Vehicle sound field control method and device, computer equipment and storage medium

Abstract

The invention relates to the technical field of vehicle control, and discloses a sound field control method and device of a vehicle, computer equipment and a storage medium, the method comprises the following steps: in response to a scene mode trigger signal, obtaining a first real-time opening condition of a vehicle door and / or a vehicle window, determining a target sound field projection area in the vehicle based on the first real-time opening condition, generating an audio processing parameter corresponding to a loudspeaker in the vehicle; the audio signal is processed according to the audio processing parameters, and the loudspeaker is driven to project sound energy corresponding to the processed audio signal to the target sound field projection area. The target sound field projection area outside the vehicle is determined by sensing the real-time opening state of the vehicle door / window and combining the preset scene mode, and the loudspeaker parameters are adjusted; therefore, the sound energy is guided to the position of the user, the problem that the sound field in the vehicle leaks or is poor in effect when the vehicle door and the vehicle window are opened is effectively solved, and the audio experience during outdoor use is improved.

Description

Technical Field

[0001] This invention relates to the field of vehicle control technology, and more specifically to a method, device, computer equipment, and storage medium for controlling the sound field of a vehicle. Background Technology

[0002] Currently, vehicles primarily rely on standard Avas (Advanced Ventilation Vents) for sound playback in exterior entertainment scenarios. Avas speakers provide warning sounds when the vehicle is traveling at low speeds; their power is typically below 10W, their frequency response is concentrated in the mid-range, and they are omnidirectional. In outdoor scenarios such as camping, users want to utilize the vehicle's audio system for entertainment, but current vehicle audio systems do not offer sound field optimization for multiple doors and windows being open, nor do they provide the ability to adjust speaker output based on different outdoor activity areas. Overall, the primary application scenario remains in-vehicle listening.

[0003] On the one hand, Avas speakers suffer from low power and narrow frequency response, resulting in insufficient external volume, lack of low frequencies, and high-frequency distortion. They are unable to provide sufficient sound pressure and clarity in open environments, and their effective coverage range is usually less than 3 meters. On the other hand, when multiple doors or windows are open, current vehicle audio systems lack sound field coordination algorithms, which can easily lead to sound wave interference and phase cancellation, resulting in listening discomfort. Furthermore, they cannot adapt the output characteristics of each speaker channel according to the user's specific location outside the vehicle, limiting the user's actual experience in outdoor entertainment scenarios. Summary of the Invention

[0004] In view of this, embodiments of the present invention provide a method, apparatus, computer device, and storage medium for controlling the sound field of a vehicle, in order to solve the problem of how to control the sound field of a vehicle based on the opening status of the vehicle doors / windows and the preset location of the user camping or relaxing outside the vehicle.

[0005] In a first aspect, embodiments of the present invention provide a method for controlling the sound field of a vehicle, the method comprising: In response to a scene mode trigger signal, the first real-time opening status of the vehicle door and / or window is acquired, the target sound field projection area in the vehicle is determined based on the first real-time opening status, and audio processing parameters corresponding to the speakers in the vehicle are generated. The audio signal is processed according to the audio processing parameters, and the speaker is driven to project the sound energy corresponding to the processed audio signal onto the target sound field projection area.

[0006] This embodiment senses the real-time opening status of the car doors / windows, determines the target sound field projection area outside the vehicle by combining preset scene modes, and adjusts the speaker parameters to direct sound energy to the user's location. This effectively solves the problem of sound field leakage or poor effect inside the vehicle when the car doors and windows are open, and improves the audio experience when used outdoors.

[0007] In conjunction with the first aspect, in one implementation, determining the target sound field projection area in the vehicle based on the first real-time activation status and generating audio processing parameters corresponding to the speakers in the vehicle includes: Detect the opening position of the vehicle door and / or the vehicle window when they are open, and determine the target sound field projection area based on the opening position; Acquire the loudspeakers in the vehicle that can cover the target sound field projection area, and determine the output power of the loudspeakers; The sound wave path difference is estimated based on the installation location of the loudspeaker and the target sound field projection area, and the time delay compensation for the loudspeaker is determined based on the sound wave path difference. The phase control parameters for the loudspeaker are determined based on the scene mode trigger signal and the target sound field projection area; The output power, the time delay compensation, and the phase control parameters are used as the audio processing parameters.

[0008] This embodiment accurately locates the target sound field area by detecting the opening position of the car door / window, selects a speaker that can cover the area, and generates optimized audio processing parameters by setting the output power of the speaker, calculating time delay compensation based on the sound wave path difference, and setting phase parameters in combination with scene mode. This allows the sound energy to be projected more concentratedly and accurately to a specific area outside the vehicle, improving the sound field performance in open environments.

[0009] In conjunction with the first aspect, in one embodiment, processing the audio signal according to the audio processing parameters includes: The first ambient sound signal is collected by a microphone array and transmitted to the vehicle's entertainment host. The entertainment host performs instantaneous sound pressure calculation and spectrum identification on the first ambient sound signal to identify the first ambient noise and the noise type of the first ambient noise. The audio processing parameters are updated based on the identified noise type, and the audio signal is processed according to the updated audio processing parameters.

[0010] This embodiment uses a microphone array to collect and analyze ambient noise outside the vehicle in real time, identifying the intensity and type of noise. Based on this, it dynamically adjusts the audio processing parameters of the speakers to achieve intelligent cancellation and compensation of ambient noise. This ensures that even in noisy environments such as outdoor camping and leisure activities, the sound field projected onto the designated area remains clear and full, significantly improving outdoor sound quality stability and anti-interference capabilities.

[0011] In conjunction with the first aspect or its corresponding implementation, in one implementation, the step of performing instantaneous sound pressure calculation and spectrum identification on the first ambient sound signal through the entertainment host to identify the first ambient noise and the noise type of the first ambient noise includes: The first ambient sound signal is converted into a discrete first digital signal sequence at a preset sampling rate, and the first digital signal sequence is extracted based on several time windows; Within each time window, the root mean square of the first digital signal sequence is calculated to obtain the average effective value of the first ambient sound signal within each time window. The average amplitude RMS value is converted into the instantaneous sound pressure level of the first ambient sound signal; Spectral analysis of the first ambient sound signal is performed to obtain the energy proportion of the first ambient sound signal in different frequency bands; Based on the energy ratio and the instantaneous sound pressure level, the first ambient noise contained in the first ambient sound signal and the noise type of the first ambient noise are identified.

[0012] This embodiment converts ambient sound signals into digital sequences and performs windowing processing to calculate the instantaneous sound pressure level within each window. Simultaneously, it identifies the energy distribution of different frequency bands through spectrum analysis, thereby quantifying the intensity and type of ambient noise and achieving targeted cancellation of specific noises. This ensures that users can have a stable and clear sound field experience in different noise environments.

[0013] In conjunction with the first aspect or its corresponding implementation, in one implementation, updating the audio processing parameters based on the identified noise type and processing the audio signal according to the updated audio processing parameters includes: Select the basic equalization curve corresponding to the scene mode trigger signal, adjust the basic equalization curve according to the noise type and the target sound field projection area, and process the audio signal according to the updated basic equalization curve.

