Active Noise Cancellation

The SHARC ADSP-21489 is one of two new members of the fourth generation of SHARC^® Processors that now includes the ADSP-21483, ADSP-21486, ADSP-21487, ADSP-21488, ADSP-21489 and offers increased performance, hardware-based filter accelerators, audio and application-focused peripherals, and new memory configurations capable of supporting the latest surround-sound decoder algorithms. All devices are pin-compatible with each other and completely code-compatible with all prior SHARC Processors. These newest members of the fourth generation SHARC Processor family are based on a single-instruction, multiple-data (SIMD) core, which supports both 32-bit fixed-point and 32-/40-bit floating-point arithmetic formats making them particularly suitable for high-performance audio applications.

The ADSP-21489 offers the highest performance–450 MHz/2700 MFLOPs–in an LQFP package within the fourth generation SHARC Processor family. This level of performance makes the ADSP-21489 particularly well suited to address the automotive audio and industrial control segments. In addition to its high core performance, the ADSP-21489 includes additional processing blocks such as FIR, IIR, and FFT accelerators to increase the total performance of the system. There is a new feature called Variable Instruction Set Architecture (VISA) that allows the code size to be decreased by 20% to 30% and increase the memory size availability. The fourth generation DSP allows the ability to connect to external memory by providing a glueless interface to 16-bit wide SDR SDRAMs.

Fourth-generation SHARC Processors also integrate application-specific peripherals designed to simplify hardware design, minimize design risks, and ultimately reduce time to market. Grouped together, and broadly named the Digital Applications Interface (DAI), these functional blocks may be connected to each other or to external pins via the software-programmable Signal Routing Unit (SRU). The SRU is an innovative architectural feature that enables complete and flexible routing amongst DAI blocks. Peripherals connected through the SRU include but are not limited to serial ports, IDP, S/PDIF Tx/Rx, and an 8-Channel asynchronous sample rate converter block. The fourth generation SHARC allows data from the serial ports to be directly transferred to external memory by the DMA controller. Other peripherals such as SPI,UART and Two-Wire Interface are routed through a Digital Peripheral Interface (DPI).

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 Automotive Audio Bus (A²B^®) provides a multichannel, I²S/TDM link over distances of up to 15 m between nodes. It embeds bidirectional synchronous pulse-code modulation (PCM) data (for example, digital audio), clock, and synchronization signals onto a single differential wire pair. A²B supports a direct point to point connection and allows multiple, daisy-chained nodes at different locations to contribute and/or consume time division multiplexed (TDM) channel content.

A²B is a single-master, multiple-slave system where the transceiver at the host controller is the master. The master generates clock, synchronization, and framing for all slave nodes. The master A²B transceiver is programmable over a control port (I²C) for configuration and read back. An extension of this control port is embedded in the A²B data stream, which grants direct access of registers and status information on slave transceivers as well as I²C to I²C communication over distance.

The transceiver can connect directly to general-purpose digital signal processors (DSPs), field-programmable gate arrays (FPGAs), application specific integrated circuits (ASICs), microphones, analog-to-digital converters (ADCs), digital-to-analog converters (DACs), and codecs through a multichannel I²S/TDM interface. It also provides a pulse density modulation (PDM) interface for direct connection of up to four PDM digital microphones.

Finally, the transceiver also supports an A²B bus powering feature, where the master node supplies voltage and current to the slave nodes over the same daisy-chained, twisted pair wire cable as used for the communication link.

Table 1. Product Comparison Guide

Feature	AD2420/ AD2420W	AD2426/ AD2426W	AD2427/ AD2427W	AD2428/ AD2428W	AD2429/ AD2429W
Master capable	No	No	No	Yes	Yes
Number of slaves discoverable1	N/A	N/A	N/A	Up to 10	Up to 2
Functional TRX blocks	A only	A only	A + B	A + B	B only
I2S/TDM support	No	No	No	Yes	Yes
PDM microphone inputs	2 mics2	4 mics	4 mics	4 mics	4 mics
Max node to node cable length	5 m	15 m	15 m	15 m	5 m

¹ N/A means not applicable.
² PDM microphones must be connected to the DRX0/IO5 pin.

