Low-noise design method for an acoustic fan based on fan noise

Abstract

This invention relates to the field of fan, speaker, and light design and control technology, and discloses a low-noise design method for a speaker fan based on fan noise. The method includes receiving the original audio signal, generating an audio power spectrum, and extracting low-frequency transient features; calculating the instantaneous masking threshold using a psychoacoustic model, and then inversely shaping the allowable noise power spectrum of the fan motor to generate a target fan spectrum; generating a fan drive signal based on the target spectrum using a random spread spectrum algorithm, and constructing an airflow velocity vector by combining the angular velocity fed back from the motor and the duct parameters; calculating the phase compensation amount based on the airflow velocity vector to generate a phase correction signal, and then combining the low-frequency transient features with an anti-phase aerodynamic impedance compensation component to synthesize a speaker signal; finally, driving the fan motor and speaker unit to operate synchronously. This invention utilizes the auditory masking effect to achieve invisible fan operation and cancels the interference of airflow on sound waves through flow field mapping and phase pre-distortion, achieving synergistic optimization of the sound field and airflow field.

Description

Technical Field

[0001] This invention relates to the field of fan, speaker, and light design and control technology, specifically a low-noise design method for speaker fans based on fan noise. Background Technology

[0002] An audio fan light is a composite household appliance that integrates an air circulation component and an audio playback component. The speaker unit is typically located at the center of the fan motor's shaft or at the base of the light fixture. When this device is in operation, the rotation of the fan blades not only regulates indoor airflow but also creates the physical environment for the acoustic reproduction system.

[0003] In practical applications, the electromagnetic noise generated by the fan motor and the aerodynamic noise generated by the blades cutting through the air are objective sources of interference. Existing control schemes typically operate fan speed control and audio playback as two separate logics, with the fan maintaining a specific speed only based on the user-set speed or ambient temperature. This control method does not consider the energy distribution characteristics of the currently played audio content, resulting in the background noise energy of the fan potentially exceeding the audio signal energy when the audio signal is in a low-level segment or a sparse frequency range. This significantly reduces the signal-to-noise ratio at the human ear, causing musical details to be masked by noise.

[0004] Meanwhile, because the speaker unit of the audio fan light is located close to the rotating fan blades, the sound waves emitted by the speaker must pass through the high-speed turbulent flow field created by the blades to reach the listening area. The uneven distribution of air velocity and its dynamic changes over time will have a nonlinear modulation effect on sound wave propagation, causing refraction of the sound wave front or propagation delay. Existing designs do not establish an acoustic compensation mechanism for this dynamic flow field environment, resulting in phase shift or frequency response distortion when the sound waves reach the listening point, affecting the clarity of sound reproduction.

[0005] Furthermore, the continuous airflow generated by the fan and the sound waves generated by the speaker diaphragm pushing the air are physically pressure-coupled. In the absence of coordinated control, the unsteady airflow pressure from the fan may act on the speaker diaphragm surface, altering its equivalent compliance and leading to unstable diaphragm motion. Existing driving methods fail to achieve time-domain synchronization between the fan airflow pulses and the speaker's low-frequency vibrations. This not only fails to utilize the physical energy of airflow to enhance low-frequency perception but may also cause mutual cancellation due to phase mismatch, affecting the overall performance of the device. Summary of the Invention

[0006] To address the shortcomings of existing technologies, this invention provides a low-noise design method for audio fans based on fan noise, which solves the problems in existing technologies such as fan operating noise reducing the audio signal-to-noise ratio at low volumes, fan dynamic flow field altering sound wave propagation characteristics leading to phase distortion, and lack of coordination between fan airflow and speaker vibration causing mutual interference of physical fields.

[0007] To achieve the above objectives, the present invention provides the following technical solution: This invention provides a low-noise design method for audio fans based on fan noise. This method establishes a closed-loop coupling control between audio signals, fan drive, and airflow field through a digital processor.

[0008] First, the digital processor receives the raw audio signal, performs time-domain buffering and short-time Fourier transform on it to generate the audio power spectrum, and simultaneously performs time-domain envelope transient detection to extract low-frequency transient features.

[0009] Next, the digital processor retrieves the audio power spectrum and combines it with the psychoacoustic model to calculate the instantaneous masking threshold according to the masking threshold formula. This masking threshold formula is mainly calculated based on the audio power spectrum, the human ear's critical signal-to-noise ratio constant, and the frequency domain masking spread function value, and is used to quantify the ability of the audio signal to mask background noise at the current moment.

[0010] Based on the calculated instantaneous masking threshold, the permissible noise spectrum of the fan motor across the entire frequency band is reverse-shaped. That is, the amplitude data of the instantaneous masking threshold is used as the upper limit of the noise amplitude, and the fan noise energy is controlled to be lower than the upper limit, thereby generating the fan target spectrum that specifies the maximum permissible noise amplitude at each frequency point.

