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OPERATING MANUAL
ITDSP-0404D
ITDSP-0808D
ITDSP-1208D
ITDSP-1616D
ITDSP-0404Dv2
ITDSP-0808Dv2
ITDSP-1616Dv2
Digital Sound Processor
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Help File
1. Technology Overview .................................................... 5
1.1 Introduction to Technology ............................................................................................................................ 5
1.2 Audio Input Section........................................................................................................................................... 5
1.3 Audio Output Section ....................................................................................................................................... 6
1.4 Float Point DSP ................................................................................................................................................... 7
1.5 Audio Flow ........................................................................................................................................................... 9
1.6 Typical System Application ........................................................................................................................... 10
2. Hardware ................................................................. 12
2.1 Safety Instructions ........................................................................................................................................... 12
2.2 Audio Wiring Reference ................................................................................................................................. 13
2.3 Specifications .................................................................................................................................................... 15
2.4 Mechanical Data ............................................................................................................................................... 16
2.5 Front Panel ......................................................................................................................................................... 16
2.6 Rear Panel .......................................................................................................................................................... 17
3. Software ................................................................... 18
3.1 Software Installation ....................................................................................................................................... 18
3.2 Using the Software .......................................................................................................................................... 19
3.3 Custom edit processing module .................................................................................................................. 20
3.4 Audio Module Parameters............................................................................................................................. 20
3.4.1 Input Source ............................................................................................................................................... 21
3.4.2 Expander ..................................................................................................................................................... 22
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3.4.3 Compressor & Limiter ............................................................................................................................. 23
3.4.4 Auto Gain Control .................................................................................................................................... 25
3.4.5 Equalizers .................................................................................................................................................... 26
3.4.6 Figure Balancer .......................................................................................................................................... 28
3.4.7 Feedback Inhibition ................................................................................................................................. 28
3.4.8 Noise Gate .................................................................................................................................................. 31
3.4.9 Ducker .......................................................................................................................................................... 32
3.4.10 Spl ............................................................................................................................................................... 33
3.4.11AutoMixer .................................................................................................................................................. 33
3.4.12Echo Canceller .......................................................................................................................................... 37
3.4.13Noise Suppression .................................................................................................................................. 37
3.4.14 Matrix ........................................................................................................................................................ 38
3.4.15 High & Low Pass Filter .......................................................................................................................... 38
3.4.16 Delayer ...................................................................................................................................................... 39
3.4.17 Output ....................................................................................................................................................... 39
3.4.19 USB Soundcard ....................................................................................................................................... 41
3.5 Setting Menu ..................................................................................................................................................... 43
3.5.1 File Menu .................................................................................................................................................... 43
3.5.2 Device Setting ............................................................................................................................................ 44
3.5.3 GPIO Setting .............................................................................................................................................. 44
3.5.4 Group Setting ............................................................................................................................................ 47
3.5.5 Panel Setting
.............................................................................................................................................. 48
3.5.6 Dante Setting ............................................................................................................................................. 51
3.5.7Help Menu ................................................................................................................................................... 55
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4. Dante Audio .............................................................. 56
4.1 Dante Overview ................................................................................................................................................ 56
4.2 Dante Requirements .................................................. 58
4.3 Dante Network Design ................................................................................................................................... 58
4.4 Dante Modes ..................................................................................................................................................... 60
4.5 Dante Controller ............................................................................................................................................... 61
4.6 Dante Virtual Soundcard ................................................................................................................................ 61
5. Control .................................................................... 63
5.1 External Control Programmer ....................................................................................................................... 63
5.2 Control Protocol ............................................................................................................................................... 63
5.3 Serial Port-to-UDP (RS232 To UDP) ............................................................................................................ 67
5.4 ASCII control command ................................................................................................................................. 68
6. FAQs ....................................................................... 73
Appendix A: Module ID Distribution ..................................... 74
Appendix B: Module Parameter Types (1) ............................. 75
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1. Technology Overview
1.1 Introduction to Technology
DSP Series are equipped with several core technical features to facilitate the work of audio
engineers. DSP-based remote audio hardware is routed, processed and controlled via computer.
This Manual mainly introduces the techniques used to achieve the goal.
DSP Controller is a Windows-based application, which is used to conduct configuration and
control on DSP hardware. DSP Controller has 16 built-in presets, and the modules and sequences
for each preset can be flexibly designed in accordance with the designer's requirements. After
the design is finished, it can be saved for future use. The sequences and parameters of DSP
Controller's built-in processing modules accord to most of the application scenarios without any
change.
DSP Controller is a full-featured application including the parameter adjustment and
peripheral accessory settings of all modules, such as RS232, RS485, click-and-drag panel
configuration and Dante network audio control etc. The most interesting part is the user interface,
which allows the engineer to customize user interface so that the integrator can edit it or the
onsite technicians or end users who have no idea of relevant techniques can operate it. Superior
safety functions make it possible for the end users to access to the controls allowed by the
engineer or designer.
1.2 Audio Input Section
DSP supports up to 16 fixed analog audio inputs, which can be connected via removable
balanced phoenix connectors. The analog input section supports microphone or line-level signals
whose nominal levels are 0dBu, 10dBu, 20dBu, 30dBu, 40dBu and 43dBu. +48VDC phantom
power can be adopted for each input.
Preamp gain and phantom power can be conveniently controlled via DSP Controller.
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A
/D Technical Indicators:
Sampling rate: 48kHz
THD+N:≤0.001%@4dBu（A-weighting）
Dynamic range: 110dB
Audio format: 24Bit MSB TDM
Operating Manual for Digital Sound Processor (DSP) V3.0
1.3 Audio Output Section
Phase 1 of analog output section refers to D/A converter (DAC). DSP adopts advanced 24-
bit 256X sampling converter. Just like A/D converter, it also uses multi-bit architecture for
broader dynamic range; in the meantime, it is also equipped with excellent distortion just like
regular digital unit analog converter. Unit gain (0dB) is set via volume control, and the analog
output section is corrected as +4dBu with 20dB headroom. That is to say, 0dBFS digital signal is
equivalent to +24dBu output signal. If other signal levels are required, you may change the
volume to achieve it.
D/A Technical Indicators:
Sampling rate: 48kHz
THD+N:≤0.001%@4dBu（A-weighting）
Dynamic range (A-weighted): 112dB
Audio format: 24Bit MSB TDM
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1.4 Float Point DSP
DSP device adopts analog device SHARC DSP, enabling 32-bit and 40-bit floating-point
processing, which can be compared to 40-bit floating-point processing of other devices.
Floating-point processing provides prominent advantages for the users in terms of sound quality
and usability.
Fixed-Point Processing Limitations
Fixed-point processing has its own disadvantages. If there is a significant change in gain,
data loss or more severe situation may occur, including clipping distortion. For example, for the
processing of 24-bit fixed point-based audio signal, in some cases, if you attenuate the signal to
42dB, the new signal only includes 17-bit information. Due to gain attenuation, 7-bit information
will get lost forever. More worse is the clipping distortion. For a signal nearly close to 0dBFS, the
signal will be clipped at 0dBFS and the audio distortion will occur. Even if the signal level is
adjusted to below 0dBFS through post-regulation, the clipping has occurred, and the distortion
still exists. Fixed point processing can help to create some headroom above 0dBFS. By doing so,
some bits have to be abandoned. For example, if a 12dB (2 bits) headroom is created, a 24-bit
system actually only has 22 bits.
Floating-Point Processing
In the contrary, by taking advantage of floating-point processing, no matter what the signal
level is, all available bits are uniformly distributed to the signals. Basically, the floating points use
some bits as indexes to set up general signal level and distribute the remaining bits to signals
with independently stored level. As a result, no matter what kind of level (from -200dB and 200dB
below to 0dBFS above, the stored signal's accuracy is optimized without clipping distortion.
SHARC provides 32-bit and 40-bit accurate processing; through 32-bit processing, 25 bits are
distributed to storage signals no matter what its signal level is. This means that, based on at least
1-bit low level signal, its accuracy is always significantly superior to 24-bit fixed point processing.
Through expanded 40-bit accurate processing, 33-bit storage signal can be achieved.
Practical Significance
What's the practical significance of floating point processing for the users? The gain stages
between multiple modules can be ignored. If the signal level of a module is reduced by 50dB
and is then restored to its original value through another processing, data loss will not occur. In
the fixed-point system, the users must check other signal levels before sending it to A/D
converter because all digital-to-analog converters adopt fixed points. In DSP system, if you
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notice that your signal has been clipped before it is outputted and transmitted to the digital-to-
analog converter, you may close it off immediately at the output section to correct the situation.
By using the fixed point system, you have to search each processing module to find the clipping
source.
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1.5 Audio Flow
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1.6 Typical System Application
Conference Amplifying System: The processor can be connected to capacitor microphone, local
output amplifier and speaker so as to decorate and process with the signals via the speaker on
output channel. The output signal can also transmitted to the recording device with Dante interface
via such interface, such as computers with Dante virtual soundcard. By adopting a simple control
panel, volume adjustment, scenario call and other functions can be achieved via UDP control
processor.
Dante Application: Dante network breaks through space limitations, and has a wide range of
application scenarios. It can connect all devices that support Dante protocol to the same local area
network. By cascading 2 DSPs, it supports the access of up to 32 analog microphones and 32 Dante
microphones. Each processor will conduct first-level auto mixing on its own microphones, and the
second-level auto mixing is then made on the mixed signal via the auto mixer of main processor so
that more microphones can be connected to build a larger auto mixing matrix. The two-level auto
mixing working mode can satisfy general application scenarios.
