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Wholesale professional digital audio processor ADSP-21571 digital DSP processor FIR 512 speaker manager with EQ audio system PC

Wholesale professional digital audio processor ADSP-21571 digital DSP processor FIR 512 speaker manager with EQ audio system PC

Regular price $562.42 USD
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SPECIFICATIONS

A/D: AK5552

Battery Included: No

Brand Name: NoEnName_Null

Certification: None

DAC decoding: CS4398

FIR: With FIR

FIR order: 512

Floating-point kernel: SHARC+DSP processor, supporting 64-bit floating-point optimized FIR and IIR algorithms

Gain range: ±20dB

Hign-concerned Chemical: None

Input/output gain: -80dB~12dB

Liquid crystal screen: 4.58 inches

Model Number: ADSP-21571

Origin: Mainland China

Sampling rate: 32-bit 768Khz

Signal-to-noise ratio: Over 120dB, ultra-low THD+N,118dB

Support: 192KHz sampling rate and 24-bit low noise

Support Bluetooth: No

System language: English/Chinese

Type: Speakers




3 in and 6 out



The sockets available include: European sockets, American sockets, British sockets, and Chinese standard sockets. You can make a note for me or leave a message.

Some countries may have tariffs. If you think the tariffs are too high, you can communicate with me. Many regions support UPS and DHL express delivery, which will arrive within 5 to 10 days. Whether to buy or not, you are welcome to inquire. Thank you!





448 digital audio processor

The series digital audio processor is a high-performance digital DSP processor that supports multi-channel analog signal routing. Users can connect to a computer via USB or RJ45 Ethernet for real-time control of the upper computer. The simple and friendly PC software interface presents the operation to users in a more intuitive and understandable way.

The input channels include 5 segments of DEQ, 31 segments of GEQ, and 15 segments of PEQ. The output channels feature 15 segments of PEQ and FIR functions. The channels are equipped with limiters and adjustable noise gates. The built-in 30 factory presets are suitable for one-click layout in most audio systems. Users can also customize and save for invocation. The PC software is equipped with a signal generator for convenient on-site speaker debugging and calibration. At the same time, this machine is equipped with a 4.58-inch ultra-high-definition LCD screen. Supports both Chinese and English display, providing a considerable and convenient human-computer interaction operation experience.

The CPU adopts the ADSP-21571 digital audio processing chip from Analog Devices (ADI) of the United States. It is a dual-core SHARC+DSP processor based on the Arm Cortex-A5 high-performance floating-point core architecture, supporting 64-bit floating-point optimized FIR and IIR algorithms. The A/D section employs the AK5552 analog-to-digital conversion chip, supporting a 32-bit 768Khz sampling rate. It features a differential filter circuit input design, effectively ensuring high resolution and noise filtering of the input signal. Additionally, it offers a professional-level 118dB signal-to-noise ratio, effectively suppressing background noise issues in digital audio processing circuits

The DAC decoding uses the CS4398 chip designed and produced by Cirrus Logic, an audio chip manufacturer in the United States. It supports a sampling rate of 192KHz and 24-bit low noise, low distortion, and high dynamic output. It has a signal-to-noise ratio of more than 120dB and ultra-low THD+N, meeting the playback requirements of various mainstream audio sources at present.



Function

Input gain: -80dB to 12dB

Output gain: -80dB to 12dB

Input delay: 0 to 680ms

Output delay: 0 to 680ms

Input equalizer type: DEQ/GEQ/PEQ

Output equalizer type: PEQ


Graphic equalizer: 31 inputs per channel;

Gain range: ±20dB


Dynamic equalizer: 5 inputs per channel; Support bypass

Threshold: -35dB to 20dB

Frequency range: 20Hz to 20Khz

Ratio: 1:1 to 10:1; Startup time: 2ms to 500ms; Release time: 8ms to 2000ms

Q value range: 0.3 to 15

Gain range: ±20dB

Dynamic equalizer working mode: Cut Above/Cut Below/Boost Above/Boost Below


Parametric equalizer: 15 inputs per channel, 15 outputs per channel; Support bypass

The Q value range of the parametric equalizer is 0.3 to 15

Parametric equalizer types: PEQ/High-shelf/LOW-shelf/AP1/AP2

High and low pass filtering

Butterworth slope: 12/18/24 42/48/30/36 / db/Oct

Bessel slope: 12/18/24 42/48/30/36 / db/Oct

Lin Kui Bates - Rayleigh: 12/18/24 42/48/30/36 / db/Oct


Input noise gate

Threshold: -135dB to -39dB;

Startup time: 2ms to 500ms;

Holding time: 2ms to 500ms;

Release time: 8ms to 2000ms


Output limiter

Threshold: -50dB to 20dB;

