An acoustic array is a system composed of multiple acoustic sensors (microphones or speaker units) arranged in a specific geometric configuration. This structural design enables the array to perform complex functions unattainable by a single sensor.

The core advantage of an acoustic array lies in its spatial processing capability. By coordinating the operation of multiple sensors, it enables precise control or analysis of sound propagation direction and coverage area. Acoustic arrays can focus sound in specific directions for transmission/reception, enhance target signals, suppress noise interference, and control the direction and range of sound propagation. Based on the type of sensors used and their primary function, acoustic arrays are mainly divided into two categories:

Microphone Array: Focuses on sound reception, collection, and analysis.
Speaker Array: Focuses on sound radiation, playback, and control.

A microphone array draws inspiration from the binaural principle humans use to locate sound sources based on time differences. It achieves powerful spatial acoustic processing by synchronously sampling sound signals with multiple microphones and employing advanced signal processing techniques (primarily beamforming). The core functions of a microphone array are mainly manifested in the following three aspects:

Sound Source Localization

This function aims to determine the precise spatial coordinates of a sound event.

Since sound travels at a finite speed, the sound waves from a single source reach microphones at different positions in the array at slightly different times. This difference is called the "Time Difference of Arrival" (TDOA) or "time delay".

The core idea of beamforming is to apply adjustable "artificial time delays" to compensate each microphone channel. By adjusting these compensation values, the system attempts to align (make in-phase) the signals originating from a hypothesized direction across all microphone channels. When the signals from all channels are aligned and summed, the total output power for that direction reaches its maximum.

The system scans different points in space to find the combination of "artificial time delays" that maximizes the output power. Based on this optimal compensation combination and the known geometric structure of the microphone array, the actual spatial position of the sound source causing that time difference can be calculated. This process essentially uses time delay information for spatial inversion.
 

Key Application Scenarios:

1. Intelligent Traffic Enforcement: Whistle capture systems accurately locate vehicles violating horn regulations; illegal modified vehicle capture systems track the source position of roaring exhaust noise.

2. Industrial Equipment Monitoring: Real-time location of abnormal noise points (e.g., from bearings, gearboxes, pipes) in factories for predictive maintenance and fault diagnosis (e.g., detecting bearing wear, specific frequency noise from gas leaks).

3. Environmental Noise Monitoring: Community or urban noise monitoring systems quickly identify and locate nuisance noise sources (e.g., construction noise, entertainment venue noise), improving enforcement efficiency.


Directional Sound Pickup

This function aims to enhance the sound signal from a specific target direction while suppressing interfering noise and ambient sounds from other directions, thereby improving the signal-to-noise ratio (SNR) of the target sound.

Directional pickup also relies on beamforming technology. The system pre-sets (or dynamically tracks) the direction of the target sound source and calculates the optimal "artificial time delay" compensation values for that direction.

After applying these compensation values, sound signals from the target direction are aligned across the microphone channels and significantly enhanced through in-phase summation. Sound signals from non-target directions (including interfering noise and diffuse ambient sound), unable to be aligned by this specific set of delay compensations, undergo varying degrees of cancellation or attenuation during summation. This significantly improves the clarity and intelligibility of speech or sound from the target direction, achieving "directional focusing of sound".
 

Typical Application Scenarios:

1. Far-field Voice Interaction: Smart speakers, smart TVs, and video conferencing systems can clearly pick up user voice commands or speech from across the room (typically several meters away), reducing the impact of ambient noise.

2. High-definition Conference Recording & Speaker Separation: In meeting rooms, systems can directionally pick up sound from specific speakers (e.g., the chairperson, the current speaker), or form independent pickup beams for speakers in different locations, enabling "speaker separation" recording for clearer and more traceable meeting minutes.

3. Outdoor Professional Recording: Effectively suppresses background interference like wind noise and traffic noise in noisy outdoor environments (e.g., news reporting sites, wildlife observation) to clearly capture sound from specific target objects (e.g., interviewees, specific animals).

4. Security Surveillance: Works with cameras to directionally pick up specific sounds (e.g., abnormal cries for help, breaking glass) within a monitored area, enhancing surveillance effectiveness.


Far-field High-definition Pickup and Dereverberation

The main challenge for far-field pickup is reverberation interference. Dereverberation technology aims to eliminate or reduce reverberant sound caused by room reflections, preserving and enhancing the direct sound to improve clarity in long-distance pickup.

When microphones are far from the sound source, the strength of the direct sound signal weakens, while the energy contribution of reverberation formed by multiple reflections off walls, ceilings, floors, etc., significantly increases. Severe reverberation causes blurred speech and syllable smearing, greatly reducing speech recognition accuracy and auditory clarity.

To enhance long-distance pickup clarity, the core lies in applying dereverberation technology, which aims to effectively suppress or eliminate these harmful reverberant components while preserving and enhancing the direct sound signal from the source. A widely used and highly effective solution is Multi-Channel Linear Prediction (MCLP). The core insight of this method leverages the fundamental statistical differences, particularly in sparsity, between the real speech signal (mainly direct sound and early reflections) and late reverberation – real speech signals typically exhibit higher sparsity (more concentrated energy distribution) in the time-frequency domain than late reverberation. By analyzing the spatial correlation between multi-channel signals and reverberation characteristics, the MCLP method establishes a linear prediction model to estimate and separate the reverberant components, ultimately outputting significantly clearer speech signals.

