avl: 3db to -12db mode1 avflt dsts sidetone dvst: 0db to -60db avrg: 0/6/12/18db avr: 3db to -2db automatic gain control eq1en eq2en dveq1: 0db to -15db dv1g: 0/6/12/18db dv2: 0db to -15db dcb2 5-band parametric eq 5-band parametric eq dveq2: 0db to -15db multiband alc excursion limiter dv1: 0db to -15db mode1 dvflt dalen mixdar dacl daren dacr + noise gate mix mix mixdal mix audio/ filters audio/ voice filters audio/ voice filters datasheet.in stereo audio codec with flexsound technology MAX9888 69 table 8. sidetone register digital audio interfaces the ic includes two separate playback signal paths and one record signal path. digital audio interface 1 (dai1) is used to transmit the recorded stereo audio signal and receive a stereo audio signal for playback. digital audio interface 2 (dai2) is used to receive a second stereo audio signal. use dai1 for all full-duplex operations and for all voice signals. use dai2 for music and to mix two playback audio signals. the digital audio interfaces are separate from the audio ports to enable either interface to communicate with any external device connected to the audio ports. each audio interface can be configured in a variety of formats including left justified, i 2 s, pcm, and time divi - sion multiplexed (tdm). tdm mode supports up to 4 mono audio slots in each frame. the ic can use up to 2 mono slots per interface, leaving the remaining two slots available for another device. table 9 shows how to configure the device for common digital audio formats. figures 16 and 17 show examples of common audio formats. by default, sdouts1 and sdouts2 are set high impedance when the ic is not outputting data to facilitate sharing the bus. configure the interface in tdm mode using only slot 1 to transmit and receive mono pcm voice data. the ics digital audio interfaces support both adc to dac loop-through and digital loopback. loop-through allows the signal converted by the adc to be routed to the dac for playback. the signal is routed from the record path to the playback path in the digital audio interface to allow the ics full complement of digital signal processing to be used. loopback allows digital register bit name description 0x2a 7 dsts sidetone source selects which adc output is fed back as sidetone. when mixing the left and right adc outputs, each is attenuated by 6db to prevent full-scale signals from clipping. 00 = sidetone disabled 01 = left adc 10 = right adc 11 = left + right adc 6 4 dvst sidetone level adjusts the sidetone signal level. all levels are referenced to the adcs full-scale output. value level (db) value level (db) 3 0x00 sidetone disabled 0x10 -30.5 0x01 -0.5 0x11 -32.5 0x02 -2.5 0x12 -34.5 0x03 -4.5 0x13 -36.5 2 0x04 -6.5 0x14 -38.5 0x05 -8.5 0x15 -40.5 0x06 -10.5 0x16 -42.5 0x07 -12.5 0x17 -44.5 1 0x08 -14.5 0x18 -46.5 0x09 -16.5 0x19 -48.5 0x0a -18.5 0x1a -50.5 0x0b -20.5 0x1b -52.5 0 0x0c -22.5 0x1c -54.5 0x0d -24.5 0x1d -56.6 0x0e -26.5 0x1e -58.5 0x0f -28.5 0x1f -60.5 datasheet.in stereo audio codec with flexsound technology MAX9888 70 figure 15. digital audio signal routing table 9. common digital audio formats x = dont care. data input to either sdins1 or sdins2 to be routed from one interface to the other for output on sdouts2 or sdouts1. both interfaces must be configured for the same sample rate, but the interface format need not be the same. this allows the ic to route audio data from one device to another, converting the data format as needed. figure 15 shows the available digital signal routing options. mode wci1/wci2 bci1/bci2 dly1/dly2 tdm1/tdm2 slotl1/slotl2 slotr1/slotr2 left justified set as desired set as desired 0 0 x x i 2 s 1 0 1 0 x x pcm x 1 x 1 0 0 tdm x 1 x 1 set as desired mas1 dai1 dai1 record path dai1 playback path dai2 playback path sel1 sel2 bclk1 bclks1 lrclk1 sdout1 sdin1 lten1 lben2 + bit clock frame clock data output data input mas1 hizoff1 mas2 dai2 bclk2 lrclk2 sdout2 sdin2 bit clock frame clock data output data input mas2 hizoff2 lrclks1 sdouts1 sdins1 dvdds1 bclks2 lrclks2 sdouts2 sdins2 dvdds2 mux lben1 port s1 port s1 datasheet.in stereo audio codec with flexsound technology MAX9888 71 table 10. digital audio interface registers register bit name description 0x14/0x1c 7 mas1/mas2 dai1/dai2 master mode in master mode, dai1/dai2 outputs lrclk and bclk. in slave mode, dai1/dai2 accept lrclk and bclk as inputs. 0 = slave mode 1 = master mode 6 wci1/wci2 dai1/dai2 word clock invert tdm1/tdm2 = 0: 0 = left-channel data is transmitted while lrclk is low. 1 = right-channel data is transmitted while lrclk is low. tdm1/tdm2 = 1: always set wci = 0. 5 bci1/bci2 dai1/dai2 bit clock invert bci1/bci2 must be set to 1 when tdm1/tdm2 = 1. 0 = sdin is accepted on the rising edge of bclk. sdout is valid on the rising edge of bclk. 1 = sdin is accepted on the falling edge of bclk. sdout is valid on the falling edge of bclk. master mode: 0 = lrclk transitions on the falling edge of bclk. 1 = lrclk transitions on the rising edge of bclk. 4 dly1/dly2 dai1/dai2 data delay dly1/dly2 has no effect when tdm1/tdm2 = 1. 0 = the most significant data bit is clocked on the first active bclk edge after an lrclk transition. 1 = the most significant data bit is clocked on the second active bclk edge after an lrclk transition. 2 tdm1/tdm2 dai1/dai2 time-division multiplex mode (tdm mode) set tdm1/tdm2 when communicating with devices that use a frame synchronization pulse on lrclk instead of a square wave. 0 = disabled 1 = enabled (bci1/bci2 must be set to 1) 1 fsw1/fsw2 dai1/dai2 wide frame sync pulse increases the width of the frame sync pulse to the full data width when tdm1/tdm2 = 1. fsw1/fsw2 has no effect when tdm1/tdm2 = 0. 0 = disabled 1 = enabled 0 ws1/ws2 dai1/dai2 audio data bit depth determines the maximum bit depth of audio being transmitted and received. data is always 16 bit when tdm1/tmd2 = 0. 0 = 16 bits 1 = 24 bits datasheet.in stereo audio codec with flexsound technology MAX9888 72 table 10. digital audio interface registers (continued) register bit name description 0x15/0x1d 7 osr1 adc oversampling ratio use the higher setting for maximum performance. use the lower setting for reduced power consumption at the expense of performance. 00 = 96x 01 = 64x 10 = reserved 11 = reserved 6 2 bsel1/ bsel2 dai1/dai2 bclk output frequency when operating in master mode, bsel1/bsel2 set the frequency of bclk. when operating in slave mode, bsel1/bsel2 have no effect. select the lowest bclk frequency that clocks all data input to the dac and output by the adc. 000 = bclk disabled 001 = 64 x lrclk 010 = 48 x lrclk 011 = 128 x lrclk (invalid for dhf1/dhf2 = 1) 100 = pclk/2 101 = pclk/4 110 = pclk/8 111 = pclk/16 1 0 0x16/0x1e 7 sel1/sel2 dai1/dai2 audio port selector selects which port is used by dai1/dai2. 00 = none 01 = port s1 10 = port s2 11 = reserved 6 5 lten1 dai1 digital loopthrough connects the output of the record signal path to the input of the playback path. data input to dai1 from an external device is mixed with the recorded audio signal. 0 = disabled 1 = enabled 4 lben1/ lben2 dai1/dai2 digital audio interface loopback lben1 routes the digital audio input to dai1 back out on dai2. lben2 routes the digital audio input to dai2 back out on dai1. selecting lben2 disables the adc output data. 0 = disabled 1 = enabled 3 dmono1/ dmono2 dai1/dai2 dac mono mix mixes the left and right digital input to mono and routes the combined signal to the left and right playback paths. the left and right input data is attenuated by 6db prior to the mono mix. 0 = disabled 1 = enabled datasheet.in stereo audio codec with flexsound technology MAX9888 73 table 10. digital audio interface registers (continued) register bit name description 0x16/0x1e 2 hizoff1/ hizoff2 disable da1/dai2 output high-impedance mode normally sdout is set high impedance between data words. set hizoff1/hizoff2 to force a level on sdout at all times. 0 = disabled 1 = enabled 1 sdoen1/ sdoen2 dai1/dai2 record path output enable dai2 outputs data only if lben1 = 1. 0 = disabled 1 = enabled 0 sdien1/ sdien2 dai1/dai2 playback path input enable 0 = disabled 1 = enabled 0x17/0x1f 7 slotl1/ slotl2 tdm left time slot selects which of the four slots is used for left data on dai1/dai2. if the same slot is selected for left and right audio, left audio is placed in the slot. 00 = slot 1 01 = slot 2 10 = slot 3 11 = slot 4 6 5 slotr1/ slotr2 tdm right time slot selects which of the four slots is used for right data on dai1/dai2. if the same slot is selected for left and right audio, left audio is placed in the slot. 00 = slot 1 01 = slot 2 10 = slot 3 11 = slot 4 4 3 slotdly1/ slotdly2 tdm slot delay adds 1 bclk cycle delay to the data in the specified tdm slot. 1xxx = slot 4 delayed x1xx = slot 3 delayed xx1x = slot 2 delayed xxx1 = slot 1 delayed 2 1 0 datasheet.in stereo audio codec with flexsound technology MAX9888 74 figure 16. non-tdm data format examples wc i_ = 0, bc i_ = 0, dly_ = 0, td m_ = 0, fs w_ = 0, ws _ = 0, hi zo ff _ = 1, sl ot l_ = 0, sl ot r_ = 0 l rclk left left left bc lk sd ou t d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 d15 d14 d13 d12 d11 d10 d9 d8 d7 d6 d5 d4 d3 d2 d1 d0 ri gh t ri gh t ri gh t s din wc i_ = 1, bc i_ = 0, dly_ = 0, td m_ = 0, fs w_ = 0, ws _ = 0, hi zo ff _ = 1, sl ot l_ = 0, sl ot r_ = 0 l rclk bc lk sd ou t s din l rclk bc lk sd ou t s din wc i_ = 0, bc i_ = 1, dly_ = 0, td m_ = 0, fs w_ = 0, ws _ = 0, hi zo ff _ = 1, sl ot l_ = 0, sl ot r_ = 0 wc i_ = 0, bc i_ = 0, dly_ = 1, td m_ = 0, fs w_ = 0, ws _ = 0, hi zo ff _ = 1, sl ot l_ = 0, sl ot r_ = 0 l rclk bc lk sd ou t ri gh t le ft s din datasheet.in stereo audio codec with flexsound technology MAX9888 75 figure 17. tdm mode data format examples lrcl k bclk sdou t sdin l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 wci_ = 0, bci_ = 1, dly_ = 0, tdm_ = 1, fsw_ = 0, ws_ = 0, hizoff_ = 0, slotl_ = 0, slotr_ = 1 l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 hi-z hi-z lrclk bclk sdout sdin lrclk bclk sdout sdin lrclk bclk sdout sdin lrcl k bclk sdou t sdin l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 wci_ = 0, bci_ = 1, dly_ = 0, tdm_ = 1, fsw_ = 1, ws_ = 0, hizoff_ = 0, slotl_ = 0, slotr_ = 1 