Echo cancellation method, electronic device, storage medium and program product

By introducing nonlinear models and adaptive filtering technology, the problem of incomplete echo cancellation in the existing technology is solved, more accurate echo simulation and elimination is achieved, and communication quality is improved.

CN120343131AActive Publication Date: 2025-07-18HONOR DEVICE CO LTD
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Patent Information

Application Number
CN202510820280.9
Authority / Receiving Office
CN · China
Patent Type
Applications(China)
Current Assignee / Owner
Filing Date
2025-06-19
Publication Date
2025-07-18
Estimated Expiration
2045-06-19

AI Technical Summary

Technical Problem

The existing echo cancellation methods are mostly based on linear models, and cannot accurately simulate the nonlinear distortion of the speaker, resulting in unsatisfactory echo cancellation effect and cannot meet the needs of high-quality communication.

Method used

Nonlinear models, including electrical, mechanical and acoustic models, are used to fully cover complex nonlinear factors in the speaker's operation process. The echo signal is accurately simulated through adaptive filtering and inverting processing and echo cancellation.

Benefits of technology

It improves the accuracy and efficiency of echo cancellation, ensures the clarity and nature of voice communication, and meets the needs of high-quality communication.

✦ Generated by Eureka AI based on patent content.

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Abstract

The embodiment of the invention discloses an echo cancellation method, electronic equipment, a storage medium and a program product. The echo cancellation method comprises the following steps: determining an echo analog signal based on a first input signal by using a nonlinear model; the nonlinear model is used for describing a nonlinear relationship between the echo analog signal and the first input signal; and performing echo cancellation processing on the second input signal according to the echo analog signal. According to the embodiment of the invention, the echo simulation signal is determined by introducing the nonlinear model, and the nonlinear model can accurately describe a complex nonlinear relationship between the echo simulation signal and the first input signal, so that an actual echo generation process is more accurately simulated, and the echo simulation signal can be closer to a real echo signal. On the basis, echo cancellation processing is performed on the second input signal picked up by the sound pickup by using the echo analog signal, so that echo components can be more accurately eliminated, the communication quality is improved, and complete and clear voice communication experience is provided for a user.
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Description

Technical Field

[0001] This application relates to the field of echo cancellation, and in particular, to a method for echo cancellation, an electronic device, a storage medium, and a program product. Background Art

[0002] In communication systems, especially in the fields of voice communication and audio processing, the echo problem has always been an important factor affecting communication quality. Echoes are mainly generated when the sound played by the speaker is picked up again by the pickup, thus mixing into the original input signal, causing both communication parties to hear repeated sounds, affecting the clarity and naturalness of the call, and reducing the communication efficiency and user experience.

[0003] Traditional echo cancellation methods estimate and cancel echo components through techniques such as linear filtering. However, their echo cancellation effect is not ideal, and obvious echo residues remain, unable to meet the requirements of high-quality communication. Summary of the Invention

[0004] The purpose of this application is to provide a method for echo cancellation, an electronic device, a storage medium, and a program product to solve the problem of inaccurate echo cancellation in the prior art.

[0005] In a first aspect, this application provides a method for echo cancellation, including: Determining an echo simulation signal based on a first input signal using a non-linear model; the non-linear model is used to describe the non-linear relationship between the echo simulation signal and the first input signal; the first input signal is the signal input to the speaker; Performing echo cancellation processing on a second input signal according to the echo simulation signal; the second input signal is the signal picked up by the pickup.

[0006] In the embodiments of this application, a non-linear model is introduced to determine the echo simulation signal. The non-linear model can accurately describe the complex non-linear relationship between the echo simulation signal and the first input signal, and thus more accurately simulate the actual echo generation process, making the echo simulation signal closer to the real echo signal. On this basis, using this echo simulation signal to perform echo cancellation processing on the second input signal picked up by the pickup can more accurately cancel the echo components, effectively overcoming problems such as incomplete echo cancellation in traditional technologies, improving communication quality, and providing users with a complete and clear voice communication experience.

[0007] In some possible implementation manners, the non-linear model includes an electrical model, a mechanical model, and an acoustic model; the electrical model is used to describe the relationship between the first input signal and the non-linear parameters of the voice coil of the speaker; the mechanical model is used to describe the relationship between the non-linear parameters of the voice coil and the non-linear parameters of the diaphragm of the speaker; the acoustic model is used to describe the relationship between the non-linear parameters of the diaphragm and the echo simulation signal.

[0008] In the embodiments of the present application, the non-linear model includes an electrical model, a mechanical model, and an acoustic model established by focusing on the non-linear characteristics of different physical levels of the speaker, comprehensively and systematically covering various complex non-linear factors in the working process of the speaker. Compared with a single model, it can more accurately capture each physical link in the generation of echoes, thereby improving the accuracy of the echo simulation signal.

