Analog Devices ADAU1381 User manual

Evaluation Board User Guide
UG-030
One Technology Way •P. O. Box 9106 •Norwood, MA 02062-9106, U.S.A. •Tel: [phone] •Fax: [phone] •www.analog.com
Using the ADAU1381 Sound Engine
Please see the last page for an important warning and disclaimers. Rev. 0 | Page 1 of 40
INTRODUCTION
This user guide explains the signal flow and parameter settings
for the ADAU1381 sound engine. The ADAU1381 is ideal for
low power portable applications, such as digital camera audio.
During the recording or playing back of audio, the sound
engine provides many signal processing features to improve
audio quality.
DIGITAL CAMERA SYSTEM OVERVIEW
Although the ADAU1381 is flexible enough to be used in
several types of portable audio applications, its design specifically
targets digital camera systems. The sound processing engine
was, therefore, designed especially with such a system in mind.
In general, digital cameras use audio processing when recording
or playing back video. When recording, one or more microphones
mounted in the camera or connected externally capture the
audio data, which is then stored in the memory along with the
video data. During playback or review mode, the audio data is
retrieved from memory and played back through a speaker
mounted in the camera or through a jack for headphones or
other external connections.
In record mode, the source is an audio transducer (microphone)
and the target is memory. In playback mode, the source is
memory and the target is an audio transducer (speaker). In
both modes, the sound engine is positioned between the source
and target, processing the signal to improve audio quality.
Because the required audio processing differs depending on the
operating mode of the camera, several audio processing modes
have been implemented in the sound engine of the ADAU1381.
AUDIO PROCESSING MODES
Record Mode
Record mode takes audio input from a microphone. Wind noise
reduction is applied to remove unwanted noise from the signal
and improve audio clarity. The enhanced stereo capture algorithm
provides improved stereo separation when microphones are
spaced close together. The six-band equalizer can be programmed
to augment bands of interest and filter out unwanted frequencies.
The dual-band dynamics processor acts as an automatic level
control, compensating for fluctuating input signal levels. The
processed signal is output to a digital storage medium.
Two record modes exist: Record A (REC A) and Record B (REC B).
The only differences between the two modes are the six-band
equalizer and the dual-band dynamics processor settings. The
two record modes allow for different audio recording profiles,
such as voice or music. The recording profile can be changed by
a single write to the RAM parameter.
Playback Mode
Playback mode takes audio input from the digital storage. The six-
band equalizer is used for frequency compensation with the output
speaker or headphones. The dual-band dynamics processor acts
as a compressor, allowing for suitable playback levels even in
noisy environments. The playback output includes a digital
volume control for output level adjustment.
SOUND ENGINE SIGNAL FLOW BLOCK DIAGRAM
RECORD
INPUT
PLAYBACK
INPUT
RECORD
OUTPUT
PLAYBACK
OUTPUT
AUDIO MODE
WIND NOISE
REDUCTION ENHANCED
STEREO
CAPTURE SIX-BAND
EQUALIZER DUAL-BAND
DYNAMIC
PROCESSOR
08356-001
Figure 1.

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TABLE OF CONTENTS
Introduction ...................................................................................... 1
Digital Camera System Overview .................................................. 1
Audio Processing Modes ................................................................. 1
Sound Engine Signal Flow Block Diagram ................................... 1
Revision History ............................................................................... 2
SigmaStudio Interface to the Sound Engine ................................. 3
SigmaStudio Interface.................................................................. 3
ADAU1381 Power-Up Sequence................................................ 3
Connecting the ADAU1381 Evaluation Board to the
Computer....................................................................................... 3
Editing the Signal Flow................................................................ 3
Controlling Parameters in Real Time........................................ 3
Output File Generation................................................................ 3
Sound Engine Signal Processing Flow........................................... 4
Description.................................................................................... 4
Inputs ............................................................................................. 4
Outputs and Mute .........................................................................4
Mode Selection ..............................................................................4
Main Page.......................................................................................4
Wind Noise Reduction Page........................................................6
Enhanced Stereo Capture Page....................................................8
Equalization Filters Page ..............................................................9
Dual-Band Compression Page.................................................. 14
SigmaStudio Tools.......................................................................... 27
Changing Sample Rate............................................................... 27
Capture Window ........................................................................ 27
Parameter Visualization Window ............................................ 27
Sequence Window...................................................................... 27
Export Parameter and Register Settings.................................. 28
SigmaStudio Help File ............................................................... 28
Full Parameter Map........................................................................ 29
REVISION HISTORY
11/09—Revision 0: Initial Version

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SIGMASTUDIO INTERFACE TO THE SOUND ENGINE
SIGMASTUDIO INTERFACE
SigmaStudio™ is a software tool that allows the user to configure
the registers and parameters of the ADAU1381 via a graphical
user interface. SigmaStudio can communicate directly with target
hardware via the EVAL-ADUSB2EBZ board, also known as the
USBi, which uses the I2C® and SPI communications protocols.
The ADAU1381 evaluation board is configured for use with the
USBi. Prototype hardware can also be configured for a USBi
connection using a 10-pin communications header.
More information on the USBi can be found in the AN-1006
application note at www.analog.com.
ADAU1381 POWER-UP SEQUENCE
When power is supplied to the ADAU1381, a boot sequence is
initiated to clear the memory to a default state. When the boot
sequence is complete, all of the sound engine parameters are set
to 0. The parameters in the ADAU1381 memory do not match the
values shown in SigmaStudio until they are overwritten.
CONNECTING THE ADAU1381 EVALUATION
BOARD TO THE COMPUTER
To connect the ADAU1381 to the computer, complete the
following steps:
1. Install SigmaStudio; refer to the evaluation board
documentation for step-by-step instructions.
2. Set up the USBi and ADAU1381 evaluation board as
described in the evaluation board documentation.
3. Connect the USBi to the PC with a USB cable and install
the drivers as described in the AN-1006 application note.
4. Connect the communications ribbon cable to the target
ADAU1381 board to initiate the built-in hardware self-
boot function of the ADAU1381.
5. Run SigmaStudio.
6. Open the ADAU1381.dspproj file, which is located in the
SigmaStudio program directory.
7. Write registers and parameters from SigmaStudio to the
hardware to enable the audio signal paths. To download all
parameters for the ADAU1381.dspproj file at once, click
Link-Compile-Download in the main toolbar.
EDITING THE SIGNAL FLOW
The signal flow of the ADAU1381 is fixed function. The
corresponding SigmaStudio project file is locked. Therefore,
no cells can be added to or deleted from the project. Only
the parameters and register settings can be modified.
CONTROLLING PARAMETERS IN REAL TIME
SigmaStudio can be used for real-time tuning of the evaluation
board or a production system via the USBi control interface.
The method for changing the parameters of each cell is described
in the help documentation for that cell.
New parameter values should always be generated within the
SigmaStudio tool. The default minimum and maximum limits
for each control should be obeyed.
OUTPUT FILE GENERATION
SigmaStudio includes built-in code and header file generation
tools that can greatly simplify integration in the host controller
of a target system. Parameter values and register settings can
easily be exported via the Export System Files command in
SigmaStudio to C-compatible output files.