l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 hi-z hi-z l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 wci_ = 0, bci_ = 1, dly_ = 0, tdm_ = 1, fsw_ = 0, ws_ = 0, hizoff_ = 1, slotl_ = 0, slotr_ = 1 wci_ = 0, bci_ = 1, dly_ = 0, tdm_ = 1, fsw_ = 0, ws_ = 0, hizoff_ = 0, slotl_ = 2, slotr_ = 3 l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 hi-z hi-z l15 l14 l13 l12 l11 l10 l9 l8 l7 l6 l5 l4 l3 l2 l1 l0 r15 r14 r13 r12 r11 r10 r9 r8 r7 r6 r5 r4 r3 r2 r1 r0 32 cycles l l l l l l l l r r r r r r r r hi-z l l l l l l l l 1 1 1 1 r r r r hi-z hi-z 16 cycles 16 cycles 16 cycles 16 cycles wci_ = 0, bci_ = 1, dly_ = 0, tdm_ = 1, fsw_ = 0, ws_ = 0, hizoff_ = 0, slotl_ = 0, slotr_ = 1 datasheet.in stereo audio codec with flexsound technology MAX9888 76 table 11. clock control registers clock control the digital signal paths in the ic require a master clock (mclk) between 10mhz and 60mhz to func - tion. internally, the MAX9888 requires a clock between 10mhz and 20mhz. a prescaler divides mclk by 1, 2, or 4 to create the internal clock (pclk). pclk is used to clock all portions of the ic. the MAX9888 includes two digital audio signal paths, both capable of supporting any sample rate from 8khz to 96khz. each path is independently configured to allow different sample rates. to accommodate a wide range of system architectures, three main clocking modes are supported: u pll mode: when operating in slave mode, enable the pll to lock onto any lrclk input. this mode requires the least configuration, but provides the lowest performance. use this mode to simplify initial setup or when normal mode and exact integer mode cannot be used. u normal mode: this mode uses a 15-bit clock divider to set the sample rate relative to pclk. this allows high flexibility in both the pclk and lrclk frequen - cies and can be used in either master or slave mode. u exact integer mode (dai1 only): in both master and slave modes, common mclk frequencies (12mhz, 13mhz, 16mhz, and 19.2mhz) can be programmed to operate in exact integer mode for both 8khz and 16khz sample rates. in these modes, the mclk and lrclk rates are selected by using the freq1 bits instead of the ni, and pll control bits. register bit name description 0x10 5 psclk mclk prescaler generates pclk, which is used by all internal circuitry. 00 = pclk disabled 01 = 10mhz p mclk p 20mhz (pclk = mclk) 10 = 20mhz p mclk p 40mhz (pclk = mclk/2) 11 = 40mhz p mclk p 60mhz (pclk = mclk/4) 4 0x11/0x19 7 sr1/sr2 dai1/dai2 sample rate used by the alc to correctly set the dual-band crossover frequency and the excursion limiter to set the predefined corner frequencies. 6 value sample rate (khz) value sample rate (khz) 0x0 reserved 0x8 48 5 0x1 8 0x9 88.2 0x2 11.025 0xa 96 0x3 16 0xb reserved 0x4 22.05 0xc reserved 4 0x5 24 0xd reserved 0x6 32 0xe reserved 0x7 44.1 0xf reserved datasheet.in stereo audio codec with flexsound technology MAX9888 77 table 11. clock control registers (continued) register bit name description 0x11 3 freq1 exact integer mode overrides pll1 and ni1 and configures a specific pclk to lrclk ratio. value sample rate value sample rate 0x0 disabled 0x8 pclk = 12mhz, lrclk = 8khz 2 0x1 reserved 0x9 pclk = 12mhz, lrclk = 16khz 0x2 reserved 0xa pclk = 13mhz, lrclk = 8khz 0x3 reserved 0xb pclk = 13mhz, lrclk = 16khz 0x4 reserved 0xc pclk = 16mhz, lrclk = 8khz 1 0x5 reserved 0xd pclk = 16mhz, lrclk = 16khz 0x6 reserved 0xe pclk = 19.2mhz, lrclk = 8khz 0x7 reserved 0xf pclk = 19.2mhz, lrclk = 16khz 0x12/0x1a 7 pll1/pll2 pll mode enable (slave mode only) pll1/pll2 enables a digital pll that locks on to the externally supplied lrclk frequency and automatically sets the lrclk divider (ni1/ni2). 0 = disabled 1 = enabled 6 ni1/ ni2 normal mode lrclk divider when pll1/pll2 = 0, the frequency of lrclk is determined by ni1/ni2. see table 12 for common ni values. 5 4 3 sample rate dhf1/dhf2 ni1/ni2 formula 2 1 8khz p lrclk p 48khz 0 0 0x13/0x1b 7 48khz < lrclk p 96khz 1 6 5 4 3 f lrclk = lrclk frequency f pclk = prescaled mclk frequency (pclk) 2 1 0 ni1[0]/ni2[0] rapid lock mode program ni1/ni2 to the nearest valid ratio and set ni1[0]/ni2[0] when pll1/pll2 = 1 to enable rapid lock mode. normally, the pll automatically calculates and dynamically adjusts ni1/ni2. when rapid lock mode is properly configured, the pll starting point is much closer to the correct value, thus speeding up lock time. wait one lrclk period after programming ni1/ni2 before setting pll1/pll2 = 1. lrclk pclk 65536 96 f ni f lrclk pclk 65536 48 f ni f datasheet.in stereo audio codec with flexsound technology MAX9888 78 table 12. common ni1/ni2 values note: values in bold are exact integers that provide maximum full-scale performance. figure 18. digital passband filtering block diagram passband filtering each digital signal path in the ic includes options for defining the path bandwidth (figure 18). the playback and record paths connected to dai1 support both voice and music filtering while the playback path connected to dai2 supports music filtering only. the voice iir filters provide greater than 70db stopband attenuation at frequencies above f s /2 to reduce aliasing. three selectable highpass filters eliminate unwanted low-frequency signals. use music mode when processing high-fidelity audio content. the music fir filters reduce power consump - tion and are linear phase to maintain stereo imaging. an optional dc-blocking filter is available to eliminate unwanted dc offset. in music mode, a second set of fir filters are available to support sample rates greater than 50khz. the filters can be independently selected for dai1 and dai2 and support both the playback and record audio paths. pclk (mhz) lrclk (khz) dhf1/2 = 0 dhf1/2 = 1 8 11.025 12 16 22.05 24 32 44.1 48 64 88.2 96 10 13a9 1b18 1d7e 2752 3631 3afb 4ea5 6c61 75f7 4ea5 6c61 75f7 11 11e0 18a2 1acf 23bf 3144 359f 477e 6287 6b3e 477e 6287 6b3e 11.2896 116a 1800 1a1f 22d4 3000 343f 45a9 6000 687d 45a9 6000 687d 12 1062 1694 1893 20c5 2d29 3127 4189 5a51 624e 4189 5a51 624e 12.288 1000 160d 1800 2000 2c1a 3000 4000 5833 6000 4000 5833 6000 13 0f20 14d8 16af 1e3f 29af 2d5f 3c7f 535f 5abe 3c7f 535f 5abe 16 0c4a 10ef 126f 1893 21de 24dd 3127 43bd 49ba 3127 43bd 49ba 16.9344 0b9c 1000 116a 1738 2000 22d4 2e71 4000 45a9 2e71 4000 45a9 18.432 0aab 0eb3 1000 1555 1d66 2000 2aab 3acd 4000 2aab 3acd 4000 20 09d5 0d8c 0ebf 13a9 1b18 1d7e 2752 3631 3afb 2752 3631 3afb adren adlen adcr adcl avlg: 0/6/12/18db avl: 3db to -12db mode1 avflt dsts sidetone dvst: 0d b to -6 0d b avrg: 0/6/12/18db avr: 3db to -2db eq1en eq2en dveq1: 0db to -15db dv1g: 0/6/12/18db dv2: 0db to -15db dcb2 5-band parametric eq 5-band parametric eq dveq2: 0db to -15db multiband alc excursion limiter dv1: 0db to -15db mode1 dvflt dalen mixdar dacl daren dacr + mix mix mixdal mix automatic gain control noise gate audio/ voice filters audio/ filters audio/ voice filters datasheet.in stereo audio codec with flexsound technology MAX9888 79 table 13. passband filtering registers register bit name description 0x18 7 mode1 dai1 passband filtering mode 0 = voice filters 1 = music filters (recommended for f s > 24khz) 6 avflt1 dai1 adc highpass filter mode 5 mode1 avflt1 4 0 see table 14 1 select a nonzero value to enable the dc-blocking filter 3 dhf1 dai1 high sample rate mode selects the sample rate range. 0 = 8khz p lrclk p 48khz 1 = 48khz p lrclk < 96khz 2 dvflt1 dai1 dac highpass filter mode 1 mode1 dvflt1 0 0 see table 14 1 select a nonzero value to enable the dc-blocking filter 0x20 3 dhf2 dai2 high sample rate mode selects the sample rate range. 0 = 8khz p lrclk p 48khz 1 = 48khz < lrclk p 96khz 0 dcb2 dai2 dc blocking filter enables a dc-blocking filter on the dai2 playback audio path. 0 = disabled 1 = enabled datasheet.in stereo audio codec with flexsound technology MAX9888 80 table 14. voice highpass filters avftl/dvflt value intended sample rate filter response 000 n/a disabled 001/011 16khz/8khz 010/100 16khz/8khz 101 8khz to 48khz 110/111 n/a reserved frequency (hz) amplitude (db) 800 600 400 200 -50 -40 -30 -20 -10 0 -60 0 1000 frequency (hz) amplitude (db) 800 600 400 200 -50 -40 -30 -20 -10 0 -60 0 1000 frequency (hz) amplitude (db) 800 600 400 200 -50 -40 -30 -20 -10 0 -60 0 1000 lrclk = 48khz datasheet.in stereo audio codec with flexsound technology MAX9888 81 figure 20. alc input vs. output examples figure 19. playback path signal processing block diagram playback path signal processing the ic playback signal path includes automatic level control (alc) and a 5-band parametric equalizer (eq) (figure 19). the dai1 and dai2 playback paths include separate alcs controlled by a single set of registers. two completely separate parametric eqs are included for the dai1 and dai2 playback paths. automatic level control the automatic level control (alc) circuit ensures maxi - mum signal amplitude without producing audible clip - ping. this is accomplished by a variable gain stage that works on a sample by sample basis to increase the gain up to 12db. a look-ahead circuit determines if the next sample exceeds full scale and reduces the gain so that the sample is exactly full scale. a programmable low signal threshold determines the minimum signal amplitude that is amplified. select a threshold that prevents the amplification of background noise. when the signal level drops below the low signal threshold, the alc reduces the gain to 0db until the sig - nal increases above the threshold. figure 20 shows an example of alc input vs. output curves. the alc can optionally be configured in dual-band mode. in this mode, the input signal is filtered into two bands with a 5khz center frequency. each band is routed through independent alcs and then summed together. in multiband mode, both bands use the same parameters. 