[0011] Subsequently, a random spread spectrum algorithm is used to generate the fan drive signal based on the fan target spectrum. This process involves analyzing the fan target spectrum to determine the currently allowed fan fundamental frequency. When the discrete spectral energy generated by the fan fundamental frequency exceeds the limit of the fan target spectrum, the random spread spectrum algorithm is activated. This algorithm uses a pseudo-random sequence to modulate the pulse width modulation signal, dispersing the spectral energy across a wide bandwidth and placing it below the envelope of the fan target spectrum.

[0012] Simultaneously, the instantaneous angular velocity is collected using the motor feedback loop, and an air velocity vector is constructed based on the flow field vector formula, combined with the fluid dynamics model and duct geometric parameters. This flow field vector formula is mainly calculated based on the fluid conversion efficiency coefficient of the fan system, the instantaneous angular velocity of the fan motor, and the duct geometric parameters determined by the spatial position, and is used to reconstruct the air movement state in the space in front of the speaker.

[0013] Furthermore, the phase compensation amount is calculated based on the air velocity vector and the phase correction formula, and a phase correction signal is generated. This phase correction formula is mainly calculated based on the frequency variable, the distance from the sound source to the listening point, the sound speed in still air, and the angle between the sound wave propagation direction and the air velocity vector, and is used to obtain the phase shift corresponding to the time difference of sound wave propagation in still air and dynamic airflow.

[0014] The digital processor uses this phase compensation amount to apply an inverse time shift to the high-frequency components of the original audio signal. Based on this, it performs a superposition operation to synthesize the speaker signal by combining low-frequency transient characteristics, that is, generating an inverse aerodynamic impedance compensation component whose phase is synchronized with the airflow pressure pulse of the fan motor, and superimposing it with the phase correction signal.

[0015] Finally, the digital-to-analog converter amplifier circuit amplifies the fan drive signal and the speaker signal, driving the fan motor and speaker unit to operate synchronously. The fan motor rotates according to the pulse current to generate controlled airflow, and the speaker unit drives the diaphragm to vibrate according to the large current signal, generating a sound field and an airflow field in physical space.

[0016] This invention provides a low-noise design method for an audio fan based on fan noise. It has the following beneficial effects: 1. This invention calculates the instantaneous masking threshold of the audio signal in real time using a psychoacoustic model, and dynamically constrains the noise power spectrum of the fan motor accordingly. Combined with a random spread spectrum algorithm, the narrowband noise energy of the fan motor is dispersed to a wideband, making its amplitude lower than the masking threshold of the current audio signal. This ensures that the fan's operating noise is effectively masked by the audio signal during audio playback, maintaining the fan's air circulation function while avoiding mechanical noise from reducing the signal-to-noise ratio of audio playback.

[0017] 2. This invention establishes a mapping relationship between fan speed and air velocity vector in front of the speaker, and applies phase pre-correction to the audio signal based on flow field data. By calculating the propagation delay difference of sound waves in dynamic airflow, time shift compensation is performed on high-frequency components in advance during the digital signal processing stage to offset the refraction and delay effects of turbulence generated by the fan blades and downward airflow on sound wave propagation, correct the phase of the sound wave reaching the listening position, and maintain the stability of the sound frequency response.

[0018] 3. This invention utilizes low-frequency transient characteristics to achieve coordinated control of speaker diaphragm movement and fan airflow pulses. By generating an anti-phase aerodynamic impedance compensation component synchronized with the low-frequency impact, the speaker is driven, ensuring that the diaphragm's motion state is phase-synchronized with the airflow pressure changes generated by the fan. The coordinated driving method uses controlled airflow as the sound wave transmission carrier, while reducing the pressure interference of airflow on the speaker surface through diaphragm compensation movement, thereby enhancing the transmission efficiency of low-frequency signals in physical space. Attached Figure Description

[0019] Figure 1 This is a flowchart of the low-noise design method for an audio fan based on fan noise according to the present invention; Figure 2 This is a flowchart of the fan drive generation and real-time airflow field vector modeling process of the present invention; Figure 3 This is a flowchart of the acoustic phase predistortion and impedance compensation based on flow field mapping according to the present invention. Detailed Implementation

[0020] The technical solutions in 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 scope of protection of the present invention.

[0021] Please see the appendix Figure 1 This invention provides a low-noise design method for an audio fan based on fan noise, comprising the following steps: Step S1: The digital processor receives the raw audio signal from the external input and performs a time-domain buffering operation on it to construct an audio data frame containing a preset duration. The digital processor performs a short-time Fourier transform on the audio data frame to convert the time-domain signal into a frequency-domain representation, obtaining the audio power spectrum at the current moment. Simultaneously, time-domain envelope transient detection is performed on the input raw audio signal to extract low-frequency transient features.