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2. Hardware
1 Safety Instructions
2.
Safety Instructions
Important safety instructions:
. R
1
ead these instructions.
2. Keep these instructions well.
3
ay attention to all warnings.
. P
4
ollow all instructions.
. F
Operating Manual for Digital Sound Processor (DSP) V3.0
. P
5
lease 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.
. P
6
lease 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
lease do not install any heat source, such as radiator, heat register, furnace or other devices
. P
(including amplifiers) that generate heat.
. P
9
lease 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.
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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, the liquid will spill over 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.
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.
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U
nbalanced Connection
RCA interface and 1/4-inch TS interface are unbalanced interfaces. A multi-stand shielding conductor
may be installed on both ends of unbalanced connection.
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2.3 Specifications
Take 16x16 as an example (including Dante).
Processor ADI SHARC 21489(x2)
Sampling rate/Digitalizing bit 48K/24bit
Input Gain 0/3/6/9/12/15/18/21/24/27/30/33/36/39/42/45/
48 dBu
Phantom Power 48V
Frequency Response (20~20KHz) ±0.2dB
Maximum Level +18dBu
THD + Noise 0.001%@4dBu
Dynamic Range 110 dB
Background Noise (A-weighted) -91 Db
Common Mode Rejection Ratio @60Hz 80dB
Channel Isolation @1KHz 108 dB
Input Impedance (Balanced Connection) 9.4KΩ
Input Impedance (Balanced Connection) 102Ω
System Delay <3ms
Working Power AC110~240V 5Hz-60Hz
Maximum Power Consumption <40W
Dimensions (Width x Depth x Height) 482 x 260 x 45mm
Shipping Weight 3KG
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2.4 Mechanical Data
Space required:
1U (width * depth * height: 18.91″ x 9.5″x 1.72″ / 48.02 cm x 24.13 cm x 4.37 cm). The reserved head
space is not included in the depth.
An additional space at least 3 inches shall be reserved for the connection of rear panel. The
reserved depth depends on the wire used and the connection mode.
Electrical Property:
Maximum universal input power: 110-240 VAC, 50/60 Hz, 40 W.
Ventilation:
The recommended highest operating ambient temperature is 30℃ / 86℉.
Make sure that there is no blockage on both sides (a gap (at least 5.08cm, 2 inches) shall be
reserved). Please do not cover the thermovent of the device with newspapers, table cloth, curtain
and other objects.
Shipping Weight:
6.6 lbs. (3 kg)
2.5 Front Panel
ower: LED power indicator;
P
STATUS: The operation status indicator of the device;
USB AUDIO: USB soundcard (1-in-1-out), which can be used to achieve recording function.
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2.6 Rear Panel
① POWER power interface: connect to 110V- 240V AC power supply, and the rocker switch
controls the power supply of the processor.
② ETHERNET network control interface: By connecting this network port, the client computer
can debug and monitor the equipment.
③ Dante network interface: used to connect Dante audio network.
④ RS232+RS485 interface: connect to the control terminal or central control equipment.
⑤ GPIO interface: Eight definable input/output logic level control ports, driven by the contro
mo
dule in the device design. It can be set in the software to meet the output level (0V or 5V)
after internal conditions trigger; Or input level triggering (0V or 5V) to control corresponding
internal functions.
⑥ OUTPUT analog signal output interface: can be connected to power amplifiers, active
speakers and other equipment.
NPUT analog signal input interface: can connect microphone, DVD and other devices.
I
⑦
l
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3. Software
.1 Software Installation
3
A Windows PC with a 1 GHz or higher processor and:
Windows 7 or later.
1 GB of free storage space.
1024 x 768 resolution.
24-bit or higher color.
2 GB or more of memory.
Network (Ethernet) interface.
CAT5 cable or existing Ethernet network
The audio processor has built-i
the IP address of the audio processor. Enter the IP address of the device in the browser address bar
to access the audio processor, find the download link and download the installation software to the
local to complete the installation. The factory default IP address of the device is: 169.254.10.227
Subnet mask: 255.255.0.0, please add the address of this network segment in the PC first, so that
the device can access normally. After the device is started, enter "http:/ /169.254.10.227/".
n control software, which can be downloaded quickly by accessing
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Before installing the PC software, please make sure that a newer version of Microsoft .Net Framework has
been installed on the PC.
ote: If the download and installation of the software fails, please use IE or Google Chrome to try, or enter
N
the browser settings and cancel the browser download pop-up prompt to try again.
3.2 Using the Software
After enabling the software, the main menu is shown as below:
Cl
ick the button in the right corner of the main menu, and find all processors on the
network automatically. The user may connect to the designated processor based on their own needs;
after the connection, the indicator will flash, and one processor supports simultaneous connection
and control of up to four users.
：
restore default scene
：
factory reset
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3.3 Custom edit processing module
Click on the edit button, input or output channel processor module right-click selection, edit dialog
box, can replace the current processing module, can delete, copy and other operations, edit good
click upload to the host. Note: when the CPU display more than 100 will turn red, this time the
resource can not be uploaded to the host, need to be re-edited.
3.4 Audio Module Parameters
There are two regulation modes for module parameters: firstly, directly click the input or output
channel modules, and enter the parameter interface of the module; secondly, right click the module
and the configuration interface will pop out. The first regulation mode is adopted for the following
module parameters.
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3.4.1 Input Source
Sensitivity: Microphone gain, six grades for your choice
(0/3/6/9/12/15/18/21/24/27/30/33/36/39/42/45/48 dBu).
Phantom Power: Provide power for external capacitor microphone, click the button if needed. Do
not enable phantom power during line input or when the power is not required, so as not to damage
the external device.
Sine Wave: Drag the frequency to generate sine wave with designated frequency (20~20 kHz). You
may regulate the output level (unit: dBFS) based on your own needs. Use a fader to adjust or click
the text field to designate a value.
White Noise: Observe it on the frequency spectrograph with constant bandwidth, which has a flat
frequency spectrum. At this time, frequency regulation will fail, and the level can be used. Each
frequency component of white noise has equivalent energy.
Pink Noise: The frequency component powers of pink noise are mainly distributed in the middle
and low frequency bands. It decreases with a speed of 3dB/Oct in the middle and low frequency
bands. At this time, frequency regulation will fail, and the level can be used.
In addition, you may also find the following menu by right clicking each fader on the main menu.
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roup Setting: Open the group setting interface swiftly.
G
Minimum and Maximum Gains: Limit the maximum and minimum of the gain of a channel. After it
is commissioned, if you do not wish that the system's stability is affected due to external factors, you
may set up a maximum gain.
3.4.2 Expander
The expander has a totally different principle from the compressor. It can expand the dynamic range
of a signal. The most fundamental difference in these two devices lies in that, the compressor works
on the signal higher than the threshold, while the expander works on the signal lower than the
threshold. The expander can turn small signal into smaller one. It can be seen from Fig.3.2 that, when
the expansion ratio reaches 1:2, the input signal 20dB lower than the threshold will generate an
output signal 40dB lower than the threshold. Thus, as shown below, the signal lower than the
threshold will extend downwards and cause smaller level. When an expansion ratio 1:20 is adopted.
The expander seems to be a noise gate in terms of the transmission features. In fact, a noise gate is
an expander with great expansion ratio.
ig.3.2 Expander
F
The expander has the following control parameters:
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Threshold: The expander can be started only when the signal exceeds this threshold (allowing the
transmission of the signal). As a matter of fact, the signal is often set as the ambient noise.
Ratio: It refers to the slope below the threshold point on the gain curve. When the slope is set at a
high level, the gate movement will start.
Starting time: It refers to the time required to start the expander when the duration of the input
signal exceeds the threshold. Shorter starting time allows us to start the expander more quickly.
Release time: It refers to the time required for the gain to be restored to a value lower than the
threshold when the input signal is lower than the threshold.
No matter the starting time or the release time, it just helps to reduce the changing speed of gain
attenuation. That is to say, the speed of the gain from -40dB to 0dB is slowed down due to the
influence of starting time. The starting time or release time is unrelated to the threshold. If the signal
changes below the threshold, the starting time and the release time will have their own respective
influence on gain attenuation; when the signal level rises above the threshold, the gain attenuation
produced by the expander will disappear in accordance with the speed controlled by the starting
time. When the gain attenuation is reduced to 0dB, the expander will stop expansion. Later, when the
signal reduces to below the threshold, the expander will start again and the release time will begin
to work.
3.4.3 Compressor & Limiter
Compressor
The compressor is used to reduce the dynamic range of the signal higher than the threshold set by
the users, and to maintain the dynamic range of the signal lower than the threshold. The compressor
has the following control parameters:
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Threshold: When the signal level is higher than the threshold, the compressor/limiter begins to
reduce the gain. Any signal that exceeds the threshold is regarded as overshoot signal, and its level
will be reduced in normal cases. To a greater extent the signal exceeds the threshold, more level is
attenuated.