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SOUND ENGINE SIGNAL PROCESSING FLOW
The sound engine processing flow of the ADAU1381 is partitioned
into multiple hierarchy pages in the SigmaStudio tool. In this
section, each page and its corresponding controls and parameters
are described in detail.
DESCRIPTION
The main page presents an overview of the signal flow, with the
processing blocks of the sound engine presented as hierarchy
boards. Using the main page controls, the audio modes and
output volumes can be modified.
INPUTS
There are four audio inputs to the sound engine: Record Input 0,
Record Input 1, Playback Input 0, and Playback Input 1. The
source of the signals on the record inputs is the ADCs or digital
microphones. Record Input 0 comes from the left ADC or Digital
Microphone Input 1 (the LMIC/LMICN/MICD1 pin), and
Record Input 1 comes from the right ADC or Digital Microphone
Input 2 (the RMIC/RMICN/MICD2 pin). The inputs to the
playback path are from the digital serial data interface. Digital
Serial Input 0 (the left channel of the DAC_SDATA/GPIO0 pin)
is connected to Playback Input 0, and Digital Serial Input 1(the
right channel of the DAC_SDATA/GPIO0 pin) is connected to
Playback Input 1.
These two input pairs are routed to the subsequent processing
blocks based on the mode selections. In REC A and REC B
modes, the record input pair is routed through the processing
algorithms; in playback mode, the playback input pair is routed
through the processing algorithms.
OUTPUTS AND MUTE
There are four audio outputs from the sound engine: Record
Output 0, Record Output 1, Playback Output 0, and Playback
Output 1. The record output signals (also labeled as Digital
Output 0 and Digital Output 1) are sent to the digital serial
data interface, and the playback output signals (also labeled as
Analog Output 0 and Analog Output 1) go to the DACs of the
ADAU1381. Playback Output 0 is sent to the left DAC, and
Playback Output 1 is sent to the right DAC.
The digital and analog outputs have separate mute settings. In
SigmaStudio, each of these is enabled by checking the appropriate
box for the mute control.
There is a single flag in the sound engine that outputs a high or
a low logic signal on the GPIO pin of the ADAU1381. This
output is set by writing either a 0 or a 1 to the GPIO parameter.
MODE SELECTION
The sound engine can be put into three modes: REC A (Record A),
REC B (Record B), or Playback. Using the settings on the two
mode selection blocks, the routing logic properly configures the
signal flow for the selected mode. The parameter settings for
each mode are shown in Table 1 .
Table 1. Record/Playback Modes
Mode Mode Selection REC Selection
REC A (Record A) 0 0
REC B (Record B) 0 1
Playback 1 Don’t care
MAIN PAGE
08356-002
Figure 2. Main Page

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Controls
Set the audio mode by typing 0 or 1 into the audioMode cell in
the default 28.0 format (see Figure 3). More information on 28.0
and other numeric formats can be found in the Numeric Formats
section of the SigmaStudio help file.
08356-003
Figure 3. audioMode Control
Record Mode A (REC A) or Record Mode B (REC B) can be
selected by typing 0 or 1 into the REC_Coeff cell in the default
28.0 format (see Figure 4).
08356-004
Figure 4. REC_Coeff Control
The playback (analog) output volume can be adjusted using the
slewvol cell. Click and drag the slider to select a value (see Figure 5).
08356-005
Figure 5. slewvol Control
Click on the slider to type the value in directly (see Figure 6).
08356-006
Figure 6. slewvol Control Direct Value Entry
Click the dmute cell to disable the record (digital) output
(see Figure 7). A check corresponds to a mute setting.
08356-007
Figure 7. dmute Control
Click the amute cell to disable the playback (analog) output
(see Figure 8). A check corresponds to a mute setting.
08356-008
Figure 8. amute Control
To manually toggle the GPIO output, type a value into the
GPIO cell (see Figure 9). This value is in 5.23 format. More
information on 5.23 and other numeric formats can be found in
the Numeric Formats section of the SigmaStudio help file.
08356-009
Figure 9. GPIO Control
Table 2. Main Page Control Settings
Setting Name Description Default Control Type
audioMode Record/playback selection 0 Function selection
REC_Coeff Selects REC A or REC B path 0 Function selection
slewvol Analog volume control with slew 0 dB Processing parameter
dmute Digital output mute using slew Enabled Processing parameter
amute Analog output mute using slew Enabled Processing parameter
GPIO Sets the GPIO pin high/low (active high) 0 Processing parameter
Parameters
The main page parameters are stored in RAM, as outlined in Table 3. These addresses can be directly accessed and modified via the
control port of the ADAU1381.
Table 3. Main Page Parameters
Address Cell Name Parameter Name DefaultValue Function Bytes
Sample Rate
Dependent?
0x0009 audioMode DCInpAlg1 0x00, 0x00, 0x00, 0x00 Set record/playback mode 4 No
0x000A REC_Coeff DCInpAlg3 0x00, 0x00,0x00, 0x00 Set record mode A or B 4 No
0x000B GPIO DCInpAlg4 0x00, 0x00, 0x00, 0x00 Set GPIO output flag 4 No
0x01B8 slewvol GainS200AlgGrow1gain_target 0x00, 0x80, 0x00, 0x00 Analog output volume control 4 No
0x07FA,
0X07FB
slewvol GainS200AlgGrow1alpha 0x00, 0x7F, 0xF2, 0x59,
0x00, 0x00, 0x0D, 0xA7
Slew rate for analog volume
control
8 Yes
0x01B6 dmute MuteSWLinSlewAlg1mute 0x00, 0x00, 0x00, 0x00 Mute digital (record) output 4 No
0x01B7 dmute MuteSWLinSlewAlg1step 0x00, 0x00, 0x40, 0x00 Slew rate for digital mute 4 Yes
0x01BA amute MuteSWLinSlewAlg2mute 0x00, 0x00, 0x00, 0x00 Mute analog (playback) output 4 No
0x01BB amute MuteSWLinSlewAlg2step 0x00, 0x00, 0x40, 0x00 Slew rate for analog mute 4 Yes