[0014] This embodiment dynamically adjusts the basic equalization curve by combining preset scene modes with real-time noise type identification, thereby adapting to the acoustic needs of different scenarios and specifically compensating for noise interference in specific frequency bands. This ensures that the sound projected onto the target area outside the vehicle maintains optimal clarity and balance, enhancing the adaptability and expressiveness of the sound field in complex noise environments.

[0015] In conjunction with the first aspect, in one embodiment, processing the audio signal according to the audio processing parameters further includes: Read the user's playback preferences from the user database pre-built in the vehicle; The audio processing parameters are updated based on the playback preference, and the audio signal is processed according to the updated audio processing parameters.

[0016] This embodiment integrates pre-stored user playback preferences into real-time adjustments of audio processing parameters, enabling sound field control to not only respond to environmental changes but also proactively adapt to the user's personalized listening habits. By combining objective acoustic control with subjective preferences, it delivers a clear, stable, and highly personalized sound experience for users in outdoor scenarios such as camping and leisure.

[0017] In conjunction with the first aspect, in one embodiment, the method further includes: The system acquires the interconnection relationships established between the vehicle and other vehicles, and selects the main audio device and sub-audio devices based on these interconnection relationships. The main audio device is controlled to send a synchronization signal to the sub-audio device, and the sound emission time of the main audio device and the sub-audio device is calibrated using the synchronization signal; The system acquires the second real-time opening status of doors and / or windows of vehicles other than the current vehicle, and adjusts the sound parameters of the main audio device and the sub-audio device based on the first real-time opening status and the second real-time opening status. The second ambient sound signal is acquired based on the microphone array of the main audio device, and instantaneous sound pressure level and spectrum identification are performed on the second ambient sound signal to identify the second ambient noise and the noise type of the second ambient noise; Adjust the equalization parameters of the main speaker and the sub-speakers based on the noise type of the second ambient noise.

[0018] This embodiment establishes a master-slave audio interconnection between vehicles and achieves time synchronization. By coordinating the adjustment of sound parameters based on the status of multiple vehicle doors / windows, a unified and coordinated extended sound field is constructed. Furthermore, the main device collects ambient noise and identifies its type, thereby synchronously optimizing the equalization parameters of each vehicle. In multi-vehicle scenarios outside the vehicle, this effectively overcomes the limitations of a single sound field and provides users with an immersive surround sound experience.

[0019] In conjunction with the first aspect, in one embodiment, the method further includes: The vehicle's infotainment system displays a visual editing interface that includes a top-down view of the vehicle and the opening and closing status of the doors; Receive the user's drag signal on the sound field element icon in the visual editing interface, and determine the custom sound field projection area in the vehicle top view according to the drag signal; Based on the custom sound field projection area, the first real-time on status, the second real-time on status, and the preset position of the speaker, calculate the first audio processing parameters of the speaker; Based on the first audio processing parameters, the speaker is invoked to play sound in order to form a first predetermined sound field within the custom sound field projection area.

[0020] In conjunction with the first aspect or its corresponding implementation, in one implementation, calculating the first audio processing parameters of the speaker based on the custom sound field projection area, the first real-time on status, the second real-time on status, and the preset position of the speaker includes: Acquire the first loudspeaker in the vehicle that can cover the custom sound field projection area, and determine the first output power of the first loudspeaker; Estimate the first sound wave path difference based on the installation position of the first speaker and the custom sound field projection area, and determine the time delay compensation for the first speaker based on the first sound wave path difference. The first phase control parameters for the first speaker are determined based on the custom sound field projection area. The first output power, the first time delay compensation, and the first phase control parameter are used as the first audio processing parameters.

[0021] Secondly, embodiments of the present invention provide a sound field control device for a vehicle, the device comprising: The parameter generation module is used to respond to the scene mode trigger signal, acquire the first real-time opening status of the car door and / or window, determine the target sound field projection area in the vehicle based on the first real-time opening status, and generate the audio processing parameters corresponding to the speakers in the vehicle. The sound field control module is used to process the audio signal according to the audio processing parameters and drive the speaker to project the sound energy corresponding to the processed audio signal onto the target sound field projection area.

[0022] Thirdly, embodiments of the present invention provide a vehicle sound field control system, which applies the vehicle sound field control method of the first aspect above or any corresponding embodiment thereof, the system comprising: The entertainment host communicates unidirectionally with microphones outside and / or inside the vehicle via an A2B line; receives real-time door and / or window opening status data from the audio amplifier via a CAN line and sends control commands for the speakers to the audio amplifier; and transmits processed audio signals unidirectionally to the audio amplifier via an A2B line. The in-vehicle system speakers receive control commands transmitted by the audio amplifier via simulated twisted-pair cables. The audio amplifier communicates unidirectionally with microphones outside and / or inside the vehicle via an A2B cable.

[0023] Fourthly, embodiments of the present invention provide a computer device, including: a memory and a processor, the memory and the processor being communicatively connected to each other, the memory storing computer instructions, and the processor executing the computer instructions to perform the vehicle sound field control method of the first aspect or any corresponding embodiment described above.

[0024] Fifthly, embodiments of the present invention provide a computer-readable storage medium storing computer instructions for causing a computer to execute the sound field control method for a vehicle according to the first aspect or any corresponding embodiment described above. Attached Figure Description

[0025] To more clearly illustrate the specific embodiments of the present invention or the technical solutions in the prior art, the drawings used in the description of the specific embodiments or the prior art will be briefly introduced below. Obviously, the drawings described below are some embodiments of the present invention. For those skilled in the art, other drawings can be obtained from these drawings without creative effort.

[0026] Figure 1 This is a structural diagram of the vehicle's sound field control system provided based on relevant technologies; Figure 2 This is a schematic flowchart of a vehicle sound field control method according to some embodiments of the present invention; Figure 3 This is a schematic diagram of the structure of a vehicle sound field control system according to an embodiment of the present invention; Figure 4 This is a schematic diagram of the interconnection structure of a vehicle sound field control system according to an embodiment of the present invention; Figure 5 This is a structural block diagram of a vehicle sound field control device according to an embodiment of the present invention; Figure 6 This is a schematic diagram of the hardware structure of a computer device according to an embodiment of the present invention. Detailed Implementation

[0027] To make the objectives, technical solutions, and advantages of the embodiments of the present invention clearer, 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, 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.

[0028] According to an embodiment of the present invention, a method for controlling the sound field of a vehicle is provided. It should be noted that the steps shown in the flowchart in the accompanying drawings can be executed in a computer system such as a set of computer-executable instructions. Furthermore, although a logical order is shown in the flowchart, in some cases, the steps shown or described may be executed in a different order than that shown here.

[0029] Currently, vehicles primarily rely on standard Avas (Advanced Ventilation Vents) for sound playback in exterior entertainment scenarios. Avas speakers provide warning sounds when the vehicle is traveling at low speeds; their power is typically below 10W, their frequency response is concentrated in the mid-range, and they are omnidirectional. In outdoor scenarios such as camping, users want to utilize the vehicle's audio system for entertainment, but current vehicle audio systems do not offer sound field optimization for multiple doors and windows being open, nor do they provide the ability to adjust speaker output based on different outdoor activity areas. Overall, the primary application scenario remains in-vehicle listening.