0 ou tput si gnal (d bf s) ou tput si gnal (d bf s) ou tput si gnal (d bf s) input si gnal (d bf s) input si gnal (d bf s) input si gnal (d bf s) low -lev el th re s hol d -1 2 0 low -lev el th re s hol d -1 2 0 low -lev el th re s hol d -1 2 0 0 0 al c wi th al ct h 00 0 al c wi th al ct h = 000 al c di sa bl ed eq1en eq2en dveq1: 0db to -15db dv1g: 0/6/12/18db dv2: 0db to -15db dcb2 dveq2: 0db to -15db multiband alc excursion limiter dv1: 0db to -15db mode1 dvflt dalen mixdar dacl daren dacr + mix mixdal mix 5-band parametric eq 5-band parametric eq audio/ filters audio/ voice filters datasheet.in stereo audio codec with flexsound technology MAX9888 82 table 15. automatic level control registers parametric equalizer the parametric eq contains five independent biquad filters with programmable gain, center frequency, and bandwidth. each biquad filter has a gain range of q12db and a center frequency range from 20hz to 20khz. use a filter q less than that shown in figure 21 to achieve ideal frequency responses. setting a higher q results in non- ideal frequency response. the biquad filters are series connected, allowing a total gain of q60db. figure 21. maximum recommended filter q vs. frequency register bit name description 0x41 7 alcen alc enable enables alc on both the dai1 and dai2 playback paths. 0 = disabled 1 = enabled 6 alcrls alc and excursion limiter release time sets the release time for both the alc and excursion limiter. see the excursion limiter section for excursion limiter release times. alc release time is defined as the time required to adjust the gain from 12db to 0db. value alc release time (s) 5 000 8 001 4 010 2 011 1 4 100 0.5 101 0.25 110 reserved 111 reserved 3 alcmb multiband enable enables dual-band processing with a 5khz center frequency. sr1 and sr2 must be configured properly to achieve the correct center frequency for each playback path. 0 = single-band alc 1 = dual-band alc 2 alcth low signal threshold selects the minimum signal level to be boosted by the alc. 000 = - j db (low-signal threshold disabled) 001 = -12db 010 = -18db 011 = -24db 100 = -30db 101 = -36db 110 = -42db 111 = -48db 1 0 center frequency (hz) maximum recommended filter q 10,000 1000 1 10 100 1000 0.1 100 100,000 f s = 8khz f s = 48khz f s = 96khz datasheet.in stereo audio codec with flexsound technology MAX9888 83 table 16. eq registers use the attenuator at the eqs input to avoid clipping the signal. the attenuator can be programmed for fixed attenuation or dynamic attenuation based on signal level. if the dynamic eq clip detection is enabled, the signal level from the eq is fed back to the attenuator circuit to determine the amount of gain reduction necessary to avoid clipping. the MAX9888 ev kit software includes a graphic inter - face for generating the eq coefficients. the coefficients are sample rate dependent and stored in registers 0x50 through 0xb3. register bit name description 0x2c/0x2e 4 eqclp1/ eqclp2 dai1/dai2 eq clip detection automatically controls the eq attenuator to prevent clipping in the eq. 0 = enabled 1 = disabled 3 dveq1/dveq2 dai1/dai2 eq attenuator provides attenuation to prevent clipping in the eq when full-scale signals are boost - ed. dveq1/dveq2 operates only when eq1en/eq2en = 1 and eqclp1/eqclp2 = 1. 2 value gain (db) value gain (db) 0x0 0 0x8 -8 0x1 -1 0x9 -9 1 0x2 -2 0xa -10 0x3 -3 0xb -11 0x4 -4 0xc -12 0 0x5 -5 0xd -13 0x6 -6 0xe -14 0x7 -7 0xf -15 0x47 7 vs2en see the click-and-pop reduction section. 6 vsen 5 zden 1 eq2en dai2 eq enable 0 = disabled 1 = enabled 0 eq1en dai1 eq enable 0 = disabled 1 = enabled datasheet.in stereo audio codec with flexsound technology MAX9888 84 figure 22. playback level control block diagram table 17. dac playback level control register playback level control the ic includes separate digital level control for the dai1 and dai2 playback audio paths. the dai1 signal path allows boost when mode1 = 0 and attenuation in any mode. the dai2 signal path allows attenuation only. register bit name description 0x2b/0x2d 7 dv1m/dv2m dai1/dai2 mute 0 = disabled 1 = enabled 5 dv1g dai1 voice mode gain dv1g only applies when mode1 = 0. 00 = 0db 01 = 6db 10 = 12db 11 = 18db 4 3 dv1/dv2 dai1/dai2 attenuation value gain (db) value gain (db) 2 0x0 0 0x8 -8 0x1 -1 0x9 -9 0x2 -2 0xa -10 1 0x3 -3 0xb -11 0x4 -4 0xc -12 0x5 -5 0xd -13 0 0x6 -6 0xe -14 0x7 -7 0xf -15 eq1en eq2en dveq1: 0db to -15db dv1g: 0/6/12/18db dv2: 0db to -15db dcb2 dveq2: 0db to -15db multiband alc excursion limiter dv1: 0db to -15db mode1 dvflt dalen mixdar dacl daren dacr + mix mixdal mix 5-band parametric eq 5-band parametric eq audio/ filters audio/ voice filters datasheet.in stereo audio codec with flexsound technology MAX9888 85 table 18. dac input mixer register figure 23. dac input mixer block diagram dac input mixers the ics stereo dac accepts input from two digital audio paths. the dac mixer routes any audio path to the left and right dacs (figure 23). register bit name description 0x21 7 mixdal left dac input mixer 1xxx = dai1 left channel x1xx = dai1 right channel xx1x = dai2 left channel xxx1 = dai2 right channel 6 5 4 3 mixdar right dac input mixer 1xxx = dai1 left channel x1xx = dai1 right channel xx1x = dai2 left channel xxx1 = dai2 right channel 2 1 0 eq1en eq2en dveq1: 0db to -15db dv1g: 0/6/12/18db dv2: 0db to -15db dcb2 5-band parametric eq 5-band parametric eq dveq2: 0db to -15db multiband alc excursion limiter dv1: 0db to -15db mode1 dvflt dalen mixdar dacl daren dacr + mix mixdal mix audio/ filters audio/ voice filters datasheet.in stereo audio codec with flexsound technology MAX9888 86 figure 24. preoutput signal path block diagram table 19. preoutput mixer registers preoutput signal path the ics preoutput mixer stage provides mixing and level adjustment for line input signals routed to the output ampli - fiers. figure 24 shows a block diagram of the preoutput signal path. 9db is added between the line input amplifiers and the output amplifiers to boost the 1v p-p maximum line input signal level to the 1v rms maximum dac signal level. preoutput mixer the ics output amplifiers each accept input from one of the three preoutput mixers. configure each pre - output mixer to mix any combination of the four line input signals. register bit name description 0x24/0x25/ 0x26 3 mixout1/ mixout2/ mixout3 preoutput mixer 1 1xxx = ina1 x1xx = ina2 (inadiff = 0) or ina2 - ina1 (inadiff = 1) xx1x = inb1 xxx1 = inb2 (inbdiff = 0) or inb2 - inb1 (inbdiff = 1) 2 1 0 inbdiff mixout3 mixout2 pgainb: +20db to -6db + mix mix pgainb: +20db to -6db inadiff pgaina: +20db to -6db pgaout1: 0d b to -2 3d b pgaout2: 0d b to -2 3d b pgaout3: 0d b to -2 3d b + pgaina: +20db to -6db mixout1 pr eo ut 1 mix pr eo ut 2 pr eo ut 3 +9 db +9 db +9 db mixrec 0db recvol: +8db to -62db recp/ rxinp recn/ rxinn spklp spkln recbyp spkbyp recen mixspl +6db spvoll: +8db to -62db spklvdd spkrp spkrn splen spren po we r/ di st or t ion li mi te r mixspr +6db spvolr: +8db to -62db hpvoll: +3db to -67db spklgnd spkrvdd spkrpgnd hpl hpsns hpr hplen mixhpl hpvolr: +3db to -67db hpren mixhpr mix mix mix mix mix battery adc datasheet.in stereo audio codec with flexsound technology MAX9888 87 table 20. preoutput pga registers figure 25. receiver amplifier block diagram preoutput pga the ics preoutput pgas allow line input signals to be attenuated to match dac output signal levels. use the 0db setting for maximum performance. receiver amplifier the ic includes a single differential receiver amplifier. the receiver amplifier is designed to drive 32 i receivers. in cases where a single transducer is used for the loudspeaker and receiver, use the spkbyp switch to route the receiver amplifier output to the left speaker outputs. register bit name description 0x35/0x36/ 0x37 3 pgaout1/ pgaout2/ pgaout3 preoutput pga level value gain (db) value gain (db) 2 0x0 0 0x8 -15 0x1 -1 0x9 -17 1 0x2 -3 0xa -19 0x3 -5 0xb -21 0x4 -7 0xc -23 0 0x5 -9 0xd mute 0x6 -11 0xe mute 0x7 -13 0xf mute pgaout1: 0d b to -2 3d b pgaout2: 0d b to -2 3d b pr eo ut 1 pr eo ut 2 +9 db +9 db dalen dacl daren dacr mixrec 0db recvol: +8db to -62db recp/ rxinp recn/ rxinn recbyp spkbyp recen mix datasheet.in stereo audio codec with flexsound technology MAX9888 88 table 21. receiver output mixer register table 22. receiver output level register receiver output mixer the ics receiver amplifier accepts input from the stereo dac and the line inputs. configure the mixer to mix any combina - tion of the available sources. when more than one signal is selected, the mixed signal is attenuated by 6db for 2 signals, 9.5db for 3 signals, or 12db for 4 signals. receiver output volume register bit name description 0x28 3 mixrec receiver output mixer 1xxx = left dac x1xx = right dac xx1x = preoutput mixer 1 xxx1 = preoutput mixer 2 2 1 0 register bit name description 0x3a 7 recm receiver output mute 0 = disabled 1 = enabled 4 recvol receiver output volume level value volume (db) value volume (db) 3 0x00 -62 0x10 -10 0x01 -58 0x11 -8 0x02 -54 0x12 -6 0x03 -50 0x13 -4 2 0x04 -46 0x14 -2 0x05 -42 0x15 0 0x06 -38 0x16 +1 0x07 -35 0x17 +2 1 0x08 -32 0x18 +3 0x09 -29 0x19 +4 0x0a -26 0x1a +5 0x0b -23 0x1b +6 0 0x0c -20 0x1c +6.5 0x0d -17 0x1d +7 0x0e -14 0x1e +7.5 0x0f -12 0x1f +8 datasheet.in stereo audio codec with flexsound technology MAX9888 89 speaker amplifiers the ic integrates a stereo filterless class d amplifier that offers much higher efficiency than class ab without the typical disadvantages. the high efficiency of a class d amplifier is due to the switching operation of the output stage transistors. in a class d amplifier, the output transistors act as current steering switches and consume negligible additional power. any power loss associated with the class d out - put stage is mostly due to the i 2 r loss of the mosfet on-resistance, and quiescent current overhead. the theoretical best efficiency of a linear amplifier is 78%, however, that efficiency is only exhibited at peak output power. under normal operating levels (typical music reproduction levels), efficiency falls below 30%, whereas the ics class d amplifier still exhibits 80% effi - ciency under the same conditions. traditional class d amplifiers require the use of exter - nal lc filters or shielding to meet en55022b and fcc electromagnetic-interference (emi) regulation standards. maxims patented active emissions limiting edge-rate control circuitry reduces emi emissions (figure 26). figure 26. emi with 15cm of speaker cable figure 27. speaker amplifier path block diagram 300 350 400 450 500 550 600 650 700 750 800 850 900 950 1000 30 60 80 100 120 140 160 180 200 220 240 260 