The ADV7612 is offered in automotive, professional (no HDCP), and industrial versions. The operating temperature range is?40?C to +85?C.The UG-216 contains critical information that must be used in conjunction with the ADV7612.The ADV7612 is a high quality Xpressview? fast switchingHDMI?-capable receiver. It incorporates a dual input HDMI-capable receiver that supports all mandatory 3D TV formats defined in HDMI 1.4a specification, HDTV formats up to 1080p 36-bit Deep Color, and display resolutions up to UXGA (1600 ? 1200 at 60 Hz).It integrates a CEC controller that supports the capabilitydiscovery and control (CDC) feature. The ADV7612 incorporates Xpressview fast switching on both input HDMI ports. Using Analog Devices, Inc., hardware-based HDCP engine that minimizes software overhead, Xpressview technology allows fast switching between both HDMI input ports in less than 1 second. Each HDMI port has dedicated 5 V detect and Hot Plug? assert pins. The HDMI receiver also includes an integrated program-mable equalizer that ensures robust operation of the interfacewith long cables.The ADV7612 offers a flexible audio output port for audio data extraction from the HDMI stream. HDMI audio formats, including SACD via DSD and HBR, are supported by ADV7612. The HDMI receiver has advanced audio functionality, such as a mute controller that prevents audible extraneous noise in the audio output.The ADV7612 contains one main component processor (CP)that processes the video signals from the HDMI receiver. It provides features such as contrast, brightness and saturation adjustments, STDI detection block, free run, and synchronization alignment controls.Fabricated in an advanced CMOS process, the ADV7612 is provided in a 14 mm ? 14 mm, 100-lead surface-mount LQFP_EP, RoHS-compliant package, and is specified over the ?40?C to +85?C temperature range.APPLICATIONS Projectors Automotive Video conferencing HDTVs AVR, HTiB Soundbars Video switches

The EVAL-MICCANVASZ microphone canvas provides a platform for the development and evaluation of microphone beamforming arrays. 15 microphones, arranged as a ring of eight, a ring of six, and a single in the center, provide flexibility and symmetry suitable to many applications.

The SSM2537 is a PDM digital input Class D power amplifier that offers higher performance than existing DAC plus Class D solutions. The SSM2537 is ideal for power sensitive applications where system noise can corrupt the small analog signal sent to the amplifier, such as mobile phones and portable media players.The SSM2537 combines an audio digital-to-analog converter (DAC), a power amplifier, and a PDM digital interface on a single chip. The integrated DAC plus analog ?-? modulator architecture enables extremely low real-world power consumption from digital audio sources with excellent audio performance. Using the SSM2537, audio can be transmitted digitally to the audio amplifier, significantly reducing the effect of noise sources such as GSM interference or other digital signals on the transmitted audio. The SSM2537 is capable of delivering 2.5 W of continuous output power with <1% THD + N driving a 4 ? load from a 5.0 V supply.The SSM2537 features a high efficiency, low noise modulation scheme that requires no external LC output filters. The closed-loop, three-level modulator design retains the benefits of an all-digital amplifier, yet enables very good PSRR and audio performance. The modulation continues to provide high efficiency even at low output power and has an SNR of 103 dB PDM input. Spread-spectrum pulse density modulation is used to provide lower EMI-radiated emissions compared with other Class D architectures.The SSM2537 has a four-state gain and sample frequency selection pin that can select two different gain settings, optimized for 3.6 V and 5 V operation. This same pin also controls the internal digital filtering and clocking, which can be set for 64 ? fS or 128 ? fS input sample rates to support both 3 MHz and 6 MHz PDM clock rates.The SSM2537 has a micropower shutdown mode with a typical shutdown current of 1 ?A for both power supplies. Shutdown is enabled automatically by gating input clock and data signals. A standby mode can be entered by applying a designated PDM stop condition sequence. The device also includes pop-and-click suppression circuitry. This suppression circuitry minimizes voltage glitches at the output when entering or leaving the low power state, reducing audible noises on activation and deactivation.The SSM2537 is specified over the industrial temperature range of ?40?C to +85?C. It has built-in thermal shutdown and output short-circuit protection. It is available in a 9-ball, 1.3 mm ? 1.3 mm wafer level chip scale package (WLCSP).Applications Mobile handsets

