D2-926XX_14 Datasheet, PDF (30/38 Pages) Intersil Corporation – Intelligent Digital Amplifier and Sound Processor

D2-926xx
Audio Processing
Audio processing is totally defined by the system firmware and
signal flow. Audio processing blocks are implemented using the
Audio Canvas III design tool software. This tool includes an
extensive library of audio processing functions that are
developed for operating in the DAE-3 and DAE-3HT device
families.
The Audio Canvas III software provides the interface to define
and build a complete audio sound processing system. Drag-and-
drop inserting of its included audio processing blocks enables
building a customized signal flow, by placing audio algorithms
into customized firmware for executing on the DAE devices.
Audio Processing Algorithms
Audio processing algorithms include all of these following
functions. Multiples of each are permitted, and there is no limit
to the order or interconnect of the algorithm blocks. Upon
completion of defining the audio processing path, the Audio
Canvas III software incorporates the blocks and builds the final
system firmware for loading onto and running the DAE devices.
Details and equations for each parameter are provided in the
Audio Canvas III User’s Manual.
INPUT SOURCE SELECTION
A source selection register defines input channel assignment of
audio presented to the DAE device’s audio input ports.
VOLUME
Volume control blocks provide level and trim adjustments within
the signal flow. Continuous adjustment through programmable
gain ranges supports attenuation to -100dB and gain to +24dB. A
single 24-bit register value provides gain setting and also
supports selectable audio phase inversion.
SHARED VOLUME
Shared Volume blocks implement multiple channels of level
attenuation. The number of channels is configurable and a single
24-bit register value equally controls all channel levels. Volume is
continuously adjustable from unity gain to -100dB.
MIXERS
Mixers provide individually-adjustable inputs that are summed
together and passed to their output. Each input mixing level is
controlled with its own 24-bit register that provides continuous
adjustment from unity gain to -100dB, along with full audio path
cut-off and optional input phase inversion supporting sum and
difference mixers. Audio block choices include 2-input, 4-input,
and configurable N-input mixers accommodating as many inputs
as desired.
ROUTER
Routers perform independent channel routing assignment,
connecting any input to any output. Number of channels is
configurable with up to 64 inputs and 12 outputs.
STEREO A/B SWITCH
The A/B Switch provides stereo routing selection to switching
either of 2 pairs of stereo inputs to its output. It operates as a
double-pole, double-throw type of switch to the audio flow.
TONE CONTROL
Tone Controls are shelving filters providing independent of gain
and frequency adjustment for bass and treble tone settings.
Frequency and gain are continuously and independently
adjustable for both the bass and treble settings, supporting gain
ranges of -14 to +14 dB.
PARAMETRIC EQUALIZERS
Parametric Equalizer (EQ) blocks provide an adjustable bandpass
or band-reject frequency response. With frequency-domain
parameter settings of frequency, gain, and bandwidth or Q,
parameters are continuously and independently adjustable. EQs
are provided as individual audio blocks, and as blocks with
groups of 3-Band and 5-Band EQs.
BIQUAD FILTER (FREQUENCY DOMAIN
CONFIGURATION)
The Biquad block is a frequency-domain-parameter-entry biquad
filter implemented as a second-order biquad algorithm, providing
configurable high-pass, low-pass, and all-pass filtering functions.
First or second filter order may be selected, and parameter
setting entries of frequency and damping coefficient are
continuously adjustable. Bypass and polarity phase inversion is
also supported.
BIQUAD FILTER (Z-DOMAIN CONFIGURATION)
The z-Domain Biquad is a second order biquad digital algorithm
that operates from direct entry of z-transform coefficients. The
filter supports individual user entry coefficients enabling nearly
any second order filter synthesis per cascadable block.
FILTER - CROSSOVER
The Crossover Filter blocks provide high-pass or low-pass filtering
using frequency domain adjustment settings. Blocks are
implemented from 2 cascaded second-order biquad elements,
with selections that directly implement Linkwitz-Riley,
Butterworth, or Bessel filter presets. Slope setting is adjustable
from 6, 12, 18 or 24dB per octave, frequency and damping
coefficients are continuously adjustable, and bypass, active or
mute functions are supported.
FIR FILTER
The FIR filter is a configurable n-tap finite impulse response filter
implementation. The number of taps and their coefficient values
are defined in the audio signal flow simply through a user-
generated list of tap coefficients for the FIR structure.
EXCURSION CONTROL
Excursion Control is a specialized algorithm that dynamically
controls audio based on frequency and level. Excursion Control
boosts the low frequency response to compensate for physically-
limited low-frequency capabilities of small loudspeakers and
subwoofers at low listening levels. As listening levels rise, it
dynamically adjusts its boost enabling an optimum sub-woofer
listening experience at all loudness levels.
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FN6787.2
July 12, 2012