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WIND NOISE REDUCTION PAGE
08356-010
Figure 10. Wind Noise Reduction Page
Description
The wind noise reduction page houses the wind noise reduction
algorithm, which uses two microphones to detect and filter wind
noise from the audio signal. Wind noise can easily overwhelm
an audio recording; this reduction algorithm can be used to
lower the effect and increase the clarity of the signal to be recorded.
The algorithm works by detecting the presence of wind noise
and smoothly enabling or disabling a high-pass filter that removes
the noise from the signal. Much of the wind noise that the
microphones pick up is at low frequencies; therefore, the cutoff
frequency of the high-pass filter should be adjusted to adequately
remove the unwanted noise.
L
ROUTPUTINPUTS
FILTERS
WIND NOISE
DETECTION
WIND NOISE
REDUCTION
08356-011
Figure 11. Wind Noise Reduction Block Diagram
Routing and Bypass
The wind noise reduction processing path is automatically enabled
on the multiplexer (MX3) when the sound engine is put into either
Record Mode A or Record Mode B. When in playback mode, this
mulitplexer bypasses the wind noise reduction algorithm. The
switch on this page (WN) can be used to manually bypass the wind
noise reduction, even in the record modes, if desired.
Controls
Three controls are recommended for in-system tuning:
frequency (Freq), Level 1, and Level 2.
The frequency control sets the detector filters. This parameter
should be tuned so that wind noise is removed, but the desired
audio signal is preserved. The frequency parameter should be
tuned while the system is presented with a constant wind noise,
such as from a fan blowing across, not directly onto, the
microphones. The value can be entered by clicking the up/
down arrows or by entering text directly in the box.
08356-012
Figure 12. Freq Control
Level 1 should be tuned while turning the wind source on and
off and simultaneously tuning the parameter setting between 0
and 100. The Level 1 setting is recommended to be between 60
and 90, but this varies depending on the application. The value
can be entered by clicking the up/down arrows or by entering
text directly in the box.
08356-013
Figure 13. Level 1 Control
Level 2 should be tuned in the same way as Level 1; its settings
range from 0 to 15, with 0 being for strong wind noise and 15 being
for a signal with a weak wind noise component. The value can be
entered by clicking the up/down arrows or by entering text directly
in the box.
08356-014
Figure 14. Level 2 Control
The WN switch manually enables or bypasses the algorithm
independently of multiplexer MX3, which allows the algorithm
to be disabled even when a record mode is active. The switch
can be changed by clicking on the appropriate radio button.
08356-015
Figure 15. WN Control

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Table 4. Wind Noise Reduction Page Control Settings
Setting Name Description Default Control Type
Freq High-pass filter setting 1000 Tune
Attack (ms) Wind noise reduction effect attack time 5 Use default
Release (ms) Wind noise reduction effect release time 2500 Use default
Eff Gain Effect gain 5 Use default
tc 1 (ms) Time constant 22 Use default
Level 1 Level of wind noise reduction 70 Tune
Level 2 Wind noise strength (0 = strong, 15 = weak) 4 Tune
WN Switch Bypass Switch to disable algorithm Enable algorithm Function selection
MX3 Mux Bypass Switch to bypass algorithm (via multiplexer) Enable algorithm Function selection
Parameters
The wind noise reduction page parameters are stored in RAM, as outlined in Table 5. These addresses can be directly accessed and
modified via the control port of the ADAU1381.
Table 5. Wind Noise Reduction Page Parameters
Address
Cell
Name Parameter Name Default Value Function Bytes
Sample Rate
Dependent?
0x0011 WNAlg WindNoiseAlg2F11 0x00, 0xE8, 0x5D, 0x19 Frequency and effect gain parameters 4 Yes
0x0012 WNAlg WindNoiseAlg2F12 0xFF, 0x95, 0xA1, 0x9C Frequency and effect gain parameters 4 Yes
0x0013 WNAlg WindNoiseAlg2F20 0x00, 0x00, 0x80, 0x53 Frequency and effect gain parameters 4 Yes
0x0014 WNAlg WindNoiseAlg2F21 0x00, 0x01, 0x00, 0xA6 Frequency and effect gain parameters 4 Yes
0x0015 WNAlg WindNoiseAlg2F30 0x00, 0xE8, 0xD0, 0x3A Frequency and effect gain parameters 4 Yes
0x0016 WNAlg WindNoiseAlg2F31 0xFE, 0x2E, 0x5F, 0x8D Frequency and effect gain parameters 4 Yes
0x0017 WNAlg WindNoiseAlg2F42 0x00, 0x80, 0x00, 0x00 Frequency and effect gain parameters 4 Yes
0x0018 WNAlg WindNoiseAlg2tc1 0x00, 0x00, 0x20, 0x00 Time constant 1 (ms) 4 Yes
0x0019 WNAlg WindNoiseAlg2tc11 0x00, 0x7F, 0xE0, 0x00 Time constant 1 (ms) 4 Yes
0x001A WNAlg WindNoiseAlg2tc2 0x00, 0x00, 0x20, 0x00 Time constant 2 (ms) 4 Yes
0x001B WNAlg WindNoiseAlg2tc22 0x00, 0x7F, 0xE0, 0x00 Time constant 2 (ms) 4 Yes
0x001C WNAlg WindNoiseAlg2Level1 0x00, 0x59, 0x99, 0x9A Level 1 4 No
0x001D WNAlg WindNoiseAlg2Level2 0x00, 0x08, 0x00, 0x00 Level 2 4 No
0x001E WNAlg WindNoiseAlg2attack 0x00, 0x00, 0x80, 0x00 Attack (ms) 4 Yes
0x001F WNAlg WindNoiseAlg2release 0x00, 0x00, 0x00, 0x40 Release (ms) 4 Yes
0x0020 WN stereomux1940ns40 0x00, 0x00, 0x00, 0x00 On/off (burst write Address 0x0020
and Address 0x0021 together)
4 No
0x0021 WN stereomux1940ns41 0x00, 0x80, 0x00, 0x00 On/off (burst write Addresses 0x0020
and Address 0x0021 together)
4 No