[0009] In some possible implementation manners, using the non-linear model to determine the echo simulation signal based on the first input signal includes: Determining the voice coil non-linear parameter corresponding to the first input signal based on the electrical model; Determining the diaphragm non-linear parameter corresponding to the voice coil non-linear parameter based on the mechanical model; Determining the echo simulation signal corresponding to the diaphragm non-linear parameter based on the acoustic model.

[0010] In the embodiments of the present application, by sequentially applying the electrical model, the mechanical model, and the acoustic model, the characteristic parameters related to the echo are gradually and accurately extracted from the first input signal, and finally a high-precision echo simulation signal is generated. This step-by-step processing method can consider the non-linear characteristics of each link of the speaker more comprehensively compared with a single model, thereby effectively improving the accuracy of echo simulation and providing data support for subsequent echo cancellation.

[0011] In some possible implementation manners, the voice coil non-linear parameter of the speaker includes the voice coil magnetic force factor, and the non-linear model further includes a first functional relationship, which is used to describe the non-linear relationship between the voice coil magnetic force factor of the speaker, time, and the voice coil displacement of the speaker.

[0012] During the sound generation process of the speaker, when the voice coil moves in the magnetic circuit, the non-uniform distribution of the magnetic circuit will cause non-linear changes in the voice coil magnetic force factor, resulting in non-linear distortion of the sound emitted by the speaker. In the embodiments of the present application, by incorporating the non-linear relationship between the voice coil magnetic force factor, the voice coil displacement, and time into the model, the model can reflect the non-linear characteristics of the voice coil magnetic force factor, and thus can more accurately capture the actual behavior of the voice coil during the dynamic working process, which helps to improve the accuracy of echo simulation, thereby enhancing the effect of echo cancellation and ensuring the clarity and naturalness of voice communication.

[0013] In some possible implementation manners, the diaphragm non-linear parameter of the speaker includes the diaphragm vibration stiffness, and the non-linear model further includes a second functional relationship, which is used to describe the non-linear relationship between the diaphragm vibration stiffness of the speaker, time, the diaphragm vibration angular frequency of the speaker, and the voice coil displacement.

[0014] During the sound generation process of the speaker, the vibration stiffness of the speaker's diaphragm changes non-linearly with the change of the vibration position, resulting in non-linear distortion in the sound emitted by the speaker. In the embodiments of the present application, by introducing the non-linear relationship between the diaphragm vibration stiffness and the diaphragm vibration angular frequency, voice coil displacement, and time into the non-linear model, the complex dynamic behavior of the diaphragm during actual operation can be more accurately simulated, and further, the non-linear stiffness change of the diaphragm under different vibration frequencies, voice coil displacements, and time conditions can be captured, thereby improving the accuracy of the echo simulation signal.

[0015] In some possible implementation manners, the diaphragm non-linear parameters of the speaker include the diaphragm vibration damping, and the non-linear model further includes a third functional relationship, which is used to describe the non-linear relationship between the diaphragm vibration damping of the speaker, time, the diaphragm vibration angular frequency of the speaker, and the voice coil velocity.

[0016] During the sound generation process of the speaker, the diaphragm vibration damping of the speaker will produce non-linear changes due to the asymmetry of the vibrating air environment during the up-and-down vibration, resulting in non-linear distortion in the sound emitted by the speaker. In the embodiments of the present application, by introducing the non-linear relationship between the diaphragm vibration damping and the diaphragm vibration angular frequency, voice coil velocity, and time into the non-linear model, the energy dissipation characteristics of the diaphragm under different vibration states and time conditions can be more accurately captured, which helps to improve the accuracy of the echo simulation signal.

[0017] In some possible implementation manners, the diaphragm non-linear parameters of the speaker include the diaphragm vibration area, and the non-linear model further includes a fourth functional relationship, which is used to describe the non-linear relationship between the diaphragm vibration area of the speaker, the diaphragm vibration angular frequency of the speaker, and the voice coil displacement.

[0018] During the sound generation process of the speaker, when the speaker diaphragm vibrates up and down, due to the asymmetric distribution of the surround up and down, the effective vibration area of the diaphragm changes inconsistently with the up-and-down vibration, resulting in non-linear distortion in the sound emitted by the speaker. In the embodiments of the present application, by introducing the non-linear relationship between the diaphragm vibration area and the diaphragm vibration angular frequency and the voice coil displacement into the non-linear model, the actual vibration area change of the diaphragm under different vibration states can be more accurately simulated, which helps to improve the accuracy of the echo simulation signal.

[0019] In some possible implementation manners, the diaphragm non-linear parameters of the speaker include the front cavity stiffness, and the non-linear model further includes a fifth functional relationship, which is used to describe the non-linear relationship between the front cavity stiffness of the speaker, time, and the voice coil displacement of the speaker.