280 300 40 30 20 10 -10 0 40 30 20 10 -10 0 frequency (mhz) amplitude (dbv/m) frequency (mhz) amplitude (dbv/m) pgaout2: 0d b to -2 3d b pgaout3: 0d b to -2 3d b pr eo ut 2 pr eo ut 3 +9 db +9 db dalen dacl daren dacr spklp spkln mixspl +6db spvoll: +8db to -62db spklvdd spklgnd spkrvdd spkrp spkrn splen spren po we r/ di st or ti on li mi te r mixspr +6db spvolr: +8db to -62db spkrpgnd mix mix battery adc datasheet.in stereo audio codec with flexsound technology MAX9888 90 table 23. speaker output mixer register table 24. speaker output mixer register speaker output mixers the ics speaker amplifiers accept input from the stereo dac and the line inputs. configure the mixer to mix any combina - tion of the available sources. when more than one signal is selected, the mixed signal is attenuated by 6db for 2 signals, 9.5db for 3 signals, or 12db for four signals. speaker output volume register bit name description 0x29 7 mixspl left speaker output mixer 1xxx = left dac x1xx = right dac xx1x = reserved xxx1 = preoutput mixer 3 6 5 4 3 mixspr right speaker output mixer 1xxx = left dac x1xx = right dac xx1x = reserved xxx1 = preoutput mixer 2 2 1 0 register bit name description 0x3b/0x3c 7 splm/sprm left/right speaker output mute 0 = disabled 1 = enabled 4 spvoll/spvolr left/right speaker output volume level value volume (db) value volume (db) 0x00 -64 0x10 -10 0x01 -59 0x11 -8 3 0x02 -55 0x12 -6 0x03 -50 0x13 -4 0x04 -46 0x14 -2 2 0x05 -42 0x15 0 0x06 -38 0x16 +1 0x07 -35 0x17 +2 1 0x08 -32 0x18 +3 0x09 -29 0x19 +4 0x0a -26 0x1a +5 0x0b -23 0x1b +6 0 0x0c -20 0x1c +6.5 0x0d -17 0x1d +7 0x0e -14 0x1e +7.5 0x0f -12 0x1f +8 datasheet.in stereo audio codec with flexsound technology MAX9888 91 figure 28. speaker amplifier signal processing block diagram speaker amplifier signal processing the ic includes signal processing to improve the sound quality of the speaker output and protect transducers from damage. an excursion limiter dynamically adjusts the highpass corner frequency, while a power limiter and distortion limiter prevent the amplifier from outputting too much distortion or power. the excursion limiter is located in the dsp while the distortion limiter and power limiter control the analog volume control (figure 28). all three limiters analyze the speaker amplifiers output signal to determine when to take action. excursion limiter the excursion limiter is a dynamic highpass filter that monitors the speaker outputs and increases the high - pass corner frequency when the speaker amplifiers out - put exceeds a predefined threshold. the filter smoothly transitions between the high and low corner frequency to prevent unwanted artifacts. the filter can operate in four different modes: u fixed frequency preset mode. the highpass corner frequency is fixed at the upper corner frequency and does not change with signal level. u fixed frequency programmable mode. the high - pass corner frequency is fixed to that specified by the programmable biquad filter. u preset dynamic mode. the highpass filter automati - cally slides between a preset upper and lower corner frequency based on output signal level. u user programmable dynamic mode. the highpass filter slides between a user-programmed biquad filter on the low side to a predefined corner frequency on the high side. the transfer function for the user-programmable biquad is: -1 -2 0 1 2 -1 -2 1 2 b b z b z h(z) 1 a z a z + + = + + the coefficients b 0 , b 1 , b 2 , a 1 , and a 2 are sample rate dependent and stored in registers 0xb4 through 0xc7. store b 0 , b 1 , and b 2 as positive numbers. store a 1 and a 2 as negated twos complement numbers. separate fil - ters can be stored for the dai1 and dai2 playback paths. the MAX9888 ev kit software includes a graphic interface for generating the user-programmable biquad coefficients. note: only change the excursion limiter settings when the signal path is disabled to prevent undesired artifacts. dalen dacl daren dacr spklp spkln mixspl +6db spvoll: +8db to -62db spklvdd spklgnd spkrvdd spkrp spkrn splen spren po we r/ di st or t ion li mi te r mixspr +6db spvolr: +8db to -62db spkrpgnd mix mix eq1en eq2en dveq1: 0db to -15db dv1g: 0/6/12/18db dv2: 0db to -15db dcb2 5-band parametric eq 5-band parametric eq dveq2: 0db to -15db multiband alc excursion limiter dv1: 0db to -15db mode1 dvflt mixdar + mix mixdal mix audio/ filters audio/ voice filters battery adc datasheet.in stereo audio codec with flexsound technology MAX9888 92 table 25. excursion limiter registers register bit name description 0x3f 6 dhpucf excursion limiter corner frequency the excursion limiter has limited sliding range and minimum corner frequencies. listed below are all the valid filter combinations. 5 lower corner frequency upper corner frequency minimum biquad corner frequency dhpucf dhplcf excursion limiter disabled 000 00 4 400hz 001 00 600hz 010 00 800hz 011 00 1khz 100 00 1 dhplcf programmable using biquad 100hz 000 11 200hz 400hz 001 01 400hz 600hz 010 10 400hz 800hz 011 10 programmable using biquad 400hz 200hz 001 11 0 programmable using biquad 600hz 300hz 010 11 programmable using biquad 800hz 400hz 011 11 programmable using biquad 1khz 500hz 100 11 0x41 6 alcrls alc and excursion limiter release time sets the release time for both the alc and excursion limiter. see the automatic level control section for alc release times. excursion limiter release time is defined as the time required to slide from the high corner frequency to the low corner frequency. value excursion limiter release time (s) 5 000 4 001 2 010 1 011 0.5 4 100 0.25 101 0.25 110 reserved 111 reserved 0x40 3 dhpth excursion limiter threshold measured at the class d speaker amplifier outputs. signals above the threshold use the upper corner frequency. signals below the threshold use the lower corner frequency. v bat must correctly reflect the voltage of spklvdd to achieve accurate thresholds. 000 = 0.34v p 001 = 0.71v p 010 = 1.30v p 011 = 1.77v p 100 = 2.33v p 101 = 3.25v p 110 = 4.25v p 111 = 4.95v p 2 1 0 datasheet.in stereo audio codec with flexsound technology MAX9888 93 power limiter the ics power limiter tracks the rms power delivered to the loudspeaker and briefly mutes the speaker amplifier output if the speaker is at risk of sustaining permanent damage. loudspeakers are typically damaged when the voice coil overheats due to extended operation above the rated power. during normal operation, heat generated in the voice coil is transferred to the speakers magnet, which transfers heat to the surrounding air. for the voice coil to overheat, both the voice coil and the magnet must over - heat. the result is that a loudspeaker can operate above its rated power for a significant time before it heats suf - ficiently to cause damage. the ics power limiter includes user-programmable time constants and power thresholds to match a wide range of loudspeakers. program the power limiters threshold to match the loudspeakers rated power handling. this can be determined through measurement or the loudspeak - ers specification. program time constant 1 to match the voice coils thermal time constant. program time constant 2 to match the magnets thermal time constant. the time constants can be determined by plotting the voice coils resistance vs. time as power is applied to the speaker. table 26. power limiter registers register bit name description 0x42 7 pwrth power limiter threshold if the rms output power from the speaker amplifiers exceeds this threshold, the out - put is briefly muted to protect the speaker. the threshold is measured in watts assum - ing an 8 i load. vbat must correctly reflect the voltage of spklvdd/spkrvdd to achieve accurate thresholds. 6 value threshold (w) value threshold (w) 0x0 power limiter disabled 0x8 0.27 5 0x1 0.05 0x9 0.35 0x2 0.06 0xa 0.48 0x3 0.09 0xb 0.72 0x4 0.11 0xc 1.00 4 0x5 0.13 0xd 1.43 0x6 0.18 0xe 1.57 0x7 0.22 0xf 1.80 2 pwrk power limiter weighting factor determines the balance between time constant 1 and 2 to match the dominance of each time constant in the loudspeaker. 1 value t1 (%) t2 (%) 000 50 50 001 62.5 37.5 010 75 25 011 87.5 12.5 0 100 100 0 101 12.5 87.5 110 25 75 111 37.5 62.5 register bit name description datasheet.in stereo audio codec with flexsound technology MAX9888 94 distortion limiter the ics distortion limiter ensures that the speaker amplifiers output does not exceed the programmed thd+n limit. the distortion limiter analyzes the class d output duty cycle to determine the percentage of the waveform that is clipped. if the distortion exceeds the programmed threshold, the output gain is reduced. table 26. power limiter registers (continued) 0x43 7 pwrt2 power limiter time constant 2 select a value that matches the thermal time constant of the loudspeakers magnet. 