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ENHANCED STEREO CAPTURE PAGE
08356-016
Figure 16. Enhanced Stereo Capture Page
Description
The enhanced stereo capture (ESC) algorithm takes a stereo record
signal and creates a wider stereo image. ESC is used as a recording
algorithm to capture an enhanced stereo image from two closely
spaced microphones.
The ESC algorithm takes two input signals from two closely
spaced microphones. The algorithm separates these two signals
and widens the stereo image. The result is a perceived widened
stereo image as if the audio was captured by microphones with
greater left/right separation. ESC is based on proprietary filtering
and a stereo balance gain that adjusts how much stereo effect is
achieved in the algorithm.
Routing and Bypass
The enhanced stereo capture path is automatically enabled on
the mux (rec_play) when the sound engine is put into either
REC A or REC B. When in playback mode, the mux bypasses
the wind noise reduction algorithm. The switch on this page
(SS) can be used to bypass the enhanced stereo capture, even in
the record modes, if desired.
Controls
The MicDistance control can be set from −10 to +10, with a
default value of 0 (see Figure 17). This control determines the
sensitivity of the ESC algorithm and directly affects the level of
stereo enhancement perceived in the recorded signal. Increasing
the enhancement too much may result in an unnatural quality
in the recorded audio. This control may vary greatly depending
on factors such as microphone selection, spacing, and housing.
Therefore, it must be tuned to fit the needs of a specific design.
08356-017
Figure 17. MicDistance Control
Right-click the slider to enter the value directly (see Figure 18).
0
8356-018
Figure 18. MicDistance Control Direct Value Entry
The SS switch allows the algorithm to be bypassed independently
of the rec_play multiplexer and the active audio mode. The
switch can be changed by clicking on the appropriate radio button.
08356-019
Figure 19. SS Control
Table 6. ESC Page Control Settings
Setting Name Description Default Control Type
MicDistance Control enhancement level 0 Tune
SS Switch Bypass Switch to disable algorithm Algorithm enabled Function selection
rec_play Mux Bypass Switch to bypass algorithm (via multiplexer) Algorithm enabled Function selection

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Parameters
The enhanced stereo capture page parameters are stored in RAM, as outlined in Table 7 . These addresses can be directly accessed and
modified via the control port of the ADAU1381.
Table 7. ESC Page Parameters
Address Cell Name Parameter Name Default Value Function Bytes
Sample Rate
Dependent?
0x0029 MicDistance Gain1940AlgNS1 0x00, 0x80, 0x00, 0x00 Gain setting related to the distance
between microphones that enhances
the perceived effect
4 No
0x002B SS stereomux1940ns30 0x00, 0x00, 0x00, 0x00 On/off (burst write Address0x002B
and Address 0x002C together)
4 No
0x002C SS stereomux1940ns31 0x00, 0x80, 0x00, 0x00 On/off (burst write Address 0x002B
and Address 0x002C together)
4 No
0x0023 Locked Cell param1 0x00, 0xCA, 0x9A, 0x58 Locked parameter (generated by
SigmaStudio)
4 Yes
0x0024 Locked Cell param2 0x0F, 0x35, 0x65, 0xA8 Locked parameter (generated by
SigmaStudio)
4 Yes
0x0025 Locked Cell param3 0x00, 0x7F, 0xAA, 0xE7 Locked parameter (generated by
SigmaStudio)
4 Yes
0x0026 Locked Cell param4 0x00, 0x08, 0x38, 0x65 Locked parameter (generated by
SigmaStudio)
4 Yes
0x0027 Locked Cell param5 0x00, 0x00, 0x00, 0x00 Locked parameter (generated by
SigmaStudio)
4 Yes
0x0028 Locked Cell param6 0x00, 0x7B, 0x1A, 0x7E Locked parameter (generated by
SigmaStudio)
4 Yes
EQUALIZATION FILTERS PAGE
08356-020
.
on this page (filtS) can be used to completely bypass
Click Show on the EQFilter cell to configure the filter bands
(see Figure 21).
Figure 20. Equalization Filters Page
Description
Equalization (EQ) filters are used to tune the frequency response of
the recorded or played back audio signal. The ADAU1381 sound
engine includes three, six-band EQ paths, one for playback and
the other two for different recording scenarios, such as music
recording and voice.
Each EQ band is implemented as a double-precision biquad
filter. These filters can be used in a wide variety of configurations,
such as low-pass, high-pass, band-pass, parametric, shelving,
peaking, tone control, and others.
Routing and Bypass
There are three, six-band EQ paths in the sound engine: one
each for Record A (REC A), Record B (REC B), and Playback
modes. Path 0 (top row) is the EQ filters for Record A (REC A),
Path 1 (middle row) is the EQ filters for Record B (REC B), and
Path 2 (bottom row) is the filters for the playback processing.
The appropriate path is automatically selected when the mode is
selected on the main page
The switch
the EQFilter, if desired.
Controls
08356-021
Figure 21. EQFilter Control with Show Button

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When Show is clicked, it displays a filter matrix with three rows
and six columns (see Figure 22).
08356-
Figure 22. EQFilter Matrix
The first row represents the six bands of the Record A (REC A)
mode, the second
022
row represents the six bands of the Record B
d
window provides access to a large variety of filters, each with its
own property pages and controls (see Figure 23 and Figure 24).
(REC B) mode, and the third row represents the six bands of the
Playback mode.
Each button in the matrix contains a single second-order biqua
filter. To individually tune a filter, click its corresponding button.
Clicking the menu at the top of the General Filter Settings
08356-023
Figure 23. Individual Filter Band Settings
08356-
Figure 24. F
024
ilter Type Selection
Click Stimulus and Probe to open the Simulated Frequency
Response window (see Figure 25 and gure 26).
More information on the various filters is available in the Help
menu within SigmaStudio.
Fi
0
8356-025
Figure 25. Stimulus Button
0
8356-026
The Simulated Frequency Response window displays a
calculated frequency response for each of the filter bands. It
shows only one EQ curve at a time, the one corresponding to
the filter mode that was last edited.
By default, the EQ curve for Record A (REC A) mode is configured
for voice recording (see Figure 27). The high-pass filter removes
low frequencies that are not necessary for voice recording. The
wide boost in the 150 Hz range amplifies the voice fundamental
frequencies, and the narrow boost near 4 kHz increases vocal clarity.
08356-027
Figure 27. Default Record A (REC A) Mode EQ Curve
By default, the EQ curve for Record B (REC B) mode is configured
for music and live concert recording (see Figure 28). The high-
pass filter removes low frequency boom and rumble from a
concert recording environment. The cut in the midbass range
around 300 Hz helps to increase the perceived level of the bass.
The low-pass filter on the high frequency range helps to reduce
ringing caused by reflections in a loud concert environment.
0
8356-028
Figure 28. Default Record B (REC B) Mode EQ Curve
Figure 26. Probe Button

