Space orientation active noise reduction system and method
The active noise reduction system, which utilizes a linear array arrangement and adaptive filtering algorithm, solves the problems of high cost, high complexity, and poor scene adaptability of existing systems. It achieves efficient noise reduction and low-frequency noise control in specific directions, making it easy to apply flexibly in various scenarios.
Patent Information
- Authority / Receiving Office
- CN · China
- Patent Type
- Applications(China)
- Current Assignee / Owner
- WUHAN UNIV OF TECH
- Filing Date
- 2026-02-05
- Publication Date
- 2026-05-08
AI Technical Summary
Existing active noise cancellation systems suffer from high hardware costs, high system complexity, difficulty in flexible deployment, and poor adaptability to different scenarios in practical applications, especially in open spaces where it is difficult to achieve ideal low-frequency noise control.
By employing a linear array arrangement, a reference microphone, an error microphone, and a secondary speaker are linearly arranged. Combined with an adaptive filtering algorithm and a DSP digital signal processor, a secondary noise control signal is generated to achieve directional noise reduction. The system has a simple structure and is easy to integrate and move.
It achieves efficient noise reduction in a specific direction, reduces hardware costs, simplifies system structure, facilitates flexible deployment in different scenarios, adapts to different noise environments, and has good low-frequency noise control capabilities.
Smart Images
Figure CN121999751A_ABST
Abstract
Description
Technical Field
[0001] This invention relates to the field of noise control technology, specifically to a spatially oriented active noise reduction system and method. Background Technology
[0002] Noise pollution has become one of the major environmental problems facing modern society. Traditional passive noise reduction technologies mainly reduce noise through sound-absorbing materials and sound insulation structures. These methods are effective for high-frequency noise, but have limited control over low-frequency noise and often require large spaces and are costly. In contrast, active noise control (ANC) technology uses loudspeakers to emit sound waves with the same amplitude but opposite phase to the noise to be controlled. It utilizes the principle of destructive interference of sound waves to actively cancel out the noise, and has significant advantages in low-frequency noise control.
[0003] However, existing active noise cancellation systems still have many limitations in practical applications. First, traditional systems often aim for full-space noise reduction, which requires multiple microphones and speaker arrays, increasing hardware costs and system complexity, and making it difficult to achieve ideal noise reduction in open spaces. Second, existing systems are usually large and complex, making them difficult to deploy and move flexibly, and unable to meet users' noise reduction needs in different scenarios. Third, most active noise cancellation systems are designed for specific application scenarios, resulting in poor scenario adaptability and difficulty in achieving good noise reduction effects in diverse scenarios such as factory workshops, medical environments, transportation vehicles, and residential areas.
[0004] Therefore, there is an urgent need for an active noise cancellation technology solution that is simple in structure, moderate in cost, easy to deploy, and has good adaptability to different scenarios, in order to meet the needs of directional noise control in practical applications. Summary of the Invention
[0005] To address the aforementioned technical problems, this invention provides a spatially oriented active noise reduction system and method, which achieves efficient noise reduction in a specific direction through a linear array arrangement. It has advantages such as simple system structure, easy integration, and strong scene adaptability.
[0006] To achieve the above objectives, the present invention adopts the following technical solution: A spatially oriented active noise reduction system includes a signal input module, a signal processing module, and a signal output module; The signal input module includes a reference microphone for acquiring target noise signals and an error microphone for acquiring residual noise signals; The signal processing module includes an analog-to-digital converter, a DSP digital signal processor, a digital-to-analog converter, and a power amplifier; The DSP digital signal processor processes the input target noise signal and residual noise signal based on an adaptive filtering algorithm to generate a secondary noise control signal; The signal output module includes a secondary speaker that emits secondary sound waves; The reference microphone, error microphone, and secondary speaker are arranged along the same straight line, and the secondary sound wave propagates along this straight line and cancels out the target noise to achieve noise reduction.
[0007] Preferably, the signal input module further includes a movable error microphone, which is electrically connected to the DSP digital signal processor via a retractable cable. The residual noise signal collected by the movable error microphone is converted by the analog-to-digital converter and then input to the DSP digital signal processor.
[0008] Preferably, a secondary speaker and a reference microphone are arranged in the same straight line direction; the error microphone is located between the reference microphone and the secondary speaker.