6 value time constant (min) value time constant (min) 0x0 disabled 0x8 3.75 0x1 0.50 0x9 5.00 5 0x2 0.67 0xa 6.66 0x3 0.89 0xb 8.88 0x4 1.19 0xc reserved 4 0x5 1.58 0xd reserved 0x6 2.11 0xe reserved 0x7 2.81 0xf reserved 3 pwrt1 power limiter time constant 1 select a value that matches the thermal time constant of the loudspeakers voice coil. 2 value time constant (s) value time constant (s) 0x0 disabled 0x8 3.75 0x1 0.50 0x9 5.00 1 0x2 0.67 0xa 6.66 0x3 0.89 0xb 8.88 0x4 1.19 0xc reserved 0 0x5 1.58 0xd reserved 0x6 2.11 0xe reserved 0x7 2.81 0xf reserved datasheet.in stereo audio codec with flexsound technology MAX9888 95 headphone amplifier the ics headphone amplifier integrates maxims directdrive architecture to eliminate the need for large dc-blocking capacitors. traditional single-supply head - phone amplifiers have outputs biased at a nominal dc voltage (typically half the supply). large coupling capacitors are needed to block this dc bias from the headphone. without these capacitors, a significant amount of dc current flows to the headphone, resulting in unnecessary power dissipation and possible damage to both the headphone and headphone amplifier. the directdrive architecture uses a charge pump to create an internal negative supply voltage. this allows the ics headphone outputs to be biased at gnd while operating from a single supply (figure 29). without a dc component, there is no need for the large dc-blocking capacitors. instead of two large (220 f f, typ) capaci- tors, the ic charge pump requires two small ceramic capacitors, conserving board space, reducing cost, and improving the frequency response of the headphone amplifier. there is a low dc voltage on the amplifier out - puts due to amplifier offset. however, the offset of the ic is typically q 0.2mv, which, when combined with a 32i load, results in less than 6 f a of dc current flow to the headphones. in addition to the cost and size disadvantages of the dc-blocking capacitors required by conventional headphone amplifiers, these capacitors limit the ampli - fiers low-frequency response and can distort the audio signal. the dc-blocking capacitor not only blocks dc, but also low-frequency audio. improving the low-fre - quency response of a conventional headphone amplifier requires increasing the capacitor size, further adding to the cost and size of the solution. due to the voltage coefficient of the capacitors used for dc blocking, they introduce significant distortion near the corner frequency of the highpass filter they create. this distortion further degrades the low-frequency audio quality. table 27. distortion limiter registers register bit name description 0x44 7 thdclp distortion limit measured in % thd+n. 6 value thd+n limit (%) value thd+n limit (%) 0x0 limiter disabled 0x8 12 5 0x1 < 1 0x9 14 0x2 1 0xa 16 0x3 2 0xb 18 4 0x4 4 0xc 20 0x5 6 0xd 21 0x6 8 0xe 22 0x7 10 0xf 24 2 thdt1 distortion limiter release time constant duration of time required for the speaker amplifiers output gain to adjust back to the nominal level after a large signal has passed. 000 = 6.2s 001 = 3.1s 010 = 1.6s 011 = 815ms 100 = 419ms 101 = 223ms 110 = 116ms 111 = 76ms 1 0 datasheet.in stereo audio codec with flexsound technology MAX9888 96 alternative approaches to eliminating the output-cou - pling capacitors involve biasing the headphone return (sleeve) to the dc bias voltage of the headphone ampli - fiers. this method raises some issues: u the sleeve is typically grounded to the chassis. using the midrail biasing approach, the sleeve must be isolated from system ground, complicating product design. u during an esd strike, the amplifiers esd structures are the only path to system ground. thus, the ampli - fier must be able to withstand the full energy from an esd strike. u when using the headphone jack as a line out to other equipment, the bias voltage on the sleeve may con - flict with the ground potential from other equipment, resulting in possible damage to the amplifiers. the ic features a low-noise charge pump to generate a negative supply for the headphone amplifier. the nominal switching frequency is well beyond the audio range, and thus does not interfere with audio signals. the switch drivers feature a controlled switching speed that minimizes noise generated by turn-on and turn-off transients. by limiting the switching speed of the charge pump, the di/dt noise caused by the parasitic trace inductance is minimized. the charge pump is active only in headphone modes. to reduce audible noise at the outputs, the ics head - phone amplifier includes headphone ground sensing. connect the sense line (hpsns) to the ground terminal of the devices headphone jack. any noise present at the headphone ground is then added to the headphone output. the result is elimination of this noise from the audible output. if ground sensing is not required, con - nect hpsns directly to ground. figure 30 shows a block diagram of the headphone output section including the headphone sense function. headphone output mixers the ics headphone amplifier accepts input from the stereo dac and the line inputs. the output of the left and right dac cannot be mixed at the headphone mixer. use mixdal/mixdar to mix the left and right audio channels before conversion. figure 29. traditional amplifier output vs. directdrive output figure 30. headphone amplifier block diagram v dd v dd /2 gnd conventional amplifier biasing scheme directdrive amplifier biasing scheme +v dd gnd -v dd (v ss ) mixhpl mixhpr mix mix pgaout2: 0db to -23db hpvoll: +3db to -67db hpvolr: +3db to -67db preout2 +9db pgaout1: 0db to -23db daren hpl hpsns hplen hpren hpr preout1 +9db dacr dalen dacl datasheet.in stereo audio codec with flexsound technology MAX9888 97 table 28. headphone output mixer register table 29. headphone output level register headphone output volume register bit name description 0x27 7 mixhpl left headphone output mixer 10xx = left dac 01xx = right dac (requires dalen = 0 for proper operation) 11xx = left dac xx1x = reserved xxx1 = preoutput mixer 1 6 5 4 3 mixhpr right headphone output mixer 10xx = left dac (requires daren = 0 for proper operation) 01xx = right dac 11xx = right dac xx1x = reserved xxx1 = preoutput mixer 2 2 1 0 register bit name description 0x38/0x39 7 hplm/hprm headphone output mute 0 = disabled 1 = enabled 4 hpvoll/hpvolr left/right headphone output volume level value volume (db) value volume (db) 0x00 -67 0x10 -15 0x01 -63 0x11 -13 0x02 -59 0x12 -11 3 0x03 -55 0x13 -9 0x04 -51 0x14 -7 0x05 -47 0x15 -5 0x06 -43 0x16 -4 2 0x07 -40 0x17 -3 0x08 -37 0x18 -2 0x09 -34 0x19 -1 1 0x0a -31 0x1a 0 0x0b -28 0x1b +1 0x0c -25 0x1c +1.5 0 0x0d -22 0x1d +2 0x0e -19 0x1e +2.5 0x0f -17 0x1f +3 datasheet.in stereo audio codec with flexsound technology MAX9888 98 output bypass switches the ic includes two output bypass switches that solve common applications problems. when a single trans - ducer is used for the loudspeaker and receiver, the need exists for two amplifiers to power the same transducer. bypass switches connect the ics receiver amplifier output to the speaker amplifiers output, allowing either amplifier to power the same transducer. in systems where an external receiver amplifier is used, route its output to the left speaker through recp/rxinp and recn/rxinn, bypassing the class d amplifier. in systems where an external amplifier drives both the receiver and the ics line input, one of the differential signals can be discon - nected from the receiver when not needed by passing it through the analog switch that connects recp/rxinp to recn/rxinn. figure 31. output bypass switch block diagrams table 30. output bypass switches register register bit name description 0x48 7 inabyp see the microphone inputs section. 4 mic2byp 1 recbyp rxinp to rxinn bypass switch shorts rxinp to rxinn allowing a signal to pass through the MAX9888. disable the receiver amplifier when recbyp = 1. 0 = disabled 1 = enabled 0 spkbyp rxin to spkl bypass switch shorts rxinp/rxinn to spklp/spkln allowing either the internal or an external receiver amplifier to power the left speaker. disable the left speaker amplifier when spkbyp = 1. 0 = disabled 1 = enabled external receiver amp recp/ rxinp re ce n splen power/distortion limiter re c byp spkbyp 10i* 10i* recn/ rxinn spklvdd spklp spkln 0db spklgnd +6db *optional 10i resistors improve distortion through the analog switch. speaker amplifier bypass using an external receiver amplifier speaker amplifier bypass using the internal receiver amplifier controlling an external receive amplifier and speaker external receiver amp recn/rxinn re ce n splen power/distortion limiter re c byp spkbyp recn/rxinn spklvdd spklp spkln 0db spklgnd +6db recp/rxinp re ce n splen power/distortion limiter re c byp spkbyp recn/rxinn spklvdd spklp spkln 0db spklgnd +6db datasheet.in stereo audio codec with flexsound technology MAX9888 99 click-and-pop reduction the ic includes extensive click-and-pop reduction cir - cuitry. the circuitry minimizes clicks and pops at turn-on, turn-off, and during volume changes. zero-crossing detection is implemented on all analog pgas and volume controls to prevent large glitches when volume changes are made. instead of making a volume change immediately, the change is made when the audio signal crosses the midpoint. if no zero-crossing occurs within the timeout window, the change is forced. volume slewing breaks up large volume changes into the smallest available step size and the steps through each step between the initial and final volume setting. when enabled, volume slewing also occurs at device turn-on and turn-off. during turn-on the volume is set to mute before the output is enabled. once the output is on, the volume ramps to the desired level. at turn-off the volume is ramped to mute before the outputs are disabled. when there is no audio signal zero-crossing detection can prevent volume slewing from occurring. enable enhanced volume slewing to prevent the volume control - ler from requesting another volume level until the previ - ous one has been set. each step in the volume ramp then occurs after a zero crossing has occurred in the audio signal or the timeout window has expired. during turn-off, enhance volume slewing is always disabled. table 31. click-and-pop reduction register register bit name description 0x49 7 vs2en enhanced volume smoothing during volume slewing, the controller waits for each step in the ramp to be applied before sending the next step. when zero-crossing detection is enabled this prevents large steps in the output volume when no zero crossings are detected. 0 = enabled 1 = disabled applies to volume changes in hpvoll, hpvolr, recvol, spvoll, and spvolr. 6 vsen volume adjustment smoothing volume changes are smoothed by stepping through intermediate steps. also ramps the volume from minimum to the programmed value at turn-on and back to minimum at turn-off. 0 = enabled 1 = disabled applies to volume changes in hpvoll, hpvolr, recvol, spvoll, and spvolr. 5 zden zero-crossing detection holds volume changes until there is a zero crossing in the audio signal. this reduces click and pop during volume changes (zipper noise). if no zero crossing is detected within 100ms, the volume change is forced. 0 = enabled 1 = disabled applies to volume changes in pgam1, pgam2, pgaouta, pgaoutb, pgaoutc, hpvoll, hpvolr, recvol, spvoll, and spvolr. 1 eq2en see the 5-band parametric eq section. 