Methods, devices, equipment, storage media and products for controlling the sound field of a call.

Abstract

This application discloses a method, apparatus, device, storage medium, and product for controlling the sound field of a call, relating to the field of sound field control technology. The method includes: receiving an ambient echo signal via a microphone, wherein the ambient echo signal is reflected back from an FM signal emitted by a first speaker, and the instantaneous frequency of the FM signal changes linearly with time; extracting a beat frequency signal between the ambient echo signal and the FM signal; calculating a target distance between the first speaker and the target call based on the beat frequency signal; and adjusting the call audio signal of the first speaker based on the target distance to counteract call sound field fluctuations caused by distance changes. This application adjusts the call audio signal of the first speaker according to the target distance to counteract call sound field fluctuations caused by distance changes, thereby obtaining a stable and consistent listening experience.

Description

Technical Field

[0001] This application relates to the field of sound field control technology, and in particular to a method, apparatus, device, storage medium and product for controlling the sound field during a call. Background Technology

[0002] In related technologies, when the distance between the user and the mobile phone changes, the sound pressure level actually received by the user will fluctuate significantly. This can be compensated for by the user manually adjusting the volume or by simple automatic gain control, but these methods cannot achieve a stable and consistent listening experience. Summary of the Invention

[0003] The main objective of this application is to provide a method, apparatus, device, storage medium, and product for controlling the sound field during a call, which aims to solve the technical problem that compensation can not achieve a stable and consistent listening experience through manual volume adjustment or simple automatic gain control.

[0004] To achieve the above objectives, this application proposes a call sound field control method, which includes: The system receives ambient echo signals via a microphone, wherein the ambient echo signals are reflected back from an FM signal emitted by a first loudspeaker, and the instantaneous frequency of the FM signal changes linearly with time. Extract the beat frequency signal between the environmental echo signal and the frequency modulation signal, and calculate the target distance between the first speaker and the target being communicated based on the beat frequency signal; Based on the target distance, the call audio signal of the first speaker is adjusted to counteract the call sound field fluctuations caused by the distance change.

[0005] In one embodiment, the step of extracting the beat frequency signal between the ambient echo signal and the frequency modulation signal, and calculating the target distance between the first speaker and the target communication object based on the beat frequency signal, includes: Time-align the ambient echo signal and the frequency modulation signal; The aligned environmental echo signal and the aligned frequency-modulated signal are mixed to generate a beat frequency signal that represents time delay information. Based on the beat frequency signal, the channel impulse response is calculated, and based on the channel impulse response, the target distance between the first speaker and the target being communicated with is calculated.

[0006] In one embodiment, the step of calculating the channel impulse response based on the beat frequency signal, and calculating the target distance between the first speaker and the target being communicated based on the channel impulse response, includes: The beat frequency signal of the time domain type is converted into a frequency domain signal, wherein different propagation paths in the frequency domain signal correspond to different frequency components; The frequency domain signal is mapped to obtain the channel impulse response. The distance corresponding to the preset signal amplitude point is selected from the channel impulse response as the target distance between the first speaker and the target being communicated.

[0007] In one embodiment, the step of adjusting the call audio signal of the first speaker based on the target distance to counteract the call sound field fluctuations caused by distance changes includes: Calculate the compensation gain based on the target distance and the preset reference distance; Based on the compensation gain, the call audio signal of the first speaker is adjusted to counteract the call sound field fluctuations caused by distance changes.

[0008] In one embodiment, the step of adjusting the call audio signal of the first speaker based on the target distance to compensate for call sound field fluctuations caused by distance changes includes: A preset complex control weight is obtained, and the second speaker is controlled to send a sound wave signal corresponding to the preset complex control weight. The sound wave signal is used to cancel the sound wave signal of the call audio signal in a preset audio leakage area. The preset complex control weight includes amplitude and phase.

[0009] In one embodiment, the step of obtaining a preset complex control weight and controlling the second speaker to send an acoustic signal corresponding to the preset complex control weight includes: A preset number of spatial sampling points are selected within the preset audio leakage area; Calculate the sound field vector of the first loudspeaker at the spatial sampling point, and the propagation vector of the second loudspeaker at the spatial sampling point; Obtain the initial complex control weights, and construct the total sound field function based on the sound field vector, the propagation vector, and the initial complex control weights; The total sound field function is iteratively optimized to obtain a preset complex control weight that minimizes the total sound field.

[0010] Furthermore, to achieve the above objectives, this application also proposes a call sound field control device, which includes: A receiving module is used to receive an ambient echo signal via a microphone, wherein the ambient echo signal is reflected back from an FM signal emitted by a first loudspeaker, and the instantaneous frequency of the FM signal changes linearly with time. The extraction module is used to extract the beat frequency signal between the environmental echo signal and the frequency modulation signal, and to calculate the target distance between the first speaker and the target being communicated based on the beat frequency signal; An adjustment module is used to adjust the call audio signal of the first speaker based on the target distance to counteract the call sound field fluctuations caused by distance changes.

[0011] In addition, to achieve the above objectives, this application also proposes a call sound field control device, the device comprising: a memory, a processor, and a computer program stored in the memory and executable on the processor, the computer program being configured to implement the steps of the call sound field control method as described above.

[0012] In addition, to achieve the above objectives, this application also proposes a storage medium, which is a computer-readable storage medium, on which a computer program is stored, and when the computer program is executed by a processor, it implements the steps of the call sound field control method described above.

[0013] In addition, to achieve the above objectives, this application also provides a computer program product, which includes a computer program that, when executed by a processor, implements the steps of the call sound field control method described above.