The SSM3302 is a fully integrated, high efficiency, stereo Class-D audio amplifier. The application circuit requires minimal externalcomponents and operates from a single 7 V to 18 V supply. The device is capable of delivering 2 ? 10 W of continuous output power into a 4 ? load (or 2 ? 8 W into 8 ?) with 98 dB. Spread spectrum pulse density modulation (PDM) is used toprovide lower EMI radiated emissions compared with other Class-D architectures. The SSM3302 includes an optional modulation select pin (ultralow EMI emission mode) that significantly reduces the radiated emissions at the Class-D outputs, particularly above 100 MHz. The SSM3302 can pass FCC Class-B emissions testing with an unshielded 20 inch cable using common-mode choke-based filtering.The fully differential input of the SSM3302 provides excellent rejection of common-mode noise on the input. The device also includes a highly flexible gain select pin that only requires one series resistor to choose a gain between 9 dB and 24 dB, with no change to the input impedance. The benefit of this is to improve gain matching between multiple SSM3302 devices within a single application compared with using external resistors to set gain.The SSM3302 includes an integrated voltage regulator thatgenerates a 5 V rail.The SSM3302 has a micropower shutdown mode with a typical shutdown current of 10 ?A. Shutdown is enabled by applying a logic low to the SD pin. The device also includes pop-and-click suppression circuitry that minimizes voltage glitches at the output during turn on and turn off, reducing audible noise during activation and deactivation.Other included features to simplify system level integration of the SSM3302 are input low-pass filtering to suppress out-ofband DAC noise interference to the pulse density modulator, fixed input impedance to simplify component selection across multiple platform production builds, and a thermal warning indicator pin.The SSM3302 is specified over the commercial temperaturerange (?40?C to +85?C). It has built-in thermal shutdown andoutput short-circuit protection. It is available in a halide-free,40-lead, 6 mm ? 6 mm lead frame chip scale package (LFCSP).APPLICATIONS Mobile computing Flat panel televisions Media docking stations Portable electronics Sound bars

The SSM3525 is a fully integrated, high efficiency, mono Class D audio amplifier with digital input and digitized output of output voltage, output current, and PVDD supply. The application circuit requires few external components and can operate from 4.5 V to 17 V (PVDD) and 1.8 V (IOVDD) supplies. It is capable of delivering 8.3 W of continuous output power into an 8 ? load (or 15.3 W into 4 ?) with <1% THD + N from a 12 V supply, or 30.2 W into an 4 ? load from a 17 V power supply, all with <1% THD + N. The SSM3525 features a high efficiency, low noise modulation scheme that requires no external inductor/capacitor (LC) output filters. This scheme continues to provide high efficiency even at low output power. It operates with 92% efficiency at 9W into an 8 ? load, 12V or 89% efficiency at 20 W into 4 ? from a 17 V supply, (should now match the table)and it has an signal-to-noise ratio (SNR) of 107 dB A weighted.Spread spectrum pulse density modulation provides lower electromagnetic interference (EMI) radiated emissions compared with other Class-D architectures, particularly above 100 MHz.The digital input eliminates the need of an external digital-to-analog converter (DAC). The SSM3525 has a micropower shutdown mode with a typical shut?down current of 90 nA at 12 V PVDD supply. Individual sense blocks can be powered down to save power when sense is not needed.The device also includes pop and click suppression circuitry that minimizes voltage glitches at the output during turn on and turn off.Current sensing is accomplished using an integrated analog-to-digital converter (ADC) and internal sense resistor. The digitized voltage and current information can be returned in various serial audio formats, including I2S, time division multiplexing (TDM) and pulse density modulation (PDM).The SSM3525 includes an integrated regulator to generate the required 5 V analog supply. Alternatively, if an external 5 V rail from a dc-to-dc converter is available, it can improve system efficiency.The SSM3525 is designed to operate with an I2C control interface and specified over the temperature range ?40?C to +85?C. It has built-in thermal shutdown and output short-circuit protection. It is available in a halide free, 23-ball, 2.26 mm ? 2.38 mm wafer-level chip scale package (WLCSP).Applications Mobile computing Portable electronics

The LTC1563-2/LTC1563-3 are a family of extremely easy-to-use, active RC lowpass filters with rail-to-rail inputs and outputs and low DC offset suitable for systems with a resolution of up to 16 bits. The LTC1563-2, with a single resistor value, gives a unity-gain Butterworth response. The LTC1563-3, with a single resistor value, gives a unity-gain Bessel response. The proprietary architecture of these parts allows for a simple resistor calculation:R = 10k (256kHz/fC); fC = Cutoff Frequencywhere fC is the desired cutoff frequency. For many applications,this formula is all that is needed to design a filter.By simply utilizing different valued resistors, gain andother responses are achieved.The LTC1563-X features a low power mode, for the lowerfrequency applications, where the supply current is reducedby an order of magnitude and a near zero powershutdown mode.The LTC1563-Xs are available in the narrow SSOP-16package (Same footprint as an SO-8 package).Applications Discrete RC Active Filters and Modules Antialiasing Filters Smoothing or Reconstruction Filters Linear Phase Filtering for Data Communication Phase Locked Loops

