KONANlabs KSP-S0808 Audio Processor User Manual

June 13, 2024
KONANlabs

KSP-S0808 Audio Processor

KONANlabs KSP-S0808 Audio Processor

Packing List:

  • Audio Processor X1
  • 12V/2A DC Power X1
  • Quick Guider X1
  • Warranty Card X1

Description:

The KONANlabs KSP-S0808 Audio Processor is a versatile device
suitable for various applications such as meeting rooms, campuses,
art performances, concerts, and other scene areas. It features a
B/S architecture server that allows access through a web browser.
This enables users to control channels and select scenarios.
Additionally, it provides PC client and platform components
download links. The system also includes a built-in lock screen
function to prevent incorrect operations.

The audio processing capabilities of the KSP-S0808 include DSP
audio processing, a built-in automatic mixer, and optional feedback
elimination. Each channel has front-stage amplification, a signal
generator, expander, compressor, and 5-stage parameter
equalization. The output per channel includes a 31-section diagram
equalizer, delay device, frequency divider, and limiter. The device
also offers full-function matrix remixing.

Other features of the KSP-S0808 include support for scenario
presets, automatic memory protection when power is off, a compact
1U whole aluminum chassis design, and a built-in automatic camera
tracking function.

Technical Parameters:

  • Input common mode suppression: 60Hz
  • Channel isolation: 1kHz
  • Size: 48225845(mm)
  • Weight: 2.5Kg
  • Power consumption
  • Operating temperature
  • Working power supply

Product Usage Instructions:

  1. Connect the 12V/2A DC power supply to the Audio Processor.

  2. Ensure the operating temperature is within the specified
    range.

  3. Connect the required audio input sources to the corresponding
    input channels.

  4. Access the device through a web browser by entering its IP
    address.

  5. Use the web interface to control the channels, select
    scenarios, and adjust settings.

  6. To achieve the desired audio output, configure the front-stage
    amplification, signal generator, expander, compressor, and
    equalization settings for each input channel.

  7. Adjust the output settings per channel, including the diagram
    equalizer, delay device, frequency divider, and limiter.

  8. If desired, utilize the full-function matrix remixing
    feature.

  9. Take advantage of the scenario presets for convenient switching
    between different audio setups.

  10. If power is accidentally turned off, the device will
    automatically protect and restore its memory settings upon power
    restoration.

KONANlabs KSP-S0808 Audio Processor User Manual
Packing List
Audio Processor X112V/2A DC PowerX1Quick Guider X1Warranty Card X1
Description
Suitable for meeting room, campus, art performance, concert and other scene areas. With a B/S architecture server, access through a web browser, not only to achieve channel control
and scenario selection, but also directly provide PC client and platform components download links. System built-in lock screen function, can effectively avoid the wrong operation. DSP audio processing, built-in automatic mixer, optional feedback elimination Input per channelFront stage amplifier, signal generator, expander, compressor, 5 stage parameter
equalization Output per channel31 Section diagram equalizer, delay device, frequency divider, limiter Full function matrix remix Support for scenario presets Automatic memory protection when power off 1U Whole aluminum chassis Built-in automatic camera tracking function

Technical Parameters
Technical Parameters DSP Chip Analog Channel Processing GPIO RS485 RJ45 Simulated maximum gain Digitalizing bit Sampling frequency Frequency response 20~20KHz Digital-to-analog dynamic range (A-weighted) Analog-to-digital dynamic range (A-weighted) Input to output dynamic range Total harmonic distortion + noise(THD+N) Ground Noise (A- weighted) Delay storage Analog input to output system delay Input impedance (balanced) Output impedance (balanced type) Maximum input level Maximum output level EIN20-20kHzA Weighted. Phantom power supply (per input)

Ti 456MHz FLOPS DSP 8 Input+8 Output Auto MixingAFC,AEC,ANC 2 Input 1 1 42dB 24bit 48k ±0.2dB 120dB 114dB 108dB 0.004% @20-20KHz ,18dBu -90dBu 2s 3ms 20K 100 +18dBuBalanced +18dBuBalanced -131dBU 48V

Input common mode suppression60Hz Channel isolation1kHz Size Weight Power consumption Operating temperature Working power supply

70dB 104dB 48225845(mm) 2.5Kg <24W -10-50 DC12V/2A

Interface Description
Front Panel

Back Panel

  1. DC12V: DC12V /2A power input interface; (2) PWR: power indicator light, which means the power supply of the equipment is normal; otherwise, the power supply is abnormal;