[0014] One or more technical solutions proposed in this application have at least the following technical effects: In related technologies, when the distance between the user and the mobile phone changes, the sound pressure level actually received by the user fluctuates significantly. Compensation is achieved through manual volume adjustment or simple automatic gain control, but these methods cannot achieve a stable and consistent listening experience. In contrast, this application receives an ambient echo signal via a microphone. This ambient echo signal is reflected back from an FM signal emitted by a first speaker, and the instantaneous frequency of the FM signal changes linearly with time. The application extracts a beat frequency signal between the ambient echo signal and the FM signal, calculates the target distance between the first speaker and the target being communicated based on the beat frequency signal, and adjusts the call audio signal of the first speaker based on the target distance to offset the call sound field fluctuations caused by distance changes. By receiving the ambient echo signal reflected back from the FM signal emitted by the first speaker via a microphone, this application can extract the beat frequency signal between the ambient echo signal and the FM signal because the instantaneous frequency of the FM signal changes linearly with time. Based on the beat frequency signal, the target distance between the speaker and the target being communicated can be calculated, and the call audio signal of the first speaker can be adjusted based on the target distance to offset the call sound field fluctuations caused by distance changes, thus achieving a stable and consistent listening experience. Attached Figure Description

[0015] The accompanying drawings, which are incorporated in and form part of this specification, illustrate embodiments consistent with this application and, together with the description, serve to explain the principles of this application.

[0016] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, for those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0017] Figure 1 This is a flowchart illustrating an embodiment of the call sound field control method of this application. Figure 2 This is a flowchart illustrating Embodiment 2 of the call sound field control method of this application; Figure 3 A data acquisition prompt diagram for the call sound field control method of this application; Figure 4 This is a schematic diagram of the module structure of the call sound field control device according to an embodiment of this application; Figure 5 This is a schematic diagram of the device structure of the hardware operating environment involved in the call sound field control method in the embodiments of this application.

[0018] The purpose, features, and advantages of this application will be further explained in conjunction with the embodiments and with reference to the accompanying drawings. Detailed Implementation

[0019] It should be understood that the specific embodiments described herein are merely illustrative of the technical solutions of this application and are not intended to limit this application.

[0020] To better understand the technical solution of this application, a detailed description will be provided below in conjunction with the accompanying drawings and specific implementation methods.

[0021] The main solution of this application embodiment is: receiving an ambient echo signal via a microphone, wherein the ambient echo signal is reflected back by an FM signal emitted by a first speaker, and the instantaneous frequency of the FM signal changes linearly with time; extracting a beat frequency signal between the ambient echo signal and the FM signal; calculating the target distance between the first speaker and the target being communicated based on the beat frequency signal; and adjusting the communication audio signal of the first speaker based on the target distance to counteract the communication sound field fluctuations caused by the distance change.

[0022] In related technologies, when the distance between the user and the mobile phone changes, the sound pressure level actually received by the user will fluctuate significantly. This can be compensated for by the user manually adjusting the volume or by simple automatic gain control, but these methods cannot achieve a stable and consistent listening experience.

[0023] After receiving the ambient echo signal reflected back from the FM signal emitted by the first speaker via the microphone, this application can extract the beat frequency signal between the ambient echo signal and the FM signal because the instantaneous frequency of the FM signal changes linearly with time. Then, based on the beat frequency signal, the target distance between the speaker and the target of the call can be calculated, and the call audio signal of the first speaker can be adjusted according to the target distance to counteract the call sound field fluctuation caused by the distance change, so as to obtain a stable and consistent listening experience.

[0024] It should be noted that the executing entity in this embodiment can be a computing service device with data processing, network communication, and program execution functions, such as a tablet computer, personal computer, or mobile phone, or an electronic device or call sound field control device capable of performing the above functions. The following description uses a call sound field control device as an example to illustrate this embodiment and the subsequent embodiments.

[0025] Based on this, embodiments of this application provide a method for controlling the sound field of a call, referring to... Figure 1 , Figure 1 This is a flowchart illustrating the first embodiment of the call sound field control method of this application.

[0026] In this embodiment, the call sound field control method includes steps S10 to S40: Step S10: Receive ambient echo signal through microphone, wherein the ambient echo signal is reflected back by an FM signal emitted by the first speaker, and the instantaneous frequency of the FM signal changes linearly with time. It should be noted that the execution entity in this embodiment is a call sound field control device. This call sound field control device can be a mobile terminal device or any terminal device capable of making external calls. The first speaker refers to the main speaker in the mobile terminal used to transmit ranging signals and play call sounds, typically a front-facing speaker, and includes at least two. The frequency modulation signal is a high-frequency linear frequency modulation (FMCW) signal, whose instantaneous frequency changes linearly with time. The call sound field control device receives environmental echo signals through a microphone. These environmental echo signals are formed by the frequency modulation signal emitted by the first speaker propagating in space and being reflected by the user or surrounding objects. The frequency modulation signal adopts a waveform structure where the instantaneous frequency changes linearly with time, i.e., a linear frequency modulation signal. The first speaker emits the frequency modulation signal whose instantaneous frequency changes linearly with time, and the microphone receives its reflected echo, thereby encoding the spatial propagation distance information in the echo signal in the form of a frequency difference.

[0027] Specifically, a frequency modulation signal can be represented as:

[0028] in, The starting frequency, The sweep bandwidth is the difference between the cutoff frequency and the starting frequency, t is the current time, and T is the duration of one complete frequency sweep of the linear frequency modulated signal.

[0029] Step S20: Extract the beat frequency signal between the ambient echo signal and the frequency modulation signal, and calculate the target distance between the first speaker and the target being communicated based on the beat frequency signal; It is understood that the frequency modulation (FM) signal refers to an ideal signal that is mathematically identical to the FM signal actually emitted by the first loudspeaker. This signal is used to compare with the received ambient echo signal to extract propagation delay information. The beat frequency signal refers to a low-frequency signal obtained by multiplying the received ambient echo signal and the FM signal by their conjugates. Different frequency components in this signal correspond to different propagation path time delays. The communication sound field control device performs conjugate multiplication of the received ambient echo signal and the FM signal to extract the beat frequency signal between them. This beat frequency signal converts the time delay information in the echo signal into frequency information. Subsequently, based on the beat frequency signal, the device calculates the propagation time delay of the sound wave from the first loudspeaker to the communication target and back to the microphone, and then converts it into the target distance between the first loudspeaker and the communication target. By extracting the beat frequency signal between the ambient echo signal and the FM signal, the time delay of sound wave propagation is mapped to the frequency components of the beat frequency signal, thereby transforming distance measurement into a frequency estimation problem.

