• Using clock synchronization and transmission technology, the audio delay is less than 5ms; adopt the uncompressed audio transmission with 48K sampling rate; use cat5e shielded cable to ensure long-distance reliable transmission of conference information, while providing perfect sound quality.
  • Built-in high-performance DSP processor, with audio matrix, howling suppression, EQ, volume, delay and other adjustment functions.
  • The audio input interface includes 1 RCA, 1 XLR, and 2 Phoenix terminals. The audio output interface includes 1 RCA, 1 XLR, and 16 Phoenix terminals.
  • Support16-channel output function, support flexible configuration as role separation output mode, simultaneous transmission output mode and phase control output mode, and support adjusting parameters such as EQ, volume and delay for each output channel.
  • The 16-channel role separation output mode enables wired or wireless units to output independently according to ID numbers; it can be used by recording or speech transcription equipment; the number of output channels can be extended by external devices.
  • The 16-channel simultaneous transmission output mode enables simultaneous interpretation audio to be output independently according to the channel number; it can be used by recording or monitoring equipment; the number of output channels can be extended by external devices.
  • The 16-channel phase control output mode, based on the original conference matrix technology, built-in nx16 audio matrix processor, can achieve 16-channel group output function. Any input source (including all input sources and online microphones) can be output to any channel according to any volume ratio.
  • Adopt TCP/IP network protocol, support C/S and B/S architecture; it can be controlled by PC software or browser.
  • Support the control of audio matrix parameters (including EQ, volume, delay, microphone sensitivity, etc.), 16-channel output mode switching, switch microphone synchronization, Chinese, English, Russian and French switching, and the role separation controller.
  • Large system capacity, the system supports a maximum of 4096 wired conference units and 300 wireless conference units. The maximum number of speakers in the system is 16 wired microphones and 8 wireless microphones.
  • Support circular hand-in-hand function to ensure that the conference can continue normally when one of the network cables is disconnected or the unit fails.
  • Support Chinese, English, Russian, French and other languages.
  • Through the PC software, you can view the information such as battery level, WiFi signal of the online wireless units; it supports one-key shutdown of all wireless units and a single wireless unit.
  • Support simultaneous interpretation function, the system can simultaneously transmit 63+1 wired simultaneous interpretation.
  • With a fire alarm linkage trigger interface, it provides fire alarm information to evacuate the venue personnel urgently.
  • With 1 RS-485 interface, support a camera to realize camera tracking, support PELCO-D, VISCA camera control protocol; work with HD camera tracking server to realize automatic camera tracking.
  • Four microphone management modes: FIFO (first in, first out), NORMAL (normal mode), VOICE (voice control mode), APPLY (application mode).
  • Support functions such as meeting sign-in, voting, evaluation, and other custom functions.
  • With a 4.3-inch full-color touch screen, it can realize parameter setting or viewing, and other touch operations.
  • Powerful ID editing function, support coding for wired units, wireless units, interpreter machines, and role separation server
  • With USB recording function, it can record and play meeting records.
  • Support 10-bandEQ adjustment function;16 multi-function output channels and 2 LINEOUT output channels support 10-band EQ adjustment function.
  • Support AP channel scanning, learn about the use of wireless channels on site, support automatic or manual configuration as the best channel, support online display of AP name list, convenient to
  • Support entering the registration code on the touch screen for host registration.
  • Support connecting with speech transcription server to realize speech transcription function
  • Support setting the master-slave dual backup function, when the master fails, it can automatically switch to the slave operation.
  • Support remote firmware upgrades through the web.
  • Support operation and maintenance management function, support log management. Real-time monitoring of equipment operating status and equipment failure information, including insufficient memory, fire alarm prompts, and id duplication, etc.
  • Support to choose VISCA, Pelcd-D protocol to connect with cameras, central control and other equipment.
Model TS-0300M
Microphone capacity Wired microphone≤4096; wireless microphone≤300
Simultaneous interpretation channel 63+1 channels
Frequency response 80~16KHz
SNR >78dB(A)
Dynamic range >80dB
THD <0.05%
Main power 100-120VAC/200-240VACbyswitch
Audio input LINEIN1: 775mVrms balanced; 2 output phoenix terminals: 775mVrms balanced; LINEIN2: 775mVrms unbalanced
Audio output LINEOUT1: 1Vrms balanced; 16 multi-function output phoenix terminals: 1Vrms balanced; LINEOUT2: 1Vrms unbalanced
Output load >1KΩ
EXTENSION port 1 for connecting conference system extension equipment
DANTE/NC port 1 for connecting to external devices with DANTE protocol
WIFI network port 1 for connecting to wireless AP
PC network port 1 for connecting to the computer
DELEGATES output interface 4 for connect conference speaking units
RS-232 interface 2 channels, 1 channel for camera tracking, 1 channel for docking external equipment
RS-485 interface 1 for camera tracking
Static power consumption 30W
Output power consumption 320W
Wired microphone connection method Special cable (6 pins)
Touch screen 4.3-inch fu×olor touch screen
Colour Black
Net weight 5.6Kg
Dimension 484×303×88mm (L×W×H)
Installation method 19-inch standard cabinet

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