[0022] Step S2: The digital processor retrieves the audio power spectrum, combines it with a preset psychoacoustic model, and calculates the frequency at the current moment using the masking threshold formula. The instantaneous masking threshold at a given location. The formula for the masking threshold is: ; In the formula: For a moment frequency The instantaneous masking threshold at the location; For a moment frequency The audio power spectrum at that location; For frequency The critical signal-to-noise ratio constant of the human ear at that location; For frequency The frequency domain masking spread function value at that location.

[0023] The digital processor performs reverse shaping on the full-band permissible noise spectrum of the fan motor based on the calculated instantaneous masking threshold, generating the fan target spectrum.

[0024] Step S3: Based on the target spectrum of the fan, the digital processor performs spectrum shaping using a random spread spectrum algorithm to generate the fan drive signal. The digital processor acquires the fan motor's operating status in real time through the motor feedback loop, obtaining the instantaneous angular velocity of the fan motor. Using the acquired instantaneous angular velocity and preset duct geometry parameters, the digital processor calculates the air velocity vector in the space in front of the speaker by substituting the flow field vector formula into a fluid dynamics model. The flow field vector formula is: ; In the formula: For a moment In spatial location Air velocity vector at that location; The fluid conversion efficiency coefficient of the fan system; For a moment The instantaneous angular velocity of the fan motor; For spatial location The determined duct geometry parameters.

[0025] Step S4: The digital processor analyzes the propagation delay characteristics of sound waves in the dynamic flow field based on the air velocity vector, substitutes the values ​​into the phase correction formula, and calculates the phase compensation amount. The phase correction formula is: ; In the formula: For a moment frequency The amount of phase compensation that needs to be applied; The straight-line distance from the center of the sound source to the preset listening point; The speed of sound in still air is the constant. The magnitude of the airflow velocity vector; These are cosine trigonometric functions; It is the angle between the direction of sound wave propagation and the air velocity vector.

[0026] The digital processor applies the calculated phase compensation amount to the original audio signal to generate a phase-corrected signal. Combining this with low-frequency transient characteristics, the digital processor performs a superposition operation on the phase-corrected signal to synthesize the final speaker signal.

[0027] Step S5: Receive the fan drive signal and speaker signal using a digital-to-analog converter power amplifier circuit, amplify them respectively, and output the amplified drive current. Drive the fan motor and speaker unit to operate synchronously, generating controlled airflow and sound waves in the physical space, forming an optimized sound field and airflow field.

[0028] Step S1: The digital processor performs buffering processing and multi-dimensional feature extraction of the audio signal. This process is specifically implemented through the following steps: Step S101: The digital processor receives the raw audio signal from the external input, performs a time-domain buffering operation on it, and constructs an audio data frame containing a preset duration.

[0029] Specifically, the digital processor receives raw audio signals from audio sources such as Bluetooth decoders, local players, or analog input interfaces via direct memory access (DMA) or interrupts. The raw audio signals are typically discrete sample sequences in Pulse Code Modulation (PCM) format.

[0030] The digital processor allocates a circular buffer in its internal memory and writes the continuously arriving sampling points into the buffer sequentially.

[0031] The above preset duration is set based on the real-time requirements and frequency resolution requirements of audio signal processing. The preset duration is usually set to 10 milliseconds to 50 milliseconds (e.g., 23.2 milliseconds, corresponding to 1024 sampling points at a sampling rate of 44.1kHz).

[0032] The digital processor extracts data from the circular buffer according to a preset frame shift step size to construct the current audio data frame. To ensure the continuity of the signal in the time domain and avoid boundary effects, an overlapping region is set between adjacent audio data frames, with the overlap ratio typically set to 50% or 75%.

[0033] Step S102: The digital processor performs a short-time Fourier transform on the audio data frame to convert the time-domain signal into a frequency-domain representation, thereby obtaining the audio power spectrum at the current moment.

[0034] Before performing the transformation, the digital processor performs a time-domain dot product of the audio data frame with the window function to reduce spectral leakage. For the selection of the window function, those skilled in the art can choose the Hanning window, Hamming window, or Blackman window according to actual needs; the principles of these are well-known in the field and will not be elaborated upon here.

[0035] The windowed data is fed into a Fast Fourier Transform (FFT) unit to calculate the complex spectrum. The digital processor further calculates the square of the modulus of each frequency component in the complex spectrum to obtain the audio power spectrum at the current moment. This audio power spectrum is stored as a two-dimensional array, representing the energy distribution at each frequency point from the DC component to the Nyquist frequency within the current time frame. This audio power spectrum serves as the basis for calculating the instantaneous masking threshold in subsequent steps, reflecting the energy concentration region of the audio signal in the frequency domain.

[0036] Step S103: The digital processor synchronously performs time-domain envelope transient detection on the input raw audio signal to extract low-frequency transient features.