Ratio: It refers to the compression ratio. The ratio decides the attenuation degree of the overshoot
signal to the threshold level. The smaller the compression ratio is, more easily the signal will be higher
than the threshold. Once the signal exceeds the threshold, the compression ratio decides the ratio of
input signal variation to output signal variation. For example, when the compression ratio is 1:2, if the
input signal is 2dB higher than the threshold, the exceeding part only changes by 1dB. A compression
ratio of 1:1 suggests that the compressor does not attenuate the signal in proportion. The adjustable
range of compression ratio is 1-20.
Starting Time & Release Time: In order to maintain natural oscillation, it is generally hoped that part
of the most original level will pass through the compression without any influence (or just minor
influence). Likewise, if there is a rapid sharp attenuation and rapid recovery in the signal gain, the
suction effect will occur. The starting and release time of the compressor is to avoid such circumstance.
The starting time can decide the speed of gain attenuation, while the release time decides the speed
of gain recovery.
Output Gain: It is also called gain compensation fader. If the compressor significantly reduces the
level of the signal, it may need to enhance the output gain to maintain the volume. Such enhancement
applies to all parts of the signal and is unrelated to other parameter settings of the compressor.
G.R. and output Level Meter: G.R. indicates the compressor's compression amount; output refers to
the output level of the signal that has passed through the compressor module. The compression
amount is displayed in an inverse level meter. If the input signal and threshold are set as -6dB and -
30dB, respectively, and the ratio is 2:1, then the compression amount is 12dB; G.R. level meter
indicates around -12dB and output indicates around -18dB.
Limiter
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he limiter only has one key task: make sure that the signal will not exceed the threshold level in any
T
case. By adjusting the compressor's control parameters, its working modes will be very similar to
those of the limiter. The core working principle of a limiter is that it really focuses on the signal below
the threshold level as well as how the gain attenuation is produced before the occurrence of
overshoot signal. The limit period consists of two processing stages: during the first stage, there is a
minor limit, but the overshoot signal will not be processed; during the second stage, if there is
overshoot signal, it will attenuate in a very fierce way.
The limiter only provides two parameters: Threshold and Release Time. In terms of signal processing,
occasional clipping will be solved via limiter, while the signal level shall be attenuated in terms of
frequent clipping.
3.4.4 Auto Gain Control
Auto gain control (AGC) is an exception of compressor. Its threshold is set at a very low level with
middle-to-slow starting time, long release time and low ratio. The purpose is to improve the signal
with uncertain level to a target level, while maintaining the dynamic range at the same time. Most of
the auto gain control includes silent detection to prevent the gain attenuation loss during the silent
period. This is the only function that distinguishes auto gain control from ordinary compressor/limiter.
Auto gain control may be adopted to normalize the level of CD players that play background
music, frontground music and music on hold, so as to eliminate the changes in the level of some
paging microphones.
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Auto gain control includes the following control parameters and switches:
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 is set as the background noise just higher than the level of
input signal.
Ratio: It refers to the ratio of the changes in the level of the input signal higher than the threshold to
the changes in the level of the output signal.
Target Threshold: It refers to the level of output signal required. If the signal is higher than the
threshold, the controller will compress the signal in proportion.
Starting Time: It refers to the response time required to control the level higher than the threshold.
Release Time: It refers to the response time required to control the level lower than the threshold.
3.4.5 Equalizers
The equalizer is mainly used to correct the frequency range that is overemphasized or gets lost, no
matter it is wide or narrow. In addition, the equalizer can also help us to narrow or widen the
frequency range, or change the amount of a component in the frequency spectrum. To get it
simplified, the equalizer can be used to change the tone of the signal.
The equalizer has the following control parameters:
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ig.3.6 Equalizer
F
Type: Parametric EQ is a default. High & low shelf filters and high & low pass filters can be selected.
Each kind of filter has different forms to achieve different functions.
High & Low Pass Filter: The reference frequency of pass-type filter is called cut-off frequency. Pass-
type filter allows the frequencies on one side of the cut-off frequency to fully pass the filter; in the
meantime, the frequencies on another side of the cut-off frequency are attenuated on a consecutive
basis. Among them, high pass filter allows the frequencies above the cut-off frequency to pass and
filter the frequencies below the cut-off frequency. To the contrary, low pass filter allows the
frequencies below the cut-off frequency to pass and also filter the frequencies above the cut-off
frequency.
High & Low Shelf Filter: It is also called shelf filter. High shelf filter means that the gain enhances or
attenuates for the frequencies above the set frequency. Low shelf filter means that the gain enhances
or attenuates for the frequencies below the set frequency. The set frequency is not 3dB cut-off
frequency but refers to the center of the failing edge or rising edge of the filter. Q value affects the
peak, and has a mathematical relation with the peak.
Frequency (Hz): It refers to the center frequency of the filter.
Gain (dB): It refers to the enhanced or attenuated decibel value of the gain at the center frequency.
Q: It refers to the quality factor of a filter. The adjustable range of Q value is 0.02-50;
When the filter is 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, and the roll-off will occur in advance.
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Each segment of equalizer has a switch, which is used to turn on or turn off the corresponding
segment. After being closed, the parameter setting will not work. The equalizer has a master switch,
which is used to enable or disenable a module.
Parametric equilibria have 5,8,12 options
3.4.6 Figure Balancer
By using constant Q value technology, each frequency point is equipped with a push-pull
potentiometer. The bandwidth of the filter remains unchanged regardless of the frequency
raised or attenuated. The common professional graphic equalizer is to divide 20 Hz~20 kHz
signals into 10,15,27,31 to adjust.
The graphical balance currently has 10,15 and 31 options
3.4.7 Feedback Inhibition
While using the feedback inhibition module, we'd better combine well with good systematic
designs and practical projects but not to replace good systematic designs. The traditional methods
shall still be used, such as limiting the number of microphones to be opened, minimizing the distance
between sound source and microphone, positioning the microphone and loudspeaker to get
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minimum feedback, and balancing the room to get flat response. Later, we can adopt feedback
inhibitor to get additional gain. The feedback inhibitor 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.
Feedback inhibition module automatically detects and inhibits the sound feedback in the sound
system. The module distinguishes feedback from expected sounds based on the characteristics of
signals. When a feedback at a certain frequency is detected, a notching filter will be automatically
added at the feedback point 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 the buzzer output, the filter may be locked up to prevent any change during the
performance period. The filter settings can be copied to a dedicated notching filter module (such as
equalizer). Eight filters are set as auto filters in an automatic cycle. In this way, those filters for
temporary use can be removed.
Each channel has a feedback inhibition. Use a mouse to drag the input module and find the
feedback inhibition module or rapidly enter the feedback inhibition module by clicking the shortcut
key on the right. If the feedback inhibition module needs to be enabled, click to enable the button
and automatically detect the feedback point, and use a narrow-band filter for elimination. Each
feedback inhibition module has 8 narrow-band filters.
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The feedback inhibition module has the following adjustable parameters:
Panic Threshold: According to this parameter, "any level 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 disenabled.
Feedback Threshold: According to this parameter, "any level lower than the threshold is absolutely
not a feedback". This may prevent the module detects feedback in a soft music or due to the noise
of low level.
Filter Depth: It refers to the maximum attenuation of a single filter. A shallow setting may prevent
too much damage caused by the filter or notching filter to the signal. It may cause worse feedback
control, especially in a large narrow resonance system.
Bandwidth: 1/10 and 1/5Oct can be chosen. A constant Q value is adopted. The filter will not
become wider due to the increase of depth. It is recommended to use the filter in the phonetic
environment. In the case of frequent feedback, the bandwidth is set at 1/5Oct because it has wider
bandwidth and greater influence.
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.
Notching Filter's Auto Mode: Auto mode is set for the notching filter. After eight filters are used
up, a new feedback is detected, and the module will check "auto" filter and use it to inhibit new
feedback. Each notching filter has three modes: auto, manual and fixed modes. When Manual mode
is set for the filter, 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. If you need to
save these feedback parameters, please click to save the preset button.
Clear: Click the button to instantly clear up all filters. It will clear up all feedback points detected
with inhibition previously. This operation is generally done when recommissioning the feedback
module.
Feedback inhibitor 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
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(b) Set up parameters for the feedback inhibition 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.
A
t 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 (such as equalizer). In this way, you may add more filter capacity.
If a speaker is included among the devices used, it is recommended to use a compressor/limiter
module for additional protection. You may set an appropriate limiter to make sure that the speaker
will not get damaged even if all notching filters are used up or the feedback inhibitor cannot
effectively control the feedback, such as in the case of excessive system gain.
3.4.8 Noise Gate
Noise gate: the main purpose of a noise gate is to attenuate signals below the threshold, and this
attenuated signal is usually noise.
Threshold: greater than pass, less than attenuation
Depth: attenuation, which determines how much the signal below the threshold is attenuated.
Start time: start time refers to the speed at which the noise door opens.
Release time: the release time is opposite to the start time, which refers to the speed of the noise
door closing.
Keep time: keep time is like this, when the signal is below the closing threshold, normally, the
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3.4.9 Ducker
Operating Manual for Digital Sound Processor (DSP) V3.0
Ducker: when the level of one channel exceeds the specified threshold, the level of the other
channel will be attenuated, which is the dodging effect.
Threshold: the reference signal begins to decay above the threshold and recovers below the
threshold.
Depth: the amount reduced by the evasive signal.
Start time: when the reference signal is above the threshold, the time to begin to attenuate the
evasive channel signal.
Recovery time: after the reference signal is below the threshold, the evasive signal returns to the
original signal size.