Applications

• Audio communication link
• Microphone arrays
  
  • Beamforming
  • Hands free and in car communication
• Active and road noise cancellation
• Audio/video conferencing systems

The Automotive Audio Bus (A²B^®) provides a multichannel, I²S/TDM link over distances of up to 15 m between nodes. It embeds bidirectional synchronous pulse-code modulation (PCM) data (for example, digital audio), clock, and synchronization signals onto a single differential wire pair. A²B supports a direct point to point connection and allows multiple, daisy-chained nodes at different locations to contribute and/or consume time division multiplexed (TDM) channel content.

A²B is a single-master, multiple-slave system where the transceiver at the host controller is the master. The master generates clock, synchronization, and framing for all slave nodes. The master A²B transceiver is programmable over a control port (I²C) for configuration and read back. An extension of this control port is embedded in the A²B data stream, which grants direct access of registers and status information on slave transceivers as well as I²C to I²C communication over distance.

The transceiver can connect directly to general-purpose digital signal processors (DSPs), field-programmable gate arrays (FPGAs), application specific integrated circuits (ASICs), microphones, analog-to-digital converters (ADCs), digital-to-analog converters (DACs), and codecs through a multichannel I²S/TDM interface. It also provides a pulse density modulation (PDM) interface for direct connection of up to four PDM digital microphones.

Finally, the transceiver also supports an A²B bus powering feature, where the master node supplies voltage and current to the slave nodes over the same daisy-chained, twisted pair wire cable as used for the communication link.

Table 1. Product Comparison Guide

Feature	AD2420/ AD2420W	AD2426/ AD2426W	AD2427/ AD2427W	AD2428/ AD2428W	AD2429/ AD2429W
Master capable	No	No	No	Yes	Yes
Number of slaves discoverable1	N/A	N/A	N/A	Up to 10	Up to 2
Functional TRX blocks	A only	A only	A + B	A + B	B only
I2S/TDM support	No	No	No	Yes	Yes
PDM microphone inputs	2 mics2	4 mics	4 mics	4 mics	4 mics
Max node to node cable length	5 m	15 m	15 m	15 m	5 m

¹ N/A means not applicable.
² PDM microphones must be connected to the DRX0/IO5 pin.

Applications

• Audio communication link
• Microphone arrays
  
  • Beamforming
  • Hands free and in car communication
• Active and road noise cancellation
• Audio/video conferencing systems

The Automotive Audio Bus (A²B^®) provides a multichannel, I²S/TDM link over distances of up to 15 m between nodes. It embeds bidirectional synchronous pulse-code modulation (PCM) data (for example, digital audio), clock, and synchronization signals onto a single differential wire pair. A²B supports a direct point to point connection and allows multiple, daisy-chained nodes at different locations to contribute and/or consume time division multiplexed (TDM) channel content.

A²B is a single-master, multiple-slave system where the transceiver at the host controller is the master. The master generates clock, synchronization, and framing for all slave nodes. The master A²B transceiver is programmable over a control port (I²C) for configuration and read back. An extension of this control port is embedded in the A²B data stream, which grants direct access of registers and status information on slave transceivers as well as I²C to I²C communication over distance.

The transceiver can connect directly to general-purpose digital signal processors (DSPs), field-programmable gate arrays (FPGAs), application specific integrated circuits (ASICs), microphones, analog-to-digital converters (ADCs), digital-to-analog converters (DACs), and codecs through a multichannel I²S/TDM interface. It also provides a pulse density modulation (PDM) interface for direct connection of up to four PDM digital microphones.

Finally, the transceiver also supports an A²B bus powering feature, where the master node supplies voltage and current to the slave nodes over the same daisy-chained, twisted pair wire cable as used for the communication link.