[0020] During the sound generation process of the speaker, most of the space in the front cavity of the speaker is occupied when the speaker diaphragm moves upward, and it recovers when the diaphragm moves downward. This results in the stiffness of the front cavity of the speaker not being a constant, causing non-linear distortion in the sound emitted by the speaker. In the embodiments of the present application, by introducing the non-linear relationship between the front cavity stiffness, the voice coil displacement, and time into the non-linear model, the dynamic characteristics of the front cavity of the speaker can be more accurately simulated. Furthermore, the changes in the front cavity stiffness under different voice coil displacements and time conditions can be captured, thereby improving the accuracy of the echo simulation signal.

[0021] In some possible implementation manners, the non-linear parameters of the speaker diaphragm include acoustic damping, and the non-linear model further includes a sixth functional relationship, which is used to describe the non-linear relationship between the acoustic damping of the speaker, time, and the voice coil velocity of the speaker.

[0022] During the sound generation process of the speaker, when the speaker diaphragm vibrates up and down, the pressures encountered by the air in the front cavity of the speaker when discharging outward and sucking inward are not consistent, resulting in non-linear changes in the front cavity acoustic damping and causing non-linear distortion in the sound emitted by the speaker. In the embodiments of the present application, by introducing the non-linear relationship between the acoustic damping, the voice coil velocity, and time into the non-linear model, the energy dissipation characteristics of the speaker under different operating states can be more accurately simulated, which helps to improve the accuracy of the echo simulation signal.

[0023] In some possible implementation manners, performing echo cancellation processing on the second input signal according to the echo simulation signal includes: Performing adaptive filtering processing on the echo simulation signal to obtain a filtered signal; Performing inverting processing on the filtered signal to obtain a first inverted signal; Performing addition processing on the first inverted signal and the second input signal to complete the echo cancellation processing.

[0024] In the embodiments of the present application, adaptive filtering can dynamically adjust the filtering parameters according to the characteristics of the actual echo signal, realizing more accurate processing of the echo simulation signal and improving the accuracy and adaptability of the echo cancellation effect. Inverting processing can generate a signal with a phase opposite to that of the echo signal, which can effectively cancel the echo when added to the original echo signal, enhancing the efficiency and accuracy of echo cancellation.

[0025] In some possible implementation manners, performing echo cancellation processing on the second input signal according to the echo simulation signal includes: Performing inverting processing on the echo simulation signal to obtain a second inverted signal; Performing addition processing on the second inverted signal and the second input signal to complete the echo cancellation processing.

[0026] In the embodiments of the present application, the second anti-phase signal generated through anti-phase processing can be directly opposite to the echo signal in phase, and can effectively cancel the echo when added, improving the echo cancellation effect, further enhancing the processing efficiency of echo cancellation, and reducing the system load and latency.

[0027] In a second aspect, the present application further provides an electronic device, including: a processor; a memory; A computer program is stored in the memory, and when the computer program is executed, the electronic device is caused to execute the method according to any one of the first aspect.

[0028] In a third aspect, the present application further provides a computer-readable storage medium, which includes a stored program, wherein when the program runs, it controls the device where the computer-readable storage medium is located to execute the method according to any one of the first aspect.

[0029] In a fourth aspect, the present application further provides a program product, which stores a program, and when the program is run by an information processing device, it causes the information processing device to execute the method according to any one of the first aspect.

[0030] In the embodiments of the present application, a non-linear model is introduced to determine the echo simulation signal. The non-linear model can accurately describe the complex non-linear relationship between the echo simulation signal and the first input signal, and thus more accurately simulate the actual echo generation process, making the echo simulation signal closer to the real echo signal. On this basis, using the echo simulation signal to perform echo cancellation processing on the second input signal picked up by the pickup can more accurately cancel the echo component, effectively overcoming problems such as incomplete echo cancellation in traditional technologies, improving the communication quality, and providing users with a complete and clear voice communication experience. BRIEF DESCRIPTION OF THE DRAWINGS