[0030] Specifically, in linear frequency modulation (LFM) signals, the instantaneous frequency changes strictly linearly with time. When the transmitted signal is reflected back from the target, the received echo signal has a time delay (τ) relative to the transmitted signal because sound waves take time to propagate in space. Since the frequency increases linearly with time, this time delay τ directly results in a fixed difference between the instantaneous frequency of the transmitted signal and the instantaneous frequency of the received signal at any given time t. This difference is the beat frequency. Because the slope of the frequency change (i.e., the sweep bandwidth divided by the frequency modulation period) is constant, the time delay τ and the resulting beat frequency... A one-to-one linear relationship is formed between them: the greater the time delay, the greater the frequency difference. Therefore, the device does not need to directly measure extremely short time intervals (which is easier in acoustics than electromagnetic waves due to the slower speed of sound, but is still limited by the sampling rate), but only needs to measure a stable frequency value.

[0031] Step S30: Based on the target distance, adjust the call audio signal of the first speaker to counteract the call sound field fluctuations caused by the distance change.

[0032] It should be noted that the call audio signal refers to the sound signal played by the mobile terminal to the user during a call. Call sound field fluctuation refers to the fluctuation in the sound pressure level actually received by the user when the distance between the user and the terminal changes, due to the attenuation of sound waves in the air. The call sound field control device, based on the physical laws of sound wave propagation, compares the current target distance with a preset reference distance, calculates the compensation gain to offset the sound pressure fluctuations caused by distance changes, and applies this gain to the call audio signal from the first speaker. Therefore, when the distance between the user and the terminal changes, the sound pressure level actually received by the user remains relatively stable, effectively reducing call sound field fluctuations caused by distance changes.

[0033] Furthermore, since the distance perception of this application only requires the device's own speaker and microphone, no additional hardware sensors are needed, and the ranging process runs in parallel with the voice call, there is less interference with the audio frequency band.

[0034] In one feasible implementation, step S20 includes: Time-align the ambient echo signal and the frequency modulation signal; Understandably, the call sound field control device performs cross-correlation calculations on the received ambient echo signal and the frequency modulation signal. By analyzing the correlation between the two signals at different time offsets, it finds the time offset point that maximizes the correlation, thereby determining the starting position of the echo signal relative to the transmitted signal, and thus achieving alignment of the two signals on the time axis.

[0035] Specifically, because the ambient echo signal contains other irrelevant frequency band signals, the call sound field control device performs bandpass filtering on the signal after obtaining it. This reduces the total power of the background noise, significantly increasing the proportion of effective signal in the ambient echo signal to the remaining signal (i.e., signal-to-noise ratio). After bandpass filtering, the ambient echo signal is cross-correlated with the reference signal to achieve synchronous positioning.

[0036] in, Let be the cross-correlation function, a function of time delay τ. For each possible time delay τ, a similarity value is calculated. s(t) is the frequency-modulated signal, and r(t) is the ambient echo signal. This is a time delay.

[0037] Specifically, at each time point t, the call sound field control device multiplies the ambient echo signal and the frequency modulation signal point by point. When the two signal waveforms are similar and aligned, their product, i.e., the similarity value, will be large (positive number multiplied by positive number, or negative number multiplied by negative number). When the two signals are uncorrelated (for example, one is a signal and the other is noise), their product, i.e., the similarity value, will tend to zero after long-term integration. By finding the τ value corresponding to the maximum value of the cross-correlation function, the ambient echo signal and the frequency modulation signal can be accurately aligned.

[0038] The aligned environmental echo signal and the aligned frequency-modulated signal are mixed to generate a beat frequency signal that represents time delay information. It should be noted that the communication sound field control device performs conjugate multiplication of the two signals, that is, multiplies the aligned environmental echo signal with the frequency-modulated signal, quantizes the time delay information in the signal into a low-frequency beat frequency signal, so that the delay of different propagation paths corresponds to different beat frequencies, thereby generating a beat frequency signal carrying time delay information, transforming the spatial distance measurement problem into a frequency domain analysis problem, and providing a signal form that can be directly analyzed in the frequency domain for subsequent channel impulse response extraction and distance calculation.

[0039] Specifically, the expression for the beat frequency signal is:

[0040] in, It is the complex conjugate of the frequency modulation signal.

[0041] Based on the beat frequency signal, the channel impulse response is calculated, and based on the channel impulse response, the target distance between the first speaker and the target being communicated with is calculated.

[0042] It is understandable that the channel impulse response is a function of the time delay and attenuation characteristics of each propagation path of the sound wave from the first speaker to the microphone, where the main peak corresponds to the direct or main reflection path. The call sound field control device obtains the corresponding channel impulse response based on the generated beat frequency signal, selects the main peak with the largest amplitude from the channel impulse response, and extracts the time delay corresponding to this main peak. This main peak is the main propagation path of the sound wave reflected from the call target. The extracted time delay is then used in conjunction with the speed of sound to calculate the target distance between the first speaker and the call target.

[0043] In one feasible implementation, the steps of calculating the channel impulse response based on the beat frequency signal and calculating the target distance between the first speaker and the target being communicated based on the channel impulse response include: The beat frequency signal of the time domain type is converted into a frequency domain signal, wherein different propagation paths in the frequency domain signal correspond to different frequency components; It should be noted that frequency components refer to the energy components corresponding to different frequencies in a frequency domain signal. The time delays of different propagation paths are mapped to different frequency values. The communication sound field control device performs a Fast Fourier Transform on the time-domain beat frequency signal, converting it to the frequency domain to obtain a frequency domain signal. Since different frequency components in the beat frequency signal correspond to different time delays, and different propagation paths have different time delays, in the converted frequency domain signal, different propagation paths are mapped to different frequency components. The magnitude of each frequency component reflects the reflection energy intensity of the corresponding path. Because this step converts the time-domain beat frequency signal into a frequency domain signal, multiple propagation paths that were originally superimposed and difficult to distinguish in the time domain are separated into their own independent frequency components in the frequency domain, thereby achieving the distinction and identification of different reflection paths.