[0037] This step aims to capture low-frequency components in the music signal that possess high-energy impact characteristics (such as the initial moment of a kick drum or bass string plucking) for subsequent triggering of aerodynamic impedance compensation. The digital processor first passes the raw audio signal through a digital low-pass filter to filter out mid-to-high frequency components, retaining the low-frequency components. The cutoff frequency of this low-pass filter is set based on the upper limit of the frequency at which the fan airflow pulse can effectively respond, for example, set to 150Hz or 200Hz.

[0038] Subsequently, the digital processor extracts the envelope of the filtered low-frequency signal. Envelope extraction can be achieved using Hilbert transform or full-wave rectification followed by smoothing filtering to obtain the instantaneous amplitude envelope of the low-frequency signal.

[0039] After obtaining the amplitude envelope, the digital processor calculates the first-order difference or slope of the envelope and compares it with a preset transient trigger threshold. When the rising slope of the envelope exceeds the transient trigger threshold and the current energy amplitude exceeds a preset silent noise threshold, a low-frequency transient event is determined to have been detected.

[0040] The aforementioned transient trigger threshold is set based on the dynamic response capability of the fan motor and the impact intensity of the target sound effect; for example, it is set to increase by 6dB per millisecond. The aforementioned silent noise threshold is set based on the noise floor level of the audio device; for example, it is set to -60dBFS.

[0041] The digital processor marks the detected transient occurrence time stamp and the peak envelope intensity at that moment as a low-frequency transient feature. This low-frequency transient feature serves as a control signal to indicate when the phase correction signal in subsequent steps should be superimposed with an anti-phase aerodynamic impedance compensation component, thereby achieving precise time-domain alignment between the fan airflow pulse and the low-frequency audio impact.

[0042] Step S2: The digital processor performs instantaneous masking threshold calculation and fan constraint setting based on psychoacoustics. This process is specifically implemented through the following steps: Step S201: The digital processor retrieves the audio power spectrum, combines it with a preset psychoacoustic model, and calculates the frequency at the current moment using the masking threshold formula. The instantaneous masking threshold at the location.

[0043] The digital processor first maps the audio power spectrum obtained in step S102 on a linear frequency scale to the Bark frequency scale to conform to the critical band characteristics of the human auditory system. For the critical band division and energy mapping methods in psychoacoustic models, those skilled in the art can use the Zwicker model or the Moore-Glasberg model, the principles of which are well-known in the field and will not be elaborated here.

[0044] After completing the frequency standard mapping, the digital processor identifies the masking components in the audio power spectrum, including single-tone components and narrowband noise components in the audio signal. Based on the identified masking components, the digital processor calculates the masking effect within each critical frequency band and quantifies the masking ability of the audio signal against background noise at the current moment using the following masking threshold formula: The masking threshold formula is: ; In the formula: For a moment frequency The instantaneous masking threshold at a given time and frequency represents the upper limit of the minimum sound pressure level that the human ear can perceive at that moment and frequency. For a moment frequency The audio power spectrum at a given location represents the energy intensity of the current audio signal. For frequency The critical signal-to-noise ratio constant for the human ear at a given frequency is set based on the physiological limit of the human ear in distinguishing signals from noise at different frequencies. For example, it is set to -2dB to -5dB near 1kHz. For frequency The frequency domain masking spread function value at a given frequency is used to describe the attenuation characteristics of masking energy spreading from the center frequency to adjacent frequency bands.

[0045] The instantaneous masking threshold calculated using this formula forms a curve that dynamically changes with frequency and time. The region below this curve is the area imperceptible to the human ear, where the sound pressure level of noise signals is below the lower limit of human hearing.

[0046] Step S202: The digital processor performs reverse shaping on the full-band permissible noise spectrum of the fan motor based on the calculated instantaneous masking threshold to generate the fan target spectrum.

[0047] Reverse shaping refers to using the instantaneous masking threshold curve as an upper limit constraint on fan noise energy. The digital processor compares the instantaneous masking threshold with a preset fan base noise model. The aforementioned fan base noise model is pre-determined and stored based on the noise spectrum characteristics (including blade passing frequency and its harmonic components, broadband turbulence noise distribution) of the fan motor at different speeds under uncontrolled conditions. For example, it includes sound power level data for each speed range from 500 rpm to 3000 rpm.

[0048] The digital processor scans point by point in the frequency domain and selects the instantaneous masking threshold curve as the envelope of the maximum allowable acoustic power spectrum of the fan output. When the audio signal is in a high dynamic range or a spectrally rich segment, the instantaneous masking threshold is higher, and the generated fan target spectrum allows for a higher energy distribution; when the audio signal is silent or in a single segment, the instantaneous masking threshold is lowered, and the generated fan target spectrum is strictly limited, controlling the fan noise energy to be suppressed below this threshold.

[0049] The final generated fan target spectrum is a frequency domain constraint signal that specifies the maximum noise amplitude that the fan drive system is allowed to generate at each frequency point within the current time frame, providing clear spectral boundary conditions for the subsequent generation of fan drive signals.

