TiGHT AV DSP-AEC-1010-DA User manual

DSP-AEC-1010-DA
Audio DSP with AEC
4-in/4-out analog, 4-in/4-out Dante and USB
All Rights Reserved
Version: DSP-AEC-1010-DA_2021V1.0

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Preface
Read this user manual carefully before using the product. Pictures shown in this manual are for reference only. Different models
and specifications are subject to real product.
This manual is only for operation instruction, please contact the local distributor for maintenance assistance. The functions
described in this version were updated till Sep 2021. In the constant effort to improve the product, we reserve the right to make
functions or parameters changes without notice or obligation. Please refer to the dealers for the latest details.
FCC Statement
This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the
instructions, may cause harmful interference to radio communications. It has been tested and found to comply with the limits for
a Class A digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection
against harmful interference in a commercial installation.
Operation of this equipment in a residential area is likely to cause interference, in which case the user at their own expense will
be required to take whatever measures may be necessary to correct the interference.
Any changes or modifications not expressly approved by the manufacture would void the user’s authority to operate the
equipment.

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Table of contents
1. Technology Overview................................................................................................................................................................5
1.1 Introduction to Technology..................................................................................................................................................5
1.2 About Dante AES67............................................................................................................................................................5
1.3 Audio Input Section.............................................................................................................................................................5
1.4 Audio Output Section..........................................................................................................................................................5
1.5 Floating Point DSP..............................................................................................................................................................6
1.6 Typical System Application .................................................................................................................................................6
2. Hardware...................................................................................................................................................................................7
2.1 Safety Instructions...............................................................................................................................................................7
2.2 Audio Wiring Reference ......................................................................................................................................................8
2.3 Specifications......................................................................................................................................................................9
2.4 Mechanical Data ...............................................................................................................................................................10
2.5 Front Panel........................................................................................................................................................................10
2.6 Rear Panel........................................................................................................................................................................10
3. Software..................................................................................................................................................................................12
3.1 Software Installation..........................................................................................................................................................12
3.2 Using the Software............................................................................................................................................................12
3.3 Audio Parameters .............................................................................................................................................................13
3.3.1 Input Source...................................................................................................................................................................13
3.3.2 Expander........................................................................................................................................................................14
3.3.3 Compressor & Limiter.....................................................................................................................................................14
3.3.4 Auto Gain Control...........................................................................................................................................................15
3.3.5 Equalizers ......................................................................................................................................................................16
3.3.6 Feedback filters..............................................................................................................................................................17
3.3.7 AutoMixer.......................................................................................................................................................................18
3.3.8 Acoustic Echo Cancellation (AEC).................................................................................................................................19
3.3.9 Automatic Noise Suppression (ANS)..............................................................................................................................20
3.3.10 Matrix ...........................................................................................................................................................................20
3.3.11 High & Low Pass Filter.................................................................................................................................................21
3.3.12 Delay............................................................................................................................................................................21
3.3.13 Output ..........................................................................................................................................................................21
3.3.14 USB Soundcard ...........................................................................................................................................................22

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3.4 Setting Menu.....................................................................................................................................................................23
3.4.1 File Menu .......................................................................................................................................................................23
3.4.2 Device Setting................................................................................................................................................................24
3.4.3 Group Setting.................................................................................................................................................................24
3.4.4 Panel Setting..................................................................................................................................................................25
3.4.5 Dante Setting .................................................................................................................................................................26
3.4.6 Help Menu......................................................................................................................................................................27
4. Control.....................................................................................................................................................................................27
4.1 External Control Programmer............................................................................................................................................27
4.2 Control Protocol ................................................................................................................................................................28
4.3 Serial Port-to-UDP (RS232 To UDP) ................................................................................................................................30
Appendix B: Module Parameter Types (1)...................................................................................................................................32
Appendix B: Module Parameter Types (2)...................................................................................................................................34
5. Customer Service....................................................................................................................................................................36