[0044] Specifically, the frequency domain signal expression is as follows: Where FFT is the formula for Fast Fourier Transform.

[0045] In the frequency domain, each frequency component corresponds to the delay information of different propagation paths, and its frequency and time delay satisfy an approximately linear relationship:

[0046] in, B is the beat frequency, B is the sweep bandwidth, and T is the frequency modulation period. This is to delay the transmission time.

[0047] The frequency domain signal is mapped to obtain the channel impulse response. The distance corresponding to the preset signal amplitude point is selected from the channel impulse response as the target distance between the first speaker and the target being communicated.

[0048] As can be understood, coordinate mapping refers to the process of converting the frequency axis in a frequency domain signal into a time axis or a distance axis. Based on the linear relationship between beat frequency and time delay, frequency components are mapped to corresponding propagation time delays or spatial distances. The preset signal amplitude point refers to the amplitude feature point pre-set in the channel impulse response for extracting the target distance. Typically, the position corresponding to the main peak with the largest amplitude is selected, representing the main reflection path. The communication sound field control device performs coordinate mapping on the converted frequency domain signal, converting the frequency components into a time axis or a distance axis according to the linear relationship between beat frequency and time delay, thereby obtaining the channel impulse response. This channel impulse response, with time or distance as the abscissa and energy amplitude as the ordinate, clearly shows the time delay and attenuation characteristics of each propagation path. The device selects the preset signal amplitude point in the channel impulse response, i.e., the distance value corresponding to the main peak with the largest amplitude, as the target distance between the first speaker and the communication target.

[0049] Furthermore, if the channel impulse response is plotted on time, the target distance can be calculated further:

[0050] Where c is the speed of sound.

[0051] Optionally, during the ranging process, the communication sound field control device performs Kalman filtering based on the channel impulse response of multiple consecutive frames to eliminate outlier delay values ​​introduced by non-target reflectors in the environment, and uses the filtered delay values ​​as the basis for calculating the target distance.

[0052] It should be noted that during the ranging process across multiple consecutive frames, the device performs Kalman filtering on the channel impulse response peak delay value extracted in each frame. This filtering algorithm establishes a state prediction model based on the physical continuity of the target's motion, fusing the delay observation value of the current frame with the filtered prediction value of the previous frame. When an outlier delay value appears in the channel impulse response of a certain frame due to interference from environmental reflectors, the Kalman filter suppresses this outlier value based on historical motion trends, outputting a smooth and continuous target delay estimate. Thus, the device uses the filtered delay value as the basis for calculating the target distance, effectively eliminating ranging interference introduced by non-target reflectors. Because this step introduces Kalman filtering during the ranging process across multiple consecutive frames, and utilizes the physical continuity of the target's motion to estimate the state of the channel impulse response peak delay, it can identify and suppress outliers by using historical motion trends when environmental reflectors (such as static objects like desktops, walls, or temporary obstructions) generate false peaks.

[0053] Furthermore, the call sound field control device can obtain the change of user distance over time through continuous frame calculation, thereby achieving real-time distance tracking.

[0054] In one feasible implementation, step S30 includes: Calculate the compensation gain based on the target distance and the preset reference distance; It should be noted that the preset reference distance This is a pre-set reference distance value, representing the spatial distance between the user and the terminal when in an ideal call state. Compensation gain refers to the amplification or attenuation coefficient applied to the output signal of the first speaker to compensate for the attenuation of sound waves due to distance changes during propagation, used to ensure that the sound pressure level at the target distance is consistent with the sound pressure level at the reference distance. The call sound field control device acquires the currently measured target distance in real time and calls upon the pre-set reference distance. Based on the physical laws of sound wave propagation in a free sound field, i.e., the sound pressure amplitude and distance approximately satisfy an inverse relationship, it calculates the compensation gain.

[0055] Specifically, the device calculates and compensates for the user's real-time distance gain. Compensation gain refers to the amplification or attenuation coefficient applied to the loudspeaker output signal to compensate for the attenuation of sound waves due to changes in distance during propagation. In the frequency domain, it can be determined based on the transfer function of the user's position (…). ) and reference transfer function Construction gain:

[0056] Where f is the frequency at the current moment, and d is the target distance.

[0057] Furthermore, the device limits the gain range to avoid over-amplification or attenuation. This gain is applied to the front speaker to keep the sound pressure level at the user's location relatively stable, thereby reducing volume fluctuations caused by user movement.

[0058] Based on the compensation gain, the call audio signal of the first speaker is adjusted to counteract the call sound field fluctuations caused by distance changes.

[0059] Understandably, the call sound field control device applies the calculated compensation gain to the call audio signal from the first speaker, dynamically adjusts the signal, and then plays the adjusted call audio signal through the first speaker, ensuring that the sound pressure level actually received by the user at the target distance remains consistent with the target hearing reference at the reference distance. This effectively cancels out the call sound field fluctuations caused by changes in the distance between the user and the terminal, achieving stable hearing on the user's side.

[0060] Step S40: Obtain a preset complex control weight, and control the second speaker to send a sound wave signal corresponding to the preset complex control weight. The sound wave signal is used to cancel the sound wave signal of the call audio signal in a preset audio leakage area. The preset complex control weight includes amplitude and phase.

[0061] It should be noted that the preset complex control weights are used to control the complex coefficients of the second speaker output. The second speaker refers to the auxiliary speaker located on the back of the mobile terminal, used for sound field adjustment. Its main function is to cancel or weaken the leakage sound field in the preset audio leakage area by generating sound waves with specific amplitude and phase. The sound wave signal refers to the acoustic signal with specific amplitude and phase generated by the second speaker according to the complex control weights, used to cancel the leakage sound field generated by the front speaker. The preset audio leakage area refers to a pre-defined spatial area behind the device, usually selected as a representative spatial point within a certain range on the back of the device, used to assess and suppress the sound energy of the call audio signal propagating backward. The call sound field control device obtains the complex control weights pre-calculated through sound field optimization. These weights include amplitude and phase components, and apply them to the second speaker, controlling the second speaker to send a sound wave signal corresponding to the weights. After propagating through space, this sound wave signal encounters the leakage sound wave of the call audio signal generated by the first speaker within the preset audio leakage area. Through amplitude matching and phase reversal settings, the two undergo canceling interference. This active sound field control method based on complex weights enables the device to suppress sound energy in the area behind the device without affecting the user's normal listening experience. Compared with passive sound leakage control methods that rely on structural shielding, it achieves a more efficient and flexible privacy protection effect.