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The default EQ curves are intended only as examples and
should be specifically tuned for the target application system.
By default, the EQ curve for Playback mode is flat, which should
be changed accordingly to compensate for nonlinearities due to
the speaker design and housing (see Figure 29).
0
8356-029
Figure 29. Default Playback Mode EQ Curve
Table 8. EQ Page Control Settings
Setting Name Description Default Control Type
EQFilter Three parallel six-band equalizers with independently controllable bands Example curves for
record and playback
Tune
filtS Switch to disable algorithm Algorithm enabled Function selection
Parameters
The equalization filters page parameters are stored in RAM, as outlined in Table 9. These addresses can be directly accessed and modified
via the control port of the ADAU1381.
Table 9. EQ Page Parameters
Address Cell Name Parameter Name Default Value Function Bytes
Sample Rate
Dependent?
0x002D EQFilter IndexSelMultBandAlg100b2 0x00, 0x7F, 0xAA, 0x50, Biquad F0, 0 20 Yes
0x002E 0xFF, 0x00, 0xAB, 0x60,
0x002F 0x00, 0x7F, 0xAA, 0x50,
0x0030 0xFF, 0x80, 0xAB, 0x20,
0x0031 0x00, 0xFF, 0x54, 0x5F
0x0032 EQFilter IndexSelMultBandAlg101b2 0x00, 0x7D, 0xBD, 0xAF, Biquad F0, 1 20 Yes
0x0033 0xFF, 0x02, 0x0A, 0x2E,
0x0034 0x00, 0x80, 0x42, 0x4A,
0x0035 0xFF, 0x82, 0x00, 0x07,
0x0036 0x00, 0xFD, 0xF5, 0xD2
0x0037 EQFilter IndexSelMultBandAlg102b2 0x00, 0x00, 0x00, 0x00, Biquad F0, 2 20 Yes
0x0038 0x00, 0x00, 0x00, 0x00,
0x0039 0x00, 0x80, 0x00, 0x00,
0x003A 0x00, 0x00, 0x00, 0x00,
0x003B 0x00, 0x00, 0x00, 0x00
0x003C EQFilter IndexSelMultBandAlg103b2 0x00, 0x00, 0x00, 0x00, Biquad F0, 3 20 Yes
0x003D 0x00, 0x00, 0x00, 0x00,
0x003E 0x00, 0x80, 0x00, 0x00,
0x003F 0x00, 0x00, 0x00, 0x00,
0x0040 0x00, 0x00, 0x00, 0x00

UG-030 Evaluation Board User Guide
Rev. 0 | Page 12 of 40
Address Cell Name Parameter Name Default Value Function Bytes
Sample Rate
Dependent?
0x0041 EQFilter IndexSelMultBandAlg104b2 0x00, 0x71, 0xCB, 0x91, Biquad F0, 4 20 Yes
0x0042 0xFF, 0x2E, 0xCC, 0xE6,
0x0043 0x00, 0x81, 0xA0, 0xD2,
0x0044 0xFF, 0x8C, 0x93, 0x9D,
0x0045 0x00, 0xD1, 0x33, 0x1A
0x0046 EQFilter IndexSelMultBandAlg105b2 0x00, 0x00, 0x00, 0x00, Biquad F0, 5 20 Yes
0x0047 0x00, 0x00, 0x00, 0x00,
0x0048 0x00, 0x80, 0x00, 0x00,
0x0049 0x00, 0x00, 0x00, 0x00,
0x004A 0x00, 0x00, 0x00, 0x00
0x004B EQFilter IndexSelMultBandAlg110b2 0x00, 0x7E, 0xFB, 0x24, Biquad F1, 0 20 Yes
0x004C 0xFF, 0x02, 0x09, 0xB7,
0x004D 0x00, 0x7E, 0xFB, 0x24,
0x004E 0xFF, 0x82, 0x06, 0xEE,
0x004F 0x00, 0xFD, 0xF3, 0x80
0x0050 EQFilter IndexSelMultBandAlg111b2 0x00, 0x7D, 0xEB, 0x86, Biquad F1, 1 20 Yes
0x0051 0xFF, 0x02, 0x83, 0x95,
0x0052 0x00, 0x7F, 0xC2, 0xF7,
0x0053 0xFF, 0x82, 0x51, 0x83,
0x0054 0x00, 0xFD, 0x7C, 0x6B
0x0055 EQFilter IndexSelMultBandAlg112b2 0x00, 0x00, 0x00, 0x00, Biquad F1, 2 20 Yes
0x0056 0x00, 0x00, 0x00, 0x00,
0x0057 0x00, 0x80, 0x00, 0x00,
0x0058 0x00, 0x00, 0x00, 0x00,
0x0059 0x00, 0x00, 0x00, 0x00
0x005A EQFilter IndexSelMultBandAlg113b2 0x00, 0x00, 0x00, 0x00, Biquad F1, 3 20 Yes
0x005B 0x00, 0x00, 0x00, 0x00,
0x005C 0x00, 0x80, 0x00, 0x00,
0x005D 0x00, 0x00, 0x00, 0x00,
0x005E 0x00, 0x00, 0x00, 0x00
0x005F EQFilter IndexSelMultBandAlg114b2 0x00, 0x00, 0x00, 0x00, Biquad F1, 4 20 Yes
0x0060 0x00, 0x00, 0x00, 0x00,
0x0061 0x00, 0x80, 0x00, 0x00,
0x0062 0x00, 0x00, 0x00, 0x00,
0x0063 0x00, 0x00, 0x00, 0x00
0x0064 EQFilter IndexSelMultBandAlg115b2 0x00, 0x4A, 0x91, 0x00, Biquad F1, 5 20 Yes
0x0065 0x00, 0x95, 0x22, 0x00,
0x0066 0x00, 0x4A, 0x91, 0x00,
0x0067 0xFF, 0xD1, 0x47, 0xB1,
0x0068 0xFF, 0x84, 0x74, 0x4F
0x0069 EQFilter IndexSelMultBandAlg120b2 0x00, 0x00, 0x00, 0x00, Biquad F2, 0 20 Yes
0x006A 0x00, 0x00, 0x00, 0x00,
0x006B 0x00, 0x80, 0x00, 0x00,
0x006C 0x00, 0x00, 0x00, 0x00,
0x006D 0x00, 0x00, 0x00, 0x00
0x006E EQFilter IndexSelMultBandAlg121b2 0x00, 0x00, 0x00, 0x00, Biquad F2, 1 20 Yes
0x006F 0x00, 0x00, 0x00, 0x00,
0x0070 0x00, 0x80, 0x00, 0x00,
0x0071 0x00, 0x00, 0x00, 0x00,
0x0072 0x00, 0x00, 0x00, 0x00