[0030] On the one hand, Avas speakers suffer from low power and narrow frequency response, resulting in insufficient external volume, lack of low frequencies, and high-frequency distortion. They are unable to provide sufficient sound pressure and clarity in open environments, and their effective coverage range is usually less than 3 meters. On the other hand, when multiple doors or windows are open, current vehicle audio systems lack sound field coordination algorithms, which can easily lead to sound wave interference and phase cancellation, resulting in listening discomfort. Furthermore, they cannot adapt the output characteristics of each speaker channel according to the user's specific location outside the vehicle, limiting the user's actual experience in outdoor entertainment scenarios.

[0031] Reference Figure 1The diagram illustrates the structure of a vehicle sound field control system provided by related technologies. The vehicle sound field control system includes an entertainment head unit, an in-vehicle audio system, and an external AVAS system. These three systems operate independently. The in-vehicle audio system includes an amplifier and in-vehicle speakers, while the external AVAS system includes an AVAS control module and AVAS speakers. The entertainment head unit communicates with the amplifier and AVAS control module via CAN bus, while audio signals are transmitted to the in-vehicle speakers and AVAS speakers respectively via analog twisted-pair cables. This separate design results in generally poor sound quality and limited coverage outside the vehicle, and the in-vehicle audio resources cannot be effectively utilized in external environments, leading to functional fragmentation and resource waste.

[0032] Based on this, this embodiment provides a method for controlling the sound field of a vehicle. Figure 2 This is a flowchart of a vehicle sound field control method according to an embodiment of the present invention, such as... Figure 2 As shown, the process includes the following steps: Step S201: In response to the scene mode trigger signal, the first real-time opening status of the car door and / or window is obtained, the target sound field projection area in the vehicle is determined based on the first real-time opening status, and audio processing parameters corresponding to the speakers in the vehicle are generated.

[0033] Step S202: The audio signal is processed according to the audio processing parameters, and the speaker is driven to project the sound energy corresponding to the processed audio signal onto the target sound field projection area.

[0034] The vehicle sound field control method provided in this embodiment responds to a scene mode trigger signal, acquires the first real-time opening status of the car door and / or window, determines the target sound field projection area in the vehicle based on the first real-time opening status, and generates audio processing parameters corresponding to the speakers in the vehicle; processes the audio signal according to the audio processing parameters, and drives the speakers to project the sound energy corresponding to the processed audio signal to the target sound field projection area. This embodiment, by sensing the real-time opening status of the car door / window, combined with a preset scene mode, determines the target sound field projection area outside the vehicle, and adjusts the speaker parameters, thereby directing the sound energy to the user's location, effectively solving the problem of sound field leakage or poor effect inside the vehicle when the car door and window are open, and improving the audio experience when used outdoors.

[0035] Reference Figure 3This diagram illustrates the structure of a vehicle sound field control system provided in an embodiment of the present invention. The vehicle's infotainment system communicates bidirectionally with the audio amplifier (containing a DSP algorithm) via CAN and A2B lines, enabling precise control of the amplifier and high-speed transmission of audio signals. When the user selects a specific scene tuning mode, such as outdoor music or tailgate theater, on the infotainment system interface, or when the sensor detects that the vehicle has entered a state suitable for outdoor acoustic configuration, audio processing parameters are generated based on the user selection, the vehicle's open status acquired in real time via the CAN bus, and the ambient noise data acquired in real time by the microphone. Subsequently, the infotainment system transmits the audio signal processed based on the audio processing parameters to the audio amplifier via the high-speed A2B bus, and simultaneously sends relevant control commands, such as equalizer parameters, delay time, and power allocation, to the audio amplifier via the CAN bus. The DSP (Digital Signal Processing) algorithm integrated within the audio amplifier performs multi-channel frequency division, phase calibration, time delay compensation, and precise power and gain allocation on the audio signal. This integrates the originally independent in-vehicle speaker systems, such as the door, dashboard, and rear speakers, with the external AVAS speakers into a unified, intelligent acoustic array. They work together to project optimized sound outwards, overcoming the limitations of a single speaker system and achieving the goal of providing high-quality, wide-range sound coverage in open environments outside the vehicle.

[0036] In step S201, in response to the scene mode trigger signal, the first real-time opening status of the car door and / or window is obtained, the target sound field projection area in the vehicle is determined based on the first real-time opening status, and the audio processing parameters corresponding to the speakers in the vehicle are generated.

[0037] In one embodiment, sensors installed throughout the vehicle detect open doors and / or windows, such as door switches and window motor Hall sensors. These sensors further identify the specific location of the open doors and / or windows, such as the left front door, the right rear window, or other combinations thereof. Based on the location of the open doors and / or windows, the specific direction and area from which sound needs to be projected outside the vehicle is determined; this area is the target sound field projection area. For example, if both doors on the left side of the vehicle are detected to be open, it can be inferred that the user's leisure area is primarily on the left side of the vehicle, thus setting the target sound field projection area in the external space on the left side of the vehicle.

[0038] Next, from the vehicle's in-vehicle audio system, which includes an amplifier and speakers, speakers whose sound wave radiation range can effectively cover the target sound field projection area are retrieved and selected. Simultaneously, appropriate output power is allocated to the selected speakers. For example, speakers located near the open doors, such as sill speakers and headrest speakers, may be preferentially activated. In one embodiment, the sound wave propagation distance from each selected speaker to the target sound field projection area is estimated. Since the sound pressure level attenuates with increasing propagation distance, speakers farther from the target area are allocated relatively higher output power to compensate for their energy loss along the propagation path; conversely, speakers closer to the target area are allocated relatively lower power to prevent excessive sound at that location. That is, in this embodiment, the output power allocated to the speakers is inversely proportional to the sound wave propagation distance from the speaker to the target sound field projection area, thereby ensuring that the sound emitted from speakers in different locations achieves approximately uniform loudness in the target area.

[0039] Because the distances from each speaker to the target point are different, the arrival times of the sound waves emitted by them will vary. Therefore, it is necessary to estimate the sound wave path difference from different speakers to the center point based on the geometric relationship between the physical installation position of each activated speaker and the center of the target sound field projection area. Based on the sound wave path difference, time delay compensation for each speaker can be calculated. For example, applying appropriate time delay compensation to speakers with shorter paths allows their sound waves to arrive at the target point almost simultaneously with those of speakers with longer paths, thereby achieving synchronous superposition of sound waves in space and enhancing the sound pressure level in that area.

[0040] Specifically, a three-dimensional coordinate system is established with the vehicle's center of gravity or the location of the infotainment system as the origin. ,in This represents the three-dimensional coordinates of each activated speaker. Indicates the first One activated speaker. Based on the opening orientation of the car doors / windows, the acoustic center point and its three-dimensional coordinates of the target sound field projection area are determined. The acoustic center point can be a preset location; for example, for the left front door opening, the acoustic center point might be set 1.5 meters above the ground on the left front of the vehicle. The distance from each activated speaker to the acoustic center point of the target sound field projection area is calculated using the following formula. :

[0041] in, This indicates the distance (in meters) from each activated loudspeaker to the acoustic center point of the target sound field projection area. This represents the three-dimensional coordinates of each activated speaker. Indicates the first One activated speaker, The three-dimensional coordinates of the acoustic center point of the target sound field projection area.