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1. Technology Overview
1.1 Introduction to Technology
The Tight AV DSP system is a fixed architecture signal processing platform with powerful signal processing and routing options.
The hardware is equipped with analog audio interface connectivity, as well as digital audio using Dante for audio over Ethernet
and USB-audio as an interface for computers.
Tight AV DSP 1.0 is a Windows-based application used to configure and control the DSP hardware. An easy-to-use GUI gives
the user fast access to all the available processing functions. The unit has 16 built-in presets that can store the complete
configuration for later recall.
The Tight AV DSP can be controlled via the software, a dedicated wall panel controller or from 3rd party systems using the API
control protocol over RS-232/485 or Ethernet.
1.2 About Dante AES67
Dante/AES67 audio networking utilize standard IP networks to transmit high-quality, uncompressed audio with near-zero latency.
It's the most economical, versatile, and easy-to-use audio networking solution, and is scalable from simple installations to large-
capacity networks running thousands of audio channels. Dante/AES67 can replace multiple analog or multicore cables with a
single affordable Ethernet cable to transmit high quality multi-channel audio safely and reliably. With Dante software, the network
can be easily expanded and reconfigured with just a few mouse clicks. Dante/AES67 is the audio networking choice of nearly
all professional audio manufacturers, with hundreds of Dante-enabled audio products now available.
For more information, please visit the Audinate website at www.audinate.com.
1.3 Audio Input Section
The DSP supports up to 4 balanced analog audio inputs available on removable phoenix connectors. The analog input section
supports microphone or line-level signals. +48VDC phantom power can be adopted for each input.
Preamp gain and phantom power can be conveniently controlled via Tight AV DSP 1.0 software.
1.4 Audio Output Section
The analog output section offers 4 balanced channels available on removable phoenix connectors. 24bit D/A conversion provides
excellent signal to noise ratio. With input and output faders set to 0dB the unit operates at unity gain. Nominal output level is+4dBu
with 20dB headroom. 0dBFS digital signal is equivalent to +24dBu.

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1.5 Floating Point DSP
The signal processing is done in an Analog Devices SHARC DSP, enabling 32-bit and 40-bit floating-point processing. Floating-
point processing provides advantages for the users in terms of sound quality and usability.
With floating point processing the internal calculations of the DSP will have higher dynamic range and avoid rounding errors that
can be a cause of noise.
1.6 Typical System Application
Meeting Room System: Tabletop or ceiling microphones, with local output to amplifier and
speakers. Wireless microphones via Dante, audio over Ethernet.
The output signal can also transmit to a recording device with Dante interface e.g., a computer
with Dante virtual soundcard. Volume control and preset recall can be done from a wall panel
controller.
Dante Application: Dante enables easy integration of compatible devices over a network. Possible sources and destinations
include microphones, mixers, music players and amplifiers, recorders and streaming devices.

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2. Hardware
2.1 Safety Instructions
Safety Instructions
Important safety instructions:
1. Read these instructions.
2. Keep these instructions well.
3. Pay attention to all warnings.
4. Follow all instructions.
5. Please keep the device away from water. The device shall not be exposed to water drips or water splashes; make sure that
there is no object with liquid near the device, such as vase.
6. Please use dry cloth to clean up the device.
7. Please do not block the vent. Please get the device installed based on the manufacturer's instructions.
8. Please do not install any heat source, such as radiator, heat register, furnace or other devices (including amplifiers) that
generate heat.
9. Please adopt protective grounding connection to connect the device to the power socket. Please do not use polarized plug
or grounding plug. A polarized plug has two leaves, and one is wider than another. A grounding plug has two leaves and a
third ground terminal. The wide leaf or third ground terminal can provide safety for the users. If the plug provided does not
accord to the power socket, please contact the electrician to replace the old socket with a new one.
10. Protect the power cord so that it will not be tramped or protruded, especially the plug, the socket and the connections of
cord and device.
11. Please use the accessories designated by the manufacturer.
12. Please only use the cart, the tripod, the holder or the desk designated by the manufacturer or sold together with the device.
When using the cart, please take care with the mobile cart/device to avoid injury from rollover.
13. Please unplug the device during a thunderstorm or during the idle period.
14. Please find qualified maintenance personnel to deal with all maintenance problems. When the device gets damaged in any
manner, the maintenance is required. For example, the power cord gets damaged, liquid spill or the object falls into the
device; the device is exposed to the rainwater or moisture; the operations are not correct, or the device falls off.
The lightning logo (an equilateral triangle with an arrow) is used to make the users aware of the uninsulated "dangerous voltage"
within the product shell, which may cause electric shock. An equilateral triangle with an exclamation mark is adopted to make
the users understand the importance of the operations and maintenance instructions given in the appendixes attached to the
product.
Warning: In order to prevent electric shock, please do not use a polarized plug provided by a device with an extension cord.
The socket outlet cannot be inserted except for sharp end.

