ADAPTIVE SOUND FIELD CONTROLLER with MEASUREMENT MICROPHONE
Strong performance processing capacity
Sound quality similar to analog audio
Extremely simple test operation
FPGA powerful DSP processing
High frequency calculation of ARM core
Standardized network protocol connection
SK46 is new intelligent on-site acoustic test processing tool.
The adaptive sound field processor is based on a world-leading audio research and development platform. This product is the first to implement 4096 high-order real-time FIR acoustic phase calibration processor using SINECORE technology and FPGA algorithm, effectively solving the impact of system delay caused by traditional DSP audio processing mode on sound quality transmission, achieving ultra clear sampling/High order operations/Accurate calibration; The 3D standard is used to optimize the on-site sound reinforcement effect and truely achieve "what you GET is what you SEE".
FPGA high-end real-time FIR acoustic phase calibration technology. Significantly improve the clarity of the mixing, and accurately achieve a flat sound frequency response.
The traditional DSP processing system will additionally produce more phase problems during the processing and the secret of the SK46 core technology is hidden in the SineCore audio processing platform. It uses an FPGA-based high-end real-time FIR acoustic phase calibration system, which breaks the barrier between digital audio algorithm technology and delay, and perfectly reproduces the essence of sound through the 3D standards of sampling accuracy, sampling frequency and phase accuracy.
SK46 optimizes your acoustic environment by enhancing the clarity of sound images and restoring the transparency of sound. The system provides a variety of interfaces compact in design and simple in use, suitable for the needs of sound field measurement and calibration in small and medium auditory rooms, studios, theaters, halls and large live performances. Achieve more transparent, high-resolution high-frequency performance and more powerful bass restore.
FIR filters are characterized by linear phase, which adapts to the unique requirement of audio - that is, changing the frequency domain without phase change.
However, due to the characteristics of DSP serial, FIR filters of order 10,000 or more cannot adapt to the real-time performance of audio requirements because of the long delay time. The new delay brings new phase problems, so high-order FIR and waveforms have performance problems due to DSP. Even because of the complexity of processing algorithm coefficients, some manufacturers even use the CPU of a PC to calculate coefficients and process FIR, which can cause greater system latency.
|Analog Input||4 x XLR (L/R), +24 dBu max|
|Analog Output||6 x XLR (L/R; subwoofer+bass+mid+treble), +24 dBu max|
|Digital Input||1 x AES/EBU@75 Ohms|
|Word Clock Input||1 x lnput@75 Ohms 3Vpp on BNC 32-192k Hz|
|Word Clock Output||1 x Output@75Ohms 3Vpp on BNC 32-192k Hz|
|D/A Converter||Dynamic range: 120cb THO+ N: -107 dB|
|A/D Converter||Dynamic range: 120cb THO+ N: -110 dB|
|Sampling Rate||44.1, 48, 88.2, 96, 176.4, 192k Hz|
|Electrical Requirement||AC input: 95-245 V; Power consumption: 20W max|
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