[0031] Figure 1 FIG. is a schematic diagram of the top appearance of a mobile phone with a top pickup hole provided by the embodiments of the present application; Figure 2 FIG. is a schematic diagram of the top appearance of a mobile phone without a top pickup hole provided by the embodiments of the present application; Figure 3 FIG. is a schematic diagram of the appearance of a mobile phone with a top hole-free design provided by the embodiments of the present application; Figure 4 FIG. is a schematic diagram of the internal structure of a mobile phone in which the pickup and the speaker share a narrow slit provided by the embodiments of the present application; Figure 5 FIG. is a schematic diagram of an echo cancellation method in the prior art; Figure 6 FIG. is a schematic diagram of the principle of an echo cancellation method in the prior art; Figure 7 Schematic diagram of the structure of an electronic device provided by an embodiment of the present application; Figure 8 Block diagram of the software structure of an electronic device provided by an embodiment of the present application; Figure 9 Flowchart of a method for echo cancellation provided by an embodiment of the present application; Figure 10 Flowchart of the process of echo cancellation of an input signal provided by an embodiment of the present application; Figure 11 For Figure 9 Schematic diagram of the process of step S01 in a method for echo cancellation provided; Figure 12 Schematic diagram of the structure of the first loudspeaker provided by an embodiment of the present application; Figure 13 Schematic diagram of the structural relationship between the diaphragm and the voice coil of a loudspeaker provided by an embodiment of the present application; Figure 14 Schematic diagram of the structure of the second loudspeaker provided by an embodiment of the present application; Figure 15 Schematic diagram of the structure of a loudspeaker diaphragm provided by an embodiment of the present application; Figure 16 Schematic diagram of the structure of the third loudspeaker provided by an embodiment of the present application; Figure 17 Schematic diagram of the change in air pressure during the sound generation process of a loudspeaker provided by an embodiment of the present application; Figure 18 Schematic diagram of an equivalent circuit model of a loudspeaker provided by an embodiment of the present application; Figure 19 Comparison effect diagram of the residual time-frequency spectrum provided by an embodiment of the present application; Figure 20 Comparison effect diagram of the residual frequency spectrum provided by an embodiment of the present application. Detailed implementation manners

[0032] For a better understanding of the technical solutions of the present application, the embodiments of the present application will be described in detail below with reference to the accompanying drawings.

[0033] It should be clear that the described embodiments are only a part of the embodiments of the present application, rather than all of the embodiments. All other embodiments obtained by those of ordinary skill in the art based on the embodiments in the present application without creative efforts shall fall within the scope of protection of the present application.

[0034] The terms used in the embodiments of the present application are for the purpose of describing specific embodiments only and are not intended to limit the present application. The singular forms "a", "an" and "the" used in the embodiments of the present application and the appended claims are also intended to include the plural forms unless the context clearly indicates otherwise.

[0035] It should be understood that the term "and / or" used herein is merely a description of the associated relationship of the associated objects, indicating that there can be three relationships. For example, A and / or B can represent: A exists alone, A and B exist simultaneously, and B exists alone. In addition, the character " / " herein generally represents an "or" relationship between the associated objects before and after.

[0036] The method for echo cancellation provided by the embodiments of the present application will be described below.

[0037] In a communication system, especially in the fields of voice communication and audio processing, the echo problem has always been an important factor affecting communication quality. The generation of echo mainly stems from the sound played by the speaker being picked up again by the pickup, thus mixing into the original input signal, resulting in both communication parties hearing repeated sounds, affecting the clarity and naturalness of the call, and reducing the communication efficiency and user experience.

[0038] Here, an example is given to illustrate the iterative process of the hardware design in the mobile phone acoustic system: See Figure 1 , which is a schematic diagram of the top appearance of a mobile phone with a top pickup hole provided by the embodiments of the present application. As Figure 1 shown, in order to ensure the pickup function, the mobile phone is provided with a speaker sound outlet hole and a pickup pickup hole at the top at the same time. This design results in the asymmetry of the holes at the top of the mobile phone, affecting the aesthetics of the top of the mobile phone.

[0039] See Figure 2 , which is a schematic diagram of the top appearance of a mobile phone without a top pickup hole provided by the embodiments of the present application. As Figure 2 shown, the mobile phone cancels the top pickup hole. By sharing the hole position of the pickup and the speaker, the problem of asymmetry of the holes at the top of the mobile phone is solved.

[0040] See Figure 3 , which is a schematic diagram of the appearance of a mobile phone with a holeless top provided by the embodiments of the present application. As Figure 3 shown, the mobile phone cancels the top hole position. Instead, by reserving a narrow slit above the screen, the speaker and the pickup share the narrow slit to realize sound output and pickup respectively, solving the problem that the hole position at the top of the mobile phone affects the aesthetics.

[0041] Figures 1 to 3The iterative process of the hardware design in the mobile phone acoustic system shown indicates that in the development process of mobile phones, aesthetics has been gradually elevated to a higher priority, and this design approach has led to new challenges for acoustic performance, especially in echo suppression. Specifically, as the speaker and the microphone share the same hole or even the same narrow slit, the echo energy picked up by the microphone is relatively higher, which further increases the difficulty of echo suppression. In Figure 1 the speaker sound outlet hole and the microphone sound pickup hole are separated, but the distance between the sound outlet hole and the sound pickup hole is relatively close. The sound emitted by the speaker through the sound outlet hole will be picked up by the microphone through the sound pickup hole, resulting in the generation of echo. In Figure 2 and Figure 3 the microphone and the speaker share the same hole or the same narrow slit, resulting in poor isolation between the speaker and the microphone, making it easier for the microphone to pick up the sound emitted by the speaker, further increasing the difficulty of echo suppression.