Hold time: hold time refers to how long the dodging on the dodging channel remains after the
control signal is below the threshold.
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3.4.10 Spl
Noise gain compensator: automatically adjust output volume according to ambient noise
induction and processing.
Maximum gain: the maximum amount that can be adjusted.
Minimum gain: the minimum amount that can be adjusted.
Gain sensing ratio: the ratio of lifting or attenuation.
Speed: the speed of lifting or attenuation.
Trim： gain.
Noise threshold: greater than the starting gain, less than the reduction.
Distance: distance between reference and local signals
3.4.11 AutoMixer
In a conference room, if several microphones are opened to the same gain level and there is only one
person speaking, the microphone effect may be not clear. Other microphones will pick up noise and
reverb in the room. When these signals are mixed with normal microphone signals, audio output
quality will be greatly reduced and the whole amplifying system will easily get squealing, which will
result in sound transmission gain. To solve the issue, other unused microphones shall be closed.
Automixer may close them at a higher speed than manual operation.
There is a built-in gain share automixer inside the processor. It supports up to 32-channel audio signal
output. There is a direct output at each channel of auto mixing matrix, which is only influenced by
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channel mute instead of auto gain and channel fader. Channels suitable for fixed volume like the
channel of background music need to keep at a fixed level without being controlled by automix. For
example, it shall keep microphone at normal open. Meanwhile, its gain will not be influenced by
automix. At this point, users may directly adjust the output of the channel in output matrix route as
well as turn off automix button of the channel. Its gain will not be adjusted and gains at other channels
will not be influenced by the signal level at the channel.
There are two groups of control parameters in automix module: main control parameters and channel
control parameters.
(1) Main control parameters
ick the bottom button to open or close automix
Cl
Gain: control main output volume of aumomix
Slope: The slope control influences the attenuation of lower level. If the slope
is higher, the attenuation of lower level channel will rise. The slope control and
the ratio control at the expander have the same working mode. It is suggested
that the value be set at or around 2.0. if it is set at 1.0, the effect equals to
closing automix at all channels. if it is set at 3.0, the action will result in larger
gain adjustment, which may bring unnatural effect. The bigger the value, the
more the channel is opened and the more the total attenuation. When the
slope is set at 2.0, ideal gain share may be realized, so it is the preferred value
in use.
Response Time: Faster response time may ensure head letters of speeches
not be cut off. Slower response time allows smooth operation. Practices show
that the best effect will be produced when response time is between 100ms
and 1000ms. The design of auto gains aims to faster microphones turning on
than turning off. Therefore, head letters of speeches will not be cut off even
the response time is 100ms. If it is set at several seconds, then there will be a
longer hold time of the response time of automixer, previous active channel
will be saved at open status in several seconds.
(2) Channel control parameters
Auto Mix: Each channel has an automix on/off button which must be
turned on for channels need to participate in automix. It may also be
closed, and then the channel will not participate in automix.
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Gain fader adjustment may increase/decrease the volume
riority setting may give priority treatment to high priority
M
ute: Both channel mute and fader are behind auto gain. If the channel
level is bigger, the level gain of other channels may also be reduced
even the channel mute is on.
Gain:
proportion in automix.
Priority: P
channels than low priority channels, and thus automix algorithm will be
affected. Priority parameter ranges from 0 to 10. The bigger the value,
the higher the priority.
Both channel mute and fader are behind auto gain. Any adjustment made towards these two
parameters won’t influence the operation of automix. For example, If the channel level is bigger,
the level gain of other channels may also be reduced even the channel mute is on. Channel mute
shall be turned on and automix shall be turned off to set signal mute and prevent its influence on
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. Click the
textbox and input a dB value to control channel level precisely.
Priority control allows high priority channels to cover low priority channels, and thus 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. Value increasing means priority increasing.
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 a 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 at 5.
Note: In some settings, users shall be very careful when using ultra priority differences between
channels, such as priority of 0 and 10. If channels with ultra high priority recognize signals like
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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. If the issue happened during
installation and commissioning, users may consider installing a noise gate or expander between
automixers at the highest priority channels. Meanwhile, they shall set threshold at the level that it
won’t be opened by the noise gate or expander.
Also add a threshold type automatic mixer option:
The threshold automatic mixer is based on the noise gate development, each channel has a noise
gate, which is either open or closed.
Gain: the overall output gain of the automatic mixer.
Hold time: the Hold function can adjust the length of time the channel remains open after the
speaker stops speaking. This function ensures that a brief pause between words and phrases does
not cause the channel to close.
off gain： When the sound cannot pass through the threshold, the set value of off gain will be added
to the sound signal.
Sensitivity: sensitivity control, when the channel above the adaptive noise threshold must be
conducted on the channel. If the value of this setting is increased, the channel is easily switched on
at very quiet noise.
NOM Atten：NOM attenuation push can affect the magnitude of attenuation of all channels when
more microphones are opened. This parameter reflects the attenuation when two microphones are
switched on, and the additional attenuation that will increase after each double number of open
microphones. When set to 3 dB, for example, opening two microphones attenuates 3 dB,6 when
opening four dB,9 when opening eight By the same token, the number of microphones turned on
doubles, and the output attenuation increases by 3 dB.
NOM Limit： set the maximum number of open microphones
Noise threshold: greater than this value open, less than this value attenuation
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3.4.12 Echo Canceller
Acoustic echo canceller (AEC for short) is a kind of digital audio signal processing technology. It is
used in audio/video conference when conferees in local conference room are talking with one or
more speakers certain distance away. AEC program increases remote speaker’s phonetic
intelligibility via cancelling acoustic echo generated in local room.
Echo cancellation module for remote calls can be used to carry out local amplification of remote voice
signals and attenuate the interference caused by acoustic echo. Its basic operation principle is
simulating echo channel, estimating possible echo generated by remote signals and then minus the
estimated signal from input signal of microphones, and thus there will be no echo generated in input
voice signal to achieve the goal of cancelling echo.
There is only an echo cancellation module in DSP Controller. Two local input and remote output
mixers are preset to realize multichannel signal participating echo cancellation as shown in the figure.
A parameter can be adjusted:
Non-linear filter (NLP): Three types including Conservative, Moderate and Aggressive can be selected
to determine echo suppression level.
te: The settings of echo cancellation module shall be used cooperatively with matrix module
No
setting signal router.
3.4.13 Noise Suppression
Noise suppression module can remove non-human voice effectively. It may distinguish human voice
from non-human voice and treat the latter as noise. After its processing, only human voice is
theoretically left for an audio file consisting both human voice and noise.
There is only an echo cancellation module in DSP Controller. Multichannel mixers are preset to realize
multichannel participating noise cancellation as shown in the figure.
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y three
ggressive(15dB) for selection. dB
here refers to the decibel of suppression
more harm on voice generated, which is
Suppression level: There are totall
levels including Mild(6dB)，Medium(10dB)
and A
noise reduction. The bigger the value, the
inevitable.
3.4.14 Matrix
Matrix has dual operation functions including router and sound mixing. As shown in the figure, the
horizontal direction indicates the input channel and the vertical direction indicates output channel.
One-to-one input and output is the default setting. If it’s necessary to mix voices of channel 1 and
channel 2 and then output to channel 1, users only need to click 1s on both horizontal and vertical
directions at output channel 1. If input 1 and 2 participate in automixing, then the output will not be
influenced by it. Similarly, after setting automixing, echo cancellation and noise suppression module,
users also need to set matrix to get correct signal route relation.
3.4.15 High & Low Pass Filter
Each output channel provides high- and low-pass modules which consist of high- and low-pass filters.
Each filter has four kinds of parameters as follows:
Frequency: The cutoff frequency of filters. The cutoff frequency of Bessel and Butterworth is defined
at -3 dB, and the cutoff frequency of Linkwitz-riley is defined at -6dB.
Gain: Gain setting influences the promotion and attenuation of full band.
Type: There are three types of filters including Bessel, Butterworth and Linkwitz-riley. Butterworth has
the flattest passband.
Slope: It refers to attenuation values of transition zone of filters. There are totally 8 attenuation values
including 6, 12, 18, 24, 30, 36, 42 and 48dB/Oct. For example, 24dB/Oct indicates that the attenuation
range is 24dB for each octave difference existed in frequency in transition zone.
Users may click bottom activate button to activate high- or low-pass module.
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le in modules
.4.16 Delayer
3
3
.4.17 Output
Activate button: Activate appointed delay modu
and insert it into audio signal path to increase fixed delay time for
signals.
Millisecond: Set delay time of delayer. The value ranges from 1 to
1, 200 milliseconds. Both meter and feet are alternate units for
millisecond.
I
nvert: 180-degree audio signal phase invert.
Mute: Set mute/unmute.
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Similarly, users can use right button to set part menu at output channels, which can be carried out
based on requirements.
.4.18 Camera Tracking
3
Voice tracking set tracking threshold: detect microphone input signal greater than or equal to
tracking threshold, the system automatically enable tracking parameters
Default Mike: when all Macs are not input, turn the camera to the default MIC set position reaction
time: the maximum intermittent time of the valid signal. If the microphone is used to speak, the
reaction time is set to 3 seconds, and the signal is still considered to be valid for 3 S in the middle of
the speech, and more than 3 S, is regarded as invalid. Cut back the default Mike time: the shortest
speaking time required for the camera to switch to a valid position. If the microphone is used to
speak, the speaking time must be longer than the "switching time ", the channel signal is considered
to be valid, and then the camera automatically turns to the set position. Usually "switching time" is
greater than "reaction time ". Wheel gap: the number of times each time a camera switch command
is sent: the number of times a camera switch command is sent, such as 0 for special processing, only
once triggered.
