ANALOG DEVICES UG-030 Service Manual

Evaluation Board User Guide
UG-030
One Technology Way P. O . Box 9106 Norwood, MA 02062-9106, U.S.A. Tel : 781.329.4700 Fax : 781.461.3113 www.analog.com
Using the ADAU1381 Sound Engine
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
DUAL-BAND
DYNAMIC
PROCESSOR
RECORD OUTPUT
PLAYBACK
OUTPUT
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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 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.
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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 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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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.
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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
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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).
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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.
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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
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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.
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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
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WIND NOISE REDUCTION PAGE

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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.
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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,
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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
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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.
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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.
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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 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
0x0021 WN stereomux1940ns41 0x00, 0x80, 0x00, 0x00
Name Parameter Name Default Value Function Bytes
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?
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ENHANCED STEREO CAPTURE PAGE

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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
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Right-click the slider to enter the value directly (see Figure 18).
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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.
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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 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
0x0029 MicDistance Gain1940AlgNS1 0x00, 0x80, 0x00, 0x00
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.
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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
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When Show is clicked, it displays a filter matrix with three rows and six columns (see Figure 22).
022
Figure 22. EQFilter Matrix
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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.
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Figure 23. Individual Filter Band Settings
Figure 24. F
ilter Type Selection
024
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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
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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.
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Figure 28. Default Record B (REC B) Mode EQ Curve
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Figure 26. Probe Button
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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).
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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
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
Dependent?
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Sample Rate
Address Cell Name Parameter Name Default Value Function Bytes
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
Dependent?
Rev. 0 | Page 12 of 40
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