[0062] In this embodiment, the compensation gain calculated based on the spatial propagation model is directly applied to the call audio signal of the first speaker, thereby achieving precise physical compensation for sound wave propagation attenuation. This dynamic adjustment method based on real-time target distance ensures that the sound pressure level actually received by the user remains stable near the preset reference hearing level regardless of changes in the distance between the user and the terminal. Compared to traditional automatic gain control based on signal level, this method more fundamentally solves the problem of volume fluctuations caused by distance changes, significantly improving the consistency and comfort of the user's hearing during calls.

[0063] Based on the first embodiment of this application, in the second embodiment of this application, the content that is the same as or similar to that in Embodiment 1 above can be referred to the above description, and will not be repeated hereafter. (Refer to...) Figure 2 , Figure 2 This is a flowchart illustrating Embodiment 2 of the call sound field control method of this application. Based on this, steps A01 to A04 are included before step S40: Step A01: Select a preset number of spatial sampling points in the preset audio leakage area; As can be understood, spatial sampling points refer to representative discrete spatial locations selected within a preset audio leakage area. These points are used to approximately describe the sound field distribution of the entire leakage area, and the overall sound energy level of the leakage area is characterized by sound field calculations at these points. The call sound field control device selects a preset number of spatial sampling points within the preset audio leakage area. These sampling points are distributed discretely in the leakage area behind the device to approximately represent the sound field characteristics of the entire continuous spatial region.

[0064] Step A02: Calculate the sound field vector of the first loudspeaker at the spatial sampling point and the propagation vector of the second loudspeaker at the spatial sampling point; It should be noted that the sound field vector refers to the set of sound pressure values ​​generated by the first speaker at each spatial sampling point within the preset audio leakage area, used to describe the sound field distribution characteristics generated by the first speaker in this area. The propagation vector refers to the set of propagation transfer functions between the second speaker and each spatial sampling point within the preset audio leakage area, used to describe the amplitude attenuation and phase change characteristics of the sound waves emitted by the second speaker as they propagate to each sampling point. The call sound field control device calculates the sound field vector of the first speaker at each spatial sampling point within the preset audio leakage area, and the propagation vector of the second speaker at each spatial sampling point, based on the sound wave propagation model.

[0065] Specifically, the device can approximate the propagation relationship using the Green's function. The Green's function is commonly used to describe the propagation response of a unit point sound source in space, that is, the transfer function from the sound source location to any point in space:

[0066] in, For wave number, Let be the frequency of the sound wave, and r be the straight-line distance from the point sound source to the spatial sampling point. The imaginary unit is used to conveniently represent the amplitude and phase of a sine wave in complex form.

[0067] Step A03: Obtain the initial complex control weights, and construct the total sound field function based on the sound field vector, the propagation vector, and the initial complex control weights; It is understandable that the initial complex control weights refer to the initial variables or control parameters to be solved during the sound field optimization process, and their form is: ,in, Represents the range, The phase is represented by the total sound field function. The total sound field function is a mathematical expression describing the total sound pressure at each spatial sampling point within a preset audio leakage area. It is composed of the superposition of the leakage sound field generated by the first speaker and the control sound field generated by the second speaker, and is used to characterize the overall sound energy distribution within the leakage area. The call sound field control device obtains the initial complex control weights as the control variables to be solved, and constructs the total sound field function based on the calculated sound field vectors of the first speaker at the spatial sampling points, the propagation vectors of the second speaker at the spatial sampling points, and the initial complex control weights.

[0068] Specifically, the total sound field function is:

[0069] The sound field generated by the front speaker is The sound field generated by the rear speakers is .

[0070] Step A04: Iteratively optimize the total sound field function to obtain a preset complex control weight that minimizes the total sound field.

[0071] It should be noted that the call sound field control device uses the pre-constructed total sound field function as the objective function and the initial complex control weights as the starting point, employing an iterative optimization algorithm to solve the problem. In each iteration, the device calculates the total sound field function value based on the current complex control weights and adjusts the amplitude and phase components according to the optimization criteria, gradually reducing the total sound field. After multiple iterations, the device converges to the optimal solution that minimizes the total sound field function, and outputs this optimal solution as the preset complex control weights.

[0072] Specifically, with the objective of minimizing the total acoustic energy in the leakage area, the control weights are solved:

[0073] in, This represents the sound field vector of the front loudspeaker in the leakage region. This represents the propagation vector of the rear speaker. The conjugate transpose is defined by transposing the complex vector Hs and taking the complex conjugate of each element.

[0074] Optionally, when the call sound field control device solves for the complex control weights of the second speaker, it uses the current compensation gain of the first speaker as a constraint condition to constrain the maximum amplitude of the control weights, thereby preventing the output power of the rear speaker from exceeding the power range of the first speaker.

[0075] Understandably, the maximum amplitude refers to the upper limit of the amplitude component of the complex control weights, used to limit the sound pressure level output by the second speaker, preventing it from exceeding the speaker's driving capability or becoming severely unbalanced with the power of the first speaker. When the call sound field control device iteratively optimizes the complex control weights of the second speaker, it incorporates the current compensation gain of the first speaker as a constraint into the optimization process. Specifically, the device determines the output power level of the first speaker at the current moment based on the compensation gain corresponding to the current target distance, and uses this to set the maximum amplitude upper limit of the second speaker's control weights. This ensures that the output power of the rear speaker and the power of the first speaker remain within a reasonable ratio range, preventing the output power of the rear speaker from far exceeding that of the first speaker due to excessive pursuit of leakage suppression.