0 eq1en datasheet.in stereo audio codec with flexsound technology MAX9888 100 jack detection the ic features jack detection that can detect the inser - tion and removal of a jack as well as the load type. when a jack is detected, an interrupt on irq can be triggered to alert the microcontroller of the event. figure 32 shows the typical configuration for jack detection. jack insertion to detect a jack insertion, the ic must have a power supply and micbias should be disabled. set jdeten to enable jack detection circuitry and apply a pullup cur - rent to jacksns. set jdwk to minimize supply current. clear jdwk to differentiate between headsets with a microphone and headphones without a microphone. the voltage on jacksns is equal to spklvdd as long as no load is applied to jacksns. table 32 shows the change in jksns that occurs when a jack is inserted. accessory button detection after jack insertion, the MAX9888 can detect button presses on accessories that include a microphone and a switch that shorts the microphone signal to ground. set jdeten to enable jack detection circuitry. a pullup current is automatically applied to jacksns if micbias is disabled. clear jdwk to allow differentiation between the microphone load and a short to ground. button presses can be detected both when micbias is enabled and disabled. table 33 shows the change in jksns that occurs when the accessory button is pressed. figure 32. typical configuration for jack detection table 32. change in jksns upon jack insertion table 33. change in jksns upon button press jack type jdwk = 1 jdwk = 0 jksns: 11 00 jksns: 11 00 jksns: 11 00 jksns: 11 01 jack type micbias enabled or disabled jksns: 01 00 hpl micbias jacksns mic1p hpr gnd gnd r l mic gnd r l mic gnd r l datasheet.in stereo audio codec with flexsound technology MAX9888 101 jack removal the ic detects jack removal by monitoring jacksns for transitions to the 11 state. set jdeten to enable jack detection circuitry. a pullup current is automatically applied to jacksns if micbias is disabled. set jdwk to minimize supply current if button detection is not required. table 34 shows the change in jksns that occurs when a jack is removed. table 34. change in jksns upon jack removal table 35. jack detection registers jack type jdwk = 1 and micbias disabled jdwk = 0 or micbias enabled jksns: 00 11 jksns: 00 11 jksns: 00 11 jksns: 01 11 register bit name description 0x02 (read only) 7 jksns jacksns state reports the status of jacksns when jdeten = 1. value mode description 00 mben = 1 v jacksns < 0.1 x v micbias mben = 0 v jacksns < 0.1 x v spklvdd 01 mben = 1 0.1 x v micbias < v jacksns < 0.95 x v micbias 6 mben = 0 0.1 x v spklvdd < v jacksns < 0.95 x v spklvdd 10 mben = 1 reserved mben = 0 reserved 11 mben = 1 0.95 x v micbias < v jacksns mben = 0 0.95 x v spklvdd < v jacksns 0x49 7 jdeten jack detection enable 0 = disabled 1 = enabled 1 jdeb jack detection debounce configures the debounce time for setting jdet. 00 = 25ms 01 = 50ms 10 = 100ms 11 = 200ms 0 0x4c 7 shdn see the power management section. 6 vbaten see the battery measurement section. 1 jdwk jacksns pullup when jdwk = 1 jacksns is slow to increase in voltage. set jdwk = 0 before setting jdeten = 1 to prevent false detection. valid when mbias = 0 or shdn = 0. 0 = 2.4ki to spklvdd (allows microphone detection) 1 = 5fa to spklvdd (minimizes supply current) gnd gnd r l mic gnd r l datasheet.in stereo audio codec with flexsound technology MAX9888 102 battery measurement the ic measures the voltage applied to spklvdd (typi - cally the battery voltage) and reports the value in regis - ter 0x03. this value is also used by the speaker limiter circuitry to set accurate thresholds. when the battery measurement function is disabled, the battery voltage is user programmable. table 36. battery measurement registers register bit name description 0x03 4 vbat battery voltage read vbat when v baten = 1 to determine v spklvdd . program vbat when v baten = 0 to allow proper speaker amplifier signal processing. calculate the battery voltage using the following formula: v battery = 2.55v + [vbat/10] 3 2 1 0 0x4c 7 shdn see the power management section. 6 vbaten battery measurement enable enables an internal adc to measure v spklvdd . 0 = disabled (register 0x03 readable and writeable) 1 = enabled (register 0x03 read only) 1 jdwk see the headset detection section. datasheet.in stereo audio codec with flexsound technology MAX9888 103 table 37. status and interrupt registers device status the ic uses register 0x00 and irq to report the status of various device functions. the status register bits are set when their respective events occur, and cleared upon reading the register. device status can be determined either by poling register 0x00 or configuring the irq to pull low when specific events occur. irq is an open- drain output that requires a pullup resistor for proper operation. register 0x0f determines which bits in the status register trigger irq to pull low. register bit name description 0x00 (read only) 7 cld full scale 0 = all digital signals are less than full scale. 1 = the dac or adc signal path has reached or exceeded full scale. this typically indicates clipping. 6 sld volume slew complete sld reports that any of the programmable-gain arrays or volume controllers has completed slewing from a previous setting to a new programmed setting. if multiple gain arrays or volume controllers are changed at the same time, the sld flag is set after the last volume slew completes. sld also reports when the digital audio interface soft-start or soft-stop process has completed. mclk is required for proper sld operation. 0 = no volume slewing sequences have completed since the status register was last read. 1 = volume slewing complete. 5 ulk digital audio interface unlocked 0 = both digital audio interfaces are operating normally. 1 = either digital audio interface is configured incorrectly or receiving invalid data. 1 jdet jack configuration change jdet reports changes to any bit in the jack status register (0x02). changes to the jack status bits are debounced before setting jdet. the debounce period is programmable using the jdeb bits. jdet is always set the first time jdeten or shdn is set the first time power is applied to the ic. read the status register following such an event to clear jdet and allow for proper jack detection. 0 = no change in jack configuration. 1 = jack configuration has changed. 0x0f 7 icld full-scale interrupt enable 0 = disabled 1 = enabled 6 isld volume slew complete interrupt enable 0 = disabled 1 = enabled 5 iulk digital audio interface unlocked interrupt enable 0 = disabled 1 = enabled 1 ijdet jack configuration change interrupt enable 0 = disabled 1 = enabled datasheet.in stereo audio codec with flexsound technology MAX9888 104 table 38. device revision register figure 33. start, stop, and repeated start conditions i 2 c serial interface the ic features an i 2 c/smbus k -compatible, 2-wire serial interface comprising a serial-data line (sda) and a serial-clock line (scl). sda and scl facilitate com - munication between the ic and the master at clock rates up to 400khz. figure 5 shows the 2-wire interface timing diagram. the master generates scl and initiates data transfer on the bus. the master device writes data to the ic by transmitting the proper slave address followed by the register address and then the data word. each trans - mit sequence is framed by a start (s) or repeated start (sr) condition and a stop (p) condition. each word transmitted to the ic is 8 bits long and is followed by an acknowledge clock pulse. a master reading data from the ic transmits the proper slave address followed by a series of nine scl pulses. the ic transmits data on sda in sync with the master-generated scl pulses. the master acknowledges receipt of each byte of data. each read sequence is framed by a start or repeated start condition, a not acknowledge, and a stop condi - tion. sda operates as both an input and an open-drain output. a pullup resistor, typically greater than 500 i, is required on sda. scl operates only as an input. a pullup resistor, typically greater than 500 i , is required on scl if there are multiple masters on the bus, or if the single master has an open-drain scl output. series resistors in line with sda and scl are optional. series resistors protect the digital inputs of the ic from high voltage spikes on the bus lines, and minimize crosstalk and undershoot of the bus signals. bit transfer one data bit is transferred during each scl cycle. the data on sda must remain stable during the high period of the scl pulse. changes in sda while scl is high are con - trol signals (see the start and stop conditions section). start and stop conditions sda and scl idle high when the bus is not in use. a master initiates communication by issuing a start condition. a start condition is a high-to-low transition on sda with scl high. a stop condition is a low-to- high transition on sda while scl is high (figure 33). a start condition from the master signals the beginning of a transmission to the ic. the master terminates trans - mission, and frees the bus, by issuing a stop condition. the bus remains active if a repeated start condition is generated instead of a stop condition. early stop conditions the ic recognizes a stop condition at any point during data transmission except if the stop condition occurs in the same high pulse as a start condition. for proper operation, do not send a stop condition during the same scl high pulse as the start condition. smbus is a trademark of intel corp. device revision register bit name description 0xff (read only) 7 rev device revision code rev is always set to 0x43. 