Call control method and apparatus, communication device, and computer-readable storage medium

Abstract

The application provides a call control method and device, a communication device and a computer readable storage medium. The method comprises the following steps: acquiring a far-end incoming call signal and an audio acquisition signal collected by a microphone array; acquiring a user audio direction, and extracting a user audio signal and a noise signal in the audio acquisition signal according to the user audio direction; sending the user audio signal as an uplink signal to the far end; and determining an audio playing parameter corresponding to the far-end incoming call signal according to the noise signal. The microphone array is arranged to collect audio, so that the audio acquisition signal can combine audio in multiple directions, and then the data of user speech is determined according to the user audio direction, the user audio signal is extracted, the intelligibility of the user audio signal sent to the far end in communication is improved, meanwhile, the audio playing parameter is determined based on the noise signal, so that the intelligibility of the far-end incoming call signal heard by the user can be improved, the intelligibility of the call process is realized, and the influence of environmental noise on the call is reduced.

Description

Technical Field

[0001] This invention relates to the field of communications, and more particularly to a call control method, apparatus, communication equipment, and computer-readable storage medium. Background Technology

[0002] With the widespread use of mobile communication devices, users are making calls more frequently in complex acoustic environments, and their requirements for call quality are also increasing. However, in complex acoustic environments such as subways, streets, and shopping malls, noise interference in the environment can have a serious impact on call quality. Summary of the Invention

[0003] The main objective of this invention is to provide several embodiments of a call control method, apparatus, communication device, and computer-readable storage medium, wherein at least one embodiment can solve the problem that noise interference in the environment can seriously affect call quality in the prior art.

[0004] To achieve the above objectives, the present invention provides a call control method, the method comprising the following steps: Acquire the incoming call signal from the remote end and the audio signal collected by the microphone array; Obtain the user's audio direction, and extract the user's audio signal and noise signal from the audio acquisition signal based on the user's audio direction; The user's audio signal is sent as an uplink signal to the remote end; The audio playback parameters corresponding to the remote incoming call signal are determined based on the noise signal.

[0005] Optionally, the process of obtaining the user's audio direction includes: In the previous control cycle, the audio direction of the previous control cycle is estimated based on the audio acquisition signal of the previous cycle. Calculate the angular velocity corresponding to the audio direction of the previous cycle; Based on the audio direction of the previous cycle, the direction of the next control cycle is predicted according to the directional angular velocity to obtain the user audio direction.

[0006] Optionally, the step of extracting the user audio signal and noise signal from the audio acquisition signal based on the user audio direction includes: Determine the interference direction corresponding to the user audio direction, wherein the interference direction is the direction of the interference sound source in the audio acquisition signal; The dynamic beamforming weights corresponding to the audio acquisition signal are determined based on the user audio direction and the interference direction. The dynamic beamforming weights indicate the enhancement of the audio acquisition data in the user audio direction and the suppression in the interference direction. The user audio signal is obtained by extracting the audio acquisition signal using the dynamic beamforming weights.

[0007] Optionally, determining the dynamic beamforming weights corresponding to the audio acquisition signal based on the user audio direction and the interference direction, wherein the dynamic beamforming weights indicate the enhancement of the audio acquisition data in the user audio direction and the suppression of the interference direction, includes: A maximum gain constraint is constructed for the user's audio direction; A zero-trap constraint is constructed for the direction of the disturbance; A low-gain constraint is constructed for the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction, and the gain corresponding to the low-gain constraint is a preset background gain; The dynamic beamforming weights are generated by combining the maximum gain constraint, the null constraint, and the low gain constraint.

[0008] Optionally, constructing a null constraint on the disturbance direction includes: Obtain the direction of disturbance in multiple consecutive control cycles; Calculate the standard deviation of the fluctuation in the direction of the interference; Determine the width of the zero-trap angle corresponding to the standard deviation of the fluctuation, wherein the standard deviation of the fluctuation is positively correlated with the width of the zero-trap angle; Based on the zero-trap width, construct the zero-trap constraint corresponding to the interference direction.

[0009] Optionally, the step of extracting the user audio signal and noise signal from the audio acquisition signal based on the user audio direction includes: Determine the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction; Obtain noise data corresponding to the noise direction from the audio acquisition signal; Calculate the noise intensity corresponding to the noise data; The noise intensity is taken as the noise signal.

[0010] Optionally, determining the audio playback parameters corresponding to the distant incoming call signal based on the noise signal includes: Determine the noise intensity corresponding to the noise signal; Determine whether the noise intensity is greater than the noise threshold; If the noise intensity is greater than the noise threshold, the call volume corresponding to the audio playback parameter is adjusted, wherein the call volume is positively correlated with the noise intensity.