[0042] Next, select one of the activated speakers as a reference speaker. For example, select the speaker furthest from the acoustic center point, and denot the distance as . Calculate the acoustic path difference of each of the other loudspeakers relative to the reference loudspeaker. :

[0043] in, This represents the sound wave path difference (in meters) between each loudspeaker and the reference loudspeaker. This indicates the distance (in meters) from each activated loudspeaker to the acoustic center point of the target sound field projection area. This indicates the distance (in meters) from the reference loudspeaker to the acoustic center point of the target sound field projection area.

[0044] The time delay compensation is calculated using the following formula:

[0045] Wherein, represents time delay compensation (unit: seconds). This represents the sound wave path difference (in meters) between each loudspeaker and the reference loudspeaker. This represents the speed of sound in the air (unit: meters per second), and the standard value is usually taken as 343 meters per second.

[0046] When a user selects a specific scene tuning mode on the entertainment system interface, such as outdoor music or tailgate theater, the system receives a scene mode trigger signal from the user. Different specific scene tuning modes may have different requirements for the width and depth of the sound field. Therefore, it is necessary to combine the location of the target area and adjust the phase relationship (i.e., the starting position of the waveform) of the sound waves emitted by different speakers to form a sound energy distribution map in the vehicle's interior and exterior space. This concentrates the sound energy in the target sound field projection area, while significantly suppressing energy in other unnecessary areas. For example, it can make the sound beam more focused on a point or create a wider sound field perception in a certain area. The output power, time delay compensation, and phase control parameters are integrated to form audio processing parameters. These parameters are then sent to the vehicle's audio amplifier, which drives the corresponding speakers to project the sound energy corresponding to the processed audio signal into the target sound field projection area. The specific steps include the following: The system detects the opening position of open doors and / or windows, and determines the target sound field projection area based on the opening position; it acquires the speakers in the vehicle that can cover the target sound field projection area and determines the output power of the speakers; it estimates the sound wave path difference based on the installation position of the activated speakers and the target sound field projection area, and determines the time delay compensation for the speakers based on the sound wave path difference; it determines the phase control parameters for the speakers based on the scene mode trigger signal and the target sound field projection area; and it uses the output power, time delay compensation, and phase control parameters as audio processing parameters.

[0047] This embodiment accurately locates the target sound field area by detecting the opening position of the car door / window, selects a speaker that can cover the area, and generates optimized audio processing parameters by setting the output power of the speaker, calculating time delay compensation based on the sound wave path difference, and setting phase parameters in combination with scene mode. This allows the sound energy to be projected more concentratedly and accurately to a specific area outside the vehicle, improving the sound field performance in open environments.

[0048] In step S202, the audio signal is processed according to the audio processing parameters, and the speaker is driven to project the sound energy corresponding to the processed audio signal onto the target sound field projection area.

[0049] In one embodiment, a first ambient sound signal is acquired in real time using a microphone array installed outside the vehicle. The first ambient sound signal is a mixture of all sounds near the target sound field projection area, including the audio to be played, wind noise, traffic noise, and human conversations. The acquired first ambient sound signal is transmitted to the vehicle's infotainment system, which then deconstructs and identifies the first ambient sound signal.

[0050] Specifically, the continuous first ambient sound signal acquired by the microphone array is converted into a discrete first digital signal sequence at a preset sampling rate (e.g., 44.1kHz or 48kHz). Then, the first digital signal sequence is divided into several short subsequences according to a preset time window, for example, the time window length of each subsequence is 20 milliseconds.

[0051] Within each time window, the root mean square (RMS) of the subsequence within the window is calculated to obtain the average effective value of the signal amplitude during that time period.

[0052] in, This represents the average effective value of the amplitude of each subsequence within the time window. It is the length of the time window. It is the start time of the time window. This represents a subsequence within a time window.

[0053] Further convert the average amplitude RMS value into instantaneous sound pressure level in decibels:

[0054] in, It is a reference sound pressure level, usually taken as... (micropascals) This represents the average effective value of the amplitude of each subsequence within the time window. This indicates the instantaneous sound pressure level.

[0055] At the same time, spectral analysis is performed on the first ambient sound signal. For example, the first ambient sound signal can be decomposed from a time domain signal into a frequency domain signal using a fast Fourier transform algorithm, thereby obtaining the energy proportion of the first ambient sound signal in different frequency bands (such as low frequency band, mid frequency band, and high frequency band).

[0056] Furthermore, based on the instantaneous sound pressure level and energy ratio, the first environmental noise component contained in the first environmental sound signal is identified by comparing it with a pre-stored template of typical noise characteristics, and its noise type is further determined. For example, if the signal is found to have highly concentrated energy and a high sound pressure level in the low-frequency band (e.g., 50-200Hz), it can be identified as "wind noise"; if it has highly concentrated energy and a high sound pressure level in the mid-to-high frequency band (e.g., 1kHz-4kHz), it may be identified as "noisy human voices" or "traffic noise".

[0057] If the main noise type is identified as low-frequency wind noise, the gain of the mid-to-high frequency components in the audio signal will be increased accordingly. That is, in the equalizer settings, the mid-to-high frequency band will be boosted, and the phase control parameters will be adjusted to make the sound beam narrower in order to penetrate low-frequency interference.

[0058] If the primary noise type is identified as mid-to-high frequency human voice noise, the overall output power of the audio signal is increased, and the equalization curve is adjusted to enhance the clarity of the core frequency band, such as the speech band 300Hz-3.4kHz. Finally, the audio signal to be played is processed and reproduced according to the updated audio processing parameters. Specifically, the following steps are included: The first ambient sound signal is acquired by a microphone array and transmitted to the vehicle's entertainment system. The entertainment system performs instantaneous sound pressure calculation and spectrum identification on the first ambient sound signal to identify the first ambient noise and its noise type. Based on the identified noise type, the audio processing parameters are updated, and the audio signal is processed according to the updated audio processing parameters.

[0059] This embodiment uses a microphone array to collect and analyze ambient noise outside the vehicle in real time, identifying the intensity and type of noise. Based on this, it dynamically adjusts the audio processing parameters of the speakers to achieve intelligent cancellation and compensation of ambient noise. This ensures that even in noisy environments such as outdoor camping and leisure activities, the sound field projected onto the designated area remains clear and full, significantly improving outdoor sound quality stability and anti-interference capabilities.

[0060] Specifically, the entertainment host performs instantaneous sound pressure level calculation and spectrum identification on the first ambient sound signal to identify the first ambient noise and its noise type, including: The first ambient sound signal is converted into a discrete first digital signal sequence at a preset sampling rate, and the first digital signal sequence is truncated based on several time windows. Within each time window, the root mean square (RMS) of the first digital signal sequence is calculated to obtain the average effective value of the first ambient sound signal within each time window. The average effective value of the amplitude is converted into the instantaneous sound pressure level of the first ambient sound signal. The spectrum analysis of the first ambient sound signal is performed to obtain the energy proportion of the first ambient sound signal in different frequency bands. Based on the energy proportion and the instantaneous sound pressure level, the first ambient noise and the noise type of the first ambient noise contained in the first ambient sound signal are identified.