6 5 4 3 2 1 0 scl sda s sr p datasheet.in stereo audio codec with flexsound technology MAX9888 105 slave address the slave address is defined as the seven most signifi - cant bits (msbs) followed by the read/write bit. for the ic, the seven most significant bits are 0010000. setting the read/write bit to 1 (slave address = 0x21) configures the ic for read mode. setting the read/write bit to 0 (slave address = 0x20) configures the ic for write mode. the address is the first byte of information sent to the ic after the start condition. acknowledge the acknowledge bit (ack) is a clocked 9th bit that the ic uses to handshake receipt each byte of data when in write mode (figure 34). the ic pulls down sda dur - ing the entire master-generated 9th clock pulse if the previous byte is successfully received. monitoring ack allows for detection of unsuccessful data transfers. an unsuccessful data transfer occurs if a receiving device is busy or if a system fault has occurred. in the event of an unsuccessful data transfer, the bus master retries communication. the master pulls down sda during the 9th clock cycle to acknowledge receipt of data when the ic is in read mode. an acknowledge is sent by the master after each read byte to allow data transfer to continue. a not acknowledge is sent when the master reads the final byte of data from the ic, followed by a stop condition. write data format a write to the ic includes transmission of a start condi - tion, the slave address with the r/ w bit set to 0, one byte of data to configure the internal register address pointer, one or more bytes of data, and a stop condition. figure 35 illustrates the proper frame format for writing one byte of data to the ic. figure 35 illustrates the frame format for writing n-bytes of data to the ic. figure 34. acknowledge figure 35. writing one byte of data to the ic figure 36. writing n-bytes of data to the ic 1 scl start condition sda 2 8 9 clock pulse for acknowledgment acknowledge not acknowledge 1 byte autoincrement internal register address pointer acknowledge from MAX9888 b7 b6 b5 b4 b3 b2 b1 b0 acknowledge from MAX9888 acknowledge from MAX9888 s o a a a p slave address r/ w register addres s data byte acknowledge from MAX9888 slave address register address data byte 1 autoincrement internal register address pointer 1 byte acknowledge from MAX9888 acknowledge from MAX9888 acknowledge from MAX9888 b7 b6 b5 b4 b3 b2 b1 b0 b7 b6 b5 b4 b3 b2 b1 b0 s o a a a data byte n 1 byte p a r/w datasheet.in stereo audio codec with flexsound technology MAX9888 106 the slave address with the r/ w bit set to 0 indicates that the master intends to write data to the ic. the ic acknowledges receipt of the address byte during the master-generated 9th scl pulse. the second byte transmitted from the master configures the ics internal register address pointer. the pointer tells the ic where to write the next byte of data. an acknowledge pulse is sent by the ic upon receipt of the address pointer data. the third byte sent to the ic contains the data that is written to the chosen register. an acknowledge pulse from the ic signals receipt of the data byte. the address pointer autoincrements to the next register address after each received data byte. this autoincrement feature allows a master to write to sequential registers within one continuous frame. the master signals the end of transmission by issuing a stop condition. register addresses greater than 0xc7 are reserved. do not write to these addresses. read data format send the slave address with the r/ w bit set to 1 to initi- ate a read operation. the ic acknowledges receipt of its slave address by pulling sda low during the 9th scl clock pulse. a start command followed by a read com - mand resets the address pointer to register 0x00. the first byte transmitted from the ic is the content of register 0x00. transmitted data is valid on the rising edge of scl. the address pointer autoincrements after each read data byte. this autoincrement feature allows all registers to be read sequentially within one continu - ous frame. a stop condition can be issued after any number of read data bytes. if a stop condition is issued followed by another read operation, the first data byte to be read is from register 0x00. the address pointer can be preset to a specific register before a read command is issued. the master presets the address pointer by first sending the ics slave address with the r/ w bit set to 0 followed by the register address. a repeated start condition is then sent fol - lowed by the slave address with the r/ w bit set to 1. the ic then transmits the contents of the specified register. the address pointer autoincrements after transmitting the first byte. the master acknowledges receipt of each read byte during the acknowledge clock pulse. the master must acknowledge all correctly received bytes except the last byte. the final byte must be followed by a not acknowl - edge from the master and then a stop condition. figure 37 illustrates the frame format for reading one byte from the ic. figure 38 illustrates the frame format for reading multiple bytes from the ic. figure 37. reading one byte of data from the ic figure 38. reading n bytes of data from the ic acknowledge from MAX9888 acknowledge from MAX9888 acknowledge from MAX9888 not acknowledge from master autoincrement internal register address pointer 1 byte p repeated start s o a a sr 1 a a slave address register address slave address data byte r/ w r/ w acknowledge from MAX9888 acknowledge from MAX9888 acknowledge from MAX9888 autoincrement internal register address pointer 1 byte repeated start s o a a sr 1 a a slave address register address slave address data byte r/ w r/ w datasheet.in stereo audio codec with flexsound technology MAX9888 107 applications information typical operating circuits figures 39 and 40 provide example operating circuits for the ic. the external components shown are the minimum required for the ic to operate. additional components may be required by the application. figure 39. typical application circuit using analog microphone inputs and the bypass switch analog microphones and bypass switch 2.2ki 1ki 1ki irq 1.8v to 5.5v 10ki to microcontroller 10mhz to 60mhz clock input digital audio port 1 i 2 c control port microphone output to baseband jacksns headset microphone 1ff 1ff 1ff 1ff 2.2ff 1ff handset microphone 1ff 1ff digital audio port 2 jacksns bypass switch input lrclks2 bclks2 sdins2 sdouts2 jacksns rec p/ rxinp rec n/ rxinn spklp spkln spkrp spkrn hpr hpl hpsns ref preg reg mclk bclks1 lrclks1 sdins1 sdouts1 sda scl mic1 p/ digmicdata mic1 n/ digmicclk micbias mic2p mic2n ina1/extmicp ina2/extmicn inb1 inb2 line input 1ff 1ff 1ff 1ff agnd hpgnd spkrgnd spklgnd c1n hpvss c1p dgnd dvdds1 dvdd hpvdd avdd spklvdd spkrvdd dvdds2 1f f 1ff 10ff 1.8v to 3.6v 2.8v to 5.5v 1.8v 1.8v to 3.6v 1ff 1ff 1ff 1ff 1ff 8i 8i MAX9888 datasheet.in stereo audio codec with flexsound technology MAX9888 108 figure 40. typical application circuit using the digital microphone input and receiver amplifier digital microphones and receiver amplifier irq 10ki to microcontroller 10mhz to 60mhz clock input digital audio port 1 i 2 c control port 1ff 1ff 2.2ff digital audio port 2 jacksns lrclks2 bclks2 sdins2 sdouts2 jacksns recp / rxinp rec n/ rxinn spklp spkln spkrp spkrn hpr hpl hpsns ref preg reg mclk bclks1 lrclks1 sdins1 sdouts1 sda scl mic1 p/ digmicdata mic1 n / digmicclk micbias mic2p mic2n ina1 / extmicp ina2 / extmicn inb1 inb2 2.2ki digital mic 1 digital mic 2 jacksns headset microphone 1ff 1ff 1ff data clock data clock 1ff 1ff agnd hpgnd spkrgnd spklgnd c1n hpvss c1p dgnd line input 1ff 1ff line input 1ff 1ff 32i dvdds1 dvdd hpvdd avdd spklvdd spkrvdd dvdds2 1f f 1ff 10ff 1.8v to 3.6v 1.8v 1.8v to 3.6v 1ff 1ff 1ff 1ff 1ff 2.8v to 5.5v 1.8v to 5.5v MAX9888 8i 8i datasheet.in stereo audio codec with flexsound technology MAX9888 109 filterless class d operation traditional class d amplifiers require an output filter to recover the audio signal from the amplifiers output. the filters add cost, increase the solution size of the amplifier, and can decrease efficiency and thd+n performance. the traditional pwm scheme uses large differential output swings (2 x v dd peak to peak) and causes large ripple currents. any parasitic resistance in the filter components results in a loss of power, lowering the efficiency. the ic does not require an output filter. the device relies on the inherent inductance of the speaker coil and the natural filtering of both the speaker and the human ear to recover the audio component of the square-wave out - put. eliminating the output filter results in a smaller, less costly, more efficient solution. because the frequency of the ic output is well beyond the bandwidth of most speakers, voice coil move - ment due to the square-wave frequency is very small. although this movement is small, a speaker not designed to handle the additional power can be damaged. for optimum results, use a speaker with a series inductance > 10 f h. typical 8i speakers exhibit series inductances in the 20 f h to 100f h range. rf susceptibility gsm radios transmit using time-division multiple access (tdma) with 217hz intervals. the result is an rf signal with strong amplitude modulation at 217hz and its har - monics that is easily demodulated by audio amplifiers. the ic is designed specifically to reject rf signals; how - ever, pcb layout has a large impact on the susceptibility of the end product. in rf applications, improvements to both layout and com - ponent selection decrease the ics susceptibility to rf noise and prevent rf signals from being demodulated into audible noise. trace lengths should be kept below 1/4 of the wavelength of the rf frequency of interest. minimizing the trace lengths prevents them from functioning as anten - nas and coupling rf signals into the ic. the wavelength ( l ) in meters is given by: l = c/f where c = 3 x 10 8 m/s, and f = the rf frequency of interest. route audio signals on middle layers of the pcb to allow ground planes above and below to shield them from rf interference. ideally, the top and bottom layers of the pcb should primarily be ground planes to create effec - tive shielding. additional rf immunity can also be obtained by rely - ing on the self-resonant frequency of capacitors as it exhibits a frequency response similar to a notch filter. depending on the manufacturer, 10pf to 20pf capaci - tors typically exhibit self resonance at the rf frequencies of interest. these capacitors, when placed at the input pins, can effectively shunt the rf noise to ground. for these capacitors to be effective, they must have a low- impedance, low-inductance path to the ground plane. avoid using microvias to connect to the ground plane whenever possible as these vias do not conduct well at rf frequencies. startup/shutdown sequencing