Table 1. Product Comparison Guide

Feature	AD2420/ AD2420W	AD2426/ AD2426W	AD2427/ AD2427W	AD2428/ AD2428W	AD2429/ AD2429W
Master capable	No	No	No	Yes	Yes
Number of slaves discoverable1	N/A	N/A	N/A	Up to 10	Up to 2
Functional TRX blocks	A only	A only	A + B	A + B	B only
I2S/TDM support	No	No	No	Yes	Yes
PDM microphone inputs	2 mics2	4 mics	4 mics	4 mics	4 mics
Max node to node cable length	5 m	15 m	15 m	15 m	5 m

¹ N/A means not applicable.
² PDM microphones must be connected to the DRX0/IO5 pin.

Applications

• Audio communication link
• Microphone arrays
  
  • Beamforming
  • Hands free and in car communication
• Active and road noise cancellation
• Audio/video conferencing systems

The ADXL316 is a small, thin, low power, complete 3-axis accelerometer with signal conditioned voltage outputs. The product measures acceleration with a minimum measurement range of ±16 g. It can measure the static acceleration of gravity in tilt sensing applications, as well as dynamic acceleration resulting from motion, shock, or vibration.

The user selects the bandwidth of the accelerometer using the C_X, C_Y, and C_Z capacitors at the X_OUT, Y_OUT, and Z_OUT pins. Bandwidths can be selected to suit the application, with a range of 0.5 Hz to 1600 Hz for the x and y axes, and a range of 0.5 Hz to 550 Hz for the z axis.

The ADXL316 is available in a small, low profile, 4 mm × 4 mm × 1.45 mm, 12-lead, plastic lead frame chip scale package (LFCSP).

Applications

• Cost sensitive, low power, motion and tilt sensing applications
  • Mobile devices
  • Gaming systems
  • Disk drive protection
  • Image stabilization
  • Active noise control (ANC)
• Sports and health devices

The ADAU1977 incorporates four high performance analog-todigital converters (ADCs) with direct-coupled inputs capable of 10 V rms. The ADC uses multibit sigma-delta (Σ-Δ) architecture with continuous time front end for low EMI. The ADCs can be connected to the electret microphone (ECM) directly and provide the bias for powering the microphone. Built-in diagnostic circuitry detects faults on input lines and includes comprehensive diagnostics for faults on microphone inputs. The faults reported are short to battery, short to microphone bias, short to ground, short between positive and negative input pins, and open input terminals. In addition, each diagnostic fault is available as an IRQ flag for ease in system design. An I2C/SPI control port is also included. The ADAU1977 uses only a single 3.3 V supply. The part internally generates the microphone bias voltage. The microphone bias is programmable in a few steps from 5 V to 9 V. The low power architecture reduces the power consumption. An on-chip PLL can derive the master clock from an external clock input or frame clock (sample rate clock). When fed with a frame clock, the PLL eliminates the need for a separate high frequency master clock in the system. The ADAU1977 is available in a 40-lead LFCSP package.

APPLICATIONS

• Automotive audio systems
• Active Noise Cancellation System

The ADAU1978 incorporates four high performance, analog-to-digital converters (ADCs) with 2 V rms capable ac-coupled inputs. The ADCs use a multibit sigma-delta (Σ-Δ) architecture with continuous time front end for low EMI. An I²C/serial peripheral interface (SPI) control port is included that allows a microcontroller to adjust volume and many other parameters. The ADAU1978 uses only a single 3.3 V supply. The part internally generates the required digital DVDD supply. The low power architecture reduces the power consumption. The ADAU1978 is available in a 40-lead LFCSP package. The on-chip PLL can derive the master clock from an external clock input or frame clock (sample rate clock). When fed with the frame clock, it eliminates the need for a separate high frequency master clock in the system.

Note that throughout this data sheet, multifunction pins, such as SCL/CCLK, are referred to either by the entire pin name or by a single function of the pin, for example, CCLK, when only that function is relevant.

APPLICATIONS

• Automotive audio systems
• Active noise cancellation system