VigilLink VL-DAAP-1 User manual

VL-DAAP-1
Audio Processor
VER 1.1

Thank you for purchasing this product
For optimum performance and safety, please read these instructions carefully before connecting,
operating, or adjusting this product. Please keep this manual for future reference.
A surge protection device recommended
This product contains sensitive electrical components that may be damaged by electrical
spikes, surges, electric shocks, lightning strikes, etc. The use of surge protection systems is highly
recommended to protect and extend the life of your equipment.
Table of Contents
1. Introduction.....................................................................................................................
1
2. Features...........................................................................................................................
1
3. Package Contents. ........................................................................................................
2
4. Specifications. ................................................................................................................
2
5. Operation Controls and Functions................................................................................
3
5.1. Front Panel. ...........................................................................................................
3
5.2. Rear Panel.............................................................................................................
4
6. Host Computer Control ..................................................................................................
4
6.1. Connection ...........................................................................................................
4
6.2. Volume Meter .......................................................................................................
6
6.3. Input Settings.........................................................................................................
7
6.4. Expander. ..............................................................................................................
8
6.5. Equalizer ................................................................................................................
9
6.6. Compressor...........................................................................................................
10
6.7. Auto Gain Control ................................................................................................
12
6.8. Auto Mixer.............................................................................................................
13
6.9. Adaptive Feedback Cancellation .....................................................................
14
6.10. Adaptive Echo Cancellation ............................................................................
16
6.11. Adaptive Noise Suppression. ............................................................................
17
6.12. Matrix Mixer. .......................................................................................................
18
6.13. Output Settings ...................................................................................................
19
6.14. Two-way Crossover............................................................................................
20
6.15. Delay. ..................................................................................................................
21
6.16. Limiter. .................................................................................................................
22
6.17. Scene Management..........................................................................................
24
6.18. Serial Settings......................................................................................................
24
6.19. Control Commands............................................................................................
25
7. Troubles and Solutions. ..................................................................................................
26
8. Application Example. ....................................................................................................
26

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1. Introduction
This digital audio processor is typically used for video conferences, distant learning, and
telemedicine. It features 4-ch MIC inputs, 6-ch linear inputs, and 10-ch linear outputs. Each
analog I/O channel has a knob to control the volume. A DANTE port is also provided to
ensure low latency in the audio process. Apart from HDMI audio embedding and de-
embedding, the product can also process audio signals with algorithms, such as, full-
band Adaptive Echo Cancellation (AEC), Adaptive Noise Suppression (ANS), Automatic
Gain Control (AGC), and Auto Mixer, to output a clear, clean, and resonant sound with a
high Signal-to-Noise ratio. Concise but intelligent, the processor is designed to be applied
in scenarios without additional software assistance for debugging. It is ready to use after
installation, and perfect for project implementation and testing.
The product can be applied in a diverse range of installations and applications across
industries, such as smart system integration in the small-medium sized conference room,
instruction recording and distance teaching in education, court trial recording and virtual
court trial in the judiciary, surgery recording, and video consultation in healthcare service,
andcommand center establishment in governmental projects.
2. Features
☆HDMI 2.0b compliant
☆Provide 4-ch balanced MIC inputs, 6-ch balanced linear inputs, and 10-ch balanced
linear outputs
☆Support HDMI audio signal input, loop out, audio embedding and de-embedding
☆Provide a standard DANTE network audio interface
☆Support adaptive feedback suppression function
☆Support the full-band adaptive acoustic echo cancellation technology
☆Dynamic adaptive noise reduction technology is provided to reduce noise with signal level
up to 18dB
☆Auto Mixer function is provided to set the order of priority when multiple microphones are input
at one time
☆Inclusive of Digital signal processing modules such as Expander, Equalizer, Compressor,
Auto Gain Control, Limiter, High Pass Filter, Low Pass Filter, and Delay
☆Support volume control, meter, scene control, etc.
☆Capable to switch matrix routings
☆48V phantom power supply for 4-ch MIC inputs
☆48KHz sampling rate, 24-bit for A/D or D/A conversion
☆Separated I/O volume knobs
☆Support monitoring function
☆Compatible to run on Win 7 and Win 10, with standard RJ45 interface control
☆Support RS-232 serial commands control