[0042] See Figure 4 for the schematic structural diagram of the internal acoustic system of the holeless mobile phone shown in Figure 3 . As shown in Figure 4 , when the sound emitted by the speaker core outputs from the appearance narrow slit through the front cavity sound outlet pipe, due to the too-close distance between the front cavity sound outlet pipe and the microphone sound pickup pipe, part of the sound will be picked up by the microphone from the appearance narrow slit through the sound pickup pipe, thus increasing the echo energy picked up by the microphone. Especially in a scheme with a built-in subwoofer in a mobile phone, the microphone is located in the open rear cavity of the speaker, and part of the sound emitted by the speaker core can be directly picked up by the microphone through the rear cavity, further increasing the echo energy picked up by the microphone and resulting in a sudden increase in the difficulty of echo suppression.

[0043] To solve the echo problem, in some possible embodiments, echo cancellation processing is performed through an adaptive filter.

[0044] See Figure 5 for the schematic diagram of an echo cancellation method in an embodiment provided by this application. As shown in Figure 5 , when the speaker plays sound, it converts the received electrical signal (the far-end signal shown in the figure) into a sound signal (the speaker output signal shown in the figure) and outputs it. To ensure that the microphone only picks up the voice signal of the proximal speaker, the echo signal generated by the speaker in the input signal picked up by the microphone is eliminated through an adaptive filter.

[0045] See Figure 6 for the schematic principle diagram of the echo cancellation method shown in Figure 5 . As shown in Figure 6As shown in the figure, an adaptive filter is used to perform adaptive filtering on the electrical signal input to the speaker, and an inverter is used to invert the processed electrical signal to obtain a signal that is inverse to the echo signal. Then, this signal is superimposed on the signal picked up by the pickup to eliminate the echo.

[0046] However, the above echo cancellation methods are mostly based on linear models, which pre-assume a linear relationship between the echo signal and the signal input to the speaker, and estimate and cancel the echo component through a linear filter. However, in actual applications, the generation of echo is often accompanied by non-linear distortion of the speaker. These non-linear factors make the relationship between the actual echo signal and the input signal not a simple linear relationship. Existing echo cancellation technologies based on linear models cannot accurately simulate and cancel the echo when facing this non-linear distortion, resulting in an unsatisfactory echo cancellation effect, obvious echo residue, and inability to meet the requirements of high-quality communication.

[0047] In view of this, the embodiments of the present application provide a method for echo cancellation to solve the above technical problems.

[0048] The following describes the electronic device applicable to the echo cancellation method provided by the present application and the specific process of this method in conjunction with embodiments.

[0049] The echo cancellation method provided by the embodiments of the present application can be applied to devices capable of screen touch, such as mobile phones, tablet computers, personal computers (PCs), personal digital assistants (PDAs), smart watches, netbooks, wearable electronic devices, augmented reality (AR) devices, virtual reality (VR) devices, in-vehicle devices, smart cars, smart speakers, robots, smart glasses, smart TVs, etc.

[0050] Exemplarily, refer to Figure 7 , which is a schematic structural diagram of an electronic device provided by the embodiments of the present application. As Figure 7As shown, the electronic device 100 may include a display screen 194, a processor 110, an external memory interface 120, an internal memory 121, a universal serial bus (USB) interface 130, a charging management module 140, a power management module 141, a battery 142, an antenna 1, an antenna 2, a mobile communication module 150, a wireless communication module 160, an audio module 170, a speaker 170A, a receiver 170B, a microphone 170C, a headphone jack 170D, a sensor module 180, a motor 191, an indicator 192, a camera 193, and a subscriber identification module (SIM) card interface 195, etc. Among them, the sensor module 180 may include a touch sensor 180F.

[0051] It can be understood that the structure illustrated in the embodiments of the present application does not constitute a specific limitation on the electronic device 100. In some other possible embodiments of the present application, the electronic device 100 may include more or fewer components than shown, or combine certain components, or split certain components, or have different component arrangements. The illustrated components may be implemented in hardware, software, or a combination of software and hardware.

[0052] The display screen 194, also known as the display panel or screen, is used to display images, videos, etc. The display screen 194 includes a display panel. The display panel may adopt an OLED display panel. In some possible embodiments, the electronic device 100 may include one or N display screens 194, where N is a positive integer greater than 1.

[0053] The processor 110 may include one or more processing units. For example, the processor 110 may include an application processor (AP), a modem processor, a controller, a video codec, a digital signal processor (DSP), a baseband processor, and / or a neural-network processing unit (NPU), etc. Among them, different processing units may be independent devices or integrated in one or more processors.