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ike sets the microphone number: generally corresponding to the input channel of the device, that
M
is, the channel number connected by the Mike. Priority: the smaller the priority series, the higher the
priority level, when the priority is the same, according to the trigger priority order; if two Macs speak
at the same time, The camera automatically rotates to the preset bit corresponding to a Mike with a
small priority (that is, high priority) or sends a command corresponding to a Mike with a small priority
(that is, high priority); but if the two Macs have the same priority, the first checked signal shall prevail.
preset point, serial port number, camera address, protocol and camera related, must correspond to
the actual connection of the camera. Enable: when the device detects an input signal (usually when
someone speaks), it automatically sends the corresponding command to the defined serial port, and
then it can set the command in advance. But without checking enable Custom Command, the device
does not send automatically, but can still click the send button to send input box command to the
specified serial port. Click Save to save the parameter to the device, when the channel's Mike is
associated with the corresponding camera address. Then the "enable microphone setting" option is
used to determine whether the microphone setting is a camera debugging interface when tracking
is enabled. Finally, the parameters of this part will be saved on the camera.
3.4.19 USB Soundcard
USB soundcard is used to carry out two functions, realizing recording and broadcasting and
teleconferences using personal computers. After going through echo and noise cancellation modules,
USB voice may access to teleconferences easily. USB broadcasting function on the software interface
can only be used for recording and broadcasting function.
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ong playing information, double click to
① S
enter playlist
② Next song
③ Pause
④ Song volume adjustment
⑤ Play
⑥ Prev. song
⑦ Sound recording list
⑧ Sound recording volume adjustment
⑨ Stop recording
⑩ Start recording
Soundcard Setting
USB cable with double ends of Type-A can be used to connect DSP processor and computer host.
For initial connection, “Found New Hardwar” will pop up on computer screen, and the driver will
be installed automatically. After installation, USB soundcard will appear in computer soundcard list
as shown in the figure. Users may select USB soundcard in soundcard setting at software playlist.
Users may operate song files in playlist as well as save them as playlist. They may also directly open
them when using the device next time. As shown in the figure, click at the bottom of the playlist
to open file folder and select songs to be played, to clear the playlist and to enter the interface
of soundcard setting.
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3.5 Setting Menu
3.5.1 File Menu
offline mode, click the pop up file dialog and open an existed default document with suffix *.uma,
In
or right click on the default document to open DSP.exe.
“Save as” refers to saving presets on the application to local hard risk to realize easy copy and
store.
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3.5.2 Device Setting
Operating Manual for Digital Sound Processor (DSP) V3.0
nformation like device name, network address and SERBAUD can be set on device setting. The
I
maximum length of the device name is 16 characters or 5 Chinese characters.
Default startup: Two startup preset modes are available for selection. One is any preset from 16
presets acting as startup preset. Each boot will start with it. Another is selecting previous upload
preset and taking the last preset before power outage as the preset of next startup.
3.5.3 GPIO Setting
Open the software interface of GPIO settings. The device has total 8 GPIOs that allow independent
input or output configuration.
Input GPIOs have preset, router, gain, mute, command, analog-to digital gain for selection.
Output GPIOs have preset, level, mute and command for selection
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Trigger types: high level trigger/low level
trigger/high level trigger, low level
cancellation/low level trigger, high level
cancellation, i.e., rising edge/falling edge
trigger/rising edge trigger, falling edge
reset: It will change to preset when jump
Input GPIO Setting
Preset
Routing Trigger types: same as above
cancellation/falling edge trigger, rising edge
cancellation
P
type of hardware GPIO port and trigger type
of software setting are consistent.
Input & output: select input channel mixing
corresponding to output channel.
Carry out mixing/cancel mixing action when
the trigger condition is satisfied.
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digital gain is very useful when
potentiometer analogs and adjusts
Gain Trigger types: same as above
Channels: select input or output channel
Step length: increase step length in dB based
on original gain got by the channel.
Mute/Cancel
Mute
Trigger types: same as above
Channels: select input or output channel
Command Trigger types: same as above
Command: The command code will be sent
through RS232 when the trigger condition is
satisfied
Analog-to-
digital Gain
Analog-to-
connecting potentiometer externally. It may
adjust input or output channel gain. It looks
Output GPIO Setting
46
like rotary encoder. The difference between
them is
voltage and current while encoder is digital
and transmit the binary codes of 0 and 1.
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reset: Corresponding GPIO port outputs
evel: GPIO outputs high/low level when
hannel: Appointed input/output channel.
Preset high/low level is output when the
channel is muted. On the contrary, the
Preset Output types: high level/low level
P
high level or low level when changing to it.
Level Output types: high level/low level
Channels: appointed input or output channel
L
appointed channel level reaches preset level
threshold.
Mute Output types: high level/low level
C
opposite level will be output when cancelling
mute.
3.5.4 Group Setting
The grouping interface is divided into two labels: input and output, and grouping can be set under
each label. A channel can only participate in one group. Under the same group, their channel volume
adjustment and mute adjustment are synchronized. The biggest difference from the Link function is
that other module parameters are not synchronized.
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maximum number of passes. The
maximum number of channels for the
device depends on the model you
Each group can select 1-device
purchased. The channel is set as a group,
which will be distinguished by a color on
the Home screen.
Th
e relationship between groups and link: The channel participating in group won’t participate in
LINK, which means group’s priority is higher than LINK. The difference between groups and LINK is
that groups can only control channel gain and mute while LINK links all parameters at the channel.
3.5.5 Panel Setting
Panel setting include two panel types which are button and OLED panels. Use cables to connect
multiple physical panels with DSP device via panel setting, and then achieve the aim of panel
controlling DSP device by setting panel easily.
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ffline device: It suits for offline edit status. Firstly, commissioning engineer configures panel
O
parameters locally, and then download it to online panel. It is no doubt that the panel can be edited
directly online. Drag offline device from the column of online panel to panel design area and then
double click to edit it.
Please note that there is a small circle on both panel and device. Click the circle and then draw a line,
select target device, and then the connection between two devices are established by this way.
Double click the panel in design area to enter panel configuration interface. The configuration of two
panels is going to be described below. After finishing configuration, click the toolbar download icon
to download the panel configuration to hardware.
OLED Screen:
OLED screen consists of a 1.3” OLED screen and a knob. OLED screen display strategy is classified
according to menu. There are totally three types of menus including menu, buttons and presets.
Double click a OLED screen in design area of the screen to enter its detailed setting as shown below.
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C
lick "add menu" to pop up menu selection box, choose corresponding menu and confirm it. After
finishing the setting of software menu configuration, click toolbar download icon to download the
configuration to panel hardware.
Operation steps of panel:
1.Display panel name and IP address on main interface and turn the knob left or right to switch menu.
2. Press the button on the knob, and the second row on menu interface starts to flash, which indicates
that it enters edit mode.
3. Turn the knob left or right to change value.
4. Press the button on the knob again to edit mode and go back to menu mode.
Key Panel:
There are 8 keys and one knob on key panel. The knob is used to adjust gains, and 8 keys can be
used to realize different functions through programming. There are four types of key functions,
including volume adjustment, mute, preset and command. Drag an item in function area to appointed
key to finish programming of the key.
Similarly, after finishing all programming, users may use the emulation button to check whether the
configuration is correct.
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nstructions for panel operation indicators
I
1. Key indicator stays on, which indicates the key is configured with mute function.
. Ke
2
y indicator keeps flashing, which indicate the key is configured with gain function. The configured
knob also uses and adjusts gain of the channel. 13 indicators around the knob indicate gains. They
are on or off with gain size. All 13 indicators off indicates gain of -72dB while all on indicates gain of
12dB.
3.A sudden flash happens when pressing the key indicator indicates the key is configured with preset
or command function.
Command function: The command data come from central control command. Please refer to section
5.
3.5.6 Dante Setting
Note: Before Dante setting, please check whether the computer network card has been inserted into
Dante network. If it is needed to set up a device lock or to check the signal indicators of Ultimo, virtual
soundcard and other devices, please use DanteController. On this software interface, you can only
check the signal indicator of Brooklyn.
Dante routing setting interface is similar to that of DanteController. It provides routing, channel
information, network setting and other information. Dante device's receiving channels are displayed
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on the left part of the device, while Dante device's sending channels are displayed on the right part
of the device.
On the routing interface, the small boxes at the intersection of sending and receiving channels
indicate that routing relationship can be created. Green icon will appear at the intersection of
matrix after single click. Users may see a grey icon for a very short time period at the very beginning,
which indicates that routing is in process.
Warning icon or error icon will appear if there is a routing problem. If several devices are
subscribed at the same time, yellow icon may appear temporarily.
Note: Route and locked devices can’t be created, but existed route can be deleted or replaced.
Cancel Audio Subscriptions
Users may click subscribed intersections to cancel audio subscriptions. Subscribing icons will be
removed and restore original small boxes.