[0009] Preferably, the sound wave emission direction of the secondary loudspeaker is the same as the sound wave propagation direction of the noise source.
[0010] Preferably, the signal processing module further includes a filter, which performs filtering processing on the signals acquired by the reference microphone and the error microphone.
[0011] Preferably, the DSP digital signal processor uses an ADSP-21489 chip, and the signal processing module further includes an audio interface chip and an I2S serial interface that are compatible with the ADSP-21489 chip.
[0012] Preferably, both the reference microphone and the error microphone employ a MAX9814 audio automatic gain control module, the input sound pressure level of which is in the range of 35dB to 86dB.
[0013] Preferably, the adaptive filtering algorithm is a minimum mean square filtering algorithm.
[0014] This invention also provides a spatially oriented active noise reduction method, applicable to the above-mentioned system, comprising the following steps: Step S1: Acquire the target noise signal through the reference microphone and the residual noise signal through the error microphone; Step S2: Convert the target noise signal and the residual noise signal into digital signals using an analog-to-digital converter and input them into a DSP digital signal processor; Step S3: The DSP digital signal processor processes the digital signal based on an adaptive filtering algorithm to generate a secondary noise control signal; Step S4: Convert the secondary noise control signal into an analog signal using a digital-to-analog converter, amplify it using a power amplifier, and then output it to the secondary speaker; Step S5: The secondary speaker emits a secondary sound wave, which cancels out the target noise to achieve noise reduction; wherein the reference microphone, error microphone, and secondary speaker are arranged along the same straight line, and the secondary sound wave propagates along this straight line.
[0015] Preferably, in step S2, the target noise signal and the residual noise signal are filtered by a filter before being converted by the input analog-to-digital converter, and then amplified by a power amplifier.
[0016] Compared with the prior art, the beneficial effects of the present invention include at least the following: First, by employing a linear array arrangement, the effective range of active noise reduction is focused on a specific straight line, avoiding the complexity of traditional full-space noise reduction systems, significantly simplifying the system structure, and reducing hardware costs. This directional noise reduction strategy is particularly suitable for application scenarios with clearly defined noise source directions, enabling effective noise reduction over a large spatial area with fewer hardware resources.
[0017] Secondly, the system adopts a DSP-based digital signal processing architecture with highly integrated hardware modules. The entire system can be assembled in a small device, facilitating flexible deployment and mobility in different scenarios. The design of a movable error microphone further enhances the system's flexibility, allowing users to place it at the desired spatial location according to actual needs and monitor the noise reduction effect at that location in real time.
[0018] Third, the system employs an adaptive filtering algorithm that can automatically track changes in noise characteristics and adapt to different acoustic environments and noise types. This allows the system to achieve good noise reduction effects in various scenarios such as factory workshops, medical equipment rooms, car interiors, and residential buildings without the need for complex pre-tuning for specific scenarios, demonstrating broad application prospects.
[0019] Fourth, compared to traditional passive noise reduction techniques, this invention has better control over low-frequency noise and can achieve noise suppression over a wide frequency range. At the same time, the directional noise reduction strategy avoids interference with sound fields in non-target directions, offering unique advantages in applications where maintaining sound clarity in certain directions is required. Attached Figure Description
[0020] Figure 1 This is a structural block diagram of a spatially oriented active noise reduction system according to an embodiment of the present invention; Figure 2 This is a flowchart of a spatially oriented active noise reduction method according to an embodiment of the present invention. Detailed Implementation
[0021] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the protection scope of the present invention.
[0022] Example 1 like Figure 1 As shown, this embodiment provides a spatially oriented active noise reduction system. The system has a secondary speaker, an error microphone, and a reference microphone arranged sequentially in a straight line. This linear array configuration enables active control of noise propagating in a specific direction.
[0023] The system mainly consists of three functional modules: the signal input module is responsible for acquiring noise signals, the signal processing module is responsible for processing signals, and the signal output module is responsible for generating secondary sound waves.
[0024] In the signal input module, a reference microphone is positioned at the rear of the system, with its acoustic axis aligned with the noise source, for directional acquisition of the target noise signal to be controlled. The reference microphone utilizes the MAX9814 audio automatic gain control module, which integrates microphone bias circuitry, a low-noise preamplifier, automatic gain control circuitry, and output driver circuitry within a compact 41mm × 49mm size. The MAX9814 module supports a wide input sound pressure level range of 35dB to 86dB, and its fast automatic gain response ensures accurate capture of signal characteristics even during sudden changes in noise intensity. The analog audio signal acquired by the reference microphone is first filtered for frequency band selection and noise suppression, then moderately amplified by a power amplifier, and finally converted into a digital signal by an analog-to-digital converter before being sent to the DSP processor.