Evaluation Board User Guide UG-030
Rev. 0 | Page 13 of 40
Address Cell Name Parameter Name Default Value Function Bytes
Sample Rate
Dependent?
0x0073 EQFilter IndexSelMultBandAlg122b2 0x00, 0x00, 0x00, 0x00, Biquad F2, 2 20 Yes
0x0074 0x00, 0x00, 0x00, 0x00,
0x0075 0x00, 0x80, 0x00, 0x00,
0x0076 0x00, 0x00, 0x00, 0x00,
0x0077 0x00, 0x00, 0x00, 0x00
0x0078 EQFilter IndexSelMultBandAlg123b2 0x00, 0x00, 0x00, 0x00, Biquad F2, 3 20 Yes
0x0079 0x00, 0x00, 0x00, 0x00,
0x007A 0x00, 0x80, 0x00, 0x00,
0x007B 0x00, 0x00, 0x00, 0x00,
0x007C 0x00, 0x00, 0x00, 0x00
0x007D EQFilter IndexSelMultBandAlg124b2 0x00, 0x00, 0x00, 0x00, Biquad F2, 4 20 Yes
0x007E 0x00, 0x00, 0x00, 0x00,
0x007F 0x00, 0x80, 0x00, 0x00,
0x0080 0x00, 0x00, 0x00, 0x00,
0x0081 0x00, 0x00, 0x00, 0x00
0x0082 EQFilter IndexSelMultBandAlg125b2 0x00, 0x00, 0x00, 0x00, Biquad F2, 5 20 Yes
0x0083 0x00, 0x00, 0x00, 0x00,
0x0084 0x00, 0x80, 0x00, 0x00,
0x0085 0x00, 0x00, 0x00, 0x00,
0x0086 0x00, 0x00, 0x00, 0x00
0x008E filtS stereomux1940ns10 0x00, 0x00, 0x00, 0x00 On/off (burst write
Address 0x008E and
Address 0x008F together)
4 No
0x008F filtS stereomux1940ns11 0x00, 0x80, 0x00, 0x00 On/off (burst write
Address 0x008E and
Address 0x008F together)
4 No

UG-030 Evaluation Board User Guide
Rev. 0 | Page 14 of 40
DUAL-BAND COMPRESSION PAGE
08356-030
Figure 30. Dual-Band Compression Page
08356-031
Description
The dual-band compression page contains dynamic processors
designed to alter the dynamic range of the audio signal during
record or playback. To provide high audio quality, the input
signal is sent into a crossover network that divides it into high
and low bands. Each band is detected and processed individually.
The end result is that a sudden peak in one band (for example, a
kick drum in the low band) will not cause a dip in the overall
signal level.
Available Curves
By default, the record modes are configured with an automatic
level control (ALC) curve, and the playback mode is configured
with a standard compressor curve with a threshold of −6 dB and a
ratio of 2:1. The Record A (REC A) mode curve is an example
of hard ALC compression (see Figure 31), and the Record B (REC B)
mode curve is an example of smoothed ALC compression
(see Figure 32). The playback mode compressor curve has
moderate compression starting at a threshold of −8 dB
(see Figure 33). These default curves are intended only to be
examples. The desired curve varies greatly depending on the
application and other factors in the system. Therefore, unique
curves should be created during the
Figure 31. Default Record A (REC A) Mode Compressor Curve
08356-032
tuning process.
Figure 31, Figure 32, and Figure 33 show examples of compressor
curves. The curves represent a transfer function, with the
horizontal axis representing input in dB and the vertical axis
representing the resulting output in dB.
Figure 32. Default Record B (REC B) Mode Compressor Curve

Evaluation Board User Guide UG-030
Rev. 0 | Page 15 of 40
0
8356-033
Figure 33. Default Playback Mode Compressor Curve
Curve Shift
The compressor curves can be shifted to the right using the
ShiftLR control. This changes the input gain of the audio and
detects signals routed to the compressor. The default value of
0 dB represents no shift from the original curve. Decreasing this
value translates the compressor curves directly to the right by a
corresponding amount. Note that there is no graphical difference
shown on the compressor curve graphs, but the compressor curve
points effectively shift directly to the right as the value of the
slider decreases. The maximum shift allowed is 12 dB.
For a classic compression curve (linear compression ratio for
low amplitudes and a ratio greater than 1 after a certain
threshold), the ShiftLR control effectively increases the
threshold value. The ShiftLR control allows a curve to be
shifted at run time without requiring a download of new
compression table parameters via the control port.
Figure 34 shows an example of curve shift being applied to an
example compression curve with a gate below −80 dB, a linear
transfer function between −80 dB and −28 dB, and a compression
ratio of 2:1 for input amplitudes greater than −28 dB. The example
curve is shown furthest to the left. Shifted curves for −3 dB, −6 dB,
−9 dB, and −12 dB are shown to the right of the example curve.
For all shifted curves, the compression threshold remains constant,
but the gate threshold changes. The output gain for the linear
section of the input range decreases as the curve shifts to the right.
20
0
–20
–40
–60
–80
–100
–113
–90 0–20–40–60–80
OUTPUT LEVEL (dB)
INPUT LEVEL (dB)
08356-034
Figure 34. Curve Shift L/R Example
Detection Range Shift
The detection range control shifts the range over which the
compressor operates. The algorithm typically handles inputs
ranging from −90 dB to +6 dB. Any inputs outside of this range
have a linear input-to-output relationship, effectively ignoring
the compression curve. In applications where inputs to the
compressor greater than 6 dB are expected, the detection range
can be shifted to accommodate the input signal range. The
default shift of −12 dB changes the detection range’s lower
bound to −78 dB and its upper bound to +18 dB. This curve
shift must correspond to the compressors Adjust Gain Curve
setting, shown in Figure 39.
The compression curve displayed in the compression curve graph
represents a detection range shift of 0 dB. Decreasing the detection
range effectively shifts the curve upward and to the right.
The detection range shift should be determined during system
tuning and should not be altered when the system is in operation.
Figure 35 shows an example of detection range shift being
applied to an example compression curve with a gate below
−80 dB, a linear transfer function between −80 dB and −28 dB,
and a compression ratio of 2:1 for input amplitudes greater than
−28 dB. The example curve is shown furthest to the bottom and
the left. Shifted curves for −3 dB, −6 dB, −9 dB, and −12 dB are
shown above and to the right of the example curve. For all shifted
curves, both the compression and gate thresholds increase, but
the linear section of the input range remains linear.
20
0
–20
–40
–60
–80
–100
–113
–90 0–20–40–60–80
OUTPUT LEVEL (dB)
INPUT LEVEL (dB)
08356-035
Figure 35. Detection Range Shift Example
Routing and Bypass
The dual-band compression algorithm is enabled by default in
all audio modes. The compS switch allows the dual-band
compressor to be bypassed manually.