[0050] See appendix Figure 2 Step S3: The digital processor performs fan drive generation and real-time airflow field vector modeling. This process is specifically implemented through the following steps: Step S301: The digital processor performs spectrum shaping processing based on the target spectrum of the fan using a random spread spectrum algorithm to generate the fan drive signal.

[0051] This step aims to generate an electrical signal capable of driving the fan motor while producing acoustic characteristics that conform to the auditory masking conditions determined in step S2. The digital processor first analyzes the fan target spectrum to determine the currently allowed fan fundamental frequency (corresponding to the rotational speed) and the maximum upper limit of acoustic energy. If the discrete spectral energy generated by a single fixed-frequency drive signal (such as conventional PWM) exceeds the limit of the fan target spectrum at the corresponding frequency, the digital processor initiates a random spread spectrum algorithm.

[0052] Specifically, the random spread spectrum algorithm is implemented by introducing pseudo-random jitter near the center switching frequency. The digital processor uses a linear feedback shift register (LFSR) to generate a pseudo-random sequence and modulates this sequence onto the period parameters of the pulse width modulation (PWM) signal. This disperses the spectral energy of the fan drive signal from concentrated narrowband spikes to a wider frequency band, thereby reducing the peak power spectral density at each frequency point and placing it below the envelope of the fan's target spectrum. The generated fan drive signal is then sent to the motor drive circuit.

[0053] Step S302: The digital processor collects the operating status of the fan motor in real time through the motor feedback loop and obtains the instantaneous angular velocity of the fan motor.

[0054] The motor feedback loop is connected to the Hall sensor of the fan motor or a sensorless position detection circuit is used to achieve this. The digital processor uses an input capture unit to record the time difference of the Hall signal's transition edges, or calculates the real-time physical speed of the fan rotor by observing the zero-crossing period of the back EMF. The digital processor converts the calculated physical speed (in RPM) into instantaneous angular velocity in radians, which is then used as input parameters for flow field calculations.

[0055] Step S303: The digital processor uses the acquired instantaneous angular velocity and preset air duct geometric parameters to calculate the air velocity vector in the space in front of the speaker by substituting the flow field vector formula into the fluid dynamics model.

[0056] This step aims to establish a mathematical mapping between physical rotational speed and spatial flow field distribution, thereby quantifying the impact of fan airflow on the sound wave propagation medium. The preset duct geometry parameters are based on the aerodynamic structural design of the speaker fan light (such as the angle of attack and chord length distribution of the fan blades, the guiding characteristics of the light chassis, and the axial distance between the speaker unit and the fan blade plane). These parameters are derived from pre-calibrated vector field data obtained through computational fluid dynamics (CFD) simulation or experimental measurement, and are stored in the digital processor's memory in the form of a three-dimensional look-up table. These parameters describe the diffusion pattern and attenuation law of the airflow in the sound field region in front of the speaker at a unit rotational speed.

[0057] The digital processor inputs the real-time acquired instantaneous angular velocity into the hydrodynamic model and uses the following flow field vector formula to calculate the air velocity vector at a specific spatial location at the current moment. The flow field vector formula is: ; In the formula: For a moment In spatial location The air velocity vector at a point contains information about the velocity magnitude and direction, representing the motion state of the air medium at that point. The fluid conversion efficiency coefficient of the fan system is a constant set based on the aerodynamic characteristics of the fan blades and the flow resistance characteristics of the air duct, for example, a dimensionless value between 0.3 and 0.6. For a moment The instantaneous angular velocity of the fan motor; For spatial location The determined duct geometry parameters, specifically the flow field shape factor, describe the normalized velocity field in space. The distribution weights at each location.

[0058] Using this formula, the digital processor can reconstruct the dynamic flow field map along the sound propagation path in front of the speaker in real time. This airflow vector data is stored in a cache for later use by the acoustic phase correction module.

[0059] See appendix Figure 3 Step S4: The digital processor performs acoustic phase pre-distortion and impedance compensation based on flow field mapping. This process aims to eliminate the nonlinear interference of airflow on sound wave propagation and actively utilize airflow characteristics to enhance the low-frequency response. Specifically, it is achieved through the following steps: Step S401: The digital processor analyzes the propagation delay characteristics of sound waves in the dynamic flow field based on the air velocity vector, substitutes them into the phase correction formula for calculation, and obtains the phase compensation amount.

[0060] The digital processor first reads the air velocity vector calculated in step S303. Since the speed of sound propagation in air is not constant but is superimposed with the flow velocity of the medium itself, the actual time it takes for the sound wave to reach the listening point will deviate from the theoretical time when a non-zero velocity field generated by a fan exists in front of the speaker. The digital processor uses the flow field vector data to calculate the difference between the sound wave's propagation time in still air and its propagation time in the current dynamic airflow, and converts this time difference into a phase shift angle corresponding to the frequency.