[0025] The error microphone is located on the straight line between the reference microphone and the secondary speaker, and is arranged collinearly with the reference microphone. The error microphone also uses the MAX9814 module, and its main function is to collect the residual noise signal after noise reduction, providing error feedback for the adaptive algorithm. When the noise reduction system is running, the sound field at the location of the error microphone is formed by the superposition of the original noise and the canceling sound waves emitted by the secondary speaker; the error signal reflects the quality of the current noise reduction effect. The signal collected by the error microphone is also filtered, amplified, and converted from analog to digital before being sent to the DSP processor.
[0026] To enhance system flexibility, this embodiment also includes a movable error microphone. This microphone is connected to the DSP processor via a 10-meter retractable cable, allowing the user to move it to a location of interest and monitor the actual noise reduction effect at that location. For example, in an office environment, a user might want to achieve optimal noise reduction at their workstation; in this case, the movable microphone can be placed at that location, and the system will optimize control parameters based on the residual noise feedback from that location. The signal processing link for the movable error microphone is the same as that for the fixed error microphone, and its output digital signal is input to the DSP processor as additional feedback information.
[0027] The signal processing module is the core of the entire system. Its main components include the ADSP-21489 digital signal processor, audio interface chip, I2S serial interface, analog-to-digital / digital-to-analog converter, and matching power amplifier circuit. The ADSP-21489 is a high-performance floating-point DSP chip capable of meeting the computational requirements of real-time audio signal processing. The active noise control algorithm embedded in this chip adopts the minimum mean square (FxLMS) filtering algorithm framework.
[0028] The basic principle of the FxLMS algorithm is as follows: First, a preliminary control signal is generated by passing a reference microphone signal through an adaptive filter. This control signal is then passed through an auxiliary filter and compared with the error microphone signal to calculate the error signal. Next, the coefficients of the adaptive filter are adjusted based on the error signal to gradually reduce the mean square value of the error signal, ultimately minimizing residual noise. In this embodiment, the transfer function of the auxiliary filter is obtained by offline measurement of the acoustic transfer characteristics from the secondary speaker to the error microphone and is pre-stored in the DSP processor. During system operation, the coefficients of the adaptive filter are continuously updated according to the error signal, enabling the system to automatically adapt to changes in noise characteristics and the acoustic environment.
[0029] The secondary noise control signal generated by the DSP processor is a digital audio stream. This signal is converted into an analog voltage signal by a digital-to-analog converter and then amplified by a power amplifier to a power level sufficient to drive the speaker. The gain of the power amplifier is designed based on the sensitivity of the secondary speaker and the expected noise reduction, ensuring that the secondary sound wave has a sufficient sound pressure level to cancel out the target noise.
[0030] The core component of the signal output module is the secondary speaker, which is installed at the front end of the system. Its sound wave radiation direction is consistent with the sound wave propagation direction of the noise source, and it is aligned with the reference microphone and error microphone. The secondary speaker receives amplified secondary control signals, generating secondary sound waves in space that have the same frequency and amplitude as the target noise but are out of phase. These out-of-phase sound waves propagate along the straight line of the system layout and undergo acoustic interference when they meet the target noise propagating in the same direction. According to the principle of wave superposition, the two out-of-phase sound waves superimpose in space, resulting in destructive interference and a significant reduction in the amplitude of the synthesized sound wave, thus achieving noise reduction.
[0031] In this linear array configuration, the system's noise reduction effect forms a narrow, elongated quiet zone along a straight line. The length and width of the quiet zone depend on the noise frequency, the intensity of the secondary sound source, and the distance between the reference microphone and the secondary speaker. For low-frequency noise, due to the longer wavelength and weaker directivity of the sound wave, directional noise reduction can cover a relatively large spatial range; while for high-frequency noise, although the noise reduction area is more concentrated, good noise reduction effect can still be achieved in the target direction.