[0011] To achieve the above objectives, the present invention also provides a call control device, the call control device comprising: The first acquisition module is used to acquire the incoming call signal from the remote end and the audio acquisition signal collected by the microphone array; The second acquisition module is used to acquire the user's audio direction and extract the user's audio signal and noise signal from the audio acquisition signal based on the user's audio direction. The first transmitting module is used to transmit the user audio signal as an uplink signal to the remote end; The first determining module is used to determine the audio playback parameters corresponding to the remote incoming call signal based on the noise signal.

[0012] To achieve the above objectives, the present invention also provides a communication device, the communication device including a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the computer program, when executed by the processor, implements the steps of the call control method as described above.

[0013] To achieve the above objectives, the present invention also provides a computer-readable storage medium storing a computer program, which, when executed by a processor, implements the steps of the call control method described above.

[0014] This invention proposes a call control method, apparatus, communication device, and computer-readable storage medium. The method involves acquiring a remote incoming call signal and an audio acquisition signal collected by a microphone array; acquiring the user's audio direction; and extracting the user's audio signal and noise signal from the audio acquisition signal based on the user's audio direction; sending the user's audio signal as an uplink signal to the remote end; and determining the audio playback parameters corresponding to the remote incoming call signal based on the noise signal. By setting the microphone array to acquire audio, the audio acquisition signal can combine audio from multiple directions. Furthermore, by determining the user's speaking data based on the user's audio direction, the user's audio signal can be extracted, thereby improving the clarity of the user's audio signal sent to the remote end during communication. Simultaneously, determining the audio playback parameters based on the noise signal improves the clarity of the user hearing the remote incoming call signal, achieving clear call communication and reducing the impact of environmental noise on the call. Attached Figure Description

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

[0016] To more clearly illustrate the technical solutions in the embodiments of the present invention 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 the first embodiment of the call control method of the present invention; Figure 2 This is a detailed flowchart of the call control method of the present invention; Figure 3 This is a schematic diagram of the module structure of the communication device of the present invention. Detailed Implementation

[0018] It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention. To enable those skilled in the art to better understand the present application, the technical solutions in the embodiments of the present application will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only a part of the embodiments of the present application, and not all of them. Based on the embodiments in this application, all other embodiments obtained by those skilled in the art without creative effort should fall within the scope of protection of this application.

[0019] This invention provides a call control method, referring to... Figure 1 , Figure 1 This is a flowchart illustrating the first embodiment of the call control method of the present invention, the method comprising the following steps: Step S10: Acquire the incoming call signal from the remote end and the audio acquisition signal collected by the microphone array; The remote incoming call signal is the signal sent by the communicating party during a call.

[0020] If the current communication device is communication device A, and the remote device is communication device B; when communication device A and communication device B are having a conversation, for communication device A, the voice signal received from communication device B during the conversation is the remote incoming call signal or downlink signal, and the voice signal sent to communication device B is the uplink signal.

[0021] A microphone array is an array of multiple microphones set up on a communication device; the number of microphones and the array structure can be set according to actual needs.

[0022] The audio acquisition signal is the collection of user audio signals acquired by each microphone in the microphone array; the specific acquisition frequency can be set according to actual needs, such as acquiring once every 10ms.

[0023] Step S20: Obtain the user's audio direction, and extract the user's audio signal and noise signal from the audio acquisition signal based on the user's audio direction; The user audio direction is the direction of the user's voice source relative to the communication device.

[0024] The direction can be characterized based on the angle of arrival; the direction of subsequent interference and noise can be represented in the same way; the calculation method of the angle of arrival can be set according to actual needs.

[0025] The user audio signal is the signal of the corresponding user's voice source.

[0026] The noise signal is the signal corresponding to noise in the environment.

[0027] It is understandable that at any given moment, when a user speaks, the relative relationship between the user and the microphone array is fixed. Therefore, the user audio signal corresponding to the user's speaking sound source can be determined based on the user's audio direction, and the noise components outside the user audio signal can also be determined based on the audio components in directions other than the user's audio direction. Therefore, based on the user's audio direction, it is possible to extract the user audio signal and noise signal from the audio acquisition signal.

[0028] Step S30: Send the user audio signal as an uplink signal to the remote end; After obtaining the user's audio signal, the user's audio signal is sent as an uplink signal to the remote device.

[0029] It is understandable that the user audio signal is extracted specifically based on the user's audio direction. Therefore, the user audio signal filters out a large amount of noise from non-user audio directions, which means it has higher clarity than the audio acquisition signal. Using the user audio signal as an uplink signal can make the audio heard by the user of the remote communication device clearer.

[0030] Step S40: Determine the audio playback parameters corresponding to the remote incoming call signal based on the noise signal.

[0031] Audio playback parameters are parameters related to playing incoming call signals from a distant location during a call; the specific type of audio playback parameters can be set according to actual needs, such as call volume.