Subscription Status
Pr
ocessing Subscription is in processing
Sub
scribed Connection established
52
W
arning Subscription is not processed normally because sending device is
invisible on network
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The device is processing subscription of other channels. In most
Error Send error, for example, no enough bandwidth on the network
Coming soon
cases, many channels are subscribed at once.
Users may view information like device’s IP address and version on device information.
Double click device name on routing interface to enter detailed settings of the device as shown in
below figure.
hannel names can be modified on sending and receiving labels. Channel naming rules are as follows:
C
The maximum total length for all DSP device names is 16 characters. The name length of devices
supported by Dante is as more as 31 characters. Therefore, please ensure that name length of Dante
device names and channel names is no more than 16 characters when routing by using the interface,
or DSP Controller will cut the process off, which will result in incorrect subscription.
Names are case-insensitive. “Guitar” and “guitar” are the same name.
Valid characters include A-Z，a-z，0-9 and‘-’.
Device names can't start or end with ‘-’.
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Device names shall also be unique on network.
Any characters can be used for naming sending channel labels, except '=', '.’and '@'.
Sending channel labels must be unique in the device.
Receiving channel naming has the same rules with sending channels.
Device Configuration:
Device configuration refers to device name modification, audio sample rate and delay. Device names
must be modified follow device name modification rules. Delay needs to be emphasized. In Dante
network, compensation is needed for various delays at receiving end. There is device setting delay
(the interface delay) at each receiving end. The delay refers to time difference between samples
received at receiving end and broadcasted. The default delay for Dante device is 1ms, which is enough
for large-scale networks.
However, automatic negotiation will be carried out at sending and receiving ends when establishing
connections, which ensures delay time is enough to prevent packet loss.
For example, Ultimo devices support minimum 1ms delay. If the delay for a faster device like PCIe
card is set at 0.25s and the device is establishing connection with an Ultimo device, then the delay of
subscription will be 1ms which is the minimum delay supported by subscriptions. If minimum delay
possibly reaches 1s in megabyte network, then subscription error may occurred when transmitting in
the condition that delay time no more than 1s.
Network Configuration:
Network configuration refers to network IP address, mask and gateway settings. Brooklyn supports
redundancy mode and exchange mode settings.
Redundancy Mode
Many Dante devices have two network ports named “Primary” and “Secondary”. “Primary”
port connects physical network. If “Secondary” port has been used, then “Secondary” port shall
connect another physical network. “Secondary” port can’t communicate with “Primary” port.
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ulticast Stream:
M
What is stream? Dante audio routing automatically creates streams. A stream moves audio data from
several channels from receiving ends to one or more receiving ends. Unicast streams are given to
single receiving devices while multicast streams are given to multiple receiving devices. Multicast
streams may be created or configured manually through the interface, but it uses network bandwidth
whether there is receiving device or not. Meanwhile, it doesn’t need extra bandwidth when more
receiving ends are added.
As shown in the figure, select multicast stream label page, check device channel, click create, and
then created multicast stream will display in the right list of the interface. It can also be deleted when
users don’t need it. A stream mostly include four channes by default. If more than four channels are
checked, they will be divided to several streams automatically.
3.5.7 Help Menu
(1) Central command
O
pen central control command window and click parameters to be controlled on interface, then the
window will display current command immediately. Copy the command and then use UDP or RS232
to send the command to devices.
(2) Device upgrade
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Device upgrade can be carried out through UDP. Connect the device, click setting-help-device
upgrade. A file choice box will be popped up at this point, and then choose processor upgrade file
(*.bin).
(3) About
Display version number, tech support contact information and copyright info., etc.
4. Dante Audio
.1 Dante Overview
4
Dante technology launched by Audinate provides high performance digital media network to meet
professional on-spot sound amplification, audio/video device installation and high sound quality and
high performance requirements of broadcasting and recording systems.
Dante aims to fully develop performance of nowadays and future network devices. It may provide
media transmission mechanism which can eliminate design limit of many traditional audio network.
Dante makes it easier to create a steady and flexible digital audio network but meanwhile it's
performance is almost not limited. Dante network can be designed with a network speed mixing Gbps
and 100Mbps as well as supports audios with different sampling rate and depth, and even allows
network area design with different delays.
Dante is based on internet protocol, not only Ethernet. Dante uses standard IP through Ethernet, so
it may run on cheap, existing computer network hardware. Meanwhile, by virtues of standard QoS,
Dante may share installed network with other data and computer flow.
Dante provides sample accurate synchronization as well as lower delay required by professional audio.
Dante network center is independent of similar synchronized audios and allows complete
synchronous broadcasting through different audio channels, devices and networks and even among
several switch hop counts.
Dante makes network a real plug-and-play process and allows automated device search and .
configuration. Compatible devices of Dante will automatically set their own network configuration
and notify themselves and channels on network. It simplifies complex and error-prone setup assembly
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programs and replace "magic numbers". Network devices and their input and output signals can be
renamed to make them easier to be understood.
Dante is not limited to allow audio channel configuration and transmission. It also provides
mechanism through its IP network to send or receive control and surveillance information, including
the unique information of the device and controls appointed and developed by specific device
manufacturers. Relying on its solid foundation and existing and evolving network standard, Dante
can provide technology with proactive level, otherwise, it cannot be used in other types of digital
audio transmission. From the beginning, Dante was designed for gigabit network. Moreover, Dante
existing today has included the emerging AVB network standard. The continuous evolution of its
network technology is an inseparable part of Dante's development.
Dante technology can be used for hardware and software to be installed and APIs referring to design
and development. For more details, please visit Audinate website www.audinate.com.
Features:
Based on current IP-based network technologies including IEEE 802.3 and UDP/IP.
Use existing Ethernet hardware.
Use standard VoIP (IP voice technology) type QoS (quality of services) to integrate existing networks.
Integrate Ethernet network speed, and improve it from 100Mbit to 1Gb.
DSP’s digital audio frequency is 24 bits and its sampling rate is 48kHz. Dante itself may deeply
integrate sampling rate and depth on the same network at the same time.
Dante of DSP devices supports network delay as low as 0.25 millisecond.
For device search, Plug and Play operation of ER. Automatic configuration.
Label-based routing. Renamable data stream.
The channel number of Dante included in the DSP processor is determined according to the type of
device purchased. There are many versions including 8x8 and 16x16.
Use Virtual Sound Card software downloaded from Audinate to connect PC or Mac to Dante network
directly.
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4.2 Dante Requirements
CAT6 cables are used for all inner connections of Dante.
If flow control is conducted in the same network, then 30% of available broadband shall be reserved.
When using the reservation method, a 100Mbit link may process channels as more as 48x48. A 1Gbps
link may process channels as more as 512x512 when sampling rate is 48kHz.
Daisy chain and uplink shall be Gbps.
Repeater not supported.
Commercial-type managed switches shall be used when over 10 local units or units over 100 meters
away connect. Switches must support following functions.
oS with four queues.
1. Q
2. DSCP with strict priority.
4.3 Dante Network Design
There are two network designs can be used for typical Dante network topology.
10 or less non-redundant units
For systems configured maximum 10 non-redundant units, please connect your pc to Ethernet port,
connect remaining Ethernet ports with daisy chains, and then connect Dante ports with same daisy
chains without specially using Dante switch or configuration. All units shall run under switched port
mode.
ore than 10 units or their distance between is over 100 meters
M
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For systems equipped with more than 10 units or theirs distance is over 100 meters, please connect
your pc or Ethernet ports on all units to one Ethernet switch and the Dante main port to the second
Ethernet switch. All units shall run under switch port mode.
D
elay
Dante network delay can be set through Dante Flow Manager on toolbar.
Network delay will increase as the number of switch increase. It’s best to have less than 2 switches
so as to realize a minimum delay. The system allows to use as more as ten switches, but delay will be
increased accordingly. Maximum actual delay value must be always used. Delay is not decisive in
many installations, for example, when sending audio to acoustic isolation chamber. In many cases,
the highest delay can be selected to reduce the overall network traffic and minimize audio output
decline on overload network. If low delay is the key, then numerical values consistent with the
quantity of switches of the whole network shall be selected. If these two units are connected with
daisy chain or single gigabit switch, then the delay of 0.25mS is normally safe. Otherwise, the delay
of 0.5 or 1.0mS can be selected according to your network topology.
In Dante, network delay changes may gain compensation at user receiving ends. Each receiver has a
Rx delay setting which is also in Dante settings of device information in DSP Controller. The setting
limits delays between timestamps of incoming audio samples and sample broadcasting time.
The default delay for most Dante devices is 1Ms, which is enough for a large-scale network. The
network consists of a gigabit network core (the hop count between switches is up to 10 times) and
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100M links connected to Dante devices. Smaller gigabit networks may use delay with lower value (as
low as 200 microseconds).
Note: Dante delay refers to system-level network delay setting, which means there are no extra delays
or differences among subsequent all switch hops, such as non-additional delays. As long as (system-
level network) delay settings have long enough time to go through the entire network path, data
packets may reach the end of chains before buffer ending. Then Dante uses its clock calibration
mechanism (PTP) to calibrate output accurately. Users may compare self-clocking between any two
Dante units. Their delays are all within 100Ns or less than 1% during sampling period (up to 10 units
have been proved). Therefore, multi-point output on daisy chains will align with samples. Each output
has a specific delay from each output.
Note 2: Dante’s actual delays are intervals of three sampling times or 0.06mS longer than displayed
time set by Dante delays of Composer, which is caused by DSP sampling butters and inevitable.