[0054] Among them, the controller may be the nerve center and command center of the electronic device 100. The controller may generate operation control signals according to the instruction operation code and timing signal to complete the control of fetching and executing instructions.

[0055] A memory may also be provided in the processor 110 for storing instructions and data. In some possible embodiments, the memory in the processor 110 is a cache memory. This memory may store instructions or data that the processor 110 has just used or recycled. If the processor 110 needs to use the instruction or data again, it can be directly called from the memory. This avoids repeated accesses, reduces the waiting time of the processor 110, and thus improves the efficiency of the system.

[0056] In some possible embodiments, the processor 110 may include one or more interfaces. The interfaces may include an inter-integrated circuit (I2C) interface, an inter-integrated circuit sound (I2S) interface, a pulse code modulation (PCM) interface, a universal asynchronous receiver / transmitter (UART) interface, a mobile industry processor interface (MIPI), a general-purpose input / output (GPIO) interface, a subscriber identity module (SIM) interface, and / or a universal serial bus (USB) interface, etc.

[0057] It can be understood that the interface connection relationships between the modules illustrated in the embodiments of the present application are only illustrative and do not constitute a structural limitation on the electronic device 100. In some other possible embodiments of the present application, the electronic device 100 may also adopt different interface connection methods in the above embodiments, or a combination of multiple interface connection methods.

[0058] The internal memory 121 can be used to store computer-executable program codes, and the executable program codes include instructions. The internal memory 121 can include a program storage area and a data storage area. Among them, the program storage area can store the operating system, application programs required for at least one function (such as network connection control function, etc.). The data storage area can store data created during the use of the electronic device 100. In addition, the internal memory 121 can include high-speed random access memory, and can also include non-volatile memory, such as at least one magnetic disk storage device, flash memory device, universal flash storage (UFS), etc. The processor 110 executes various functional applications and data processing of the electronic device 100 by running the instructions stored in the internal memory 121 and / or the instructions stored in the memory provided in the processor.

[0059] The electronic device 100 can implement audio functions through the audio module 170, speaker 170A, receiver 170B, microphone 170C, headphone jack 170D, and application processor, etc. For example, music playback, recording, etc.

[0060] The audio module 170 is used to convert digital audio information into an analog audio signal for output, and is also used to convert analog audio input into a digital audio signal. The audio module 170 can also be used for encoding and decoding audio signals. In some possible embodiments, the audio module 170 can be disposed in the processor 110, or some functional modules of the audio module 170 can be disposed in the processor 110.

[0061] The speaker 170A, also called a "loudspeaker", is used to convert an audio electrical signal into a sound signal. The electronic device 100 can listen to music or hands-free calls through the speaker 170A.

[0062] The receiver 170B, also called a "handset", is used to convert an audio electrical signal into a sound signal. When the electronic device 100 answers a call or a voice message, the voice can be listened to by bringing the receiver 170B close to the human ear.

[0063] The microphone 170C, also called a "microphone", "transmitter", is used to convert a sound signal into an electrical signal. When making a call or sending a voice message, the user can speak by bringing the mouth close to the microphone 170C to input the sound signal into the microphone 170C. The electronic device 100 can be provided with at least one microphone 170C. In some other possible embodiments, the electronic device 100 can be provided with two microphones 170C, which can not only collect sound signals but also implement a noise reduction function. In some other possible embodiments, the electronic device 100 can also be provided with three, four or more microphones 170C to implement sound signal collection, noise reduction, and can also identify the sound source to implement a directional recording function, etc.

[0064] The headphone jack 170D is used to connect a wired headphone. The headphone jack 170D can be a USB interface 130, or a 3.5 mm open mobile terminal platform (OMTP) standard interface, or a cellular telecommunications industry association of the USA (CTIA) standard interface.

[0065] See Figure 8 , which is a software structure block diagram of an electronic device provided by an embodiment of the present application. The software system of the electronic device 100 can adopt a layered architecture, an event-driven architecture, a microkernel architecture, a microservices architecture, or a cloud architecture. In the embodiment of the present application, the Android system with a layered architecture is taken as an example to exemplarily illustrate the software structure of the electronic device 100. The layered architecture divides the software into several layers, and each layer has a clear role and division of labor. The layers communicate with each other through software interfaces. As Figure 8 shown, in the embodiment of the present application, the Android system includes an application layer, a framework layer, a hardware abstraction layer, and a kernel layer from top to bottom.