to ensure proper device initialization and minimal click- and-pop, program the ics shdn = 1 after configuring all registers. table 39 lists an example startup sequence for the device. to shut down the ic, simply set shdn = 0. table 39. example startup sequence sequence description registers 1 ensure shdn = 0 0x4c 2 configure clocks 0x10 to 0x13, 0x19 to 0x1b 3 configure digital audio interface 0x14 to 0x17, 0x1c to 0x1f 4 configure digital signal processing 0x18, 0x20, 0x3d to 0x44 5 load coefficients 0x50 to 0xc7 6 configure mixers 0x21 to 0x29 7 configure gain and volume controls 0x2a to 0x3c 8 configure miscellaneous functions 0x45 to 0x49 9 enable desired functions 0x4a, 0x4b 10 set shdn = 1 0x4c datasheet.in stereo audio codec with flexsound technology MAX9888 110 while many configuration options in the ic can be made while the device is operating, some registers should only be adjusted when the corresponding audio path is disabled. table 40 lists the registers that are sensitive during operation. either disable the corresponding audio path or set shdn = 0 while changing these registers. component selection optional ferrite bead filter in applications where speaker leads exceed 20mm, additional emi suppression can be achieved by using a filter constructed from a ferrite bead and a capacitor to ground (figure 41). use a ferrite bead with low dc resis - tance, high-frequency (> 600mhz) impedance between 100i and 600i , and rated for at least 1a. the capacitor value varies based on the ferrite bead chosen and the actual speaker lead length. select a capacitor less than 1nf based on emi performance. input capacitor an input capacitor, c in , in conjunction with the input impedance of the ic line inputs forms a highpass filter that removes the dc bias from an incoming analog signal. the ac coupling capacitor allows the amplifier to automatically bias the signal to an optimum dc level. assuming zero-source impedance, the -3db point of the highpass filter is given by: -3db in in 1 f 2 r c = choose c in so that f -3db is well below the lowest fre - quency of interest. for best audio quality use capacitors whose dielectrics have low-voltage coefficients, such as tantalum or aluminum electrolytic. capacitors with high- voltage coefficients, such as ceramics, may result in increased distortion at low frequencies. charge-pump capacitor selection use capacitors with an esr less than 100m i for optimum performance. low-esr ceramic capacitors minimize the output resistance of the charge pump. most surface- mount ceramic capacitors satisfy the esr requirement. for best performance over the extended temperature range, select capacitors with an x7r dielectric. table 40. registers that are sensitive to changes during operation figure 41. optional class d ferrite bead filter register description 0x10 to 0x13, 0x19 to 0x1b clock control registers 0x14 to 0x17, 0x1c to 0x1f digital audio interface configuration 0x18, 0x20 digital passband filters 0x24 to 0x29 analog mixers 0x50 to 0xc7 digital signal processing coefficients MAX9888 spk_p spk_n datasheet.in stereo audio codec with flexsound technology MAX9888 111 charge-pump flying capacitor the value of the flying capacitor (connected between c1n and c1p) affects the output resistance of the charge pump. a value that is too small degrades the devices ability to provide sufficient current drive, which leads to a loss of output voltage. increasing the value of the flying capacitor reduces the charge-pump output resistance to an extent. above 1 f f, the on-resistance of the internal switches and the esr of external charge- pump capacitors dominate. charge-pump holding capacitor the holding capacitor (bypassing hpvss) value and esr directly affect the ripple at hpvss. increasing the capacitors value reduces output ripple. likewise, decreasing the esr reduces both ripple and output resistance. lower capacitance values can be used in systems with low maximum output power levels. see the output power vs. load resistance graph in the typical operating characteristics section for more information unused pins table 41 shows how to connect the ics pins when unused. table 41. unused pins name connection name connection spkrp unconnected inb1 unconnected spkrvdd always connected ina2/micextn unconnected spklvdd always connect lrclks2 unconnected spklp unconnected mclk always connect recn/rxinn unconnected sdins2 agnd hpvdd always connect irq unconnected c1p unconnected mic1p/digmicdata unconnected hpgnd agnd ina1/micextp unconnected spkrn unconnected dgnd always connect spkrgnd always connect bclks2 unconnected spklgnd always connect sda always connect spkln unconnected scl always connect recp/rxinp unconnected reg always connect c1n unconnected ref always connect hpl unconnected mic1n/digmicclk unconnected hpvss unconnected mic2p unconnected sdins1 agnd sdouts2 unconnected lrclks1 unconnected dvdds2 dvdd hpsns agnd dvdd always connect inb2 unconnected avdd always connect hpr unconnected preg always connect dvdds1 dvdd agnd always connect sdouts1 unconnected micbias unconnected bclks1 unconnected mic2n unconnected jacksns unconnected datasheet.in stereo audio codec with flexsound technology MAX9888 112 recommended pcb routing the ic uses a 63-bump wlp package. figure 42 provides an example of how to connect to all active bumps using 3 layers of the pcb. to ensure uninter - rupted ground returns, use layer 2 as a connecting layer between layer 1 and layer 2 and flood the remaining area with ground. supply bypassing, layout, and grounding proper layout and grounding are essential for optimum performance. when designing a pcb for the ic, parti - tion the circuitry so that the analog sections of the ic are separated from the digital sections. this ensures that the analog audio traces are not routed near digital traces. use a large continuous ground plane on a dedicated layer of the pcb to minimize loop areas. connect agnd, dgnd, hpgnd, spklgnd, and spkrgnd directly to the ground plane using the shortest trace length pos - sible. proper grounding improves audio performance, minimizes crosstalk between channels, and prevents any digital noise from coupling into the analog audio signals. ground the bypass capacitors on micbias, reg, preg, and ref directly to the ground plane with minimum trace length. also be sure to minimize the path length to agnd. bypass avdd directly to agnd. connect all digital i/o termination to the ground plane with minimum path length to dgnd. bypass dvdd, dvdds1, and dvdds2 directly to dgnd. place the capacitor between c1p and c1n as close as possible to the ic to minimize trace length from c1p to c1n. inductance and resistance added between c1p and c1n reduce the output power of the headphone amplifier. bypass hpvss with a capacitor located close to hpvss with a short trace length to hpgnd. close decoupling of hpvss minimizes supply ripple and maxi - mizes output power from the headphone amplifier. hpsns senses ground noise on the headphone jack and adds the same noise to the output audio signal, thereby making the output (headphone output minus ground) noise free. connect hpsns to the headphone jack shield to ensure accurate pickup of headphone ground noise. bypass spklvdd and spkrvdd to spklgnd and spkrgnd, respectively, with as little trace length as possible. connect spklp, spkln, spkrp, and spkrn to the stereo speakers using the shortest traces pos - sible. reducing trace length minimizes radiated emi. route spklp/spkln and spkrp/spkrn as differential pairs on the pcb to minimize loop area, thereby the inductance of the circuit. if filter components are used on the speaker outputs, be sure to locate them as close as possible to the ic to ensure maximum effectiveness. minimize the trace length from any ground-connected passive components to spklgnd and spkrgnd to further minimize radiated emi. figure 42. suggested routing layer 1 layer 2 layer 3 datasheet.in stereo audio codec with flexsound technology MAX9888 113 route microphone signals from the microphone to the ic as a differential pair, ensuring that the positive and nega - tive signals follow the same path as closely as possible with equal trace length. when using single-ended micro - phones or other single-ended audio sources, ground the negative microphone input as close as possible to the audio source and then treat the positive and negative traces as differential pairs. an evaluation kit (ev kit) is available to provide an exam - ple layout for the ic. the ev kit allows quick setup of the ic and includes easy-to-use software allowing all internal registers to be controlled. wlp applications information for the latest application details on wlp construction, dimensions, tape carrier information, pcb techniques, bump-pad layout, and recommended reflow tempera - ture profile, as well as the latest information on reliability testing results, refer to the application note 1891: wafer- level packaging (wlp) and its applications . figure 43 shows the dimensions of the wlp balls used on the ic. figure 43. wlp ball dimensions 0.24m m 0.21m m datasheet.in stereo audio codec with flexsound technology MAX9888 114 package information for the latest package outline information and land patterns, go to www.maxim-ic.com/packages. note that a +, #, or - in the package code indicates rohs status only. package drawings may show a different suffix character, but the drawing pertains to the package regardless of rohs status. package type package code outline n o. land pattern no. 63 wlp w633a3+1 21-0462 datasheet.in stereo audio codec with flexsound technology MAX9888 maxim cannot assume responsibility for use of any circuitry other than circuitry entirely embodied in a maxim product. no circuit patent licenses are implied. maxim reserves the right to change the circuitry and specifications without notice at any time. maxim integrated products, 120 san gabriel drive, sunnyvale, ca 94086 408-737-7600 115 ? 2010 maxim integrated products maxim is a registered trademark of maxim integrated products, inc. revision history revision number revision date description pages changed 0 6/10 initial release datasheet.in |
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