[0076] Furthermore, during the iterative optimization process, frequency band-related limiting processing is applied to the amplitude component of the complex control weights, wherein the limiting threshold is independently set according to the compensation gain frequency response of the first loudspeaker at each frequency point.

[0077] Specifically, because sound propagation attenuation and the speaker's own frequency response are not consistent across the entire frequency band, the compensation gain may differ in different frequency bands. Therefore, during the iterative optimization process of solving the complex control weights of the second speaker, the device applies frequency band-dependent limiting processing to the amplitude component. Specifically, the device obtains the compensation gain frequency response of the first speaker, i.e., the specific values ​​of the compensation gain at different frequency points, and based on this, sets independent limiting thresholds for each frequency component of the second speaker's control weights. In the low-frequency band, due to the strong diffraction capability of sound waves and the wide distribution of leakage energy, the device sets a higher limiting threshold to ensure leakage suppression. In the high-frequency band, considering the low sensitivity of the human ear to backflow leakage and the limited high-frequency driving capability of the speaker, the device sets a lower limiting threshold to avoid over-driving and distortion. Thus, the device achieves refined power constraint in the frequency domain, ensuring that the output of the second speaker matches the actual operating state of the front speaker across the entire frequency band.

[0078] In this embodiment, iterative optimization is performed with the total sound field function as the objective. The complex control weight that minimizes the total sound energy in the leakage region is precisely solved using mathematical methods. This enables the control sound wave output by the second speaker to achieve optimal matching and destructive interference with the leakage sound wave of the first speaker in terms of amplitude and phase.

[0079] All user-related data involved in this application (e.g., user distance data) were obtained with the user's permission or consent; that is, when this application is applied to specific products or technologies, user permission is required to obtain and process the relevant data, and the processing of the relevant data must comply with the relevant laws, regulations and regulatory standards of the relevant countries and regions.

[0080] For example, when it is necessary to obtain a user's target distance, a prompt to obtain the target distance can be displayed on the user's terminal. After receiving confirmation from the user regarding the prompt, the terminal can obtain the user's target distance. Figure 3 The data retrieval prompt shown is shown in the image.

[0081] It should be noted that the above examples are only for understanding this application and do not constitute a limitation on the call sound field control method of this application. Any simple modifications based on this technical concept are within the protection scope of this application.

[0082] This application also provides a call sound field control device, please refer to... Figure 4 The call sound field control device includes: The receiving module 10 is used to receive an ambient echo signal via a microphone, wherein the ambient echo signal is reflected back by an FM signal emitted by a first speaker, and the instantaneous frequency of the FM signal changes linearly with time. Extraction module 20 is used to extract the beat frequency signal between the environmental echo signal and the frequency modulation signal, and calculate the target distance between the first speaker and the target of the call based on the beat frequency signal; The adjustment module 30 is used to adjust the call audio signal of the first speaker based on the target distance to counteract the call sound field fluctuation caused by the distance change.

[0083] Optionally, the extraction module includes: The alignment submodule is used to time-align the ambient echo signal and the frequency modulation signal; to perform frequency mixing processing on the aligned ambient echo signal and the aligned frequency modulation signal to generate a beat frequency signal characterizing time delay information; to calculate the channel impulse response based on the beat frequency signal; and to calculate the target distance between the first speaker and the target being communicated based on the channel impulse response.

[0084] Optionally, the alignment submodule includes: The mapping unit is used to convert the time-domain beat frequency signal into a frequency-domain signal, wherein different propagation paths in the frequency-domain signal correspond to different frequency components; to perform coordinate mapping on the frequency-domain signal to obtain a channel impulse response; and to select the distance corresponding to a preset signal amplitude point from the channel impulse response as the target distance between the first speaker and the target for communication.

[0085] Optionally, the adjustment module includes: The adjustment submodule is used to calculate the compensation gain based on the target distance and the preset reference distance; and to adjust the call audio signal of the first speaker based on the compensation gain to offset the call sound field fluctuation caused by the distance change.

[0086] The control submodule is used to obtain a preset complex control weight and control the second speaker to send a sound wave signal corresponding to the preset complex control weight. The sound wave signal is used to cancel the sound wave signal of the call audio signal in a preset audio leakage area. The preset complex control weight includes amplitude and phase.

[0087] Optionally, the control submodule includes: An optimization unit is configured to select a preset number of spatial sampling points in the preset audio leakage area; calculate the sound field vector of the first speaker at the spatial sampling points and the propagation vector of the second speaker at the spatial sampling points; obtain initial complex control weights; construct a total sound field function based on the sound field vector, the propagation vector and the initial complex control weights; and iteratively optimize the total sound field function to obtain a preset complex control weight that minimizes the total sound field.

[0088] The call sound field control device provided in this application, employing the call sound field control method in the above embodiments, can solve the technical problem of call sound field control. Compared with the prior art, the beneficial effects of the call sound field control device provided in this application are the same as those of the call sound field control method provided in the above embodiments, and other technical features in the call sound field control device are the same as those disclosed in the methods of the above embodiments, and will not be repeated here.

[0089] This application provides a call sound field control device, which includes: at least one processor; and a memory communicatively connected to the at least one processor; wherein the memory stores instructions executable by the at least one processor, and the instructions are executed by the at least one processor to enable the at least one processor to perform the call sound field control method in the first embodiment described above.

[0090] The following is for reference. Figure 5 The diagram illustrates a structural schematic of a call sound field control device suitable for implementing embodiments of this application. The call sound field control device in the embodiments of this application may include, but is not limited to, mobile terminals such as mobile phones, laptops, tablets, digital broadcast receivers, PDAs (Personal Digital Assistants), PMPs (Portable Media Players), in-vehicle terminals (e.g., in-vehicle navigation terminals), and fixed terminals such as digital TVs and desktop computers. Figure 5 The illustrated call sound field control device is merely an example and should not impose any limitations on the functionality and scope of use of the embodiments of this application.