[0061] This embodiment converts ambient sound signals into digital sequences and performs windowing processing to calculate the instantaneous sound pressure level within each window. Simultaneously, it identifies the energy distribution of different frequency bands through spectrum analysis, thereby quantifying the intensity and type of ambient noise and achieving targeted cancellation of specific noises. This ensures that users can have a stable and clear sound field experience in different noise environments.

[0062] In one embodiment, a corresponding base equalizer curve is first retrieved from a pre-stored equalizer curve database based on the received scene mode trigger signal. An equalizer curve is a set of settings that apply specific gain or attenuation to different frequency components; its graphical representation is a curve. Different scene modes presuppose different acoustic styles; for example, "camping mode" might presuppose a curve emphasizing vocal clarity and naturalness, while "party mode" might presuppose a curve enhancing bass rhythm and overall energy. Furthermore, the base equalizer curve is dynamically updated based on the identified noise type and target sound field projection area.

[0063] Specifically, the identified noise types are mapped to their characteristic frequency bands. In frequency bands where ambient noise energy is concentrated, the gain of that frequency band or the key frequency band that conflicts with it in the audio signal is increased. For example, when the main noise type is identified as low-frequency wind noise, it can be determined that wind noise is masking the mid-low frequency details in the audio. Therefore, based on the basic equalization curve, a gain is applied to the core mid-frequency vocal band of 300Hz-2.5kHz, such as increasing it by 3dB, to ensure that the main melody of speech and instruments can clearly penetrate the low-frequency boom. When mid-high frequency noisy vocals are identified, the gain in the 2kHz-4kHz band of the basic equalization curve is increased to enhance the brightness of the audio signal, making it stand out more in competition with ambient vocals.

[0064] On the other hand, considering the acoustic characteristics of the target sound field projection area, high-frequency components attenuate faster in air when sound propagates in open spaces. Therefore, based on the previously estimated sound wave propagation distance, when updating the equalization curve, compensation is first applied to the high-frequency portion (e.g., above 4kHz) of the basic equalization curve to offset the high-frequency loss during propagation to the target area, ensuring that the sound reaching the user's ear maintains its original frequency balance. Finally, targeted compensation adjustments based on noise type and transmission loss compensation adjustments based on the target area are applied together to the basic equalization curve selected based on the scene mode trigger signal, thereby synthesizing a basic equalization curve updated based on the current time, current environment, and current position. Specifically, the following steps are included: Select the scene mode to trigger the corresponding basic equalization curve, adjust the basic equalization curve according to the noise type and the target sound field projection area, and process the audio signal according to the updated basic equalization curve.

[0065] This embodiment dynamically adjusts the basic equalization curve by combining preset scene modes with real-time noise type identification, thereby adapting to the acoustic needs of different scenarios and specifically compensating for noise interference in specific frequency bands. This ensures that the sound projected onto the target area outside the vehicle maintains optimal clarity and balance, enhancing the adaptability and expressiveness of the sound field in complex noise environments.

[0066] In another embodiment, the vehicle's infotainment system can first store a pre-built user database, which can be synchronized with the user's cloud account. The user database stores one or more playback preference profiles for each identifiable user, which can be associated via a key, Bluetooth connection to a mobile phone, or facial recognition. Therefore, the user's playback preferences can be read from the user database, and these preferences may include the user's personalized equalizer curve, preferred sound pressure level range, and content type association settings.

[0067] The personalized equalizer curve is a specific frequency response curve generated by the user through historical manual settings or by the system learning their listening behavior over a long period. For example, one user might prefer bass enhancement, i.e., a fixed gain in the 50-150Hz frequency range, while another user might prefer prominent vocals, i.e., an boost in the 1kHz-3kHz frequency range. The preferred sound pressure level range refers to the user's accustomed listening volume range, such as preferring 65 dB in a quiet environment and 75 dB in a noisy environment. Content type-related settings refer to the different sound field modes or equalizer preferences that users may preset for different types of audio content, such as "podcasts," "classical music," and "rock music."

[0068] Specifically, after determining the basic audio processing parameters based on the environment and scenario, the user's playback preference profile can be retrieved from the user database. This profile is then integrated with the audio processing parameters, essentially a weighted combination of objective acoustic control requirements and subjective, personalized preferences. The specific steps include: Read the user's playback preferences from the pre-built user database in the vehicle; update the audio processing parameters based on the playback preferences; and process the audio signal according to the updated audio processing parameters.

[0069] Specifically, for output power, the output power value calculated based on ambient noise can be fine-tuned by incorporating the user's preferred sound pressure level range, ensuring the final volume is sufficient to penetrate ambient noise while also conforming to the user's consistent listening habits. For the equalization curve, a weighted superposition operation is performed between the base equalization curve determined based on the scene and noise type and the user's personalized equalization curve, or user preference can be prioritized in specific key frequency bands. For example, even if the mid-frequency is boosted to compensate for wind noise, the updated curve can still retain the low-frequency gain characteristics for users who have explicitly set a "deep bass" preference.

[0070] This embodiment integrates pre-stored user playback preferences into real-time adjustments of audio processing parameters, enabling sound field control to not only respond to environmental changes but also proactively adapt to the user's personalized listening habits. By combining objective acoustic control with subjective preferences, it delivers a clear, stable, and highly personalized sound experience for users in outdoor scenarios such as camping and leisure.

[0071] Reference Figure 4This diagram illustrates the interconnection structure of a vehicle sound field control system provided in an embodiment of the present invention. In scenarios such as convoy assembly and multi-vehicle camping, this embodiment can also form a unified, synchronized, and adaptive large-scale extended sound field. Specifically, the vehicle's infotainment system can wirelessly connect to a portable speaker via Bluetooth or in-vehicle Wi-Fi. The portable speaker can supplement the main sound unit and be placed in a more distant location, such as near a campfire or inside a tent, to further extend the sound field coverage or provide supplementary sound in specific areas, achieving a more immersive listening experience. Especially when the vehicle is open, the portable speaker and the vehicle system can work together to form a wider sound field. When multiple vehicles (such as car A and car B) are camping in the same area, their infotainment systems can interconnect via Bluetooth or in-vehicle Wi-Fi. In this interconnection mode, each vehicle is assigned a corresponding role; for example, one vehicle can act as the main sound source and control center, while other vehicles act as auxiliary sound units, collectively constructing a larger and more magnificent outdoor sound field.

[0072] After establishing the master-slave relationship, the master audio device sends a synchronization signal to the sub-audio devices. This synchronization signal can be a high-precision timestamp or a specific synchronization audio pulse. The synchronization signal is used to calibrate the firing time of the master and sub-audio devices, eliminating asynchrony caused by inherent differences in processing and communication delays between different vehicle audio systems. Furthermore, appropriate time delay compensation can be assigned to the sub-devices to ensure that the sound emitted by all vehicle speakers is precisely synchronized at the target location in space, avoiding interference between sound waves due to time differences, thereby guaranteeing the basic auditory consistency of the combined sound field.

[0073] Furthermore, the system acquires the second real-time opening status of doors and / or windows of vehicles other than the current vehicle. The main device collects opening status data of all connected vehicle doors and / or windows to determine the extended sound field projection area after interconnection. The opening status data includes the first real-time opening status of the current vehicle's doors and / or windows, and the second real-time opening status of doors and / or windows of vehicles other than the current vehicle. Based on the position of each vehicle relative to the extended sound field projection area, the power output of its audio system and the phase parameters of its speakers are adjusted. For details, please refer to the aforementioned embodiments, which will not be repeated here.