UG-030 Evaluation Board User Guide
Rev. 0 | Page 16 of 40
Controls
The ShiftLR control shifts the compression curve horizontally.
The slider can be dragged to change the value (see Figure 36).
The default value of 0 dB indicates that the transfer function
displayed in the compression curve editor matches the processing
in the sound engine. Decreasing the value of the control shifts
the curve to the right.
08356-036
Figure 36. ShiftLR Control
Right-click the slider to type the value in directly (see Figure 37).
08356-037
Figure 37. ShiftLR Control Direct Value Entry
The Det_Range control shifts the compression curve diagonally
(see Figure 38). The Det_Range value can be controlled by
dragging the slider or by entering the value manually by right-
clicking the slider. The Det_Range control is only intended to
take on the following values: 0 dB, −3 dB, −6 dB, −9 dB, and −12 dB.
08356-038
Figure 38. Det_Range Control
When the detection range is modified, the compressors must
also be configured to match. By default, the compressors are
configured for a detection range shift of −12 dB. To change the
compressor detection range, click on a compressor cell, select
the Adjust Gain Curve option, and select the value matching
the setting of the Det_Range control (see Figure 39). Complete
this process for both the high-band and low-band compressors.
08356-039
Figure 39. Changing the Detection Curve of a Compressor
Each frequency band (low and high) is fed into a stereo
compressor matrix (see Figure 40). Each matrix contains three
compressors, one for each audio mode. The left column
corresponds to Record A (REC A) mode, the center column
corresponds to Record B (REC B) mode, and the right column
corresponds to Playback mode.
08356-040
Figure 40. HighPass_Comp Control
Each compressor column contains a Post Gain control that adjusts
the amount of gain applied to the signal at the output of the
compressor (see Figure 41).
08356-041
Figure 41. Post Gain Control
The Hold (ms) control sets the duration that the gain reduction
ratio of the compressor is held after it is set by the input signal
(see Figure 42).
08356-042
Figure 42. Hold (ms) Control

Evaluation Board User Guide UG-030
Rev. 0 | Page 17 of 40
3).
The Decay (dB/s) control sets the speed by which the gain
reduction ratio decays after the hold duration expires
(see Figure 4
Click on a point within the graph to display the Compression
Curve Point Option menu. This is where large points can be
added, removed, or fine-tuned (see Figure 47).
08356-047
08356-04
3
Figure 43. Decay (dB/s) Control
Click the Soft Knee button to smooth the corners (also known
as knees) of the Compression Curve (see Figure 44).
08356-04
4
Figure 44. Soft Knee Button
Click the Show Graph button to display the Compression
Curve graphical editor (see Figure 45).
Figure 47. Compression Curve Point Option Menu
Click on a set point value to display a dialog box where the
coordinates of the point can be entered manually (see Figure 48).
08356-045
08356-048
Figure 45. Show Graph Button
The Compression Curve editor displays a graphical representation
of the input/output gain transfer function, which is a curve with
33 points (see Figure 46). The horizontal axis represents the input
level, and the vertical axis represents the output level. Each large
point can be dragged to a new position on the graph.
Figure 48. Compression Curve Point Direct Value Entry
The compS switch allows the dual-band compressors to be
bypassed (see Figure 49). Click on the appropriate radio button
to change the switch.
08356-046
0
8356-049
Figure 49. compS Control
Figure 46. Compression Curve Editor
Table 10. Dual-Band Compression Page Control Settings
Setting Name Description Default Control Type
ShiftLR Shift curve left/right 0 Tune
Det_Range Fixed gain for extended detection range −12 Tune
CrossHi Crossover for high frequencies N/A Locked
CrossLo Crossover for low frequencies N/A Locked
HiDet Filter Filter for high frequency detector N/A Locked
LoDet Filter Filter for low frequency detector N/A Locked
HIGH-PASS COMPRESSOR
Post Gain (dB) Gain applied to the output of the compressor 0 Tune
Hold (ms) Duration that the gain reduction ratio of the compressor
is held after it is set by the input signal
2 Tune
Decay (dB/s) Speed that the gain reduction ratio of the compressor
decreases after the hold time expires
2 Tune
Soft Knee Smooths the compression curve Active Tune
Graph Editor Graphical entry of compression curve (input/output
gain transfer function)
Default compression
curves
Tune

UG-030 Evaluation Board User Guide
Rev. 0 | Page 18 of 40
Setting Name Description Default Control Type
LOW-PASS COMPRESSOR
Post Gain (dB) Gain applied to the output of the compressor 0 Tune
Hold (ms) Duration that the gain reduction ratio of the compressor
is held after it is set by the input signal
2 Tune
Decay (dB/s) Speed that the gain reduction ratio of the compressor
decreases after the hold time expires
2 Tune
Soft Knee Smooths the compression curve Active Tune
Graph Editor Graphical entry of compression curve (input/output
gain transfer function)
Default compression
curves
Tune
compS Switch/mux bypass Algorithm enabled Function selection
Parameters
The dual-band compression page parameters are stored in RAM, as outlined in Table 11. These addresses can be directly accessed and
modified via the control port of the ADAU1381.
Table 11. Dual-Band Compression Page Parameters
Address Cell Name Parameter Name DefaultValue Function Bytes
Sample Rate
Dependent?
0x0090 ShiftLR Gain1940AlgNS4 0x00, 0x80, 0x00, 0x00 Shift curve left/right;
Address 0x0090 and
Address 0x0091
must contain
the same value
4 No
0x0091 ShiftLR Gain1940AlgNS5 0x00, 0x80, 0x00, 0x00 Shift curve left/right;
Address 0x0090 and
Address 0x0091
must contain
the same value
4 No
0x0092 Det_Range Gain1940AlgNS3 0x00, 0x20, 0x26, 0xF3 On/off (burst write
Address 0x0092 and
Address 0x0093
together)
4 No
0x0093 Det_Range Gain1940AlgNS2 0x00, 0x20, 0x26, 0xF3 On/off (burst write
Address 0x0092 and
Address 0x0093
together)
4 No
0x00A2 CrossHi EQwSubDualDP32B1 0x00, 0x7B, 0xEB, 0x74 Crossover HPF filter
coefficient
4 Yes
0x00A3 CrossHi EQwSubDualDP31B1 0x0F, 0x08, 0x29, 0x18 Crossover HPF filter
coefficient
4 Yes
0x00A4 CrossHi EQwSubDualDP30B1 0x00, 0x7B, 0xEB, 0x74 Crossover HPF filter
coefficient
4 Yes
0x00A5 CrossHi EQwSubDualDP32A1 0x0F, 0x88, 0x07, 0xC9 Crossover HPF filter
coefficient
4 Yes
0x00A6 CrossHi EQwSubDualDP31A1 0x00, 0xF7, 0xB5, 0x9A Crossover HPF filter
coefficient
4 Yes
0x00A7 CrossHi EQwSubDualDP32B2 0x00, 0x7B, 0xEB, 0x74 Crossover HPF filter
coefficient
4 Yes
0x00A8 CrossHi EQwSubDualDP31B2 0x0F, 0x08, 0x29, 0x18 Crossover HPF filter
coefficient
4 Yes
0x00A9 CrossHi EQwSubDualDP30B2 0x00, 0x7B, 0xEB, 0x74 Crossover HPF filter
coefficient
4 Yes
0x00AA CrossHi EQwSubDualDP32A2 0x0F, 0x88, 0x07, 0xC9 Crossover HPF filter
coefficient
4 Yes
0x00AB CrossHi EQwSubDualDP31A2 0x00, 0xF7, 0xB5, 0x9A Crossover HPF filter
coefficient
4 Yes