[0032] The system in this embodiment can be mounted in a compact handheld or desktop device. Thanks to the highly integrated hardware modules, the overall system size can be kept small, making it easy to carry and deploy. Users can adjust the system's orientation according to the location of the noise source, ensuring the reference microphone is pointed at the noise source and the secondary speaker is oriented towards the area requiring noise reduction, thereby achieving the best noise reduction effect.
[0033] Example 2 This embodiment provides a spatially oriented active noise reduction method based on a spatially oriented active noise reduction system according to Embodiment 1, such as... Figure 2 As shown, the method includes the following steps: Step S1: After system startup, the host computer sends a noise reduction command to the DSP processor to initialize the working status of each module. The reference microphone begins to directionally acquire noise signals within the target space, continuously monitoring changes in the ambient sound field. The output of the reference microphone is a time-varying audio signal containing the time and frequency domain characteristics of the noise to be controlled. Simultaneously, the error microphone and the movable error microphone also begin acquiring sound field signals at their respective locations, preparing for subsequent adaptive processing.
[0034] Step S2: The signals acquired by the reference microphone and error microphone are first preprocessed through filters. The filter design is determined based on the frequency characteristics of the target noise. For example, for low-frequency noise control, a high-pass filter can be used to filter out DC components and extremely low-frequency interference; for noise in a specific frequency band, a band-pass filter can be used to select the target frequency band. The filtered signal is then amplified by a power amplifier to adjust the amplitude, increasing the weak microphone output signal to a level suitable for the analog-to-digital converter (ADC) input requirements. The amplified analog signal is then converted into a digital sampling sequence by a high-precision ADC. The converted digital signal is transmitted to the internal memory of the DSP processor via the audio interface and I2S bus.
[0035] Step S3: The FxLMS algorithm embedded in the DSP processor performs real-time processing on the received reference noise signal and error noise signal. The algorithm first convolves the reference signal with the current adaptive filter coefficients to generate a preliminary control signal. Then, this control signal is filtered through a pre-established auxiliary filter to obtain the expected secondary acoustic signal at the error microphone location. Next, the expected signal is compared with the residual noise signal actually acquired by the error microphone to calculate the error value. Based on this error value, the algorithm updates the coefficients of the adaptive filter using the least mean square criterion, ensuring that the control signal at the next moment can better cancel the target noise. This iterative process is performed continuously with the audio sampling period as the time step, enabling the system to track changes in noise characteristics in real time and dynamically optimize the noise reduction effect. After convergence, the coefficients of the adaptive filter will stabilize in an optimal or near-optimal state, at which point the generated secondary control signal can cancel the target noise to the greatest extent.
[0036] Step S4: The secondary control signal output by the DSP processor is converted back to the analog domain by a digital-to-analog converter (DAC), becoming a continuous analog voltage signal. Since the output level of the DAC is typically low and insufficient to directly drive the speaker, it needs to be amplified by a power amplifier. The power amplifier amplifies the weak voltage signal into an audio signal with sufficient power, which is then output to the secondary speaker. The amplifier design must ensure linearity and low distortion during signal amplification to ensure that the secondary sound wave accurately reproduces the waveform characteristics calculated by the DSP.
[0037] Step S5: The secondary speaker receives the amplified control signal and converts the electrical signal into sound waves that propagate in space. Since the secondary speaker is positioned in the same direction as the noise source propagation, and its emitted sound waves are out of phase with the target noise, destructive interference occurs when the two sound waves meet in space. Within the interference region, the amplitude superposition of the sound waves approaches zero, significantly reducing the noise level in that area. The actual noise reduction effect can be monitored in real time using the signal from the error microphone. If the residual noise remains high, the adaptive algorithm will continue to adjust the filter parameters to further optimize the noise reduction performance. This closed-loop feedback control process enables the system to maintain a consistently good noise reduction effect, automatically adjusting and adapting even when the characteristics of the noise source change or the environmental acoustic conditions change.
[0038] In practical applications, users can adjust system parameters according to specific needs. For example, when the noise frequency is known, the frequency response of the filter can be optimized to improve the noise reduction effect in a specific frequency band; in scenarios requiring coverage of a larger space, the power of the secondary speakers can be increased or an array configuration can be used. Furthermore, by changing the distance between the reference microphone and the secondary speakers, the causality constraint and noise reduction bandwidth of the system can be adjusted to adapt to different types of noise control requirements.