Technical Implementation:

1. Modeling: MCLP utilizes the spatial correlation between signals from multiple microphone channels and the acoustic characteristics of reverberation (e.g., models of Room Impulse Response - RIR) to establish a linear prediction model.

2. Prediction & Separation: This model is used to predict the reverberant components in a microphone signal at the current time (primarily using past signal information). The prediction is based on the specific correlation patterns reverberation exhibits across multiple channels.

3. Estimation & Suppression: The predicted reverberant components are subtracted from the original microphone signal, resulting in the estimated, relatively clean direct sound signal (dereverberated speech).
 

Speaker array technology consists of a group of speaker units arranged in a specific geometric pattern (e.g., straight line, curve, plane) and working cooperatively. Through independent and precise control of the signal (amplitude and phase/delay) for each unit in the array, it achieves active control over the sound radiation pattern, overcoming the limitations of traditional point-source sound reinforcement. The core application functions of a speaker array are mainly reflected in two aspects: directional sound reinforcement and constant sound pressure level coverage.

Directional Sound Reinforcement

Directional sound reinforcement aims to concentrate sound energy radiation towards a specific target area or direction, reducing energy leakage and reflections into non-target areas.

Multiple speaker units forming an array are physically equivalent to increasing the effective size (aperture) of the sound source. According to acoustic principles, a larger sound source size inherently provides better directional control (i.e., more concentrated sound energy).

To achieve more precise and flexible directional control, sound field reconstruction techniques are employed. A dedicated digital filter can be designed for each speaker unit in the array. These filters independently adjust the amplitude (gain) and phase (delay) of the audio signal fed to each unit.

By precisely controlling the amplitude and phase relationship of the signals for each unit, the overall sound waves generated by the array can be guided to undergo constructive interference (reinforcement) and destructive interference (cancellation) in space. This precisely "steers" or "focuses" the main lobe of the sound wave (the direction of strongest energy) towards the desired target direction.
 

Effects & Applications:

1. Energy Focusing: Efficiently projects sound energy to specific areas (e.g., audience seating), avoiding energy waste on non-target areas like walls and ceilings. Improves sound reinforcement efficiency and reduces reverberation interference.

2. Zoned Sound Reinforcement: In large open spaces (e.g., museums, exhibition halls, airport terminals), plays different audio content for different zones (e.g., in front of different exhibits, at different boarding gates) with minimal mutual interference.

3. Avoiding Noise Pollution: Enables directional announcements (e.g., public square notices, bus stop announcements) near noise-sensitive areas (e.g., libraries, hospital wards, residential zones), strictly confining the sound to the target area without affecting adjacent quiet zones.

4. Creating Private Audio Zones: Forms localized audible areas at specific points (e.g., exhibit explanation points, information kiosks), while adjacent areas hear almost no sound.



Constant Sound Pressure Level Coverage

Constant Sound Pressure Level Coverage aims to solve the problem of rapid sound pressure level (SPL) attenuation with distance in traditional point-source sound systems. It achieves uniform SPL distribution over a large longitudinal depth (from front to back rows), ensuring consistent volume for all listeners.

According to sound wave propagation in a free field (inverse square law), SPL from a point source decreases by approximately 6 dB when propagation distance doubles. This causes front-row listeners in deep venues (e.g., theaters, churches, auditoriums, stadiums) to perceive sound as excessively loud, while back-row listeners find it too soft.
 

A linear array structure effectively solves longitudinal sound field uniformity. Its core lies in independent control of amplitude (volume) and phase (delay) for each speaker unit. Through precise amplitude and phase adjustment, it uses sound wave interference to achieve near-constant total sound pressure over longitudinal depth. The specific strategy is:

1. Constructive Summation at Far End (Back Rows): Precisely control the phase relationship of sound waves arriving at distant positions (e.g., back rows) to make them as in-phase as possible. In-phase waves sum constructively, significantly boosting SPL to compensate for natural attenuation.

2. Destructive Cancellation at Near End (Front Rows): Precisely control the phase relationship of sound waves arriving at close positions (e.g., front rows) to make them partially out-of-phase. Out-of-phase waves cancel destructively, reducing SPL in this area.

By carefully designing amplitude weighting and phase delay for each unit (upper units have higher output and less delay; lower units have lower output and more delay), sound wave interference maintains near-constant total SPL over large longitudinal distances from near to far. This overcomes the inverse square law limitation. Widely used in large venues (e.g., performance halls, theaters, churches, conference centers, stadiums, train stations) requiring uniform coverage in depth, ensuring clear, comfortable, and consistent volume for all audience members.
 

Acoustic phased array technology arranges multiple acoustic sensors (microphones or speaker units) in a specific geometric structure to achieve functions that a single sensor cannot accomplish. Its core is divided into two categories: microphone array and speaker array. Microphone array uses beamforming technology to process signals collected synchronously by multiple microphones. Speaker array reshapes the sound field radiation by independently and accurately controlling the amplitude and phase (delay) of each unit signal in the array.

Are you experiencing the problem of uneven sound field distribution? Need to achieve precise directional sound reinforcement? Welcome to contact us for customized solutions.