[0091] like Figure 5As shown, the call sound field control device may include a processing unit 1001 (e.g., a central processing unit, a graphics processing unit, etc.), which can perform various appropriate actions and processes according to a program stored in a read-only memory (ROM) 1002 or a program loaded from a storage device 1003 into a random access memory (RAM) 1004. The RAM 1004 also stores various programs and data required for the operation of the call sound field control device. The processing unit 1001, ROM 1002, and RAM 1004 are interconnected via a bus 1005. An input / output (I / O) interface 1006 is also connected to the bus. Typically, the following systems can be connected to the I / O interface 1006: input devices 1007 including, for example, a touchscreen, touchpad, keyboard, mouse, image sensor, microphone, accelerometer, gyroscope, etc.; output devices 1008 including, for example, a liquid crystal display (LCD), speaker, vibrator, etc.; storage devices 1003 including, for example, magnetic tape, hard disk, etc.; and communication devices 1009. The communication device 1009 allows the call sound field control device to communicate wirelessly or wiredly with other devices to exchange data. Although call sound field control devices with various systems are shown in the figures, it should be understood that it is not required to implement or possess all of the systems shown. More or fewer systems may be implemented alternatively.

[0092] Specifically, according to the embodiments disclosed in this application, the processes described above with reference to the flowcharts can be implemented as computer software programs. For example, embodiments disclosed in this application include a computer program product comprising a computer program carried on a computer-readable medium, the computer program containing program code for performing the methods shown in the flowcharts. In such embodiments, the computer program can be downloaded and installed from a network via a communication device, or installed from storage device 1003, or installed from ROM 1002. When the computer program is executed by processing device 1001, it performs the functions defined in the methods of the embodiments disclosed in this application.

[0093] The call sound field control device provided in this application, employing the call sound field control method in the above embodiments, can solve the technical problem of call sound field control. Compared with the prior art, the beneficial effects of the call sound field control device provided in this application are the same as those of the call sound field control method provided in the above embodiments, and other technical features in this call sound field control device are the same as those disclosed in the method of the previous embodiment, and will not be repeated here.

[0094] It should be understood that the various parts disclosed in this application can be implemented using hardware, software, firmware, or a combination thereof. In the description of the above embodiments, specific features, structures, materials, or characteristics can be combined in any suitable manner in one or more embodiments or examples.

[0095] The above description is merely a specific embodiment of this application, but the scope of protection of this application is not limited thereto. Any variations or substitutions that can be easily conceived by those skilled in the art within the scope of the technology disclosed in this application should be included within the scope of protection of this application. Therefore, the scope of protection of this application should be determined by the scope of the claims.

[0096] This application provides a computer-readable storage medium having computer-readable program instructions (i.e., a computer program) stored thereon, the computer-readable program instructions being used to execute the call sound field control method in the above embodiments.

[0097] The computer-readable storage medium provided in this application may be, for example, a USB flash drive, but is not limited to, electrical, magnetic, optical, electromagnetic, infrared, or semiconductor systems, devices, or any combination thereof. More specific examples of computer-readable storage media may include, but are not limited to: electrical connections having one or more wires, portable computer disks, hard disks, random access memory (RAM), read-only memory (ROM), erasable programmable read-only memory (EPROM or flash memory), optical fiber, portable compact disk read-only memory (CD-ROM), optical storage devices, magnetic storage devices, or any suitable combination thereof. In this embodiment, the computer-readable storage medium may be any tangible medium containing or storing a program that can be used by or in conjunction with an instruction execution system, system, or device. The program code contained on the computer-readable storage medium may be transmitted using any suitable medium, including but not limited to: wires, optical cables, RF (Radio Frequency), etc., or any suitable combination thereof.

[0098] The aforementioned computer-readable storage medium may be included in the call sound field control device; or it may exist independently and not assembled into the call sound field control device.

[0099] The aforementioned computer-readable storage medium carries one or more programs that, when executed by the call sound field control device, cause the call sound field control device to: receive an ambient echo signal via a microphone, wherein the ambient echo signal is reflected back from an FM signal emitted by a first speaker, and the instantaneous frequency of the FM signal changes linearly with time; extract a beat frequency signal between the ambient echo signal and the FM signal; calculate a target distance between the first speaker and the call target based on the beat frequency signal; and adjust the call audio signal of the first speaker based on the target distance to counteract call sound field fluctuations caused by distance changes.

[0100] Computer program code for performing the operations of this application can be written in one or more programming languages ​​or a combination thereof, including object-oriented programming languages ​​such as Java, Smalltalk, and C++, and conventional procedural programming languages ​​such as the "C" language or similar programming languages. The program code can be executed entirely on the user's computer, partially on the user's computer, as a standalone software package, partially on the user's computer and partially on a remote computer, or entirely on a remote computer or server. In cases involving remote computers, the remote computer can be connected to the user's computer via any type of network—including a Local Area Network (LAN) or a Wide Area Network (WAN)—or can be connected to an external computer (e.g., via the Internet using an Internet service provider).

[0101] The flowcharts and block diagrams in the accompanying drawings illustrate the architecture, functionality, and operation of possible implementations of systems, methods, and computer program products according to various embodiments of this application. In this regard, each block in a flowchart or block diagram may represent a module, segment, or portion of code containing one or more executable instructions for implementing a specified logical function. It should also be noted that in some alternative implementations, the functions indicated in the blocks may occur in a different order than those indicated in the drawings. For example, two consecutively indicated blocks may actually be executed substantially in parallel, and they may sometimes be executed in reverse order, depending on the functions involved. It should also be noted that each block in the block diagrams and / or flowcharts, and combinations of blocks in the block diagrams and / or flowcharts, can be implemented using a dedicated hardware-based system that performs the specified function or operation, or using a combination of dedicated hardware and computer instructions.

[0102] The modules described in the embodiments of this application can be implemented in software or hardware. The names of the modules do not necessarily limit the functionality of the unit itself.

[0103] The readable storage medium provided in this application is a computer-readable storage medium that stores computer-readable program instructions (i.e., a computer program) for executing the above-described call sound field control method, thereby solving the technical problem of call sound field control. Compared with the prior art, the beneficial effects of the computer-readable storage medium provided in this application are the same as the beneficial effects of the call sound field control method provided in the above embodiments, and will not be repeated here.