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3. Package Contents
①1 x Digital Audio Processor
②1 x AC 100~240V 50/60Hz Power Cord
③22 x Phoenix Connector (3-pin, 3.81mm)
④1 x User Manual
4. Specifications
Technical
HDMI Compliance
HDMI 2.0b
Amplitude-frequency
(20Hz~20KHZ@+4dBu)
MIC input: -0.5 / -0.5dB
Linear input: -0.2 / -0.2dB
THD+N (1KHZ@+4dBu)
≤0.01%
EIN (MIC input)
≤ -125dBu
SNR (linear input)
≥90dB
Dynamic Range
≥100dB
Channel Level Difference
-0.5 / -0.5dB
Channel Isolation
≥80dB
Channel Phase Difference
≤0.5°
Max Input Level
20dBu
Max MIC Gain
40dB
Input Impedance
20KΩ
Output Impedance
300Ω
Sampling Frequency 48KHZ
A/D and D/A Conversion
24Bit
Phantom Power +48 VDC
Connection
Inputs
4 × Balanced MIC [3-pin phoenix connector]
6 × Balanced LINE [3-pin phoenix connector] or
3 × Stereo Audio [3-pin phoenix connector]
Outputs
10 × Balanced LINE [3-pin phoenix connector] or
5 × Stereo audio [3-pin phoenix connector]
1 × Stereo audio [3.5mm L/R]
Controls
1 × LAN [RJ45]
1 × RS-232 [3-pin phoenix connector]
Digital Audio Interfaces
1 × HDMI IN [Type A 19-pin female]
1 × HDMI OUT [Type A 19-pin female]
1 × Dante [RJ45]

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Mechanical
Housing
Metal Enclosure
Color
Black
Dimensions
482mm (W)×250mm (D)×45mm (H)
Weight
2.9kg
Power Supply
AC 100 - 240V 50/60Hz
Power Consumption
11W(Max)
Operating Temperature
0°C ~ 40°C / 32°F ~ 104°F
Storage Temperature -20°C ~ 60°C / -4°F ~ 140°F
Relative Humidity 20~90% RH (non-condensing)
5. Operation Controls and Functions
5.1 Front Panel
No. Name Function Description
1 INPUT VOLUME
Knobs to control the volume of MIC and linear inputs.
Use a screwdriver in the clockwise direction to turn up the
volume; counterclockwise to turn it down.
2
Power LED
When the device is powered on, the red LED will light on.
3STATUS LED
When the device runs at normal, the green STATUS LED will
flash.
4 HEADPHONE
3.5mm headphone listening port.
Note: this listening port outputs the same audio as OUTPUT
LINE 5. If you want to enable the listening
function, you need
to select the corresponding input channel in the AUTO
MIXER module. For details, please refer to “6.8 Auto Mixer”.
5OUTPUT
VOLUME
Knobs to control the volume of linear outputs.
Use a screwdriver in the clockwise direction to turn up the
volume; counterclockwise to turn it down.
Note that turning the I/O knobs allows you to turn up/down the stereo volume, for
instance,if you turn the knob of LINE 1 clockwise, the volume both for LINE L1 and LINE
R1 will
be turned up at the same time.

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5.2 Rear Panel
No. Name Function Description
1INPUT
4-ch MIC inputs and 6-ch balanced linear inputs are used for
connecting audio source devices via phoenix connectors.
2OUTPUT
10-ch balanced linear outputs, used for connecting Amplifier
or Speaker via phoenix connectors.
3HDMI INPUT
HDMI signal input port, used for connecting HDMI source
device via HDMI cable.
4HDMI OUTPUT
HDMI signal output port, used for connecting HDMI display
device via HDMI cable.
5
DANTE
Standard DANTE network audio transmission interface.
6
RS-232
Used for connecting PC or control system via phoenix
connector. RS-232 (1) is used for transmitting RS-232 control
commands; RS-232 (2) for firmware upgrades.
7LAN Standard RJ45 interface for network connection.
8
BOOT 2-PIN
DIP Switch
Used for firmware upgrade.
Note: it can only be used when the device is in a factory.
Both2 pins should be switched to “1” when the device is in
service; if not, the device won’t kick off.
9
DIP Switch
Function reserved.
10
Power Switch
& Power Port
Power switch and AC 100-240V power input port.
6. Host Computer Control
6.1 Connection
The audio processor admits the control of a host computer in the windows system (WIN
7 orWIN 10). Take a cable to connect to the LAN port of the processor with the computer
fornetwork connection as following steps instructed.
Step 1: Install the “HDP-AU88P. exe” application on the host computer and change the
computer’s IP address to ensure that the IP addresses of the host computer and the
processor(Default IP address: 192.168.0.199, Subnet mask: 255.255.255.0) are within the
same network segment, as shown in the figure below.

