The SSM6322 is an integrated, dual-channel audio amplifier solution that interfaces directly with audio DAC/CODEC, maximizing the fidelity of high fidelity audio signal chains. The highly efficient design of the SSM6322 delivers outstandingaudio performance while minimizing power dissipation for maximum battery life in portable applications.The SSM6322 features ?121 dB THD + N at 1 kHz, along with very low output noise from 20 Hz to 20 kHz. The low power operation, high peak output current, and high PSRR make the SSM6322 an ideal candidate for applications that require high fidelity audio, high dynamic range, precision, and low power. This highly integrated drive solution also reduces development time while reducing board space and minimizing external components.The SSM6322 is available in a 24-lead LFCSP package. The SSM6322 operates over the industrial temperature of ?40?C to +85?C.Applications High fidelity headphone driver Mobile phones Bluetooth speakers & headphones Gaming Notebooks and Tablets A/V receivers Professional audio equipment Audio test equipment Automobile infotainment system

The ADMV8432 is a monolithic microwave integrated circuit (MMIC), tunable band-pass filter that features a user selectable pass-band frequency. The 3 dB filter bandwidth is >17% of the center frequency (fCENTER). Additionally, fCENTER can be varied between 16.6 GHz to 30.1 GHz by applying an analog tuning voltage between 0 V to 15 V. The usable pass band corner frequencies (fCORNER) span from 15.1 GHz to 32 GHz. This tunable filter can be used as a much smaller alternative to physically large switched filter banks and cavity tuned filters. This tunable filter has excellent microphonics due to the monolithic design and provides a dynamically adjustable solution in advanced communications applications.APPLICATIONS Test and measurement equipment Military radar and electronic warfare (EW) systems Video satellite (VSAT) communications?

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

Demonstration circuit DC675C is for the evaluation of filter circuits using an LT1568. The LT1568 is a dual 2nd order active-RCfilterbuildingblockwithprecision ±0.75% capacitors and low noise op amps with 180MHz GBW trimmed to ±10% maximum variation. The ±10% GBW variation of the LT1568 op amps allows for minimizing the higher frequency error by decreasing resistor values. The cutoff or center frequency (fC) range of an LT1568 filter is 200kHz to 10MHz (5MHz for a bandpass filter). The low limit of 200kHz was chosen only to minimize resistor noise and DC offsets (using external capacitors the fC frequency can be less than 200kHz). For testing and evaluation, the DC675C assembly is configured as a single 4th order, 500kHz narrow passband bandpass filter. For other possible LT1568 configurations, the DC675C has unused pads for 0805 surface mount resistors and capacitors preconfigured with PCB traces to allow for the following high accuracy LT1568 filter circuits: 4th order lowpass filter 5th order lowpass filter 4th order narrow passband bandpass 4th order wide passband bandpass 4th order highpass filter Refer to the LT1568 data sheet for additional information about filter circuit configurations.

The ADAU1401A is a complete, single-chip audio system with 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.The signal processing of the ADAU1401A 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 ADAU1401A 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.The ADAU1401A 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 ADAU1401A 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 ADAU1401A 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 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

The ADAU7112 converts stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be two microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. The ADAU7112 is specified over the commercial temperature range (?40?C to +85?C). The ADAU7112 is available in a halide-free, 9-ball, 1.26 mm ? 1.26 mm, 0.4 mm pitch, wafer level chip scale package (WLCSP).Applications? Mobile computing Portable electronics Consumer electronics Professional electronics