(3) SYS: Status indicator light. The flashing light indicates normal operation of the device; otherwise, the device fails; (4) ETHERNET: 10M/100M ETHERNET interface; (5) RESET: RESET to factory setting, long press for 5 seconds. (6) RS232: Support central control command and camera tracking,RX: receive data,TX: send data,G: ground wire; (7) RS485: Supporting camera tracking; (8) GPIO: GPIO control; (8) OUTPUT: Analog Output; (9) INPUT: Analog Input
Use and operation steps
1 Web control and software download
The factory default IP address of the device is: 192.168.1.200 Subnet mask: 255.255.255.0. Please add the address of the network segment to the PC first, so that the device can connect normally.
After the device is started, use a web browser to access the address “http://192.168.1.200/”, as shown in the figure below:

Control: Control channel parameters and enable and disable each processor Scene: Quickly recall and save device scenes. Download: The download link provides the download of the .Net framework PC software. The PC software supports XP, Win7, and Win8 operating systems.
Before installing the PC software, please make sure that Microsoft .Net Framework 3.5 or above has been installed on the PC.
When installing the software, some systems (such as WIN8) will pop up a prompt: “User Account Control Information”, please click the “OK” button to increase the software authority.

2 System flow
Signal processing flow chart Standard configuration Input: test signal/mute/expander/5 band equalizer/compressor/auto gain Output: delayer/frequency divider/31-band graphic equalizer/limiter/output inversion/mute

Advanced configuration

AFC,AEC,ANC

3 Menu bar and status bar
3.1 File

  1. New: Create a new scene with factory configuration parameters that are available offline only. 2) open: Open the local saved scene, which is available offline only. 3) Save as: Save the current configuration as a file locally, only available offline. 4) Exit: Close the software. 5) language switching: simplified, traditional and English languages are supported.

3.2 Central Command The central command generator can convert frequently used operations into a 16-character command code to facilitate calls to external devices.
Each of these commands contains three different sets of parameters.Control command types: scene, input, output, mix, parameter equalizer, graphic equalizer, expander, compressor, Automatic gain, delay, frequency divider, limiter.
3.3 Device settings Device settings include user settings, network settings, serial port settings, scene settings, camera tracking, and GPIO.

  1. User settings
    a. The initial user name of the device is admin/password 123456, The administrator can add, delete, and modify all user information; ordinary users can only modify personal information. b. Modify user: First select the user to be modified in the user list, the user name and password edit box will display the information of the currently selected user, enter new information, and click the “modify” button. c. Delete user: select the row to be deleted in the user list and click the “Delete” button to delete the user.d D.Add user: Select the empty row in the left list, and enter the new user’s information in the user name and password edit box on the right (should be empty), and click the “Add” button to add a new user.

  2. Network settings
    View and modify the network address information of the device, enter the IP address, subnet mask, and gateway in the corresponding position, and click the Apply button to complete the modification

  3. Serial port settings
    View and modify the serial port information of the current device. Click “Apply” button to modify the serial port information of the current device after setting;If you want to return to the default value, directly click the “All Reset” button, and the items cannot be empty when setting.

  4. Scene settings
    a. Modify name: modify the selected scene name. b. Upload scene: upload the scene on the PC side and overwrite the selected scene. c. Save scene: Save the current running parameters to the selected scene. d. Save as: Save the currently running parameters to the PC in a scene. e. Load scene: enable the currently selected scene, usually used for scene change.

f. Restore factory settings: restore all scene configurations to the default configuration.
This device supports two ways to save scenes offline and online. Offline save is to save the set scene on the PC, which is convenient for subsequent recall and scene copying between different devices. Saving a scene online is to save the scene directly to the device, which can be recalled directly after turning on the device next time
.4 DSP module
4.1 Input Setting
The input signal can be an analog signal or a test signal generated inside the device. If it is a network version with Dante, it can also be a network digital signal;
The analog signal can be selected by adjusting the sensitivity to adjust the input; from -60~0, every 3dB; Mute: the channel is muted when selected; Invert: The signal phase is processed by 180 degrees. Phantom power supply: used for condenser microphone power supply, please do not turn on line input or non-condenser microphone to prevent burnt; Test signal: including sine, pink, and white noise. Enabling the test signal system will automatically shield the analog input signal;

4.2 Expander
The Expander is a dynamic range of input that increases according to the user’s requirements. When the input signal is less than “threshold”, the expander compresses the input signal according to the set “ratio”, output level = threshold-(threshold-input level)/ratio; when the input signal is greater than “threshold”, then According to 1:1 output, output level = input level. Pass-through/Enable: Whether the extender is valid. Ratio: The number of decibels that the expander input signal changes dynamically/the number of decibels that the expander output signal changes dynamically. Start-up time: the time required for an input signal less than the “threshold” of the expander to enter the expanded state to output according to the set expansion ratio. Recovery time: The time required for the input signal to return from the expanded state to the original nonexpanded state.