[0032] It is understandable that noise signals indicate the noise level of the environment, and environmental noise can interfere with a user's ability to hear incoming calls from a distance. Therefore, in this embodiment, audio playback parameters are adjusted based on noise signals to adaptively play incoming calls from a distance based on the actual environmental noise levels, thereby reducing the interference of noise on the incoming calls from a distance.

[0033] This embodiment uses a microphone array to collect audio, enabling the audio signal to be combined with audio from multiple directions. By determining the direction of the user's audio, the data of the user's speech is obtained, thereby extracting the user's audio signal and improving the clarity of the user's audio signal sent to the remote end during communication. At the same time, the audio playback parameters are determined based on the noise signal, which improves the clarity of the user's hearing of the incoming signal from the remote end, achieving a clear call process and reducing the impact of environmental noise on the call.

[0034] Further details will follow. Figure 2 In the second embodiment of the call control method of the present invention based on the first embodiment of the present invention, the step before step S20 includes: Step S50: In the previous control cycle, the audio direction of the previous control cycle is estimated based on the audio acquisition signal of the previous cycle. Step S60: Calculate the directional angular velocity corresponding to the audio direction of the previous cycle; Step S70: Based on the audio direction of the previous cycle, predict the direction of the next control cycle based on the directional angular velocity to obtain the user audio direction.

[0035] The control cycle is the complete cycle in which this method completes one round of audio acquisition, uplink signal transmission, and playback of incoming call signals from the remote end.

[0036] The specific length of the control cycle can be set based on actual needs, such as 10ms.

[0037] The previous control cycle is the control cycle preceding the current control cycle.

[0038] The user audio direction is the user audio direction in the current control cycle.

[0039] The audio acquisition signal of the previous cycle is the audio acquisition signal obtained in the previous control cycle.

[0040] The audio direction of the previous cycle is the direction of the user's voice source estimated based on the audio acquisition signal of the previous cycle. The specific estimation method can be set according to actual needs. For example, multi-source direction estimation is performed on the audio acquisition signal to obtain the set of arrival angles of all significant sound sources. The direction of the sound source with the closest predicted trajectory and the highest energy corresponding to the directional angular velocity in the set of arrival angles is selected as the audio direction of the previous cycle.

[0041] The directional angular velocity is the rate of change of the sound source angle between two adjacent control cycles; therefore:

[0042] Where v(t) is the directional angular velocity; θ(t) is the audio direction estimated in the t-th cycle, i.e., the user's audio direction; θ(t-1) is the audio direction estimated in the (t-1)-th cycle; and Δt is the length of the control cycle.

[0043] It should be noted that due to the randomness of user movement, the determination of directional angular velocity may become jittery. To avoid this problem, the directional angular velocity can be limited and smoothed to suppress jitter, such as by setting |v(t)|≤v max ;v max It indicates the maximum permissible directional angular velocity; the specific value can be set based on actual needs.

[0044] The directional angular velocity indicates how fast the user's voice source moves relative to the microphone array; the audio direction of the previous cycle is the relative position between the voice source and the microphone array in the previous cycle. Therefore, the directional angular velocity can be used to determine the user's audio direction for the next cycle, i.e., the current control cycle.

[0045] Where, θ next For the user's audio direction; θ current β represents the audio direction of the previous cycle; β is the trajectory prediction look-ahead coefficient, used to adjust the dependence on directional angular velocity and control the aggressiveness-conservatism of the prediction step size; the larger β is, the more look-ahead and sensitive the system is in tracking target motion; the smaller β is, the more conservative the prediction and the more resistant to jitter. The specific value of β can be set according to actual needs, such as β∈[0.6, 1.0]; this value range can allow the system to achieve a balance between fast-moving tracking and prediction stability.

[0046] Understandably, in practical applications, the relative relationship between the location of the speaking sound source and the microphone array changes constantly with the user's actions. Therefore, if the user's audio direction is determined based on the audio acquisition data collected in the current control cycle, the determination of the user's audio direction will lag behind the accuracy due to the execution delay and the one-sidedness of single-frame data detection.

[0047] In this embodiment, the user's audio direction is obtained by using the angular velocity of the previous control cycle and the audio direction of the previous cycle. First, the audio direction of the previous cycle has a time of one cycle to complete the estimation. At the same time, the angular velocity reflects the user's motion path to a certain extent. Therefore, by combining the two, the possible user audio direction of the current control cycle can be determined in advance, and accurate prediction can be made based on the user's motion inertia, thereby improving the accuracy of the user's audio direction while avoiding lag.

[0048] Furthermore, in the third embodiment of the call control method of the present invention based on the first embodiment of the present invention, step S20 includes the following steps: Step S21: Determine the interference direction corresponding to the user audio direction, wherein the interference direction is the direction of the interference sound source in the audio acquisition signal; Step S22: Determine the dynamic beamforming weight corresponding to the audio acquisition signal based on the user audio direction and the interference direction. The dynamic beamforming weight indicates the enhancement of the audio acquisition data in the user audio direction and the suppression in the interference direction. Step S23: Extract the user audio signal from the audio acquisition signal using the dynamic beamforming weights.