[0061] The digital processor uses a phase correction formula to calculate the specific phase compensation amount. The phase correction formula is: ; In the formula: For a moment frequency The amount of phase compensation that needs to be applied; This is the straight-line distance from the center of the sound source to the preset listening point. This distance is preset based on typical usage scenarios (such as desktop near-field listening or living room far-field listening) or calibrated by an external ranging sensor, for example, set to 0.8 meters to 1.5 meters. is the speed of sound in still air, usually taken as 343 m / s (at 20°C). The magnitude of the airflow velocity vector is used here to represent the magnitude of the velocity. These are cosine trigonometric functions; It is the angle between the direction of sound wave propagation and the air velocity vector, which is determined by the geometric angle between the speaker axis and the fan exhaust duct.

[0062] Through this calculation, the digital processor obtains the precise phase correction values ​​required for each frequency component in the current frame of the audio signal.

[0063] Step S402: The digital processor applies the calculated phase compensation amount to the original audio signal to generate a phase correction signal.

[0064] The digital processor constructs a phase adjustment processing unit (all-pass filter or complex domain multiplier) with linear phase characteristics. The digital processor uses the phase compensation obtained in step S401 as the group delay parameter of the filter or the phase angle parameter of the complex coefficients.

[0065] The digital processor inputs the original audio signal (or its frequency domain form) from step S101 into the filter or multiplier. After processing, the high-frequency components of the audio signal that are significantly affected by airflow acceleration or obstruction are pre-delayed or advanced in reverse, thereby canceling the distortion effect of airflow on the sound wave front in physical space. The generated phase-corrected signal maintains the integrity of the waveform in the time domain, but the phase information has been pre-distorted for the current airflow field.

[0066] Step S403: The digital processor combines the low-frequency transient characteristics and performs a superposition operation on the phase correction signal to synthesize the final speaker signal.

[0067] This step aims to use the movement of the speaker diaphragm to balance or enhance the airflow pulses generated by the fan. The digital processor reads the low-frequency transient features (including the impact moment and energy envelope) extracted in step S103. When the low-frequency transient features indicate the presence of a high-energy low-frequency impact (such as a drumbeat), the digital processor generates an anti-phase aerodynamic impedance compensation component.

[0068] The inverse aerodynamic impedance compensation component is a sub-bass or DC offset modulated signal whose waveform is consistent with the envelope shape of the low-frequency transient characteristics, but whose phase is opposite to the phase of the airflow pressure pulse generated by the fan (for impedance balancing) or set to a specific co-phase (for pulse enhancement).

[0069] The digital processor linearly superimposes the anti-phase aerodynamic impedance compensation component with the phase correction signal generated in step S402 in the time domain.

[0070] The superimposed signals constitute the final speaker signal. This signal not only contains high-fidelity audio content but also the driving commands that control the speaker diaphragm to perform a long-stroke movement. When this signal drives the speaker, the diaphragm's movement precisely matches the airflow pulses of the fan. This results in a superimposed air momentum at the listening position during low-frequency bursts, where the speaker's action of pushing air and the airflow delivered by the fan combine to create a physical effect that enhances the sound pressure of the airflow on the low-frequency listening experience, thus producing an airflow-like sound effect.

[0071] Step S5: The digital-to-analog converter amplifier circuit, together with the fan motor and speaker unit, performs the construction of the sound-air coordinated physical field. This process is specifically achieved through the following steps: Step S501: Receive the fan drive signal and speaker signal using the digital-to-analog converter power amplifier circuit, perform power amplification processing on them respectively, and output the amplified drive current.

[0072] The digital-to-analog converter (DAC) circuit includes two independent drive channels: an audio drive channel and a motor drive channel. The digital processor transmits the speaker signal to the DAC in the audio drive channel via an internal bus or I2S interface, and transmits the fan drive signal to the gate driver in the motor drive channel via a general purpose timer (GPT) interface or PWM interface.

[0073] In the audio drive channel, a digital-to-analog converter (DAC) converts the discrete digital speaker signal into a continuous analog voltage signal. This analog signal then enters a power amplifier (e.g., a Class D amplifier), which converts the voltage signal into a large current signal capable of driving the speaker voice coil. This current signal retains the phase correction characteristics and aerodynamic impedance compensation components applied in step S4.

[0074] In the motor drive channel, the gate driver controls the on / off state of the H-bridge or three-phase inverter bridge power transistors according to the spread-spectrum PWM logic in the fan drive signal, generating a pulse current to drive the fan motor. The spectral characteristics of this pulse current strictly follow the broadband distribution characteristics determined in step S301, avoiding electromagnetic excitation at a single frequency.

[0075] Step S502: Drive the fan motor and speaker unit to operate synchronously, generating controlled airflow and sound waves in the physical space, ultimately forming an optimized sound field and airflow field.