[0039] This embodiment of the method fully leverages the advantages of directional noise reduction, focusing the noise reduction effect in a specific direction, thus avoiding the complexity and high cost of traditional all-space noise reduction systems. This method is suitable for scenarios where the location and direction of noise sources are relatively clear, such as work areas next to factory assembly lines, residential buildings along roads, specific wards in hospitals, and the passenger space inside cars. By rationally configuring the spatial layout of the system, significant noise reduction effects can be achieved at a lower cost in these scenarios, improving the quality of the acoustic environment.
[0040] The technical features of the above embodiments can be combined arbitrarily. For the sake of brevity, not all possible combinations of the technical features in the above embodiments are described; only preferred embodiments of the present invention are illustrated. The descriptions are relatively specific and detailed, but they should not be construed as limiting the scope of the present invention. As long as the combination of these technical features does not contradict each other, it should be considered within the scope of this specification.
[0041] It should be noted that those skilled in the art can make various modifications and improvements without departing from the inventive concept, and these all fall within the scope of protection of this invention. Therefore, the scope of protection of this invention should be determined by the appended claims.
Claims
1. A spatially oriented active noise reduction system, characterized in that: It includes a signal input module, a signal processing module, and a signal output module; The signal input module includes a reference microphone for acquiring target noise signals and an error microphone for acquiring residual noise signals; The signal processing module includes an analog-to-digital converter, a DSP digital signal processor, a digital-to-analog converter, and a power amplifier; The DSP digital signal processor processes the input target noise signal and residual noise signal based on an adaptive filtering algorithm to generate a secondary noise control signal; The signal output module includes a secondary speaker that emits secondary sound waves; The reference microphone, error microphone, and secondary speaker are arranged along the same straight line, and the secondary sound wave propagates along this straight line and cancels out the target noise to achieve noise reduction.
2. The spatially oriented active noise reduction system according to claim 1, characterized in that: The signal input module also includes a movable error microphone, which is electrically connected to the DSP digital signal processor via a retractable cable. The residual noise signal collected by the movable error microphone is converted by the analog-to-digital converter and then input to the DSP digital signal processor.
3. The spatially oriented active noise reduction system according to claim 1, characterized in that: A secondary speaker and a reference microphone are arranged in the same straight line direction; the error microphone is located between the reference microphone and the secondary speaker.
4. The spatially oriented active noise reduction system according to claim 1, characterized in that: The sound wave emission direction of the secondary loudspeaker is the same as the sound wave propagation direction of the noise source.
5. A spatially oriented active noise reduction system according to claim 1, characterized in that: The signal processing module further includes a filter, which filters the signals acquired by the reference microphone and the error microphone.
6. The spatially oriented active noise reduction system according to claim 1, characterized in that: The DSP digital signal processor uses the ADSP-21489 chip, and the signal processing module also includes an audio interface chip and an I2S serial interface that are compatible with the ADSP-21489 chip.
7. The spatially oriented active noise reduction system according to claim 1, characterized in that: Both the reference microphone and the error microphone use the MAX9814 audio automatic gain control module, and the input sound pressure level range of the MAX9814 audio automatic gain control module is 35dB to 86dB.
8. A spatially oriented active noise reduction system according to claim 1, characterized in that: The adaptive filtering algorithm is the minimum mean square filtering algorithm.
9. A spatially oriented active noise reduction method, applicable to the spatially oriented active noise reduction system described in any one of claims 1 to 8, characterized in that: Includes the following steps: Step S1: Acquire the target noise signal through the reference microphone and the residual noise signal through the error microphone; Step S2: Convert the target noise signal and the residual noise signal into digital signals using an analog-to-digital converter and input them into a DSP digital signal processor; Step S3: The DSP digital signal processor processes the digital signal based on an adaptive filtering algorithm to generate a secondary noise control signal; Step S4: Convert the secondary noise control signal into an analog signal using a digital-to-analog converter, amplify it using a power amplifier, and then output it to the secondary speaker; Step S5: The secondary speaker emits a secondary sound wave, which cancels out the target noise to achieve noise reduction; wherein the reference microphone, error microphone and secondary speaker are arranged along the same straight line, and the secondary sound wave propagates along this straight line.
10. A spatially oriented active noise reduction method according to claim 9, characterized in that: In step S2, the target noise signal and the residual noise signal are filtered by a filter and then amplified by a power amplifier before being converted by the input analog-to-digital converter.