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2.2 Audio Wiring Reference
Balanced Connection
Any of these interfaces may occur on both sides of the connection.
Note: For one XLR interface, the female connects to the output device and the male connects to the input device.
Unbalanced Connection
RCA interface and 1/4-inch TS interface are unbalanced interfaces. A multi-strand shielding conductor may be installed on both
ends of unbalanced connection.

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2.3 Specifications
Processor
ADI SHARC 21489
Sample rate/bit depth
48kHz/24bit
Input gain steps
0/6/12/18/24/30/36/42/48dB
Phantom Power
48V
Frequency Response (20Hz~20KHz)
±0.5dB
Maximum Level
+18dBu
THD + Noise
0.003%@4dBu
Dynamic Range
110dB
Background Noise (A-weighted)
-91dBb
Common Mode Rejection Ratio @60Hz
80dB
Channel Isolation @1KHz
108 dB
Input Impedance (Balanced Connection)
5.6kΩ
Output Impedance (Balanced Connection)
102Ω
System Delay
<3ms
Power requirements
PoE or 12V
Power adapter
AC110~240V 5Hz-60Hz, 12V DC / 2A output
Maximum Power Consumption
10W
Dimensions (Width x Depth x Height)
215 x 162 x 44mm
Shipping Weight 2kg

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2.4 Mechanical Data
Space required:
1U (W x D x H: 215 x 162 x 44mm).
At least 70mm should be reserved for the connections on the rear panel. Reserved depth depends on the wire used and the
connection mode.
Ventilation:
The recommended highest operating ambient temperature is 30℃/ 86℉.
Make sure that there is no blockage of the venting holes on both sides of the unit. A gap of at least 5cm shall be reserved).
Electrical Property:
The unit can be powered by PoE (802.3af) or with a 12V power adapter.
Universal input power adapter: AC110~240V 5Hz-60Hz, 12V DC / 2A output.
Shipping Weight:
(2 kg)
2.5 Front Panel
Power: LED power indicator.
Run: The operation status indicator of the device.
USB AUDIO: USB soundcard for recording and playback from computer.
2.6 Rear Panel

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Power Source:
Power connector: 12V DC / 2A
Dante/PoE
100 Base-T Ethernet connector, provide up to 8 (4x4) channel Dante network audio. Uses the UltimoX4 chip. This interface is
also used for communication with PC software and any third-party external controller.
Dante
100 Base-T Ethernet connector, switched network interface (not for redundant Dante use)
RS485
Used for the serial communication port Tx = sending or data output or Rx = receiving or data input that
connects to a third-party control device.
RS485 can be used for voice tracking control (or other output commands), or for bus input control. A
central command can be used to conveniently integrate it into your software.
Port setting: 115200 baud (default), 8 data bits, 1 stop bit, no parity, no flow control.
RS232
Used for serial communication with external devices. Port Tx = sending data, and Rx = receiving data
input from a third-party control device.
Port setting: 115200 baud (default), 8 data bits, 1 stop bit, no parity, no flow control.