.4 Dante Modes
4
When Dante devices are still in exchange mode, issues like modes can’t be changed or audio loss
may occur if cables are used to connect devices and use them for redundancy mode. When devices
switch from redundancy mode to exchange mode, please use following steps and vice versa.
1
network. In other words, in condition that direct connection between external switches or two devices,
then users only connect it to main socket. In condition that there is no external switch and more tha
t
chain. Do not insert the last device to the first device to avoid circulation created between them.
2. Enter Setting ->Dante Setting -> NetworkConfig in DSP Controller. Properly select redundant
networks or switch ports.
4. Turn off system power
5
mode, then users now may connect main socket and secondary socket between two devices or singl
s
en Dante devices are in exchange mode instead of redundancy mode, please connect their
. Wh
wo devices are needed, then connect main socket to the secondary port of next device with daisy
nnect Dante network to new modes depending on the circumstances. If switching to redundancy
. Co
witch.
n
e
6
rn on system power
. Tu
7. Complete mode changes.
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4.5 Dante Controller
Dante Controller is a software application provided by Audinate, allowing user to configure or send
audios by routing in Dante network. It may be used to run Windows XP, Vista and Windows7 in pcs
and OSX10.5 and 10.6 in Apple Mac. Once users install Dante Controller in pcs and connect it with
Dante network, they may use Dante Controller to conduct following things.
View all Dante compatible audio devices and their network channels;
View clock and network settings of Dante compatible devices;
Send audios by routing on these devices and view current audio routing status;
Change labels of audio channels from numbers to suitable names;
Customize receiving delay (delay before broadcasting);
Save audio routing preset;
Apply saved preset;
View and set configuration options of each device, including:
1
ange device name;
. Ch
. Ch
2
ange sampling rate and clock settings;
3. View detailed network information;
4.Introduce a device web page to upgrade firmware and license information (where there is support).
Please visit Audinate website to download Dante Controller or get help.
.6 Dante Virtual Soundcard
4
Dante Virtual Soundcard is an application software that can be purchased from Audinate. It may
change customer’s pc or Mac to devices compatible with Dante. Standard Ethernet can be used to
transmit or receive Dante audios, but require no additional hardware. The latest version uses standard
Core Audio ((Mac OS X) of Dante Virtual Soundcard or ASIO (Windows) audio ports of Steinberg can
be used for application of any audio.
Once users install Dante Virtual Soundcard on a pc or Mac and connect it to Dante network, users
may
View and change existing audio sampling rate;
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Customize receiving delay (receiving devices shall use delays before audio broadcasting);
View and set Ethernet ports of computers and view details of network ports;
Startup and shut off Dante Virtual Soundcard;
Select available audio channel numbers on Dante Virtual Soundcard.
For Windows system, users may
View and set specific parameters of ASIO.
Note: Users must install Dante Controller on pcs or Macs in Dante network to control audios and
sending them by routing. It can be installed on the same computer with Dante Virtual Soundcard.
Please visit Audinate website to download Dante Virtual Soundcard or get help.
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5. Control
5.1 External Control Programmer
External control programmer supports UDP and RS232 and controls protocols covering all controls
parameters of processor, including parameter controls, parameter acquisition and preset calling.
When UDP controls are used, the default port is 50000. Ports can be set in “Device Setting” via
upper computer software.
When using RS232 controls are used, the default baud rate is 115200, digit bits is 8, and stop bit is 1,
no parity bit. Similarly, they can be set in “Device Setting”. The interval between messages shall
keep more than 100mS for RS232 sending.
If central control needs reply, please turn on central control reply switch in “Device Setting”.
Control Protocol
5
.2
Because of historical reasons, the latest control protocol adopts variable-length and is fully
compatible with old fixed-length control protocols. In protocols, the fourth byte is used to distinguish
versions. 0- indicates V1 version (previous versions) and 1- indicates V2 version (current protocol
version).
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The difference between V1 and V2 is V1 may control all processing module parameters, but one
command can only control one parameter. If a parameter is needed to control continuous multiple
channels, then V2 version shall be used. In other words, in condition that users need to press one key
on key panel to trigger GPIO output high-/low-level of devices, or send a command via RS232/RS485,
then V2 version will be the best choice.
Software coding rules (total 12 bytes)
byte1 byte2 byte3 byte4 byte5~132
0xb3 Message
Type
Length Version
No.
Data
V1:
Information types (byte2): There are three information types including 0x21 (parameter controls),
0x22 (parameter acquisition) and 0x13 (scenario switch).
Length (byte3): invalid.
0x21 (parameter control):
At this point, Databyte5~12 are respectively
byte 5~6 byte 7~8 byte 9~10 byte 11~12
Module ID Parameter Type Parameter 1 Parameter 2
Please refer to Appendix A to get the distribution of Module ID (byte5~6).
Please refer to Appendix B for Parameter types (byte7~8).
When Parameter 1 (byte9~10) has only one parameter, then only parameter 1 is valid, such as control
compressor switch.
Parameter 2 (byte11~12) only valid when there are two parameters, such as control output channel
1 mute. Parameter value 1 shall be filled in input channel number from 0. Parameter value 2 shall be
filled in 1 (mute).
Exception: Matrix routing has three parameters. The first one is input channel number, the second
one is output channel number, and the third one is routing switch. At this point, byte9 of parameter
value 1 shall be filled in input channel numbers, byte 10 shall be filled in output channel number, and
parameter 2 shall be filled in routing switch.
0x22 (Parameter Acquisition):
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Parameter acquisition rules are the same with parameter controls. The difference between them is
values acquired shall be filled in parameter 1 and parameter 2.
0x13 (Scenario Switch):
Users only need to fill scenario numbers (0~15) in byte5 and 0 in byte6~12.
Note: Central control command of V1 version can acquire code through software menu bar of PC.
For customized development, please use this protocol rule.
V2:
Message types (byte2): There are three message types (byte2) including 0x21（parameter controls）,
0x22（parameter acquisition）, 0x13（scenario switch）, 0x74（other controls） and 0x6e（Dante
routing）.
Length (byte3): Fill in corresponding data section length based on information type. The length can
be longer when actual sending is carried out. Total data volume can be got through adding 4 byte
header information to data length.
1. Parameter Control (0x21)
At this point, the formats of data section are as follows.
byte5 byte6 byte7 byte8 byte9~72
Input/Output Start Channel End Channel Parameter Type Parameter Value
byte5: It indicate control input or output channel, 0x2- input channels and 0x1-output channels
byte6-7: They indicate start and end channel numbers. Channel numbers start from 0.
byte8: This kind of parameter is the same with V1 version. Please refer to Appendix B.
byte9-40: Fill in parameter values of start to end channels. It shall be filled in from the ninth byte.
Each parameter value shall take two bytes.
2. Parameter Acquisition (0x22)
Data section format is the same with parameter controls. Parameter values may not be filled in.
acquired parameters will be filled in this position.
3. Scenario Switch (0x13)
byte5: Fill in scenario numbers (0-15).
byte6-8: Fill in 0.
4. Other Controls (0x74)
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Other controls include but not limited to GPIO, RS232, RS485 and central control replies. The protocol
formats are as follows.
GPIO:
byte5 byte6 byte7 byte8 byte9 byte10 byte11 byte12
Control
Type
Data
Length
Reserved Reserved GPIO
Direction
Start GPIO End GPIO Value
The controlling type for byte5 is 1.
The data length of byte6 is fixed as four bytes.
Byte9 GPIO direction, set input or output. Value 0 indicates input, and value 1 indicates output.
Byte10-11 start GPIO and end GPIO. DSP devices totally have eight GPIOs, which are indicated with
number 0-7.
Byte12 is determined according to byte9GPIO direction. The field shall be filled in high (1) /low (0)
level for output settings. The field is a return field to read GPIO level value on devices for input
settings.
RS232/RS485:
byte5 byte6 byte7 byte8 byte9-132
Control Type Data
Reserved Reserved Data
Length
Byte5 is 2 when controlling type RS232, and 3 for RS485.
The data length of byte6 refers to data length that shall be sent via RS232/485 currently.
Byte9-132 shall be filled in data sent by RS232/485.
Central control replies:
byte5 byte6 byte7 byte8 byte9
Control Type Data
Reserved Reserved Reply Switch
Length
Byte5 controlling type is 4.
The data length of byte6 is 1.
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When byte9 is 1, it means turning on central control replies switch; and 0 means turning off replies.
5. Dante Routing (0x6e)
Data section formats are:
byte5 byte6 byte7 byte8 byte9-24 byte25-40
Dante
Channel No.
Byte5 Dante channel numbers. The difference is, Dante channel numbers start from 1.
Byte6 Dante channel, subscribe/cancel subscription of appointed channels of Dante devices indicated
with byte25-49. The appointed channels are indicated with byte9-24 channel names.
.3 Serial Port-to-UDP (RS232 To UDP)
5
DSP devices support RS232 translating into UDP. The protocol formats are as follows.
4bytes prefix 4bytes 2bytes 1byte 1byte 128bytes
UDP: IP Address Port Data Length Reserved Data
After receiving the protocol format data packet, RS232 send data in the protocol to appointed IP
Routing
Switch
Reserved Reserved Subscribed
channel name
Subscribed
device name
addresses and devices at ports.