[0066] The application layer (Application, App) may include a series of application program packages. For example, the application program package may include application programs such as a gallery, a file manager, a chart, a calendar, a map, a favorite, a message, a shopping, a note, and a contact list. In the embodiment of the present application, the application program further includes communication applications such as a phone, a video conference, and a voice conference. The communication application is used to determine an echo simulation signal based on a first input signal by using a non-linear model during a communication process; the non-linear model is used to describe the non-linear relationship between the echo simulation signal and the first input signal; the first input signal is a signal input to a speaker; the second input signal is processed for echo cancellation according to the echo simulation signal; the second input signal is a signal picked up by a pickup.

[0067] The framework layer (Framework, FWK) provides an application programming interface (API) and a programming framework for the application programs in the application layer, including some predefined functions. In Figure 2Among them, the application framework layer may include a window manager, a content provider, a view system, a telephone manager, and a resource manager. The window manager is used to manage window programs. The window manager can obtain the display screen size, determine whether there is a status bar, lock the screen, capture the screen, etc. The content provider is used to store and obtain data, and make this data accessible to application programs. The data may include videos, images, audio, incoming and outgoing calls, browsing history and bookmarks, phone books, etc. The view system includes visual controls, such as controls for displaying text, controls for displaying pictures, etc. The view system can be used to build application programs. The display interface can be composed of one or more views. The telephone manager is used to provide the communication function of the electronic device 100. For example, the management of call status (including connection, disconnection, etc.). The resource manager provides various resources for application programs, such as localized strings, icons, pictures, layout files, audio files, and so on.

[0068] The Hardware Abstract Layer (HAL) is an interface layer located between the operating system kernel and the hardware circuit. Its purpose is to abstract the hardware. It hides the hardware interface details of a specific platform, provides a virtual hardware platform for the operating system, makes it hardware-independent, and can be ported to multiple platforms. For example, HAL can provide virtual hardware for one or more audio processors, that is, an audio processing module, which can obtain audio data from the underlying pickups and speakers and process the audio data. In some embodiments provided in the present application, the hardware abstract layer can use a non-linear model to determine an echo simulation signal based on a first input signal through the audio processing module, and perform echo cancellation processing on a second input signal collected from the hardware according to the echo simulation signal to support the efficient operation and function implementation of communication applications.

[0069] The kernel layer is the layer between the hardware and the software. In some embodiments provided in the present application, the kernel layer may include a pickup driver, an audio processor driver, a speaker driver, etc. Among them, the speaker driver is used to drive the speaker to convert the first input signal in the form of an electrical signal received into a sound signal. The pickup driver is used to drive the pickup to pick up the second input signal in different environments.

[0070] For the sake of easy understanding, the following embodiments of the present application will use an electronic device with Figure 7 and Figure 8 the structure shown as an example, and in combination with the accompanying drawings and application scenarios, specifically elaborate on the method for echo cancellation provided by the embodiments of the present application.

[0071] See Figure 9 , which is a schematic flowchart of a method for echo cancellation provided by an embodiment of the present application. It can be understood that Figure 9In the method for echo cancellation, the timing of each step is only an example and does not impose any limitation on the execution order of actions. For example, Figure 9 As shown, when applied to an electronic device, the main steps of the echo cancellation method provided by the embodiments of the present application include: S01: Determine an echo simulation signal based on a first input signal using a non-linear model.

[0072] Among them, the non-linear model is used to describe the non-linear relationship between the echo simulation signal and the first input signal; the first input signal is the signal input to the speaker.

[0073] In the embodiments of the present application, the non-linear model is a mathematical model that can more accurately describe the complex non-linear relationship between the first input signal input to the speaker and the generated echo simulation signal. Compared with the traditional linear model, the non-linear model is no longer limited to simple linear functions, but fully considers the non-linear distortion of the signal caused by the non-linear characteristics of the speaker itself. Through the non-linear model, an echo simulation signal closer to the actual echo signal can be more accurately simulated, providing a more reliable basis for subsequent echo cancellation processing.

[0074] S02: Perform echo cancellation processing on a second input signal according to the echo simulation signal.

[0075] In the embodiments of the present application, the second input signal is the signal picked up by the pickup, which contains interference components such as the user's voice and echo. The echo simulation signal is generated based on the first input signal using a non-linear model and is used to approximate the actually generated echo. Echo cancellation processing refers to separating the actual echo signal from the second input signal based on the echo simulation signal, leaving a relatively pure user voice signal to improve the quality of voice communication and thus enhance the user experience.

[0076] For the specific implementation of step S02, reference can be made to the embodiments shown in subsequent steps S21 to S23 and steps S31 to S32, which will not be elaborated here.