[0104] This application also provides a computer program product, including a computer program that, when executed by a processor, implements the steps of the call sound field control method described above.

[0105] The computer program product provided in this application can solve the technical problem of call sound field control. Compared with the prior art, the beneficial effects of the computer program product provided in this application are the same as the beneficial effects of the call sound field control method provided in the above embodiments, and will not be repeated here.

[0106] The above description is only a part of the embodiments of this application and does not limit the scope of protection of this application. All equivalent structural transformations made under the technical concept of this application and using the content of this application specification and drawings, or direct / indirect applications in other related technical fields, are included in the scope of protection of this application.

Claims

1. A method for controlling the sound field during a call, characterized in that, The call sound field control method includes: The system receives ambient echo signals via a microphone, wherein the ambient echo signals are reflected back from an FM signal emitted by a first loudspeaker, and the instantaneous frequency of the FM signal changes linearly with time. Extract the beat frequency signal between the environmental echo signal and the frequency modulation signal, and calculate the target distance between the first speaker and the target being communicated based on the beat frequency signal; Based on the target distance, the call audio signal of the first speaker is adjusted to counteract the call sound field fluctuations caused by the distance change; Select a preset number of spatial sampling points within the preset audio leakage area; Calculate the sound field vector of the first loudspeaker at the spatial sampling point, and the propagation vector of the second loudspeaker at the spatial sampling point; Obtain the initial complex control weights, and construct the total sound field function based on the sound field vector, the propagation vector, and the initial complex control weights; The total sound field function is iteratively optimized to obtain a preset complex control weight that minimizes the total sound field. The magnitude of the preset complex control weight satisfies a preset constraint condition, which is determined based on the current compensation gain of the first loudspeaker. Obtain a preset complex control weight, and control the second speaker to send a sound wave signal corresponding to the preset complex control weight, wherein the sound wave signal is used to cancel the sound wave signal of the call audio signal in a preset audio leakage area, and the preset complex control weight includes amplitude and phase; The step of iteratively optimizing the total sound field function includes: Obtain the target values ​​of the compensation gain at different frequency points; Based on the target value, a corresponding limiting threshold is set for each frequency component in the preset complex control weight, and frequency band-related limiting processing is applied based on the limiting threshold.

2. The call sound field control method as described in claim 1, characterized in that, The step of extracting the beat frequency signal between the environmental echo signal and the frequency modulation signal, and calculating the target distance between the first speaker and the target for communication based on the beat frequency signal, includes: Time-align the ambient echo signal and the frequency modulation signal; The aligned environmental echo signal and the aligned frequency-modulated signal are mixed to generate a beat frequency signal that represents time delay information. Based on the beat frequency signal, the channel impulse response is calculated, and based on the channel impulse response, the target distance between the first speaker and the target being communicated with is calculated.

3. The call sound field control method as described in claim 2, characterized in that, The steps of calculating the channel impulse response based on the beat frequency signal, and calculating the target distance between the first speaker and the target being communicated based on the channel impulse response, include: The beat frequency signal of the time domain type is converted into a frequency domain signal, wherein different propagation paths in the frequency domain signal correspond to different frequency components; The frequency domain signal is mapped to obtain the channel impulse response. The distance corresponding to the preset signal amplitude point is selected from the channel impulse response as the target distance between the first speaker and the target being communicated.

4. The call sound field control method as described in claim 1, characterized in that, The step of adjusting the call audio signal of the first speaker based on the target distance to counteract the call sound field fluctuations caused by distance changes includes: Calculate the compensation gain based on the target distance and the preset reference distance; Based on the compensation gain, the call audio signal of the first speaker is adjusted to counteract the call sound field fluctuations caused by distance changes.

5. A communication sound field control device, characterized in that, The device includes: A receiving module is used to receive an ambient echo signal via a microphone, wherein the ambient echo signal is reflected back from an FM signal emitted by a first loudspeaker, and the instantaneous frequency of the FM signal changes linearly with time. The extraction module is used to extract the beat frequency signal between the environmental echo signal and the frequency modulation signal, and to calculate the target distance between the first speaker and the target being communicated based on the beat frequency signal; An adjustment module is used to adjust the call audio signal of the first speaker based on the target distance to counteract the call sound field fluctuations caused by distance changes; A control submodule is used to obtain a preset complex control weight and control the second speaker to send a sound wave signal corresponding to the preset complex control weight. The sound wave signal is used to cancel the sound wave signal of the call audio signal in a preset audio leakage area. The preset complex control weight includes amplitude and phase. The control submodule includes: An optimization unit is configured to select a preset number of spatial sampling points in the preset audio leakage area; calculate the sound field vector of the first speaker at the spatial sampling points and the propagation vector of the second speaker at the spatial sampling points; obtain initial complex control weights; construct a total sound field function based on the sound field vector, the propagation vector, and the initial complex control weights; iteratively optimize the total sound field function to obtain a preset complex control weight that minimizes the total sound field, wherein the magnitude of the preset complex control weights satisfies a preset constraint condition, and the preset constraint condition is determined based on the current compensation gain of the first speaker; The optimization unit is also used to obtain the target value of the compensation gain at different frequency points; based on the target value, set the corresponding limiting threshold for each frequency component in the preset complex control weight, and apply band-related limiting processing based on the limiting threshold.

6. A communication sound field control device, characterized in that, The device includes: a memory, a processor, and a computer program stored in the memory and executable on the processor, the computer program being configured to implement the steps of the call sound field control method as described in any one of claims 1 to 4.

7. A storage medium, characterized in that, The storage medium is a computer-readable storage medium, and a computer program is stored on the storage medium. When the computer program is executed by a processor, it implements the steps of the call sound field control method as described in any one of claims 1 to 4.

8. A computer program product, characterized in that, The computer program product includes a computer program that, when executed by a processor, implements the steps of the call sound field control method as described in any one of claims 1 to 4.

Images