[0049] Interference sources are the sources of interfering sounds in the environment, such as pedestrians talking or vehicles honking their horns.

[0050] The method for determining the direction of interference can be set based on actual needs. For example, multi-source direction estimation can be performed on the audio acquisition signal to obtain the set of arrival angles of all significant sound sources. After removing the user's audio direction, all other directions are the interference directions.

[0051] The dynamic beamforming weight is a dynamic beamforming weight vector under the spatial attention mechanism, which assigns differentiated gains to different spatial domains.

[0052] In order to obtain a user audio signal with reduced noise, this embodiment sets dynamic beamforming weights to differentiate the gain settings for audio in different directions, thereby enhancing the user audio direction component in the audio acquisition signal and suppressing the interference direction component, thus achieving the reduction or removal of interference audio.

[0053] After the dynamic beamforming weights are determined, the audio acquisition signal is extracted using these weights. This allows for the extraction of the user audio signal enhanced by the dynamic beamforming weights with high gain, while interference audio components are not extracted, thus achieving interference removal in the extracted user audio signal. Specifically:

[0054] Among them, Y up W(t) represents the user's audio signal; W(t) represents the dynamic beamforming weight; H indicates the conjugate transpose; and X(t) represents the audio acquisition signal.

[0055] Further, step S22 includes the following steps: Step S221: Construct a maximum gain constraint for the user audio direction; Step S222: Construct zero-trap constraints for the interference direction; Step S223: Construct a low-gain constraint for the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction, and the gain corresponding to the low-gain constraint is a preset background gain; Step S224: Combine the maximum gain constraint, the null constraint, and the low gain constraint to generate the dynamic beamforming weights.

[0056] The maximum gain constraint is used to set no attenuation constraint on the audio signal in the user's audio direction, thereby ensuring the complete preservation of the user's voice; specifically, the normalized gain corresponding to the user's audio direction can be set to 1.

[0057] Null constraints are used to construct zero limits for the direction of interference, thereby eliminating the audio in the direction of interference; specifically, the gain corresponding to the direction of interference can be set to 0 or a very small gain close to 0.

[0058] When setting zero-limit constraints in practice, zero traps can be constructed in several directions with the strongest energy in the interference direction; the specific number can be set according to actual needs, such as 1 to 2; at this time, the direction in which zero traps are constructed is the interference direction, and the direction in which zero traps are not constructed is the noise direction.

[0059] If no sound source is detected in the current control cycle, zero trap constraints can be omitted.

[0060] The low-gain constraint is a constraint that preserves low gain for audio in the direction of noise.

[0061] The preset background gain is a small pre-set gain threshold value, and the specific value can be set according to actual needs, such as 0.1~0.3.

[0062] It is understandable that setting zero-confinement constraints in all directions other than the user's audio direction would result in a dead sound in the audio. Therefore, in this embodiment, the direction of interference with the strongest energy is subject to zero-confinement constraints to eliminate more obvious noise interference. At the same time, a small value is retained for weaker background noise to provide an ambient noise floor, thereby avoiding a dead sound in the audio.

[0063] In this embodiment, by setting corresponding constraints for the user's audio direction, interference direction, and noise direction respectively, it is possible to achieve stable enhancement of human voice, suppression of interference, and appropriate preservation of ambient background sound; thereby achieving effective noise reduction while eliminating the feeling of dead silence in the call and improving the user experience.

[0064] Further, step S222 includes the following steps: Step S2221: Calculate the standard deviation of the fluctuation in the direction of the interference; Step S2222: Determine the width of the zero-trap angle corresponding to the standard deviation of the fluctuation, wherein the standard deviation of the fluctuation is positively correlated with the width of the zero-trap angle; Step S2223: Construct the zero-dimple constraint corresponding to the interference direction based on the zero-dimple angle width.

[0065] In practical applications, the direction of interference fluctuates as the user moves and the interference source moves. Using a fixed zero-concave angle width cannot adapt to scenarios where the interference source is moving back and forth, which can easily cause noise to leak from the edge of the zero-concave, resulting in a significant decrease in noise reduction effect.

[0066] To address this issue, this embodiment incorporates a dynamic zero-contour width adjustment.

[0067] The zero-trap width is the angular range within the zero-trap constraint that is used to suppress interference signals; that is, the zero-trap constraint is used to constrain signals within the zero-trap width range.

[0068] The standard deviation of fluctuation is the statistical standard deviation of the direction of disturbance; the standard deviation of fluctuation can be calculated by combining the disturbance directions collected from multiple control cycles; the specific number of control cycles involved can be set based on actual needs.