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3. Software
3.1 Software Installation
A Windows PC with a processor of 1 GHz or higher and:
Windows 7 or higher version.
1 GB free storage space.
1024 x 768 resolution.
24 bit or higher color.
2GB or higher memory.
Network (Ethernet) port.
1. Download software and install files.
2. Double click the downloaded file and install by following the instructions on the screen.
After the software is installed, read other parts of the help file or execute the software.
After the software is installed, use one of the following methods to enable the software:
1. Desktop icons:
2. Start menu:
When starting the software for the first time, it may take some time (1-15s) to start it. Please wait for a while.
3.2 Using the Software
After starting the software, the main menu is shown as below:
Click “Device List” in the top right corner of the main menu, to discover all processors on the
network automatically.

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Click “Connect” to match the device on the network with the unit in the open tab in the software. One processor supports
simultaneous connection and control of up to four users.
In the “Device List” menu there are also settings for assigning static IP-address.
3.3 Audio Parameters
There are two modes for parameter editing:
•Click the input or output channel modules and enter the parameter interface of the module.
•Right click the module and the configuration interface will pop out. The first mode is adopted for the following module
parameters.
3.3.1 Input Source
Sensitivity: Input gain in nine steps (0/6/12/18/24/30/36/42/48dB).
Phantom Power: Provides +48V DC power for condenser microphones. Do not enable phantom power for a line input or when
the power is not required, as this may damage the external device.
Sine Wave: Click to activate tone generator. Drag the frequency slider to set frequency (20~20 kHz). Drag level slider to adjust
output level (unit: dBFS). Use the slider or click the text field to set a value.
White Noise: Click to activateWhite noise generator. White noise has a flat frequency spectrum. Use level slider to adjust level.
Frequency slider has no function in this mode.
Pink Noise: Click to activate pink noise generator. Pink noise falls off by 3dB/octave. Use level slider to adjust level. Frequency
slider has no function in this mode.
Right click to copy/paste parameters between channels
Right click/Group Setting: Open the group setting interface swiftly.
Right click/Minimum and Maximum Gains: Limit the maximum and minimum of the gain of a channel.

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3.3.2 Expander
The expander will reduce the signal level by the selected ratio when the signal goes below the threshold.
Fig.3.2 Expander The
expander has the following control parameters:
Threshold: The expander is active when the signal goes below this threshold.
Ratio: Refers to the slope below the threshold point on the gain curve. Higher ratio results in more attenuation below threshold
Attack time: Refers to the time required to activate the expander when the signal falls below the threshold.
Release time: Refers to the time required for the gain to be restored when signal is above the threshold.
3.3.3 Compressor & Limiter
Compressor
The compressor is used to reduce the dynamic range of the signal higher than the threshold and maintaining the dynamic range
of the signal below the threshold.
The compressor has the following control parameters:
Threshold: The compressor is active when the signal goes above this threshold.
Ratio: Refers to the compression ratio. The ratio sets the degree of gain reduction when the signal is above the threshold level.
The adjustable range of compression ratio is 1-20.

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Attack & Release Time: Refers to the time required to activate the compressor when the signal goes above the threshold.
Output Gain: Used to compensate for the gain reduction in the compressor.
G.R. and output Level Meter: G.R. indicates the compressors amount of gain reduction. Output refers to the output level of the
signal that has passed through the compressor module. Gain reduction is displayed in an inverse level meter.
Limiter
The limiter on the outputs works in similar way as the compressor, but with a high ratio. The output level will not exceed the set
threshold.
The limiter only provides two parameters: Threshold and Release Time.
3.3.4 Auto Gain Control
The Auto Gain Control (AGC) is a dynamic processor used to adjust the signal to a set target level, while maintaining the dynamic
range.
The threshold is set at a very low level with middle-to-slow attack time, long release time and low ratio. The auto gain control
includes silent detection to prevent unwanted changes during silent periods.
Auto gain control may e.g. be used to normalize the level of background music player.
Auto gain control includes the following control parameters and switches:

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Threshold: When the signal level is lower than the threshold, the input-to-output ratio is 1:1. When the
signal level is higher than the threshold, the input-to-output ratio changes with the ratio control settings.
The threshold should be set just above the background noise level to avoid amplification of the noise.
Ratio: Refers to the ratio of the changes in level when the input signal is higher than the threshold.
Target level: Refers to the target output level. If the signal is higher than the threshold, the controller will compress the signal
proportionally.
Attack (ms): Refers to the time required to activate the AGC when the signal is above the threshold.
Release (ms): Refers to the time required for the gain to be restored when signal is below the threshold
3.3.5 Equalizers
Filter for changing the frequency balance of the signal.
The equalizer has the following control parameters:
Fig.3.6 Equalizer
Type: Parametric EQ is the default filter type. Additionally, high & low shelf filters and high & low pass filters can be selected.
High & Low Pass Filter: The high-pass filter will pass through frequencies higher than the set frequency and attenuate the
frequencies below this with a slope of 12dB/octave. Default Q of 0.71 gives a maximally flat amplitude (Butterworth). The low-
pass filter works in a similar way but attenuating above the corner frequency.
High & Low Shelf Filter: A shelving filter will amplify or attenuate the frequencies above (high shelf) or below (low shelf) the
set frequency by the amount set in the gain parameter.
Frequency (Hz): Refers to the centre frequency of the filter.
Gain (dB): Amount of gain or attenuation of the signal at the set frequency.
Q: Refers to the quality factor of a filter. The adjustable range of Q value is 0.02-50.
For a parametric EQ filter, Q value refers to the width of the bell-shaped frequency response curve on both sides of the cut-off
frequency.
When the filter is a high & low shelf filter or a high & low pass filter, if Q>0.707, there will be peaks in the filter responses. If
Q<0.707, the slope will become flatter.
The Q-factor parameter should be used with caution, as a too high value can result in a resonant filter function.
Each segment of the equalizer has a switch, used to turn on or turn off the filter. After being closed, the parameter setting will
not work. The equalizer has a master switch used to enable or disable all filters.

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3.3.6 Feedback filters
After setting proper levels for microphone and speakers, the feedback filter section can be used as a secondary precaution to
avoid unwanted feedback. The traditional methods should still be used, such as limiting the number of open microphones,
minimizing the distance between sound source and microphone, good distance between the microphone and loudspeakers, and
equalizing the room speakers to get a flat response. Later, we can adopt feedback filters to get additional gain. The feedback
filters cannot be used to magically solve the system's design defects or improve the sound transmission gain in a way exceeding
the system's physical limitations.
The feedback filter module automatically detects and prevents feedback in the sound system. The module distinguishes
feedback from expected sounds based on the characteristics of the signal. When feedback at a certain frequency is detected, a
notching filter will be automatically added at this frequency to attenuate it. During the first addition, the notching filter only
attenuates the feedback a bit. If the feedback still exists, the notching filter will continue to attenuate the feedback in accordance
with the preset parameters until the feedback disappears or reaches the maximum preset parameter. Multiple user parameters
can be used for accurate fine tuning of the effects of the module.
After tuning, the filter may be locked up to prevent any change during the performance. The filter settings can be copied to a
dedicated notching filter module (parametric equalizer). Eight filters are set as auto filters in an automatic cycle. In this way,
filters for temporary use can be removed.
Each channel has a feedback filter section. Click on/off button to enable or disable the filter section. Each feedback filter module
has 8 narrow-band filters.
The feedback filter module has the following adjustable parameters:
Panic Threshold: According to this parameter, "any signal higher than the threshold is absolutely a feedback". When a signal
level is higher than the feedback threshold, any of the following circumstances will occur:
(a) the output gain is temporarily attenuated to control the speed of feedback
(b) the output level is restricted to prevent out of control
(c) the filter's sensitivity is increased for faster detection and feedback
Once the output level is lower than the threshold, the gain will be recovered, and the sensitivity is restored to normal state. This
value refers to the peak value of digital range signal. If the value is set as 0, this function is disabled.
Feedback Threshold: Signal below this threshold will not be analysed by the feedback filter section.
Filter Depth: Refers to the maximum attenuation of a single filter. A low setting may prevent over-processing effects caused by
the filter or notching filter. It may cause worse feedback control, especially in a large narrow resonance system.
Bandwidth: 1/10 and 1/5 Octave can be chosen. The filter will not become wider due to the increase of depth. In the case of
frequent feedback, the bandwidth can be set at 1/5Oct for a wider bandwidth.
Preset: There are four built-in presets: "big music room", "small music room", "big voice room" and "small voice room". These
four presets apply to the default settings of most applications.

