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Using the ADAU1381 Sound Engine
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.
SOUND ENGINE SIGNAL FLOW BLOCK DIAGRAM
AUDIO MODE
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 sixband 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.
RECORD
INPUT
PLAYBACK
INPUT
WIND NOISE
REDUCTION
ENHANCED
STEREO
CAPTURE
SIX-BAND
EQUALIZER
Figure 1.
Please see the last page for an important warning and disclaimers. Rev. 0 | Page 1 of 40
Export Parameter and Register Settings .................................. 28
SigmaStudio Help File ............................................................... 28
Full Parameter Map ........................................................................ 29
REVISION HISTORY
11/09—Revision 0: Initial Version
Rev. 0 | Page 2 of 40
Evaluation Board User Guide UG-030
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 I
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.
2
C® and SPI communications protocols.
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 selfboot 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.
Rev. 0 | Page 3 of 40
UG-030 Evaluation Board User Guide
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
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.
1 (the
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 Tabl e 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
Figure 2. Main Page
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Evaluation Board User Guide UG-030
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).
Figure 4. REC_Coeff Control
08356-004
The playback (analog) output volume can be adjusted using the
slewvol cell. Click and drag the slider to select a value (see Figure 5).
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.
Figure 8. amute Control
08356-008
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
08356-005
Figure 5. slewvol Control
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
Sample Rate
Address Cell Name Parameter Name Default Value Function Bytes
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
Rev. 0 | Page 5 of 40
UG-030 Evaluation Board User Guide
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.
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.
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.
08356-012
FILTERS
L
R
WIND NOIS E
DETECTION
Figure 11. Wind Noise Reduction Block Diagram
WIND NOIS E
REDUCTION
OUTPUTINPUTS
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,
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
08356-011
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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Evaluation Board User Guide UG-030
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 Tabl e 5. These addresses can be directly accessed and
modified via the control port of the ADAU1381.
Table 5. Wind Noise Reduction Page Parameters
Cell
Address
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)
On/off (burst write Addresses 0x0020
and Address 0x0021 together)
4 No
4 No
Sample Rate
Dependent?
Rev. 0 | Page 7 of 40
UG-030 Evaluation Board User Guide
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.
Figure 17. MicDistance Control
08356-017
Right-click the slider to enter the value directly (see Figure 18).
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
Rev. 0 | Page 8 of 40
Evaluation Board User Guide UG-030
Parameters
The enhanced stereo capture page parameters are stored in RAM, as outlined in Tabl e 7. These addresses can be directly accessed and
modified via the control port of the ADAU1381.
Table 7. ESC Page Parameters
Sample Rate
Address Cell Name Parameter Name Default Value Function Bytes
0x002B SS stereomux1940ns30 0x00, 0x00, 0x00, 0x00
0x002C SS stereomux1940ns31 0x00, 0x80, 0x00, 0x00
0x0023 Locked Cell param1 0x00, 0xCA, 0x9A, 0x58
0x0024 Locked Cell param2 0x0F, 0x35, 0x65, 0xA8
0x0025 Locked Cell param3 0x00, 0x7F, 0xAA, 0xE7
0x0026 Locked Cell param4 0x00, 0x08, 0x38, 0x65
0x0027 Locked Cell param5 0x00, 0x00, 0x00, 0x00
0x0028 Locked Cell param6 0x00, 0x7B, 0x1A, 0x7E
Gain setting related to the distance
between microphones that enhances
the perceived effect
On/off (burst write Address0x002B
and Address 0x002C together)
On/off (burst write Address 0x002B
and Address 0x002C together)
Locked parameter (generated by
SigmaStudio)
Locked parameter (generated by
SigmaStudio)
Locked parameter (generated by
SigmaStudio)
Locked parameter (generated by
SigmaStudio)
Locked parameter (generated by
SigmaStudio)
Locked parameter (generated by
SigmaStudio)
4 No
4 No
4 No
4 Yes
4 Yes
4 Yes
4 Yes
4 Yes
4 Yes
EQUALIZATION FILTERS PAGE
Dependent?
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.
Rev. 0 | Page 9 of 40
08356-020
The appropriate path is automatically selected when the mode is
selected on the main page
The switch
on this page (filtS) can be used to completely bypass
.
the EQFilter, if desired.
Controls
Click Show on the EQFilter cell to configure the filter bands
(see Figure 21).
Figure 21. EQFilter Control with Show Button
08356-021
UG-030 Evaluation Board User Guide
When Show is clicked, it displays a filter matrix with three rows
and six columns (see Figure 22).
022
Figure 22. EQFilter Matrix
08356-
The first row represents the six bands of the Record A (REC A)
mode, the second
row represents the six bands of the Record B
(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
d
filter. To individually tune a filter, click its corresponding button.
Clicking the menu at the top of the General Filter Settings
window provides access to a large variety of filters, each with its
own property pages and controls (see Figure 23 and Figure 24).
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.
08356-023
Figure 23. Individual Filter Band Settings
Figure 24. F
ilter Type Selection
024
08356-
More information on the various filters is available in the Help
menu within SigmaStudio.
Click Stimulus and Probe to open the Simulated Frequency Response window (see Figure 25 and gure 26).
Figure 25. Stimulus Button
Fi
8356-025
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 highpass 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.
8356-028
Figure 28. Default Record B (REC B) Mode EQ Curve
8356-026
Figure 26. Probe Button
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Evaluation Board User Guide UG-030
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).
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
filtS Switch to disable algorithm Algorithm enabled Function selection
The default EQ curves are intended only as examples and
should be specifically tuned for the target application system.
Example curves for
record and playback
Tune
Parameters
The equalization filters page parameters are stored in RAM, as outlined in Tab le 9. These addresses can be directly accessed and modified
via the control port of the ADAU1381.
Table 9. EQ Page Parameters
Sample Rate
Address Cell Name Parameter Name Default Value Function Bytes