[0069] The standard deviation of fluctuation reflects the degree of change in the direction of fluctuation; the larger the standard deviation, the more drastic the change in the direction of fluctuation, and therefore, the greater the possibility of noise leakage. Therefore, in this embodiment, the standard deviation of fluctuation is set to be positively correlated with the zero-depression angle width. Thus, when the standard deviation of fluctuation is larger, a larger zero-depression angle width is set to suppress interference audio over a wider range; conversely, when the standard deviation of fluctuation is smaller, the change in the direction of fluctuation is more gradual, and therefore, the possibility of noise leakage is smaller. A smaller zero-depression angle width is set to suppress interference audio over a smaller range, achieving more precise noise reduction.

[0070] Where, Δθ null,i Δθ0 is the default value for the zero concave angle width; the specific value can be set according to actual needs, such as 8°~12°; k σ This is the broadening factor; the specific value can be set based on actual needs. σ θ，i The standard deviation of the fluctuation.

[0071] This embodiment quantifies the jitter in the direction of interference by measuring the standard deviation of the disturbance direction. It dynamically adjusts the zero-dimple angle width through a positive correlation, which can effectively avoid noise leakage and ensure stable noise reduction in scenarios with large disturbances. When the disturbance is stationary, it can maintain a narrow zero-dimple angle width to retain appropriate ambient sound, thus meeting the noise reduction needs in different scenarios.

[0072] Furthermore, in the fourth embodiment of the call control method of the present invention based on the first embodiment of the present invention, step S20 includes the following steps: Step S24, determine the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction; Step S25: Obtain noise data corresponding to the noise direction from the audio acquisition signal; Step S26: Calculate the noise intensity corresponding to the noise data; Step S27: The noise intensity is taken as the noise signal.

[0073] Noise intensity indicates the magnitude of noise in the audio acquisition signal; the specific value of noise intensity can be set based on actual needs, such as by averaging the noise energy in multiple noise directions, for example:

[0074] Where N_env(t) is the noise intensity determined in the t-th period; T indicates the index of the noise direction; Y noise (t, n) represents the noise energy in the nth noise direction.

[0075] In this embodiment, by determining the noise intensity of the noise data in the noise direction, the noise assessment deviation caused by user voices and loud noise interference in the environment is avoided, thereby obtaining an accurate noise signal.

[0076] Furthermore, in the fifth embodiment of the call control method of the present invention based on the first embodiment of the present invention, step S40 includes the following steps: Step S41: Determine the noise intensity corresponding to the noise signal; Step S42: Determine whether the noise intensity is greater than the noise threshold; Step S43: If the noise intensity is greater than the noise threshold, adjust the call volume corresponding to the audio playback parameter, wherein the call volume is positively correlated with the noise intensity.

[0077] The noise threshold is the threshold used to distinguish between quiet and noisy environments; the specific value of the noise threshold can be set based on actual needs.

[0078] If the noise intensity is less than or equal to the noise threshold, it means that the noise in the current environment is low and will not affect the user's ability to hear the incoming call signal from a distance. Therefore, there is no need to increase the call volume; the basic volume should be maintained. The specific value of the basic volume can be set according to actual needs.

[0079] If the noise intensity exceeds the noise threshold, it indicates that the current ambient noise is high, which will affect the user's ability to hear the incoming call signal from a distance. Therefore, increasing the call volume will allow the user to hear the incoming call signal from a distance more clearly; for example:

[0080] Among them, V tar (t) represents the call volume used in the t-th cycle; V base Basic volume; N th α is the noise threshold; α is the compensation coefficient, and the specific value can be set according to actual needs.

[0081] Furthermore, sudden changes in ambient noise can cause abrupt changes in call volume, affecting the user's call. To avoid this problem, a limiting and smooth adjustment method is used when adjusting the call volume, such as:

[0082] V(t) represents the smoothed call volume; γ is the weight, used to indicate the tendency for stability and variation in volume adjustment, and its specific value can be set according to actual needs; clip() indicates that the volume is limited to the device's allowed range, i.e., limited to the minimum volume V. min With maximum volume V max between.

[0083] In this embodiment, the noise intensity obtained from the dynamic beamforming weight calculation is reused, eliminating the need for additional independent noise detection calculations and reducing the device's computing power consumption. By using threshold determination to distinguish between quiet and noisy environments, the volume can be adjusted adaptively to adapt to changes in scene noise.

[0084] The overall implementation process of this application is described below: 1. Frame-level triggering and input establishment: The system clock triggers a control cycle every 10 ms.

[0085] Two data streams are accessed in parallel within the same control cycle: Uplink input: The signal acquired by the local microphone array, i.e., the audio signal; Downlink input: Remote incoming call signal.

[0086] 2. Uplink Direction Estimation: Multi-source direction estimation is performed on the acquired audio signal to obtain the angle of arrival of the target sound source and interfering sound sources, thus determining the user's audio direction and the interference direction. This result serves as the basis for subsequent prediction and beam control.