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Step 2: Double-click the “HDP-AU88P. exe” icon (see the figure below) to run the application
on the host computer.
Step 3: When the application is open, the startup page appears, and then comes the
HOMEpage, as shown below.
Step 4: Check the Network icon on the tool bar as shown below.
If there is a red X at the top-right corner of this icon as the figure above shown, it means the
host computer has not connected with the audio processor yet. In such a case, click this
icon,then a dialog box with a list of IP addresses of all connected devices will pop out, as the
following figure shows.

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Step 5: Choose the audio processor’s IP address and click CONNECT, then the processor
will relate to the host computer. After connection, the CONNECT & SETUP box (see the
figure below) will pop out. You may close it directly if no such operation is needed.
If the audio processor has related to the host computer successfully, the Networkicon will
appear in a way as shown below.
6.2 Volume Meter
The volume meter is typically to denote the signal level of the I/O volume.
The meter has 3 colors (Red, Yellow, and Green) to denote signal levels for the volume of
16-ch inputs and 16-ch outputs, in which the Red denotes that volume may be too high to
be limited and the yellow indicates the volume is within a reasonable range.

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6.3 Input Settings
As shown below, by clicking the INPUT module on the HOME page or the IN tab onthe
menu bar, you will enter the INPUT setting page.
The following figure shows the details of the INPUT setting page.
SENSITIVITY: Indicates the sensitivity level of the input signal, which could be adjusted to
ensure the sensitivity on the input channel is adaptable to the MIC or linear input (function
reserved).
INVERT: When pressed, inverts the polarity of the signal on the input channel.
PHANTOM: When pressed, turns on 48V phantom power for the channel.
MUTE: When pressed, mutes the input channel, equivalent to clicking the letter M at the
INPUT module on the HOME page.
WHITE: When pressed, generates white noises for signal testing.
PINK: When pressed, generates pink noises for signal testing.
SINE: When pressed, generates sinusoidal signals for testing, the meter below shows the
frequency of the sine wave.
FADER: Controls the signal level of the testing signal.
You can set the signal parameters of an input channel by entering its CHAN No. on the
rightpanel of the IN page. Audio process modules, such as EXPANDER, EQUALIZER,
COMPRESSOR, and AUTO GAIN CONTROL, are available on the right panel for your
setting.

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6.4 Expander
The Expander, which can effectively extend the dynamic range of the input signal, is used
to eliminate noise under the threshold level.
As shown below, clicking the EXP module on the HOME page or the EXP button on
the right panel of the IN page will give you the option to enter the EXP setting page.
or
The following figure shows complete information to configure the expander module.
THRESHOLD: The level that the signal below it will be identified as noise and attenuated,
with a range of [-60.0, 0.0] dB.
RATIO: The compression ratio for signals below the threshold, with a scale of [1.0, 20.0]. If
theratio is 2.0, it means the signal below the threshold will be reduced to ½ of the original.
KNEE: The curve setting of the inflection point of the expander, with a scale of [0.0, 20.0].
0 indicates a hard knee; others indicate a soft knee.
ATTACK: The time required by the expander to begin the Expander process once a signal
drops below the threshold. Values are squashed between [1, 500] ms.
RELEASE: The time required by the expander to stop the Expander process once a signal
is over the threshold. Values are squashed between [1, 10000] ms.

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G.R. (Gain Reduction): Indicates the amount of gain attenuation of the input signal (in dB)
in the Expander process.
ACTIVE: Activates the EXP process. The function is equivalent to clicking the green letter A
at the EXP module on the HOME page.
Clicking the pair of UP-DOWN arrows of the CHAN No. on the right panel of the IN page will
give you an option to select an input channel for the Expander configuration.
6.5 Equalizer
Each I/O channel has an 8-band parametric equalizer for voice processing, with the
frequency, gain, and bandwidth of each band adjustable.
As shown below, by clicking the EQ module on the left panel of the HOME page or
theEQ button on the IN page, you will enter the EQ setting page of the input channel.
or
As shown below, click the EQ module on the right panel of the HOME page or
the EQ button on the OUT page, and you will enter the EQ setting page of the output
channel.
or