Startup time: 2ms to 500ms;

Release time: 8ms to 2000ms;

Ratio: 1:1 to INF:1;

Gain: ±12dB

Working modes: Hard press/Medium press/soft press


FIR order: 512

FIR cut-off frequency range: 20Hz-20kHz

FIR filter types: pass-through, low-pass, high-pass, band-pass, band-stop, external import

FIR supports file formats: txt, csv

FIR attenuation range: -32dB to -96dB

FIR function types: Rectangular/Tukey/Triangular/Hanning, Hamming/Blackman/Kaiser


Others

Channel parameter copying: Supported

Channel grouping and joint debugging: Supported

Signal generator functions: sine wave/white noise/powder noise/sweep frequency; Gain range: -60dB to 6dB

Input/output channel phase: Normal phase (0°)/inverting phase (180°)

Input/output channel mute: ON/OFF

Custom user presets: 32 groups

Engineering startup count lock: Supported

Panel operation lock: Supported

System language: Chinese/English

Working indicator lights: Silent/Peak/Noise gate/limiter /-24dB to 12dB level

RJ45 network port: UDP protocol; (Only supports single machine)

RJ45 connection method: Automatically obtain the IP address assigned by the DHCP server (router/switch)


Specification parameters

Input channel: 4ch analog XLR balanced type

Output channel: 8ch analog XLR balanced type

Frequency response (20Hz-20kHz@+4dBu) : +0/-0.3dB

Total harmonic distortion (20Hz-20kHz@+4dBu) : <0.003%

Signal-to-noise ratio (S/N) : 113dB A weighting

Background noise: <-90 dbu

Dynamic range (20Hz-20kHz, 0dB) : >122 dB

Maximum gain (input to output) : 48dB

Maximum delay (input to output) : 1360ms

Channel separation (between channels @1kHz) : >80dB

Common-mode rejection ratio: 60Hz>100dB@+20dBu

Input impedance (balanced/unbalanced) : Bal:20K/Unbal:10K

Output impedance (balanced/unbalanced) : Bal:100ohm/Unba







LD336

It adopts advanced dual-core DSP technology and supports 64-bit floating-point operations. The ADC section employs a 768Khz/32-bit high-performance audio converter. All algorithms are processed by a pair of 500MHz high-performance DSPS. Digital processing includes: gain, polarity conversion, dynamic equalizer, graphic equalizer, parametric equalizer, shelf filter, time delay, frequency division, voltage limiting and signal routing. All inputs and outputs are amplified using differential balanced amplification to enhance the signal's anti-interference capability and signal integrity.

By adopting the latest FIR digital filter design, low-pass, high-pass, band-pass and band-stop filters can be designed, which largely enables the extraction of the required signals and the suppression of unwanted ones.

All audio interfaces use standard XLR interfaces.

The LD448 can store up to 32 presets. It is convenient for users to save and call at any time.

The LD448 input equalizer features 15-segment parametric equalization. 5-segment dynamic equilibrium The 31-segment illustrated equalizer largely meets various tuning needs

◎ High-definition LCD screen in both Chinese and English, with an intuitive functional operation interface

It is equipped with a built-in signal generator for convenient on-site debugging and calibration

The input is equipped with a noise gate and the output has a limiter function

Independent input and output channel level indication

Supports panel lock and engineering lock functions

Balanced input and output XLR speakers with dedicated PC control software

Frequency response (20Hz-20kHz@+4dBu):+0/-0.3dB

Total harmonic distortion (20Hz-20kHz@+4dBu):<0.003%

Signal-to-noise ratio :110dB A weighting

Background noise :<-90 dbu

Dynamic range (20Hz-20kHz, 0dB):>116 dB

The maximum gain (input-to-output) of Zudibi compared to Aizi is 48dB

Maximum delay (input to output):750ms

Channel separation (between channels @1kHz):>80dB

Common-mode rejection ratio :60Hz>100dB@+20dBu

Input impedance (balanced/unbalanced):Bal:20K/Unbal:10K

Output impedance (balanced/unbalanced):Bal:100ohm /Unbal:50ohm

Maximum input level :+20dBu

Maximum output level :+20dBu

Input gain range (adjustable):-80dB to +12dB

Output gain range (adjustable):-80dB to +12dB

A/D chip: AK5552

A/D sampling rate :768kHz

A/D converter bit width :32 bits

D/A chip :AD1955

D/A sampling rate :192kHz

The bit width of the D/A converter is 24 bits

DSP chip :ADSP-21571

DSP main frequency :500Mhz

DSP bit width :32/40, 4-bit floating-point, dual-core SHARC+ARMCortex-A5TM core


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