[0087] 3. Trajectory prediction: Calculate the angular velocity based on the direction difference between the current frame and historical frames, and perform amplitude limiting and smoothing to predict the direction of the user's audio in the next frame.

[0088] The purpose of this step is to upgrade the traditional "passive following" to "forward tracking" and reduce off-target situations in mobile scenarios.

[0089] 4. Spatial Processing and Uplink Output: Dynamic weights are generated based on the prediction results, and three types of spatial strategies are executed: User audio direction enhancement; Dynamic null trap in the direction of interference; Non-target / non-interference direction, i.e., noise direction, retains low-gain ambient sound to avoid dead silence; at the same time, noise intensity is obtained.

[0090] This results in a purified uplink signal, which is then sent to the remote end.

[0091] 5. Environmental noise extraction and parameter sharing: While processing the space, the environmental noise intensity is calculated from the noise sector.

[0092] This parameter is not used as downlink voice content, but as a downlink volume control quantity, and is directly reused in S4 to avoid redundant estimation.

[0093] 6. Downlink volume adaptive: Taking the downstream remote incoming call signal as the object of processing, volume compensation is performed based on noise intensity. When the noise intensity is greater than the noise threshold, increase the target volume. Maintain the base volume when the noise intensity is less than or equal to the noise level.

[0094] Then, a limiting and smoothing update is performed, and the downstream playback signal is output to the local speaker or headphones.

[0095] 7. Closed-loop update: After the current frame is processed, proceed to the next frame and repeat S1 to S4.

[0096] The system forms a collaborative closed loop of uplink tracking enhancement and downlink auditory compensation in consecutive frames, achieving a stable call experience that is clear to speak and hear, and does not sound lifeless in complex mobile environments.

[0097] It should be noted that, for the sake of simplicity, the foregoing method embodiments are all described as a series of actions. However, those skilled in the art should understand that this application is not limited to the described order of actions, as some steps may be performed in other orders or simultaneously according to this application. Furthermore, those skilled in the art should also understand that the embodiments described in the specification are preferred embodiments, and the actions and modules involved are not necessarily essential to this application.

[0098] Through the above description of the embodiments, those skilled in the art can clearly understand that the methods according to the above embodiments can be implemented by means of software plus necessary general-purpose hardware platforms. Of course, they can also be implemented by hardware, but in many cases the former is a better implementation method. Based on this understanding, the technical solution of this application, in essence, or the part that contributes to the prior art, can be embodied in the form of a software product. This computer software product is stored in a storage medium (such as ROM / RAM, magnetic disk, optical disk), and includes several instructions to cause a terminal device (which may be a mobile phone, computer, server, or network device, etc.) to execute the methods described in the various embodiments of this application.

[0099] This application also provides a call control device for implementing the above-described call control method, wherein the call control includes: The first acquisition module is used to acquire the incoming call signal from the remote end and the audio acquisition signal collected by the microphone array; The second acquisition module is used to acquire the user's audio direction and extract the user's audio signal and noise signal from the audio acquisition signal based on the user's audio direction. The first transmitting module is used to transmit the user audio signal as an uplink signal to the remote end; The first determining module is used to determine the audio playback parameters corresponding to the remote incoming call signal based on the noise signal.

[0100] This call control device uses a microphone array to collect audio, enabling the audio signal to be combined with audio from multiple directions. By determining the direction of the user's audio, the device extracts the user's audio signal, thereby improving the clarity of the user's audio signal sent to the remote end during communication. At the same time, it determines the audio playback parameters based on the noise signal, which improves the clarity of the incoming call signal heard by the user from the remote end, achieving a clear call process and reducing the impact of environmental noise on the call.

[0101] It should be noted that the first acquisition module in this embodiment can be used to execute step S10 in this application embodiment, the second acquisition module in this embodiment can be used to execute step S20 in this application embodiment, the first sending module in this embodiment can be used to execute step S30 in this application embodiment, and the first determining module in this embodiment can be used to execute step S40 in this application embodiment.

[0102] Furthermore, prior to obtaining the user's audio direction, the process includes: In the previous control cycle, the audio direction of the previous control cycle is estimated based on the audio acquisition signal of the previous cycle. Calculate the angular velocity corresponding to the audio direction of the previous cycle; Based on the audio direction of the previous cycle, the direction of the next control cycle is predicted according to the directional angular velocity to obtain the user audio direction.

[0103] Further, the step of extracting the user audio signal and noise signal from the audio acquisition signal based on the user audio direction includes: Determine the interference direction corresponding to the user audio direction, wherein the interference direction is the direction of the interference sound source in the audio acquisition signal; The dynamic beamforming weights corresponding to the audio acquisition signal are determined based on the user audio direction and the interference direction. The dynamic beamforming weights indicate the enhancement of the audio acquisition data in the user audio direction and the suppression in the interference direction. The user audio signal is obtained by extracting the audio acquisition signal using the dynamic beamforming weights.