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Filter Mode: Each feedback filter has three modes: dynamic, manual and fixed. When manual mode is set, the gain can also be
manually set. When Fixed mode is set, the filter always works and will not be occupied by new feedback points; it still works
when being rebooted. When all eight filters are used and new feedback is detected, the module will take one of the “Dynamic”
filters and use this to inhibit new feedback.
Clear: Click the button to instantly clear up all filters. It will clear up all feedback points found previously.
The feedback filter section can be used as a tool during the system commissioning to identify feedback points or as a preventive
measure during normal operations. If you want to get higher system transmission gain and feedback inhibition effect, it is
recommended that you debug by following the steps below:
(a) Reduce the system gain, and use the button "Clear" to reset all filter parameters
(b) Set up parameters for the feedback filter module. Also, decrease the panic threshold to reduce the feedback level.
(c) Open all microphones, and slowly increase system gain until the feedback occurs. Stop increasing system gain when
the feedback occurs.
(d) Wait for the feedback inhibition module to take effect; after the feedback disappears, continue to increase gain.
(e) Repeat the operation until the system reaches the required gain or until all filters are fully distributed
(f) Change the panic threshold to a maximum level just higher than the expected non-feedback signal.
At this time, if needed, you may set Fixed mode for each filter or save the dynamic status to deal with possible feedback during
the performance period. Additionally, you may copy the filter to the notching filter module (parametric equalizer).
3.3.7 AutoMixer
A normal mixer is simply summing the signals. For every doubling of open microphones, the total gain will double. This will
amplify noise and room noise leaking into unused microphones.
The gain sharing automixer solve this problem by maintaining a fixed total gain. Unused microphones will be attenuated while
the microphones in use will be open, sharing the available gain.
To use the Automix, the direct routing of the microphone input channel should be disabled, and the output of the Automix should
be routed to the selected destination in the matrix.
The automix function can be activated for each channel independently. Disabling the automix function for a channel will pass
the signal through like a normal mixer, and not influence the automix algorithm.
There are two groups of control parameters in the automix module: main control parameters and channel control parameters.
(1) Main control parameters
Gain: controls the main output volume of the automixer
Slope: The slope control influences the attenuation of unused channels. If the slope is higher, the level of
the unused channels will be reduced. It is suggested that the value beset at or around 2.0. A slope parameter
set at 1.0 will make the automixer function like a normal mixer. If set at 3.0, the action will result in larger
gain reduction, which may sound unnatural. The bigger the value, the more the channel is opened and the
more the total attenuation. The recommended value is around 2.0.
Response Time: Shorter response time ensure fast opening of microphones in use. Longer response time
gives a smooth operation but may cut off the start of a word if set too long. In practice the best effect will be
when response time is between 100ms and 1000ms. The autogain algorithm will open microphones faster
than closing them. Therefore, the start of words will not be cut off even with the response time at 100ms. If
set to several seconds, there will be a longer hold time of the response time in the automixer. The previous
active channel will remain open for several seconds.
Mute: master mute for the automixer
On/off: button for activating or deactivating the automixer function.

