EVAL-ADAU1761Z Analog Devices Inc, EVAL-ADAU1761Z Datasheet - Page 23
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EVAL-ADAU1761Z
Manufacturer Part Number
EVAL-ADAU1761Z
Description
Eval Board For ADAU1761
Manufacturer
Analog Devices Inc
Series
SigmaDSP®r
Specifications of EVAL-ADAU1761Z
Main Purpose
Audio, CODEC
Embedded
Yes, DSP
Utilized Ic / Part
ADAU1761
Primary Attributes
Stereo, 24-Bit, 8 ~ 96 kHz Sampling Rate, GUI Tool
Secondary Attributes
I²C and GPIO Interfaces, 2 Differential and 1 Stereo Single-Ended Analog Inputs and Outputs
Silicon Manufacturer
Analog Devices
Core Architecture
SigmaDSP
Silicon Core Number
ADAU1761
Silicon Family Name
SigmaDSP
Application Sub Type
Audio
Lead Free Status / RoHS Status
Lead free / RoHS Compliant
Available stocks
Company
Part Number
Manufacturer
Quantity
Price
Company:
Part Number:
EVAL-ADAU1761Z
Manufacturer:
Analog Devices Inc
Quantity:
135
THEORY OF OPERATION
The ADAU1761 is a low power audio codec with an integrated
stream-oriented DSP core, making it an all-in-one package that
offers high quality audio, low power, small size, and many
advanced features. The stereo ADC and stereo DAC each have
an SNR of at least +98 dB and a THD + N of at least −90 dB.
The serial data port is compatible with I
justified, and TDM modes for interfacing to digital audio data.
The operating voltage range is 1.8 V to 3.65 V, with an on-board
regulator generating the internal digital supply voltage.
The record signal path includes very flexible input configurations
that can accept differential and single-ended analog microphone
inputs as well as a digital microphone input. A microphone bias
pin provides seamless interfacing to electret microphones. Input
configurations can accept up to six single-ended analog signals
or variations of stereo differential or stereo single-ended signals
with two additional auxiliary single-ended inputs. Each input
signal has its own programmable gain amplifier (PGA) for volume
adjustment and can be routed directly to the playback path output
mixers, bypassing the ADCs. An automatic level control (ALC)
can also be implemented to keep the recording volume constant.
The ADCs and DACs are high quality, 24-bit Σ-Δ converters
that operate at selectable 64× or 128× oversampling ratios. The
base sampling rate of the converters is set by the input clock rate
and can be further scaled with the converter control register
settings. The converters can operate at sampling frequencies
from 8 kHz to 96 kHz. The ADCs and DACs also include very
fine-step digital volume controls.
The playback path allows input signals and DAC outputs to be
mixed into various output configurations. Headphone drivers
are available for a stereo headphone output, and the other output
pins are capable of differentially driving an earpiece speaker.
Capless headphone outputs are possible with the use of the
mono output as a virtual ground connection. The stereo line
outputs can be used as either single-ended or differential
outputs and as an optional mix-down mono output.
The DSP core introduces many features that make this codec
unique and optimized for audio processing. The program and
parameter RAMs can be loaded with custom audio processing
signal flow built using the SigmaStudio graphical programming
software from Analog Devices, Inc. The values stored in the
parameter RAM control individual signal processing blocks,
such as equalization filters, dynamics processors, audio delays,
and mixer levels.
2
S, left-justified, right-
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The SigmaStudio software is used to program and control the
SigmaDSP through the control port. Along with designing and
tuning a signal flow, the tools can be used to configure all of the
DSP registers. The SigmaStudio graphical interface allows any-
one with digital or analog audio processing knowledge to easily
design DSP signal flow and port it to a target application. At the
same time, it provides enough flexibility and programmability
for an experienced DSP programmer to have in-depth control
of the design. In SigmaStudio, the user can connect graphical
blocks (such as biquad filters, dynamics processors, mixers, and
delays), compile the design, and load the program and parameter
files into the ADAU1761 memory through the control port.
Signal processing blocks available in the provided libraries
include the following:
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Additional processing blocks are always being developed.
Analog Devices also provides proprietary and third-party
algorithms for applications such as matrix decoding, bass
enhancement, and surround virtualizers. Contact Analog
Devices (www.analog.com) for information about licensing
these algorithms.
The ADAU1761 can generate its internal clocks from a wide
range of input clocks by using the on-board fractional PLL.
The PLL accepts inputs from 8 MHz to 27 MHz.
The ADAU1761 is provided in a small, 32-lead, 5 mm × 5 mm
LFCSP with an exposed bottom pad.
Enhanced stereo capture
Single- and double-precision biquad filters
FIR filters
Dynamics processors with peak or rms detection for mono
and multichannel dynamics
Mixers and splitters
Tone and noise generators
Fixed and variable gain
Loudness
Delay
Stereo enhancement
Dynamic bass boost
Noise and tone sources
Level detectors
ADAU1761