[0104] Furthermore, the step of determining the dynamic beamforming weights corresponding to the audio acquisition signal based on the user audio direction and the interference direction, wherein the dynamic beamforming weights indicate the enhancement of the audio acquisition data in the user audio direction and the suppression of the interference direction, includes: A maximum gain constraint is constructed for the user's audio direction; A zero-trap constraint is constructed for the direction of the disturbance; A low-gain constraint is constructed for the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction, and the gain corresponding to the low-gain constraint is a preset background gain; The dynamic beamforming weights are generated by combining the maximum gain constraint, the null constraint, and the low gain constraint.

[0105] Furthermore, the construction of null trap constraints on the interference direction includes: Obtain the direction of disturbance in multiple consecutive control cycles; Calculate the standard deviation of the fluctuation in the direction of the interference; Determine the width of the zero-trap angle corresponding to the standard deviation of the fluctuation, wherein the standard deviation of the fluctuation is positively correlated with the width of the zero-trap angle; Based on the zero-trap width, construct the zero-trap constraint corresponding to the interference direction.

[0106] Further, the step of extracting the user audio signal and noise signal from the audio acquisition signal based on the user audio direction includes: Determine the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction; Obtain noise data corresponding to the noise direction from the audio acquisition signal; Calculate the noise intensity corresponding to the noise data; The noise intensity is taken as the noise signal.

[0107] Further, determining the audio playback parameters corresponding to the distant incoming call signal based on the noise signal includes: Determine the noise intensity corresponding to the noise signal; Determine whether the noise intensity is greater than the noise threshold; If the noise intensity is greater than the noise threshold, the call volume corresponding to the audio playback parameter is adjusted, wherein the call volume is positively correlated with the noise intensity.

[0108] Reference Figure 3 In terms of hardware structure, the communication device may include components such as a communication module 10, a memory 20, and a processor 30. In the communication device, the processor 30 is connected to both the memory 20 and the communication module 10. The memory 20 stores a computer program, which is executed by the processor 30. When the computer program is executed, it implements the steps of the above-described method embodiments.

[0109] The communication module 10 can connect to external communication devices via a network. The communication module 10 can receive requests from the external communication devices and can also send requests, instructions, and information to the external communication devices. The external communication devices can be other communication devices, servers, or IoT devices, such as televisions, etc.

[0110] The memory 20 can be used to store software programs and various data. The memory 20 may primarily include a program storage area and a data storage area. The program storage area may store the operating system, at least one application program required for a function (such as acquiring remote incoming call signals and audio acquisition signals collected by a microphone array), etc.; the data storage area may include a database, and may store data or information created based on system usage. Furthermore, the memory 20 may include high-speed random access memory, and may also include non-volatile memory, such as at least one disk storage device, flash memory device, or other volatile solid-state storage device.

[0111] The processor 30 is the control center of the communication device. It connects various parts of the communication device via various interfaces and lines. By running or executing software programs and / or modules stored in the memory 20, and by calling data stored in the memory 20, it performs various functions and processes data, thereby providing overall monitoring of the communication device. The processor 30 may include one or more processing units; optionally, the processor 30 may integrate an application processor and a modem processor. The application processor mainly handles the operating system, user interface, and applications, while the modem processor mainly handles wireless communication. It is understood that the modem processor may not be integrated into the processor 30.

[0112] although Figure 3 Not shown, but the above-described communication device may further include a circuit control module, which is connected to a power supply to ensure the normal operation of other components. Those skilled in the art will understand that... Figure 3 The communication device structure shown does not constitute a limitation on the communication device and may include more or fewer components than shown, or combine certain components, or have different component arrangements.

[0113] The present invention also proposes a computer-readable storage medium having a computer program stored thereon. The computer-readable storage medium may be... Figure 3 The memory 20 in the communication device may also be at least one of ROM (Read-Only Memory) / RAM (Random Access Memory), magnetic disk, optical disk, etc. The computer-readable storage medium includes a number of instructions to cause a terminal device with a processor (which may be a television, automobile, mobile phone, computer, server, terminal, or network device, etc.) to execute the methods described in the various embodiments of the present invention.

[0114] In this invention, the terms "first," "second," "third," "fourth," and "fifth" are used for descriptive purposes only and should not be construed as indicating or implying relative importance. Those skilled in the art can understand the specific meaning of the above terms in this invention according to the specific circumstances.