[0074] Furthermore, a second ambient sound signal is acquired using the microphone array of the main audio device. This second ambient sound signal represents the environmental noise perceived from the location of the main device within the extended sound field projection area. The main device performs instantaneous sound pressure level calculation and spectrum identification on the acquired signal, as described in the previous embodiment, to identify the second ambient noise and its type. Based on the identified noise type of the second ambient noise, the equalization parameters of the main audio device and the sub-audio devices are adjusted to generate an equalization strategy containing these parameters. This equalization strategy is then distributed to all sub-devices for execution, ensuring consistent frequency response compensation for the sound emitted by all vehicles, collectively combating environmental interference, and providing a clear and uniform listening experience throughout the entire extended sound field projection area. Specifically, the steps include the following: The system acquires the interconnection relationships established between the vehicle and other vehicles, and selects the main audio equipment and sub-audio equipment based on these relationships; it controls the main audio equipment to send synchronization signals to the sub-audio equipment, and uses these synchronization signals to calibrate the sound emission time of the main audio equipment and sub-audio equipment; it acquires the second real-time opening status of the doors and / or windows of other vehicles besides the current vehicle, and adjusts the sound emission parameters of the main audio equipment and sub-audio equipment based on the first and second real-time opening status; it collects a second ambient sound signal based on the microphone array of the main audio equipment, and performs instantaneous sound pressure calculation and spectrum identification on the second ambient sound signal to identify the noise type of the second ambient noise; and it adjusts the equalization parameters of the main audio equipment and sub-audio equipment based on the noise type of the second ambient noise.

[0075] This embodiment establishes a master-slave audio interconnection between vehicles and achieves time synchronization. By coordinating the adjustment of sound parameters based on the status of multiple vehicle doors / windows, a unified and coordinated extended sound field is constructed. Furthermore, the main device collects ambient noise and identifies its type, thereby synchronously optimizing the equalization parameters of each vehicle. In multi-vehicle scenarios outside the vehicle, this effectively overcomes the limitations of a single sound field and provides users with an immersive surround sound experience.

[0076] In one embodiment, a visual editing interface including a top-down view of the vehicle and the opening and closing status of the doors can also be displayed on the vehicle's infotainment system; a drag signal from the user on the sound field element icons in the visual editing interface can be received, and a custom sound field projection area can be determined in the top-down view of the vehicle based on the drag signal; a first audio processing parameter of the speaker can be calculated based on the custom sound field projection area, a first real-time opening status, a second real-time opening status, and the preset position of the speaker; and the speaker can be invoked to play sound based on the first audio processing parameter to form a first predetermined sound field within the custom sound field projection area.

[0077] Specifically, the vehicle's infotainment interface renders a visual editing interface that includes a top-down view of the vehicle and the opening / closing status of the doors. The outline of the top-down view maintains a fixed mapping relationship with the actual size and proportions of the vehicle. For example, when the left front door is open, the corresponding area of ​​the left front door in the top-down view will be visually indicated by highlighting, color changing, or dynamic icons, thus translating the physical opening / closing status of the vehicle into graphical interface information that the user can understand.

[0078] At least one sound field element icon is provided on the side or toolbar of the visual editing interface, such as a graphic symbol representing a sound source or sound focusing area. Users drag this icon using input devices such as touch, mouse, or knob. The system receives and parses the drag signal generated by the dragging action to determine a custom sound field projection area outside the vehicle's overhead view. This custom sound field projection area can be a circular area centered on the drag endpoint, a polygonal area defined by the user, or a specific shape area related to the dragging trajectory. Its geographic coordinates can be obtained through a preset conversion relationship between interface pixel coordinates and the actual spatial coordinates around the vehicle. The user-defined custom sound field projection area includes the desired location and spatial range of the target sound field.

[0079] Sensors installed throughout the vehicle detect open doors and / or windows. Combining this with speaker location information and a custom sound field projection area, first audio processing parameters are calculated to drive each available speaker. These parameters include, but are not limited to, a speaker call necessity determination bit, a multi-channel gain scaling factor, and a time delay parameter. The speaker call necessity determination bit determines whether a speaker makes a substantial contribution to the current sound field projection; if not, it is muted to save energy and avoid interference. The multi-channel gain scaling factor adjusts the output volume ratio of each called speaker to create a specific sound field energy distribution. The time delay parameter applies a slight time delay to the audio signal output by each speaker. Based on these first audio processing parameters, the audio signal undergoes real-time channel allocation, gain adjustment, and delay alignment, and the corresponding speaker is called to play the processed sound.

[0080] In one specific embodiment, a first audio processing parameter for the speaker is calculated based on a custom sound field projection area, a first real-time on state, a second real-time on state, and a preset position of the speaker. Specifically, a first speaker in the vehicle that can cover the custom sound field projection area is acquired, and a first output power of the first speaker is determined; a first sound wave path difference is estimated based on the installation position of the first speaker and the custom sound field projection area, and a time delay compensation for the first speaker is determined based on the first sound wave path difference; a first phase control parameter for the first speaker is determined based on the custom sound field projection area; and the first output power, the first time delay compensation, and the first phase control parameter are used as the first audio processing parameter. The calculation process in this embodiment refers to the steps described above for calculating audio processing parameters, and will not be repeated here.

[0081] This embodiment also provides a vehicle sound field control device for implementing the above embodiments and preferred embodiments; details already described will not be repeated. As used below, the term "module" can refer to a combination of software and / or hardware that performs a predetermined function. Although the device described in the following embodiments is preferably implemented in software, hardware implementation, or a combination of software and hardware, is also possible and contemplated.

[0082] This embodiment provides a sound field control device for a vehicle, such as... Figure 5 As shown, it includes: The parameter generation module 501 is used to respond to the scene mode trigger signal, obtain the first real-time opening status of the car door and / or window, determine the target sound field projection area in the vehicle based on the first real-time opening status, and generate audio processing parameters corresponding to the speakers in the vehicle.

[0083] The sound field control module 502 is used to process the audio signal according to the audio processing parameters and drive the speaker to project the sound energy corresponding to the processed audio signal to the target sound field projection area.

[0084] The parameter generation module 501 also includes: The opening orientation detection unit is used to detect the opening orientation of the car door and / or window when it is open, and to determine the target sound field projection area based on the opening orientation.

[0085] The output power determination unit is used to acquire the loudspeakers in the vehicle that can cover the target sound field projection area and determine the output power of the loudspeakers.

[0086] The delay compensation calculation unit is used to estimate the sound wave path difference based on the installation position of the activated speaker and the target sound field projection area, and to determine the time delay compensation for the speaker based on the sound wave path difference.

[0087] The phase parameter determination unit is used to determine the phase control parameters for the loudspeaker based on the scene mode trigger signal and the target sound field projection area.

[0088] The audio parameter determination unit is used to take output power, time delay compensation, and phase control parameters as audio processing parameters.

[0089] The sound field control module 502 includes: The first noise recognition unit is used to collect a first ambient sound signal through a microphone array and transmit it to the vehicle's entertainment host. The entertainment host performs instantaneous sound pressure calculation and spectrum recognition on the first ambient sound signal to identify the first ambient noise and the noise type of the first ambient noise.