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(2) Channel control parameters
AutoMixer: Each channel has an automix on/off button which must be turned on for channels to participate in the automix.
Mute: Both channel mute and fader are behind auto gain. If the channel level is increased, the gain level of other channels may
also be reduced even if the channel mute is on.
Gain: Gain fader for adjustment will increase/decrease the volume proportion in automix.
Priority: Priority setting for automix algorithm. Priority parameter ranges from 0 to 10. Higher value gives
higher priority.
Both channel mute and fader are behind auto gain. Any adjustment made in these two parameters will not
influence the operation of the automix. For example, If the channel level is higher, the level gain of other
channels may be reduced even if the channel mute is on. Channel mute shall be turned on and automix
shall be turned off to mute the signal and prevent its influence on the automix. Mute button at each channel
shall be muted and directly connect output mute when mixing sound. Channel faders also control sound
mixing level and direct output level of channels.
Priority control allows high priority channels to override low priority channels, and thus the automix algorithm
will be affected. Priority value can be set from 0 (the lowest priority) to 10 (the highest priority), and the
default value is 5 (standard priority). Users may use slider or click textbook to input a specified priority
between 0 and 10 to adjust priority.
If two channels have the same signal level, then the channel with higher priority will get more auto gain. If
there is one-unit priority between them, then the channel with higher priority will get extra sound mixing gain
of 2dB (suppose the slope of the two channels is set at 2.0). For example, if channel 1 and 2’s priorities are
respectively set at 6 and 3, and the input level of those two channels are the same, then channel 1 will get
extra sound mixing gain of 66dB than channel 2. During operation, it shall be noted that the slope setting of
main control parameters will also influence sound mixing gain difference brought by the priority weight of
channels. If the slope is set at 3.0, then one priority unit difference will result in gain difference of 4dB. If all
channels have the same priority, then their priority settings shall be set at 5.
Note: Users should be very careful when using high priority differences between channels, such as priority of 0 and 10. If
channels with high priority recognize signals like background music from speaker, then it is possible for them to mask channels
with lower priority even the former is not used. It will get worse if the slope is higher. To prevent this users may consider installing
a noise gate or expander between automixers at the highest priority channels. Threshold should be set at the level where it is
not opened by the noise gate or expander.
3.3.8 Acoustic Echo Cancellation (AEC)
To use the AEC module, select the local microphones to be processed, select the input from the far side talker and do the
appropriate routing of the AEC channel in the matrix.
Acoustic echo cancellation (AEC for short) is a type of digital audio signal processing to improve. It is used in audio/video
conferencing when talkers in the local conference room are talking with one or more speakers at a remote location. AEC
improves the remote talkers experience by cancelling acoustic echo generated in the local room.
Echo cancellation module for remote calls can be used for local amplification of remote voice signals and attenuate the
interference caused by acoustic echo. Its basic operation principle is simulating the echo channel, calculate the echo generated
by remote signals and then subtract the estimated signal from the input signal from microphones.
There is one echo cancellation module in the Tight DSP unit. Two local input and remote output mixers are preset to realize
multichannel signal participating echo cancellation as shown in the figure. One parameter can be adjusted:
Non-linear filter (NLP): Three types including Conservative, Moderate and Aggressive can be selected to determine echo
suppression levels.

Audio DSP with AEC, 4-in/4-out analog, 4-in/4-out Dante and USB
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3.3.9 Automatic Noise Suppression (ANS)
The Noise suppression module can remove noise from the microphone signal. To use the ANS, the direct routing of the
microphone input channel should be disabled, and the output of the ANS should be routed to the selected destination in the
matrix.
There is only one noise suppression module in the DSP Controller. Select the channels to be noise cancellation as shown in the
figure.
Suppression level:
There are three levels of noise reduction:
Mild(6dB), Medium(10dB) and Aggressive(15dB).
3.3.10 Matrix
The matrix is the core function for routing inputs and processing functions to the appropriate output. As shown in the figure, the
horizontal direction indicates the input channels, and the vertical direction indicates output channel. One-to-one input to outputs
is the default setting. Click a field to establish the new route and click again to remove it. Right-click to bring up the level control
for the crosspoint. When using automixing, echo cancellation and noise suppression modules use the matrix to route the
processed signal to the wanted output.