The ADAU1462/ADAU1466 are automotive qualified audio processors that far exceed the digital signal processing capabilities of earlier SigmaDSP? devices. They are pin and register compatible with each other, as well as with the ADAU1450/ADAU1451/ADAU1452 SigmaDSP processors. The restructured hardware architecture is optimized for efficient audio processing. The audio processing algorithms support a seamless combination of stream processing (sample by sample), multirate processing, and block processing paradigms. The SigmaStudio? graphical programming tool enables the creation of signal processing flows that are interactive, intuitive, and powerful. The enhanced digital signal processor (DSP) core architecture enables some types of audio processing algorithms to be executed using significantly fewer instructions than were required on previous SigmaDSP generations, leading to vastly improved code efficiency.The 1.2 V, 32-bit DSP core can run at frequencies of up to 294.912 MHz and execute up to 6144 SIMD instructions per sample at the standard sample rate of 48 kHz. Powerful clock generator hardware, including a flexible phase-locked loop (PLL) with multiple fractional integer outputs, supports all industry standard audio sample rates. Nonstandard rates over a wide range can generate up to 15 sample rates simultaneously. These clock generators, along with the on board asynchronous sample rate converters (ASRCs) and a flexible hardware audio routing matrix, make the ADAU1462/ADAU1466 ideal audio hubs that greatly simplify the design of complex multirate audio systems.The ADAU1462/ADAU1466 interface with a wide range of analog-to-digital converters (ADCs), digital-to-analog converters (DACs), digital audio devices, amplifiers, and control circuitry with highly configurable serial ports, I2C, serial peripheral interface (SPI), Sony/Philips Digital Interconnect Format (S/PDIF) interfaces, and multipurpose input/output (I/O) pins. Dedicated decimation filters can decode the pulse code modulation (PDM) output of up to four MEMS microphones.Independent slave and master I2C/SPI control ports allow the ADAU1462/ADAU1466 to be programmed and controlled by an external master device such as a microcontroller, and to program and control slave peripherals directly. Self boot functionality and the master control port enable complex standalone systems.The power efficient DSP core can execute at high computational loads while consuming only a few hundred milliwatts (mW) in typical conditions. This relatively low power consumption and small footprint make the ADAU1462/ADAU1466 ideal replacements for large, general-purpose DSPs that consume more power at the same processing load.Applications Automotive audio processing Head units Distributed processors Rear seat entertainment systems Trunk amplifiers Commercial and professional audio processing

The ADAU1701 is a complete single-chip audio system with a 28-/56-bit audio DSP, ADCs, DACs, and microcontroller-like control interfaces. Signal processing includes equalization, cross-over, bass enhancement, multiband dynamics processing, delay compensation, speaker compensation, and stereo image widening. This processing can be used to compensate for real-world limitations of speakers, amplifiers, and listening environments, providing dramatic improvements in perceived audio quality.Its signal processing is comparable to that found in high end studio equipment. Most processing is done in full 56-bit, double precision mode, resulting in very good low level signal performance. The ADAU1701 is a fully programmable DSP. The easy to use SigmaStudio? software allows the user to graphically configure a custom signal processing flow using blocks such as biquad filters, dynamics processors, level controls, and GPIO interface controls.ADAU1701 programs can be loaded on power-up either from a serial EEPROM through its own self-boot mechanism or from an external microcontroller. On power-down, the current state of the parameters can be written back to the EEPROM from the ADAU1701 to be recalled the next time the program is run.Two ?-? ADCs and four ?-? DACs provide a 98.5 dB analog input to analog output dynamic. Each ADC has a THD + N of ?83 dB, and each DAC has a THD + N of ?90 dB. Digital input and output ports allow a glueless connection to additional ADCs and DACs. The ADAU1701 communicates through an I2C? bus or a 4-wire SPI port.APPLICATIONS Multimedia speaker systems MP3 player speaker docks Automotive head units Minicomponent stereos Digital televisions Studio monitors Speaker crossovers Musical instrument effects processors In-seat sound systems (aircraft/motor coaches)

The ADAU1777 is a codec with four inputs and two outputs thatincorporates a digital processing engine to perform filtering,level control, signal level monitoring, and mixing. The pathfrom the analog input to the DSP core to the analog output isoptimized for low latency and is ideal for noise canceling headsets.With the addition of just a few passive components, a crystal,and an EEPROM for booting, the ADAU1777 provides a completeheadset solution.Note that throughout this data sheet, multifunction pins, such asSCL/SCLK, are referred to either by the entire pin name or by asingle function of the pin, for example, SCLK, when only thatfunction is relevant.Applications Noise canceling handsets, headsets, and headphones Bluetooth??active noise cancelling (ANC) handsets, headsets, and headphones Personal navigation devices Digital still and video cameras

The ADAU1860 is a codec with three inputs and one output that incorporates two digital signal processors (DSPs). The path from the analog input to the DSP core to the analog output is optimized for low latency and is ideal for noise canceling earphone. With the addition of just a few passive components, the ADAU1860 provides a complete earphone solution.APPLICATIONS Noise canceling handsets, headsets, and headphones Bluetooth active noise canceling (ANC) handsets, headsets, and headphones Personal navigation devices Digital still and video cameras Musical instrument effect processors Multimedia speaker systems Smartphones