[0090] The first parameter update unit is used to update the audio processing parameters based on the identified noise type, and to process the audio signal according to the updated audio processing parameters.

[0091] The first noise identification unit is specifically used to convert a first ambient sound signal into a discrete first digital signal sequence at a preset sampling rate, and to extract the first digital signal sequence based on several time windows; within each time window, to perform root mean square calculation on the first digital signal sequence to obtain the average effective value of the first ambient sound signal within each time window; to convert the average effective value of the amplitude into the instantaneous sound pressure level of the first ambient sound signal; to perform spectral analysis on the first ambient sound signal to obtain the energy proportion of the first ambient sound signal in different frequency bands; and to identify the first ambient noise and the noise type of the first ambient noise contained in the first ambient sound signal based on the energy proportion and the instantaneous sound pressure level.

[0092] The first parameter update unit is specifically used to select the basic equalization curve corresponding to the scene mode trigger signal, adjust the basic equalization curve according to the noise type and the target sound field projection area, and process the audio signal according to the updated basic equalization curve.

[0093] The sound field control module 502 also includes: The user preference reading unit is used to read the user's playback preferences from the user database pre-built in the vehicle.

[0094] The second parameter update unit is used to update the audio processing parameters based on playback preferences and process the audio signal according to the updated audio processing parameters.

[0095] The device also includes: The main device determination unit is used to acquire the interconnection relationship established between the vehicle and other vehicles, and select the main audio device and sub-audio devices based on the interconnection relationship.

[0096] The time calibration unit is used to control the main audio equipment to send synchronization signals to the sub-audio equipment, and to use the synchronization signals to calibrate the sound output time of the main audio equipment and the sub-audio equipment.

[0097] The sound parameter adjustment unit is used to obtain the second real-time opening status of the doors and / or windows of other vehicles besides the current vehicle, and adjust the sound parameters of the main audio equipment and the sub-audio equipment according to the first real-time opening status and the second real-time opening status. The noise recognition unit is used to collect the second ambient sound signal based on the microphone array of the main audio device, and to perform instantaneous sound pressure calculation and spectrum recognition on the second ambient sound signal to identify the noise type of the second ambient noise. The equalization parameter adjustment unit is used to adjust the equalization parameters of the main audio equipment and the sub-audio equipment based on the noise type of the second ambient noise.

[0098] The device also includes: The interface display module is used to display a visual editing interface on the vehicle's infotainment system, which includes a top-down view of the vehicle and the opening and closing status of the doors.

[0099] A custom module is used to receive drag signals from the user on the sound field element icons in the visual editing interface, and to determine the custom sound field projection area in the vehicle top view based on the drag signals.

[0100] The first audio processing module is used to calculate the first audio processing parameters of the speaker based on the custom sound field projection area, the first real-time on status, the second real-time on status, and the preset position of the speaker.

[0101] The first sound field generation module is used to call the speaker to play sound according to the first audio processing parameters, so as to form a first predetermined sound field within a custom sound field projection area.

[0102] The first audio processing module is specifically used to acquire a first speaker in the vehicle that can cover a custom sound field projection area, and determine the first output power of the first speaker; estimate the first sound wave path difference based on the installation position of the first speaker and the custom sound field projection area, and determine the time delay compensation for the first speaker based on the first sound wave path difference; determine the first phase control parameter for the first speaker based on the custom sound field projection area; and use the first output power, the first time delay compensation, and the first phase control parameter as the first audio processing parameter.

[0103] In this embodiment, the vehicle's sound field control device is presented in the form of a functional unit. Here, a unit refers to an ASIC circuit, a processor and memory that execute one or more software or fixed programs, and / or other devices that can provide the above-mentioned functions.

[0104] Further functional descriptions of the above modules and units are the same as those in the corresponding embodiments described above, and will not be repeated here.

[0105] like Figure 3 As shown, this embodiment also provides a vehicle sound field control system, which is used to implement the above embodiments and preferred embodiments; details already described will not be repeated. The system includes: The entertainment head unit communicates unidirectionally with microphones outside and / or inside the vehicle via the A2B line; receives real-time information on the opening status of doors and / or windows from the audio amplifier via the CAN line and sends control commands for the speakers to the audio amplifier; and transmits processed audio signals unidirectionally to the audio amplifier via the A2B line.

[0106] The in-vehicle speaker system receives control commands transmitted from the audio amplifier via simulated twisted-pair cables.

[0107] The audio amplifier communicates unidirectionally with microphones outside and / or inside the vehicle via an A2B cable.

[0108] This invention also provides a computer device having the above-described features. Figure 5 The vehicle's sound field control device is shown.

[0109] Please see Figure 6 , Figure 6 This is a schematic diagram of the structure of a computer device provided in an optional embodiment of the present invention, such as... Figure 6 As shown, the computer device includes one or more processors 10, memory 20, and interfaces for connecting the components, including high-speed interfaces and low-speed interfaces. The components communicate with each other via different buses and can be mounted on a common motherboard or otherwise installed as needed. The processors can process instructions executed within the computer device, including instructions stored in or on memory to display graphical information of a GUI on external input / output devices (such as display devices coupled to the interfaces). In some alternative implementations, multiple processors and / or multiple buses can be used with multiple memories and multiple memory modules, if desired. Similarly, multiple computer devices can be connected, each providing some of the necessary operations (e.g., as a server array, a group of blade servers, or a multiprocessor system). Figure 6 Take a processor 10 as an example.

[0110] Processor 10 may be a central processing unit, a network processor, or a combination thereof. Processor 10 may further include a hardware chip. The hardware chip may be an application-specific integrated circuit (ASIC), a programmable logic device (PLD), or a combination thereof. The programmable logic device may be a complex programmable logic device (CAMP), a field-programmable gate array (FPGA), a general-purpose array logic (GDA), or any combination thereof.

[0111] The memory 20 stores instructions executable by at least one processor 10 to cause at least one processor 10 to perform the method shown in the above embodiments.

[0112] The memory 20 may include a program storage area and a data storage area. The program storage area may store the operating system and applications required for at least one function; the data storage area may store data created based on the use of the computer device as shown by a landing page for an app. Furthermore, the memory 20 may include high-speed random access memory and may also include non-transitory memory, such as at least one disk storage device, flash memory device, or other non-transitory solid-state storage device. In some alternative embodiments, the memory 20 may optionally include memory remotely located relative to the processor 10, which can be connected to the computer device via a network. Examples of such networks include, but are not limited to, the Internet, intranets, local area networks, mobile communication networks, and combinations thereof.

[0113] The memory 20 may include volatile memory, such as random access memory; the memory may also include non-volatile memory, such as flash memory, hard disk or solid-state drive; the memory 20 may also include a combination of the above types of memory.

[0114] The computer device also includes a communication interface 30 for communicating with other devices or communication networks.