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The figure below shows the setting page of the EQ module.
TYPE: Provides 3 types of band/filters, with High Pass, Low Pass, and Peak optional.
FREQUENCY: In High/Low Pass Filter, it means the cut-off frequency of the EQ band;
In Peak Filter, it means the center frequency point of the EQ band; with a scale of
[20, 20000] Hz.
GAIN: In High/Low Pass Filter, fixed to 0dB; In Peak Filter, squashed between
[-24.0, +18.0] dB.
Q FACTOR: value is squashed between [0.02, 50.00].
SWITCH: Each band has a switch. When pressed, the EQ function in this band will be
disabled.
ACTIVE: Activates the EQ process. The function is equivalent to clicking the letter A at
the EQ module on the HOME page.
A pair of UP-DOWN arrows on the right panel is provided for you to select an input channel
for the Equalizer configuration.
6.6 Compressor
The Compressor is used to reduce the dynamic range of the signal above a user-determined
threshold so that output sounds could be more solid.
As shown below, click the COMP module on the HOME page or the COMP button
on the right panel of the IN page, you will enter the Compressor setting page.

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or
The figure below shows the setting page of the compressor.
THRESHOLD: The level that the signal above it will be compressed, with a scale of
[-60.0, 0.0]dB.
RATIO: The compression ratio for signals with levels above threshold, with a scale of
[1.0, 20.0]. When the ratio sets to 2.0, it means that the signal with a level below the
thresholdwill be reduced to ½ of the original.
KNEE: The curve setting of the inflection point of the compressor, with a scale of [0.0, 20.0].
0 indicates a hard knee, other values for a soft knee.
ATTACK: The time required by the compressor to begin the Compressor process once a
signal is over the threshold, with a scale of [1, 500] ms.
RELEASE: The time required by the compressor to stop the Compressor process once a
signal drops below the threshold, with a scale of [1, 10000] ms.
MAKEUP GAIN: Since the compressor can reduce the gain level, the Makeup Gain gives
youthe option to make a compensation for the gain level, with a scale of [-12.0, +18.0] dB.
G.R. (Gain Reduction): Indicates the amount of gain attenuation of the input signal (in dB).
OUT: Meter indicates the output signal level after the Compressor process.
ACTIVE: Activates the COMP process. The function is equivalent to clicking the letter A at
the COMP module on the HOME page.

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A pair of UP-DOWN arrows on the right panel is also provided for you to select any input
channel for the Compressor configuration.
6.7 Auto Gain Control
Automatic Gain Control (AGC) intends to control the audio signal within a range to extend
the distance of pick-up sounds. The AGC process aims to attenuate the signal with
excessive high signal levels and boost the signal with low signal levels up to the target
level.
As shown below, by clicking the AGC module on the HOME page or the AGC button
on the right panel of the IN page, you will enter the AGC setting page.
or
The following figure shows the setting page of the AGC module.
TARGET LEVEL: The desired output signal level, with a scale of [-20.0, 0.0] dB.
RATIO: The compression ratio for signals above the target level, with a scale of [1.0,
20.0].If the value sets to 2.0, it means that signals above the target level will be
reduced to ½ of the original.

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MAX GAIN: Indicates the maximum amount of gain that the signal with a low level can
beadded to by the AGC process, with a scale of [0.0, 12.0] dB.
THRESHOLD: Signals below the threshold will remain unchanged while those between
threshold and target level will be amplified (at a ratio determined by the MAX GAIN).
To ensure the work of the AGC process, the Target Level must be set higher than the
Threshold. The threshold is recommended to be set with a range of [-60.0, 0.0]dB.IN:
Meter indicates the detected signal level before the AGC process.
OUT: Meter indicates the output signal level after the AGC process.
ACTIVE: Activates the AGC process. The function is equivalent to clicking the green letter A
at the AGC module on the HOME page.
A pair of up-down arrows on the right panel is available for you to select an input channel for
the AGC configuration.
6.8 Auto Mixer
The AUTO MIXER is built to set the order of priority for MIC inputs when 4-ch microphones
are open at one time.
As shown below, by clicking the AUTO MIXER section on the HOME page or the
AUTOMIXER tab on the menu bar, you will enter the AUTO MIXER setting page.
The figure below shows the setting page of the Auto Mixer module.

