Nonvolatile, I2C-Compatible 64-Position, Digital Potentiometer Features: Nonvolatile memory maintains wiper settings  64-position digital potentiometer  Compact MSOP-10 (3 mm × 4.9 mm)  I2C-compatible interface VLOGIC pin provides increased interface flexibility  End-to-end resistance 1 kΩ, 10 kΩ, 50 kΩ, 100 kΩ Resistance tolerance stored in EEPROM (0.1% accuracy) Power-on EEPROM refresh time <1 ms Software write protect command  Address Decode Pin AD0 and Address Decode Pin AD1 allow four packages per bus  100-year typical data retention at 55°C  Wide operating temperature −40°C to +85°C 

Digital Potentiometer ICs 6 Channel 8-Bit Digital Potentiometer

Digital Potentiometer ICs DUAL 8-BIT POTENTIOMETER

400 MSPS 10-Bit DAC 1.8 V CMOS Direct Digital Synthesizer Features: 400 MSPS internal clock speed Integrated 10-bit DAC 32-bit tuning word Phase noise ≤ –120 dBc/Hz at 1 kHz offset (DAC output) Excellent dynamic performance >75 dB SFDR at 160 MHz (±100 kHz offset) AOUT Serial input/output (I/O) control 1.8 V power supply Software and hardware controlled power-down 48-lead TQFP/EP package PLL REFCLK multiplier (4× to 20×) Internal oscillator; can be driven by a single crystal Phase modulation capability Multichip synchronization

Digital Potentiometer ICs HEX NON VOLATILE POT WITH I/O & MEM

Telecom Interface ICs Quad E1 Transceiver

Digital Potentiometer ICs

The ADAU1442 / ADAU1445?/ ADAU1446?are enhanced audio processors that allow full flexibility in routing all input and output signals. The SigmaDSP? core features full 28-bit processing (56-bit in double-precision mode), synchronous parameter loading for ensuring filter stability, and 100% code efficiency with the SigmaStudio? tools. This DSP allows system designers to compensate for the real-world limitations of speakers, amplifiers, and listening environments, resulting in a dramatic improvement of the perceived audio quality through speaker equalization, multiband compression, limiting, and third-party branded algorithms.The flexible audio routing matrix (FARM) allows the user to multiplex inputs from multiple sources running at various sample rates to or from the SigmaDSP core. This drastically reduces the complexity of signal routing and clocking issues in the audio system. FARM includes up to eight stereo asynchronous sample rate converters (depending on the device model), Sony/ Philips Digital Interconnect Format (S/PDIF) input and output, and serial (I2S) and time division multiplexing (TDM) I/Os. Any of these inputs can be routed to the SigmaDSP core or to any of the asynchronous sample rate converters (ASRCs). Similarly, any one of the output signals can be taken from the SigmaDSP core or from any of the ASRC outputs. This routing scheme, which can be modified at any time via control registers, allows for maximum system flexibility.The ADAU1442, ADAU1445, and ADAU1446 differ only in ASRC functionality and packaging. The ADAU1442 / ADAU1445 contain 16 channels of ASRCs and are packaged in TQFP packages, whereas the ADAU1446 contains no ASRCs and is packaged in an LQFP. The ADAU1442 can handle nine clock domains, the ADAU1445 can handle three clock domains, and the ADAU1446 can handle one clock domain.The ADAU1442 / ADAU1445 / ADAU1446?can be controlled in one of two operational modes: the settings of the chip can be loaded and dynamically updated through the SPI/I2C? port, or the DSP can self-boot from an external EEPROM in a system with no microcontroller. There is also a bank of multipurpose (MP) pins that can be used as general-purpose digital I/Os or as inputs to the 4-channel auxiliary control ADC.The ADAU1442 / ADAU1445 / ADAU1446?are supported by the SigmaStudio graphical development environment. This software includes audio processing blocks such as FIR and IIR filters, dynamics processors, mixers, low level DSP functions, and third-party algorithms for fast development of custom signal flows.APPLICATIONS Automotive audio processing Head units Navigation systems Rear-seat entertainment systems DSP amplifiers (sound system amplifiers) Commercial audio processing

The ADAU1761 is a low power, stereo audio codec with integrated digital audio processing that supports stereo 48 kHz record and playback at 14 mW from a 1.8 V analog supply. The stereo audio ADCs and DACs support sample rates from 8 kHz to 96 kHz as well as a digital volume control.The SigmaDSP? core features 28-bit processing (56-bit double precision). The processor allows system designers to compensate for the real-world limitations of microphones, speakers, amplifiers, and listening environments, resulting in a dramatic improvement in the perceived audio quality through equalization, multiband compression, limiting, and third-party branded algorithms.The SigmaStudio? graphical development tool is used to program the ADAU1761. This software includes audio processing blocks such as filters, dynamics processors, mixers, and low level DSP functions for fast development of custom signal flows.The record path includes an integrated microphone bias circuit and six inputs. The inputs can be mixed and muxed before the ADC, or they can be configured to bypass the ADC. The ADAU1761 includes a stereo digital microphone input.The ADAU1761 includes five high power output drivers (two differential and three single-ended), supporting stereo head-phones, an earpiece, or other output transducer. AC-coupled or capless configurations are supported. Individual fine level controls are supported on all analog outputs. The output mixer stage allows for flexible routing of audio.APPLICATIONS Smartphones/multimedia phones Digital still cameras/digital video cameras Portable media players/portable audio players Phone accessories products

TThe SSM3582A is a fully integrated, high efficiency, digital input stereo Class-D audio amplifier. The device can operate from a single supply and requires only a few external components, significantly reducing the circuit bill of materials.A proprietary, spread spectrum ?-? modulation scheme enables direct connection to the speaker and ensures state-of-the-art analog performance while lowering radiated emissions compared to other Class-D architectures. An optional ultralow electromagnetic interference (EMI) mode significantly reduces radiated emissions above 100 MHz, enabling longer speaker cable lengths. Audio is transmitted digitally to the amplifier, minimizing the possibility of signal corruption in digital environments. The amplifier provides outstanding analog performance, with a 106 dB signal-to-noise ratio and a low 0.004% total harmonic distortion + noise (THD + N).The SSM3582A operates from a single 4.5 V to 16.5 V supply and is capable of delivering 2? 15 W rms continuously into 8 ? and 4 ? loads at The pulse code modulation (PCM) audio serial port supports most common protocols, such as I2S, left justified, and time division multiplexing (TDM), and can address up to 16 devices on a single interface, for up to 32 audio playback channels.IC operation is controlled through a dedicated I2C interface. The two ADDRx pins (2?, five-level) define up to 16 individual addresses in I2C and standalone modes and automatically set the default TDM slots attribution.A micropower shutdown mode is triggered by removing the digital audio interface clock, with a typical current of An automatic power-down feature shuts down the amplifier and the digital-to-analog converter (DAC) when no signal is present at the input, minimizing power consumption during digital silence. The device restarts when nonzero data is present at the input. Mute and unmute transitions are pop and click free.The SSM3582A is specified over the commercial temperature range of ?40?C to +85?C. The device has built in thermal shutdown and output short-circuit protection, as well as an early thermal warning with programmable gain limiting to maintain operation.The SSM3582A is available in a 40-lead, 6 mm ? 6 mm lead frame chip scale package (LFCSP), with a thermal pad to improve heat dissipation.Applications Mobile computing All in one computers Portable electronics Wireless speakers Televisions