For example, when sending data ”HELLO DSP to device port 50000 of device ”192.168.10.22”, the
protocol commands are as follows.
4 bytes prefix 4 bytes 2 bytes 1 byte 1 byte 128 bytes
0x3a504455（’:PDU’） 0x1610A8C0 0xC350 0x09 0x00 ”HELLO
DSP”
Application scenario: The function can be applied in scenarios when many central control hosts have
no network port. As shown in the figure, central control hosts transalate network commands through
serial ports to control any network device.
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RS232
5
.4 ASCII control command
Preface:
ncentrate:
Co
1, because the starting bit of the channel is 0, it can be understood that the 0-3 channel
corresponds to the IN1-4 software display channel, and 0-3 is only for example The actual
number of channels shall prevail the device model.
2. In the function on/off settings, 1 is open and 0 is off; For example, set:output#mute#0-3#1 The
last 1 stands for "turn on mute".
5.4.1 Input volume control and acquisition
set:input#gain#0-3#1
(Setting: Enter #Gain #Channel Number # to 1dB)
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get:input#gain#0-3 --> get:input#gain#0-3#1#1#1#1
(Get: Input #Gain#Channel Number) Return Information Example: get:input#gain#0-
3#1#1#1#1#1(Input 1-4 channel volume is 1/1/1/1 in turn)
5
.4.2 Output volume control and acquisition
set:output#gain#0-3#1
get:output#gain#0-3 --> get:output#gain#0-3#1#1#1#1
5.4.3 Phantom power supply control and acquisition
set:input#phant#0-3#1
get:input#phant#0-3 --> get:input#phant#0-3#1#1#1#1
5
.4.4 Input Mute Control and Acquisition
set:input#mute#0-3#1
get:input#mute#0-3 --> get:input#mute#0-3#1#1#1#1
5
.4.5 Output mute control and acquisition
set:output#mute#0-3#1
get:output#mute#0-3 -> get:output#mute#0-3#1#1#1#1
5
.4.6 Control and acquisition of sensitivity
set:input#sens#0-3#1 (for 3db, for second gear)
get:input#sens#0-3 -> get:input#sens#0-3#1#1#1#1
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5.4.7 Control acquisition of matrices: multiple inputs control a single output, and one input
controls multiple outputs
set:mixer#switch#0#0-3#1 (set input 1 to output 1 and 4 routes to open)
set:mixer#switch#0-3#0#1 (set input 1 to 4 to output 1 route open)
set:mixer#gain#0-3#0#1 (set input 1 to 4 to output 1 route gain 1db)
get:mixer#switch#0-3#0(0-3:input,0:output) -> get:mixer#switch#0-3#0#1#0#1#1
5
.4.8 Invocation and Saving of Scenarios
scene: toggle #3 (scene call, pc is displayed as scene 4)
scene:save#3 (save scene)
.4.9 Input level acquisition
5
get:input#level#0-3 -> get:input#level#0-3 #-105.4#-102.5#-105.2#-104.8(dbfs)
.4.10 Output level acquisition
5
get:output#level#0-3 -> get:output#level#0-3#-56.0#-40.8#-43.6#-46.4
.4.11 System Mute Control and Acquisition
5
set:sysctl#mute#1 (on system mute)
get:sysctl#mute -> get:sysctl#mute#1
.4.12 Settings and getting the channel name of the input and output
5
set:input#name#0#1
get:input#name#0-3 -> get:input#name#0-3#IN1#IN2#IN3#IN4
5.4.13 Input and output inverting control and acquisition
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set:input#phase#0-3#1
set:output#phase#0-3#1
get:input#phase#0-3 -> get:input#phase#0-3#1#1#1#1
get:output#phase#0-3 -> get:output#phase#0-3#1#1#1#1
.4.14 Input and output step control and acquisition
5
set:input#step#0-3#10
set:output#step#0-3#10
.4.15 Input and output link control and acquisition
5
set:input#link#0-3#1
set:output#link#0-3#1
get:input#link#0-3 -> get:input#link#0-3#1#1#1#1
get:output#link#0-3 -> get:output#link#0-3#1#1#1#1
5
.4.16 Control and acquisition of signal generators
set:input#type#0-3#1
get:input#type#0-3 -> get:input#type#0-3#1#1#1#1
5.
4.17 Restore factory settings control
set:refactory
5.4.18 Scene Reset Control
set:rescene
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5.4.19 Get the name of the scene that is set arbitrarily
set:scene#name#0-3#pre1 (PC only supports UTF-8 encoding, Chinese ANSII code will become
garbled)
get:scene#name#0-3 (scene number: 0-15) -> get:scene#name#0-3#pre1#pre1#pre1#pre1#pre1
5.4.20 Module name:input,output|mixer
item 名:
(input)mute,gain,sens,phant,type,freq,name,phase,step,link,level
(output)mute,gain,name,step,link,level
(mixer)switch,gain
(scene)toggle,save,name
(sysctl)mute
(rescene)
(refactory)
4.21 Format the instruction:
5.
set: Module name #itemname #Start Channel-End Channel # Parameter value
For example
set:input#mute#0-3#0/1
4.22 Get the instruction format
5.
get: Module name #itemname #Start Channel- End Channel
For example
get:input#mute#0-3
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6. FAQs
1. How to restore factory setting?
Connect it to computer through RS232 and run serial port software (SecureCRT is recommended to
use). The default baud rate of series ports is 115200, 8 data bits, no parity check, one stop bit. After
connecting SecureCRT to serial ports, long press to enter at terminal interface to reboot the computer
and enter bootloader boot dialog box as shown in the figure.
Command explanation:
del config: delete configuration information, such as network configurations like IP address. The
device restores to default IP 169.254.20.227 after deleting.
del secens: delete preset. All 16 presets of DSP devices restore to default values.
del all: delete all sections except the program.
Note: There may be no echo after the installation of some SecureCRTs. Please check ”Local echo”
by going to Options->Session Options, as shown in the figure.
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Appendix A: Module ID Distribution
odule Name ID Module Name ID
M
Input source 299 Output Channel 1-32
High & Low Pass
Input Channel 1-32
Expander
Input Channel 1-32
Compressor
Input Channel 1-32
Auto Gain
Input Channel 1-32
Equalizer
Input Channel 1-32
Feedback Inhibition
AutoMixer 161 Echo Canceller 162
1~32 Output Channel 1-32
Equalizer
33~64 Output Channel 1-32
Delayer
65~95 Output Channel 1-32
Limiter
97~128
129~160
167~198
199~230
231~262
263~294
Echo Cancellation 163 Noise Suppressor 164
Noise Suppression 165
M
ixer 166
Output
Sy
stem Control 296
295
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Appendix B: Module Parameter Types (1)
Module
Name
Input
Source
Parameter
Type
0x1 Gain Output 0x10 Gain
0x2 Mute 0x11 Link
0x3 Sensitivity 0x12 Channel Level
0x4 Phantom Power
0x5 Signal Generator
0x6 Signal Generator
Description Module
Name
Switch
Type
Frequency
Parameter
Type
0x1 Gain
0x2 Mute
0x3 Channel Name
Description
Compensation
0x7 Sine Wave Gain
Size
0x8 Channel Name 0x5 Sensitivity
0x9 Invert 0x6 Gain
0x10 Gain
Compensation
0x11 Link 0x8 Channel Level
0x12 Channel Level Expander 0x1 Switch
Delayer 0x1 Bypass Switch 0x2 Threshold
0x2 Millisecond 0x3 Ratio
0x4 Invert
Compensation
0x7 Link
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0x3 Microsecond 0x4 Setup Time
Equalizer 0x1 Total Equalizer
Switch
0x2 Child Segment
Switch
0x3 Frequency 0x2 Compressor
0x4 Gain 0x3 Compressor
0x5 Q Value 0x4 Setup Time
0x6 Type 0x5 Recovery Time
0x5 Release Time
Compressor 0x1 Compressor
Switch
Threshold
Ratio
0x6 Gain
Compensation
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Appendix B: Module Parameter Types (2)
Module
Name
Mixer 0x1 Mixer Switch Feedback
High &
Low Pass
Parameter
Type
0x2 Mixer Gain 0x2 Feedback Point
0x1 High Pass
0x2 High Pass Type 0x6 Preset
0x3 High Pass Slope 0x7 Clear
0x4 High Pass
0x5 High Pass Gain 0x9 Feedback
Description Module
Name
Inhibition
Switch
Frequency
Parameter
Type
0x1 Switch
0x3 Feedback Point
0x8 Panic Threshold
Description
Frequency
Gain
0x11 Low Pass Switch Auto Gain 0x1 Switch
0x12 Low Pass Type 0x2 Threshold
0x13 Low Pass Slope 0x3 Target Threshold
0x14 Low Pass
Frequency
0x15 Low Pass Gain 0x5 Setup Time
Auto Mix 0x1 Total Mute 0x6 Release Time
0x2 Total Gain Echo
Cancellation
0x3 Slope 0x2 Echo
0x4 Ratio
0x1 Echo
Cancellation Switch
Cancellation Mode
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0x4 Response Time Noise
Suppression
0x5 Channel Auto
Switch
0x6 Channel Mute System
0x1 Noise
Suppression Switch
0x2 Noise
Suppression Mode
0x1 System Mute
Control
0x7 Channel Gain 0x2 System Gain
0x8 Priority
0x9 Auto Mix Switch
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