[0115] This invention also provides a computer-readable storage medium. The methods described above according to embodiments of the invention can be implemented in hardware or firmware, or implemented as computer code that can be recorded on a storage medium, or implemented as computer code downloaded via a network and originally stored on a remote storage medium or a non-transitory machine-readable storage medium and then stored on a local storage medium. Thus, the methods described herein can be processed by software stored on a storage medium using a general-purpose computer, a dedicated processor, or programmable or dedicated hardware. The storage medium can be a magnetic disk, optical disk, read-only memory, random access memory, flash memory, hard disk, or solid-state drive, etc.; further, the storage medium can also include combinations of the above types of memory. It is understood that computers, processors, microprocessor controllers, or programmable hardware include storage components capable of storing or receiving software or computer code, which, when accessed and executed by the computer, processor, or hardware, implements the methods shown in the above embodiments.

[0116] Although embodiments of the invention have been described in conjunction with the accompanying drawings, those skilled in the art can make various modifications and variations without departing from the spirit and scope of the invention, and such modifications and variations all fall within the scope defined by the appended claims.

Claims

1. A method for controlling the sound field of a vehicle, characterized in that, The method includes: In response to a scene mode trigger signal, the first real-time opening status of the vehicle door and / or window is acquired, the target sound field projection area in the vehicle is determined based on the first real-time opening status, and audio processing parameters corresponding to the speakers in the vehicle are generated. The audio signal is processed according to the audio processing parameters, and the speaker is driven to project the sound energy corresponding to the processed audio signal onto the target sound field projection area.

2. The method according to claim 1, characterized in that, The step of determining the target sound field projection area in the vehicle based on the first real-time activation status and generating audio processing parameters corresponding to the speakers in the vehicle includes: Detect the opening position of the vehicle door and / or the vehicle window when they are open, and determine the target sound field projection area based on the opening position; Acquire the loudspeakers in the vehicle that can cover the target sound field projection area, and determine the output power of the loudspeakers; The sound wave path difference is estimated based on the installation location of the loudspeaker and the target sound field projection area, and the time delay compensation for the loudspeaker is determined based on the sound wave path difference. The phase control parameters for the loudspeaker are determined based on the scene mode trigger signal and the target sound field projection area; The output power, the time delay compensation, and the phase control parameters are used as the audio processing parameters.

3. The method according to claim 1, characterized in that, The process of processing the audio signal according to the audio processing parameters includes: The first ambient sound signal is collected by a microphone array and transmitted to the vehicle's entertainment host. The entertainment host performs instantaneous sound pressure calculation and spectrum identification on the first ambient sound signal to identify the first ambient noise and the noise type of the first ambient noise. The audio processing parameters are updated based on the identified noise type, and the audio signal is processed according to the updated audio processing parameters.

4. The method according to claim 3, characterized in that, The step of performing instantaneous sound pressure level calculation and spectrum identification on the first ambient sound signal through the entertainment host to identify the first ambient noise and the noise type of the first ambient noise includes: The first ambient sound signal is converted into a discrete first digital signal sequence at a preset sampling rate, and the first digital signal sequence is extracted based on several time windows; Within each time window, the root mean square of the first digital signal sequence is calculated to obtain the average effective value of the first ambient sound signal within each time window. The average amplitude RMS value is converted into the instantaneous sound pressure level of the first ambient sound signal; Spectral analysis of the first ambient sound signal is performed to obtain the energy proportion of the first ambient sound signal in different frequency bands; Based on the energy ratio and the instantaneous sound pressure level, the first ambient noise contained in the first ambient sound signal and the noise type of the first ambient noise are identified.

5. The method according to claim 3, characterized in that, The step of updating the audio processing parameters based on the identified noise type and processing the audio signal according to the updated audio processing parameters includes: Select the basic equalization curve corresponding to the scene mode trigger signal, adjust the basic equalization curve according to the noise type and the target sound field projection area, and process the audio signal according to the updated basic equalization curve.

6. The method according to claim 1, characterized in that, The step of processing the audio signal according to the audio processing parameters further includes: Read the user's playback preferences from the user database pre-built in the vehicle; The audio processing parameters are updated based on the playback preference, and the audio signal is processed according to the updated audio processing parameters.

7. The method according to claim 1, characterized in that, The method further includes: The system acquires the interconnection relationships established between the vehicle and other vehicles, and selects the main audio device and sub-audio devices based on these interconnection relationships. The main audio device is controlled to send a synchronization signal to the sub-audio device, and the sound emission time of the main audio device and the sub-audio device is calibrated using the synchronization signal; The system acquires the second real-time opening status of doors and / or windows of vehicles other than the current vehicle, and adjusts the sound parameters of the main audio device and the sub-audio device based on the first real-time opening status and the second real-time opening status. The second ambient sound signal is acquired based on the microphone array of the main audio device, and instantaneous sound pressure level and spectrum identification are performed on the second ambient sound signal to identify the second ambient noise and the noise type of the second ambient noise; Adjust the equalization parameters of the main speaker and the sub-speakers based on the noise type of the second ambient noise.

8. The method according to claim 1, characterized in that, The method further includes: The vehicle's infotainment system displays a visual editing interface that includes a top-down view of the vehicle and the opening and closing status of the doors; Receive the user's drag signal on the sound field element icon in the visual editing interface, and determine the custom sound field projection area in the vehicle top view according to the drag signal; Based on the custom sound field projection area, the first real-time on status, the second real-time on status, and the preset position of the speaker, calculate the first audio processing parameters of the speaker; Based on the first audio processing parameters, the speaker is invoked to play sound in order to form a first predetermined sound field within the custom sound field projection area.

9. The method according to claim 8, characterized in that, The step of calculating the first audio processing parameters of the speaker based on the custom sound field projection area, the first real-time on status, the second real-time on status, and the preset position of the speaker includes: Acquire the first loudspeaker in the vehicle that can cover the custom sound field projection area, and determine the first output power of the first loudspeaker; Estimate the first sound wave path difference based on the installation position of the first speaker and the custom sound field projection area, and determine the time delay compensation for the first speaker based on the first sound wave path difference; The first phase control parameters for the first speaker are determined based on the custom sound field projection area. The first output power, the first time delay compensation, and the first phase control parameter are used as the first audio processing parameters.

10. A sound field control device for a vehicle, characterized in that, The device includes: The parameter generation module is used to respond to the scene mode trigger signal, acquire the first real-time opening status of the car door and / or window, determine the target sound field projection area in the vehicle based on the first real-time opening status, and generate the audio processing parameters corresponding to the speakers in the vehicle. The sound field control module is used to process the audio signal according to the audio processing parameters and drive the speaker to project the sound energy corresponding to the processed audio signal onto the target sound field projection area.

11. A sound field control system for a vehicle, characterized in that, The system, employing the method of any one of claims 1 to 9, comprises: The entertainment host communicates unidirectionally with microphones outside and / or inside the vehicle via an A2B line; receives real-time door and / or window opening status data from the audio amplifier via a CAN line and sends control commands for the speakers to the audio amplifier; and transmits processed audio signals unidirectionally to the audio amplifier via an A2B line. The in-vehicle system speakers receive control commands transmitted by the audio amplifier via simulated twisted-pair cables. The audio amplifier communicates unidirectionally with microphones outside and / or inside the vehicle via an A2B cable.

12. A computer device, characterized in that, include: A memory and a processor, the memory and the processor being communicatively connected to each other, the memory storing computer instructions, the processor executing the computer instructions to perform the method of any one of claims 1 to 9.

13. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores computer instructions for causing a computer to perform the method of any one of claims 1 to 9.

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