Evaluation Board User Guide UG-030
Rev. 0 | Page 19 of 40
Address Cell Name Parameter Name DefaultValue Function Bytes
Sample Rate
Dependent?
0x0094 CrossLo EQwSubDualDP42B1 0x00, 0x00, 0x10, 0xA7 Crossover LPF filter
coefficient
4 Yes
0x0095 CrossLo EQwSubDualDP41B1 0x00, 0x00, 0x21, 0x4E Crossover LPF filter
coefficient
4 Yes
0x0096 CrossLo EQwSubDualDP40B1 0x00, 0x00, 0x10, 0xA7 Crossover LPF filter
coefficient
4 Yes
0x0097 CrossLo EQwSubDualDP42A1 0x0F, 0x88, 0x07, 0xC9 Crossover LPF filter
coefficient
4 Yes
0x0098 CrossLo EQwSubDualDP41A1 0x00, 0xF7, 0xB5, 0x9A Crossover LPF filter
coefficient
4 Yes
0x0099 CrossLo EQwSubDualDP42B2 0x00, 0x00, 0x10, 0xA7 Crossover LPF filter
coefficient
4 Yes
0x009A CrossLo EQwSubDualDP41B2 0x00, 0x00, 0x21, 0x4E Crossover LPF filter
coefficient
4 Yes
0x009B CrossLo EQwSubDualDP40B2 0x00, 0x00, 0x10, 0xA7 Crossover LPF filter
coefficient
4 Yes
0x009C CrossLo EQwSubDualDP42A2 0x0F, 0x88, 0x07, 0xC9 Crossover LPF filter
coefficient
4 Yes
0x009D CrossLo EQwSubDualDP41A2 0x00, 0xF7, 0xB5, 0x9A Crossover LPF filter
coefficient
4 Yes
0x00BE HiDet_Filter EQwSubDualDP52B1 0x00, 0x7D, 0x44, 0xE0 Crossover HPF
detection path
filter coefficient
4 Yes
0x00BF HiDet_Filter EQwSubDualDP51B1 0x0F, 0x05, 0x76, 0x40 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C0 HiDet_Filter EQwSubDualDP50B1 0x00, 0x7D, 0x44, 0xE0 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C1 HiDet_Filter EQwSubDualDP52A1 0x0F, 0x85, 0x67, 0x55 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C2 HiDet_Filter EQwSubDualDP51A1 0x00, 0xFA, 0x7A, 0xD5 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C3 HiDet_Filter EQwSubDualDP52B2 0x00, 0x7D, 0x44, 0xE0 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C4 HiDet_Filter EQwSubDualDP51B2 0x0F, 0x05, 0x76, 0x40 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C5 HiDet_Filter EQwSubDualDP50B2 0x00, 0x7D, 0x44, 0xE0 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C6 HiDet_Filter EQwSubDualDP52A2 0x0F, 0x85, 0x67, 0x55 Crossover HPF
detection path
filter coefficient
4 Yes
0x00C7 HiDet_Filter EQwSubDualDP51A2 0x00, 0xFA, 0x7A, 0xD5 Crossover HPF
detection path
filter coefficient
4 Yes
0x00B0 LoDet_Filter EQwSubDualDP62B1 0x00, 0x00, 0x24, 0xE2 Crossover LPF
detection path
filter coefficient
4 Yes
0x00B1 LoDet_Filter EQwSubDualDP61B1 0x00, 0x00, 0x49, 0xC4 Crossover LPF
detection path
filter coefficient
4 Yes

UG-030 Evaluation Board User Guide
Rev. 0 | Page 20 of 40
Address Cell Name Parameter Name DefaultValue Function Bytes
Sample Rate
Dependent?
0x00B2 LoDet_Filter EQwSubDualDP60B1 0x00, 0x00, 0x24, 0xE2 Crossover LPF
detection path
filter coefficient
4 Yes
0x00B3 LoDet_Filter EQwSubDualDP62A1 0x0F, 0x8B, 0xDA, 0xCC Crossover LPF
detection path
filter coefficient
4 Yes
0x00B4 LoDet_Filter EQwSubDualDP61A1 0x00, 0xF3, 0x91, 0xAC Crossover LPF
detection path
filter coefficient
4 Yes
0x00B5 LoDet_Filter EQwSubDualDP62B2 0x00, 0x00, 0x24, 0xE2 Crossover LPF
detection path
filter coefficient
4 Yes
0x00B6 LoDet_Filter EQwSubDualDP61B2 0x00, 0x00, 0x49, 0xC4 Crossover LPF
detection path
filter coefficient
4 Yes
0x00B7 LoDet_Filter EQwSubDualDP60B2 0x00, 0x00, 0x24, 0xE2 Crossover LPF
detection path
filter coefficient
4 Yes
0x00B8 LoDet_Filter EQwSubDualDP62A2 0x0F, 0x8B, 0xDA, 0xCC Crossover LPF
detection path
filter coefficient
4 Yes
0x00B9 LoDet_Filter EQwSubDualDP61A2 0x00, 0xF3, 0x91, 0xAC Crossover LPF
detection path
filter coefficient
4 Yes
0x00CC HighPass_Comp PeakDBCompLUTAlgPG30decay 0x00, 0x00, 0x00, 0x04 REC_Auto: decay 4 Yes
0x00CD HighPass_Comp PeakDBCompLUTAlgPG30hold 0x00, 0x00, 0x00, 0x60 REC_Auto: hold 4 Yes