[0076] The fan motor rotates under the influence of the driving current, causing the blades to cut through the air and generate airflow. Because the driving signal undergoes inverse shaping and random spread spectrum processing based on a masking threshold, the aerodynamic noise energy generated by the fan is dispersed across a wide frequency band and remains below the energy of the currently played audio signal, thus creating a low-noise airflow that is imperceptible to the ear. This airflow, based on the flow field vector model in step S303, is discharged from the speaker duct at a specific velocity and direction, forming a controlled airflow.

[0077] At the same time, the speaker unit drives the diaphragm to vibrate under the action of the driving current, radiating sound waves into space. Since the input signal has already included the phase pre-distortion compensation amount for the current controlled airflow, when the sound wave passes through the controlled airflow area, the physical distortion of its wavefront caused by airflow refraction is precisely canceled by the preset phase shift, so that the wavefront of the sound wave reaching the listening position is restored to the expected plane wave or spherical wave, ensuring the flatness of the frequency response.

[0078] Furthermore, during low-frequency transients, the fan motor's rotational speed pulses and the speaker diaphragm's large-stroke motion are precisely synchronized in time. The controlled airflow, acting as the physical carrier of sound waves, carries low-frequency sound energy to the listening area. The compensating motion of the speaker diaphragm effectively smooths out the sudden changes in local pressure caused by the airflow pulses, allowing users to perceive the airflow synchronized with the music rhythm while experiencing the impact of low-frequency sounds. This creates a synergistic sound and air field in the physical space, where the sound field and airflow field merge and reinforce each other.

[0079] The low-noise design method for an audio fan light based on fan noise provided in this invention establishes a closed-loop coupling control between audio signals, fan drive, and airflow field through a digital processor.

[0080] This method extracts the frequency domain power spectrum and time domain transient features of the original audio signal as reference data for subsequent control. A psychoacoustic model is used to calculate the instantaneous masking threshold, quantifying the lower limit of human hearing perception of noise at the current moment, and generating a fan target spectrum to constrain the operating noise of the fan motor. A digital processor combines the instantaneous angular velocity acquired from the motor feedback loop with preset duct geometry parameters, using a hydrodynamic model to calculate the airflow velocity vector in the space below the speaker fan light in real time. Simultaneously, a random spread spectrum algorithm is used to generate the fan drive signal, dispersing the fan noise energy and keeping it below the audio masking threshold.

[0081] By applying a reverse phase pre-distortion to the audio signal using the calculated airflow velocity vector, the refraction and delay interference caused by dynamic airflow on sound wave propagation is canceled out. Simultaneously, by combining low-frequency transient characteristics with aerodynamic impedance compensation components, the movement of the speaker diaphragm and the airflow pulses generated by the fan are synchronized in the time domain. A digital-to-analog converter amplifier circuit drives the fan motor and speaker unit to operate in tandem, achieving the simultaneous use of controlled airflow as a sound wave transmission carrier to enhance low-frequency listening experience while utilizing the fan for air circulation, thus constructing a low-noise and optimized sound-air synergistic physical field.

Claims

1. A low-noise design method for an audio fan based on fan noise, characterized in that, Includes the following steps: S1. The digital processor receives the original audio signal, performs time-domain buffering and short-time Fourier transform to generate the audio power spectrum, and simultaneously performs time-domain envelope transient detection to extract low-frequency transient features. S2. Retrieve the audio power spectrum and combine it with the psychoacoustic model. Calculate the instantaneous masking threshold according to the masking threshold formula. Then, perform reverse shaping on the full-band allowable noise spectrum of the fan motor based on the instantaneous masking threshold to generate the fan target spectrum. S3. Based on the target spectrum of the fan, a random spread spectrum algorithm is used to generate a fan drive signal. The instantaneous angular velocity is collected using the motor feedback loop. The air velocity vector is constructed according to the flow field vector formula in combination with the fluid dynamics model and the duct geometric parameters. S4. Calculate the phase compensation amount and generate a phase correction signal based on the air velocity vector and phase correction formula, and perform a superposition operation to synthesize a loudspeaker signal in combination with the low-frequency transient characteristics. S5. The digital-to-analog converter power amplifier circuit amplifies the power of the fan drive signal and the speaker signal, driving the fan motor and speaker unit to operate synchronously to generate a sound field and an airflow field.

2. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S1, generating the audio power spectrum specifically includes: The digital processor receives the raw audio signal from an external input, writes the raw audio signal into its internal memory, and extracts data to construct audio data frames. The audio data frame is multiplied in the time domain by a window function, and a fast Fourier transform is performed to obtain the complex spectrum. The audio power spectrum is obtained by calculating the square of the modulus of each frequency component in the complex spectrum.

3. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S1, the synchronous execution of temporal envelope transient detection to extract low-frequency transient features specifically includes: The digital processor passes the original audio signal through a digital low-pass filter to filter out mid-to-high frequency components and retain low-frequency components, and uses Hilbert transform to obtain the instantaneous amplitude envelope of the low-frequency components. Calculate the slope of the instantaneous amplitude envelope. When the slope exceeds the transient trigger threshold and the current energy amplitude exceeds the silent noise threshold, it is determined that a low-frequency transient event has been detected. Record the occurrence time and envelope peak intensity of the low-frequency transient event, and label them as the low-frequency transient feature; The transient trigger threshold is preset based on the dynamic response capability of the fan motor, and the silent noise threshold is preset based on the noise floor level of the audio device.

4. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S2, calculating the instantaneous masking threshold based on the masking threshold formula specifically includes: The digital processor uses the psychoacoustic model to map the audio power spectrum to the Barker frequency standard, and identifies the masking components in the audio power spectrum based on the psychoacoustic model; The masking effect within each critical frequency band is calculated, and the masking threshold formula is used to quantify the masking ability of the original audio signal to the background noise at the current moment, so as to obtain the instantaneous masking threshold. The psychoacoustic model is an auditory perception model established based on the critical frequency band characteristics of human hearing.

5. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S2, the process of inversely shaping the full-band permissible noise spectrum of the fan motor based on the instantaneous masking threshold to generate the fan target spectrum specifically includes: The digital processor compares the instantaneous masking threshold with a preset fan base noise model; The amplitude data of the instantaneous masking threshold is selected as the upper limit of the noise amplitude by scanning point by point on the frequency domain of the full-band allowable noise spectrum of the fan motor. When the instantaneous masking threshold is reduced, the fan noise energy is controlled to be lower than the upper limit of the noise amplitude, and the fan target spectrum with the maximum allowable noise amplitude at each frequency point is generated; The fan's basic noise model is pre-determined and stored based on the noise spectrum characteristics of the fan motor at different speeds under uncontrolled conditions.

6. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S3, generating the fan drive signal based on the fan target spectrum using a random spread spectrum algorithm specifically includes: The digital processor analyzes the target spectrum of the fan to determine the currently allowed fan base frequency; When the discrete spectral energy generated by the fan's fundamental frequency exceeds the limit of the fan's target spectrum, the random spread spectrum algorithm is activated. A pseudo-random sequence is generated using a linear feedback shift register, and the pseudo-random sequence is modulated onto the periodic parameter of a pulse width modulation signal to disperse the spectral energy across a wide bandwidth and below the envelope of the fan target spectrum, thereby generating the fan drive signal.

7. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S3, constructing the air velocity vector based on the flow field vector formula, combining the fluid dynamics model and duct geometric parameters, specifically includes: The digital processor uses the input capture unit to record the time difference of the transition edge of the Hall signal in the motor feedback loop, calculates the real-time physical speed of the fan rotor, and converts it into the instantaneous angular velocity; The instantaneous angular velocity is input into the hydrodynamic model, and the duct geometry parameters are retrieved. The air velocity vector at the current moment is calculated according to the flow field vector formula. The duct geometry parameters are vector field data that are pre-calibrated through computational fluid dynamics simulation based on the design of the fan blades and air guide structure and stored in the form of a three-dimensional lookup table. The fluid dynamics model is a mathematical model constructed based on the mapping relationship between rotational speed and flow velocity.

8. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S4, calculating the phase compensation amount and generating the phase correction signal based on the air velocity vector and phase correction formula specifically includes: The digital processor uses the air velocity vector to calculate the difference between the propagation time of sound waves in still air and the propagation time in dynamic airflow. The difference is converted into the phase compensation amount at the corresponding frequency according to the phase correction formula. A phase adjustment processing unit with linear phase characteristics is constructed. The phase compensation amount is used as a parameter to apply an inverse time shift to the high-frequency components of the original audio signal to generate the phase correction signal.

9. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, In step S4, the process of combining the low-frequency transient characteristics to perform a superposition operation to synthesize the loudspeaker signal specifically includes: The digital processor generates an anti-phase aerodynamic impedance compensation component based on the low-frequency transient characteristics. The phase of the anti-phase aerodynamic impedance compensation component is synchronized with the phase of the airflow pressure pulse generated by the fan motor. The anti-phase aerodynamic impedance compensation component and the phase correction signal are linearly superimposed in the time domain to synthesize the loudspeaker signal containing the command to control the diaphragm to perform a large stroke.

10. The low-noise design method for an audio fan based on fan noise according to claim 1, characterized in that, The S5 step specifically includes: The digital-to-analog converter power amplifier circuit converts the speaker signal into a high-current signal to drive the speaker unit through the audio drive channel, and controls the power transistor to turn on and off according to the fan drive signal through the motor drive channel to generate a pulse current to drive the fan motor. The fan motor rotates according to the pulsed current to generate controlled airflow, and the speaker unit drives the diaphragm to vibrate according to the large current signal; The controlled airflow is the physical carrier for sound wave transmission, carrying low-frequency sound wave energy. The movement of the diaphragm compensates for local pressure changes in the controlled airflow, thus constructing a sound field and an airflow field that are integrated in the physical space.

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