4.3 Compressor
The compressor is used to reduce the dynamic range of the signal above a user- defined threshold. The signal level below the threshold remains unchanged. Threshold: When the signal level is higher than the threshold, the gain will be reduced. This point is the inflection point in the input/output curve. For peak stop, the threshold that needs to be stopped is just below the peak level. Ratio: The compression ratio of input and output. Start-up time: Start with the gain of the compressor to reduce the processing speed. The shorter the startup time, the greater the instantaneous change of the signal, and the short-term gain attenuation makes the hearing unsuitable. Release time: The release time determines the gain change from moment to moment of the compressor. The quick release time increases the subjective level, while the slow release time is more useful to keep it under control. Output fader: The fader can control the output gain of the module. If the compressor reduces the signal level significantly, the output gain boost may need to maintain the perceived volume.

4.4 Equalizer
Parameter equalization Pass-through/enable: Whether the equalizer is valid. Band Pass-through/enable: Whether the section equalizer is valid. Center frequency: the center frequency that needs to be equalized. Gain: Gain/attenuation value at the frequency center point. Bandwidth: the influence range of this segment around the center frequency. The larger the value, the larger the bandwidth and the larger the influence range. Graphic equalization The gain of the 31 band frequency points can be adjusted separately to achieve the purpose of strengthening or weakening some frequency points and achieve different effects. Pass-through/Enable: Enable and disable the equalizer. Gain: The gain/attenuation of the frequency center point. Flat: Restore all frequency band gains to 0dB state. Narrowband: A kind of bandwidth, the bandwidth is lower than the ordinary bandwidth.

Normal: Common bandwidth. Broadband: The largest bandwidth.
4.5 Auto gain
The purpose of automatic gain control is to achieve the target level of signals with uncertain levels while maintaining the dynamic range of the volume. Typical usage scenarios: For example, when the user is speaking in front of the microphone, the distance between the mouth and the microphone may be too far and near, which causes the output volume to fluctuate and even feel intermittent in speech. Automatic gain is to set the threshold, output the input signal below the threshold in a ratio of 1:1, and directly increase the level above the threshold according to the ratio. After setting the target level, the sound signal can be output stably. Threshold: When the signal level is lower than the threshold, the input/output ratio is 1:1. When the signal level is higher than the threshold, input/output = ratio. Set this threshold level slightly higher than the noise ratio of your input signal. Target threshold: the required output signal level. Automatic gain control is to automatically control the amplitude of the gain by changing the input and output compression ratio. When a weak signal is input, the signal is amplified to ensure the strength of the output sound signal; when the input signal strength reaches a certain level, the signal is compressed

to reduce the sound output amplitude. 4.6 Auto Mixer
The automatic mixer is mainly used to automatically control how the traditional mixer has a large amount of voice input to output the desired result. Consider a typical conference room scene. There are ten participants, each with a microphone. If ten microphones are turned on at the same time, only one person is talking, then the output effect is definitely not ideal, because the other nine microphones pick up Sound insulation of the room, reverberation, etc., will reduce the output effect of the entire system. Each channel of the automatic mixer has an input, a gain level meter and an automatic gain, channel fader, priority, and channel mute. Channel control Each channel has an “auto” button, press to add this channel to the automatic mixing. The channel mute and fader are both automatic gain type. To mute a signal and prevent the signal from entering the automatic mixing, please turn on “Mute” and cancel “Auto”. The channel fader controls the mixing level and direct output level of the channel. Priority control PR: Allows channels with higher priority to surpass channels with lower priority, thereby affecting the automatic mixing algorithm. The control defines the priority with a value between 0 (lowest priority) and 10 (highest priority), and the default value is 5 (standard priority). If the priority of all channels is equal, set the priority of all channels to 5.

4.7 Auto Mixing/AFC/AEC/ANC
Feedback: Select the signal to be processed by the feedback canceller, and select the output channel of the processed signal in the mixer. Auto mix: Mix the signal of the selected input channel to the corresponding output channel. AM: Signal processed by the automatic mixer AFC: Signal processed by feedback canceller AEC: Signal processed by echo canceller ANC: Signal processed by noise canceller 4.8 Delayer The time interval between the signal input to the output processor.Generally used to produce reverberation or echo and other effects, can also be used for the use of large auxiliary speaker processing.

4.9 Frequency Divider High frequency pass/enable: enable and disable the high pass filter. Low frequency pass/enable: enable and disable the low pass filter. High-pass frequency: the cut-off frequency of high-pass filtering. Low-pass frequency: the cut-off frequency of low-pass filtering. 4.10 Limiter Through/Enable: Enable or disable the limiter. Threshold: the starting level of the limiter. When the signal is higher than this limit value, the limiter processing module is started. Recovery time: When the input signal is lower than this setting value, the sound channel will not be closed immediately, and the closing time will be delayed according to this setting value. During this time, as long as there is a signal higher than the “threshold” limit value, the sound channel can continue to be turned on. Compression: The difference between the signal processed by the limiter and the input signal. 4.11 Output Setting You can set the output to mute and invert.

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