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FADER: Each MIC input channel has a fader to adjust its signal level. It controls the mixed
volume on the input channel.
MUTE: Each MIC input channel has a MUTE button. Clicking it will enable you to mute the
channel.
PR (Priority): Sets the priority of the MIC input, 0 = lowest and 10 = highest priority. The
channel of higher priority will override the one of lower priority to be prioritized in the auto
mixer’s algorithm. If two or more channels are at the same priority, the channel for the main
speaker will be prioritized in the AUTO MIXER process.
AUTO: When pressed, the input MIC will be added to the auto mixing system, otherwise,
it won’t be auto mixed.
METER: Indicates the output volume after the AUTO MIXER process.
OUT: Fader to control the output volume after the AUTO MIXER process.
6.9 Adaptive Feedback Cancellation
The Adaptive Feedback Cancellation (AFC, also known as the Feed Back Suppression)
module can detect and suppress the frequency point of the acoustic feedback in the audio
field. The AFC module allows you to detect and suppress the frequency points of the
acoustic feedback for 8 bands. Parameters on this module are open to being configured.
As shown below, by clicking the AFC module on the HOME page or the AFC tab on
the menu bar, you will enter the setting page of the AFC module.

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The following figure shows the AFC setting page in detail.
MAX DEPTH: The maximum amount of gain that the module allows to reduce, with a scale
of [-20.0, 0.0] dB.
BANDWIDTH: The bandwidth of the filter, with Narrow and Wide optional.
RESET: When pressed, the filter will go back to initial settings and the module will re-detect
the frequency point of the acoustic feedback.
FIXBAND: When pressed, fixed values of parameters, such as, Frequency, Gain, and Width
will be applied to configure the filter in a fixed band. You are allowed to design the filter by
setting its parameters.
FIX ALL: When pressed, all bands will be processed in a fixed mode or USER-EDITING
mode.
BYPASSBAND: When pressed, the filter of the related band will be disabled. BYPASS
ALL: When pressed, the AFC module will be disabled, equivalent to pressing thegreen
letter A at the AFC module on the HOME page.
ANS LEVEL: The Adaptive Noise Suppression (ANS) module is provided to suppress noises
with signal levels up to 18dB. Clicking the ACTIVE button will give you the option to activate
this function.

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6.10 Adaptive Echo Cancellation
The 48kHz full-band Adaptive Echo Cancellation (AEC) algorithm is set to eliminate echoes
coming from the far end in a remote video conference.
As shown below, clicking the AEC module on the HOME page or the AEC tab on
the menu bar will allow you to enter the AEC configuration page.
The following figure shows the setting page of the AEC module.
LOCAL INPUT: Shows source signals that are available to have access to the AEC process.
AUTOMIXER refers to mixed signals after the AUTOMIXER process.
NONLINEAR PROCESSING: Determines the audio effect for both ends of the video
conference, with soft, medium, and aggressive optional.
ACTIVE: Activates the AEC process. The function is equivalent to clicking the green letter A
at the AEC module on the HOME page.
ANS LEVEL: The Adaptive Noise Suppression (ANS) function is provided to suppress
noises with signal levels up to 18dB. Pressing the ACTIVE button, the function will be
enabled.REMOTE REFERENCE: Indicates signals that are available for the AEC algorithm
to learn how to eliminate analogous echoes in practice.

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As shown in the figure above, we choose LineIn1-L and LineIn1-R as reference signals,
in such case, we need to connect the INPUT LINE (L+R) 1 port of the processor to the
terminal of the video conference and connect the OUTPUT LINE (L+R) 1 port to an external
speaker. As the reference signal after the AEC algorithm will be sent to the far end by the
terminal, we also need to configure selected input channels in the matrix routing.
The figure below gives an example of configuring the matrix routing.
6.11 Adaptive Noise Suppression
The 48kHZ full-band Adaptive Noise Suppression (ANS) algorithm can effectively increase
the signal-to-noise (S/N) ratio and suppress noises from the blower or air conditioner.
As shown below, by pressing the ANS module on the HOME page or the ANS tab
on the menu bar, you will enter the ANS configuration page.
The following figure shows the details of the ANS setting page.

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LOCAL INPUT: Shows source signals that are available to have access to the ANS process.
AUTOMIXER refers to mixed signals after the AUTOMIXER process.
ANS LEVEL: The ANS module provides 12 noise suppression levels, ranging from 6dB to
18dB.
ACTIVE: Activates the ANS process. The function is equivalent to clicking the green letter A
at the ANS module on the HOME page.
6.12 Matrix Mixer
The Matrix Mixer allows you to free choose the audio route by matrix mode.
As shown below, clicking the MATRIX MIXER section on the HOME page or the MATRIX
tab on the menu bar will enable you to enter the MATRIX MIXER configuration page.
The following figure shows the configuration page in detail.