[0115] In the description of this specification, the references to terms such as "one embodiment," "some embodiments," "example," "specific example," or "some examples," etc., indicate that a specific feature, structure, material, or characteristic described in connection with that embodiment or example is included in at least one embodiment or example of the present invention. In this specification, the illustrative expressions of the above terms do not necessarily refer to the same embodiment or example. Furthermore, the specific features, structures, materials, or characteristics described may be combined in any suitable manner in one or more embodiments or examples. Moreover, without contradiction, those skilled in the art can combine and integrate the different embodiments or examples described in this specification, as well as the features of different embodiments or examples.

[0116] Although embodiments of the present invention have been shown and described above, the scope of protection of the present invention is not limited thereto. It is understood that the above embodiments are exemplary and should not be construed as limiting the present invention. Those skilled in the art can make changes, modifications, and substitutions to the above embodiments within the scope of the present invention, and such changes, modifications, and substitutions should all be covered within the scope of protection of the present invention. Therefore, the scope of protection of the present invention should be determined by the scope of the claims.

Claims

1. A call control method, characterized in that, The call control method includes: Acquire the incoming call signal from the remote end and the audio signal collected by the microphone array; Obtain the user's audio direction, and extract the user's audio signal and noise signal from the audio acquisition signal based on the user's audio direction; The user's audio signal is sent as an uplink signal to the remote end; The audio playback parameters corresponding to the remote incoming call signal are determined based on the noise signal.

2. The call control method as described in claim 1, characterized in that, Prior to obtaining the user's audio direction, the following is included: In the previous control cycle, the audio direction of the previous control cycle is estimated based on the audio acquisition signal of the previous cycle. Calculate the angular velocity corresponding to the audio direction of the previous cycle; Based on the audio direction of the previous cycle, the direction of the next control cycle is predicted according to the directional angular velocity to obtain the user audio direction.

3. The call control method as described in claim 1, characterized in that, The step of extracting the user audio signal and noise signal from the audio acquisition signal based on the user audio direction includes: Determine the interference direction corresponding to the user audio direction, wherein the interference direction is the direction of the interference sound source in the audio acquisition signal; The dynamic beamforming weights corresponding to the audio acquisition signal are determined based on the user audio direction and the interference direction. The dynamic beamforming weights indicate the enhancement of the audio acquisition data in the user audio direction and the suppression in the interference direction. The user audio signal is obtained by extracting the audio acquisition signal through the dynamic beamforming weights.

4. The call control method as described in claim 3, characterized in that, The step of determining the dynamic beamforming weights corresponding to the audio acquisition signal based on the user audio direction and the interference direction, wherein the dynamic beamforming weights indicate the enhancement of the audio acquisition data in the user audio direction and the suppression of the interference direction, includes: A maximum gain constraint is constructed for the user's audio direction; A zero-trap constraint is constructed for the direction of the disturbance; A low-gain constraint is constructed for the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction, and the gain corresponding to the low-gain constraint is a preset background gain; The dynamic beamforming weights are generated by combining the maximum gain constraint, the null constraint, and the low gain constraint.

5. The call control method as described in claim 4, characterized in that, The construction of null constraints on the interference direction includes: Obtain the direction of disturbance in multiple consecutive control cycles; Calculate the standard deviation of the fluctuation in the direction of the interference; Determine the width of the zero-trap angle corresponding to the standard deviation of the fluctuation, wherein the standard deviation of the fluctuation is positively correlated with the width of the zero-trap angle; Based on the zero-trap width, construct the zero-trap constraint corresponding to the interference direction.

6. The call control method as described in claim 1, characterized in that, The step of extracting the user audio signal and noise signal from the audio acquisition signal based on the user audio direction includes: Determine the noise direction, wherein the noise direction is a direction other than the user audio direction and the interference direction; Obtain noise data corresponding to the noise direction from the audio acquisition signal; Calculate the noise intensity corresponding to the noise data; The noise intensity is taken as the noise signal.

7. The call control method as described in claim 1, characterized in that, The step of determining the audio playback parameters corresponding to the distant incoming call signal based on the noise signal includes: Determine the noise intensity corresponding to the noise signal; Determine whether the noise intensity is greater than the noise threshold; If the noise intensity is greater than the noise threshold, the call volume corresponding to the audio playback parameter is adjusted, wherein the call volume is positively correlated with the noise intensity.

8. A call control device, characterized in that, The call control device includes: The first acquisition module is used to acquire the incoming call signal from the remote end and the audio acquisition signal collected by the microphone array; The second acquisition module is used to acquire the user's audio direction and extract the user's audio signal and noise signal from the audio acquisition signal based on the user's audio direction. The first transmitting module is used to transmit the user audio signal as an uplink signal to the remote end; The first determining module is used to determine the audio playback parameters corresponding to the remote incoming call signal based on the noise signal.

9. A communication device, characterized in that, The communication device includes a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the computer program, when executed by the processor, implements the steps of the call control method as described in any one of claims 1 to 7.

10. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores a computer program that, when executed by a processor, implements the steps of the call control method as described in any one of claims 1 to 7.

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