[0077] Based on the above technical solutions, the embodiments of the present application determine the echo simulation signal by introducing a non-linear model. The non-linear model can accurately describe the complex non-linear relationship between the echo simulation signal and the first input signal, and thus more accurately simulate the actual echo generation process, making the echo simulation signal closer to the real echo signal. On this basis, using this echo simulation signal to perform echo cancellation processing on the second input signal picked up by the pickup can more accurately eliminate the echo component, effectively overcoming problems such as incomplete echo cancellation in traditional technologies, improving communication quality, and providing users with a complete and clear voice communication experience.

[0078] SeeFigure 10 , which is a schematic flowchart of echo cancellation for an input signal provided by an embodiment of the present application. As Figure 10 shown, in some embodiments provided by the present application, a nonlinear model is used to determine an echo simulation signal based on a first input signal, and the echo simulation signal is used to perform echo cancellation processing on a second input signal picked up by a pickup, so that the echo signal is removed from the output signal, improving the quality of voice communication.

[0079] Based on the above embodiments, in some possible embodiments, to effectively improve the accuracy of echo simulation and thus improve the accuracy of echo cancellation, more accurate echo simulation signals can be generated through multiple associated nonlinear models at different physical levels. That is, the nonlinear model can include an electrical model, a mechanical model, and an acoustic model, where: 1) The electrical model is used to describe the relationship between the first input signal and the nonlinear parameters of the voice coil of the speaker.

[0080] When the speaker is working, the first input signal (usually an electrical signal) is converted into a sound signal and output. The voice coil of the speaker plays a key role in this conversion process, but its actual performance is not completely linear. Therefore, in the embodiments of the present application, the electrical model is a mathematical model for characterizing the quantitative relationship between the first input signal and the nonlinear parameters of the speaker voice coil, which describes how the input electrical signal affects the behavior of the voice coil through the electrical characteristics of the speaker (such as nonlinear parameters such as impedance and inductance), such as the current and voltage waveforms in the voice coil and the distortion caused by nonlinear factors. So that the system can determine the nonlinear parameters of the voice coil under different input signal conditions, laying a foundation for more accurately simulating the echo signal subsequently.

[0081] In some possible embodiments, the establishment of the electrical model requires a comprehensive measurement and analysis of the electrical characteristics of the speaker. This includes measuring the changes in parameters such as the impedance and inductance of the speaker, as well as the current and voltage waveforms in the voice coil under input signals of different frequencies and amplitudes. Through these experimental data, circuit analysis theory and nonlinear system modeling methods can be used, such as using mathematical tools such as polynomial models and Volterra series to fit the relationship between the input signal and the nonlinear parameters of the voice coil.

[0082] 2) The mechanical model is used to describe the relationship between the nonlinear parameters of the voice coil and the nonlinear parameters of the speaker diaphragm.

[0083] In the embodiments of the present application, when the speaker is working, there is a close mechanical connection between the movement of the voice coil and the vibration of the diaphragm. The mechanical model is a mathematical model used to accurately describe the interaction relationship between the nonlinear parameters of the voice coil and the nonlinear parameters of the diaphragm. It covers the mechanical connection characteristics between the voice coil and the diaphragm, the force transmission and conversion process, and the nonlinear effects caused by factors such as material properties and geometric shapes. Through the mechanical model, it can accurately reflect how the movement of the voice coil affects the vibration state of the diaphragm through complex mechanical relationships, thereby providing a key mechanical basis for more realistically simulating the acoustic output of the speaker.

[0084] In some possible embodiments, the establishment of the mechanical model requires detailed mechanical property tests on the voice coil and diaphragm of the speaker. This includes measuring the mechanical responses (such as displacement, velocity, acceleration, etc.) of the voice coil under different currents and different frequencies, as well as the changes in mechanical parameters of the diaphragm under different vibration modes. For example, professional equipment such as a laser vibrometer and a dynamic mechanical analyzer can be used to obtain high-precision data. Then, combined with the experimental data, numerical modeling methods (such as finite element analysis, nonlinear regression, etc.) are used to construct the mechanical model. During the model establishment process, factors such as the connection stiffness and nonlinear damping between the voice coil and the diaphragm need to be considered, and the model parameters are continuously adjusted to make the output of the model match the actual test results.

[0085] 3) The acoustic model is used to describe the relationship between the nonlinear parameters of the diaphragm and the echo simulation signal.

[0086] As a key component for the speaker to generate sound, the nonlinear vibration characteristics of the diaphragm play a decisive role in the quality and characteristics of the output sound. However, traditional echo simulation methods often simplify this complex acoustic process and ignore the fine influence of the nonlinear parameters of the diaphragm on the sound wave form, resulting in a deviation between the simulated echo signal and the actual echo. Therefore, in the embodiments of the present application, the acoustic model converts the complex nonlinear vibration characteristics of the diaphragm into quantifiable acoustic parameters, and then accurately describes how these acoustic parameters change with the nonlinear vibration of the diaphragm, and finally generates an echo simulation signal. It comprehensively considers the influence of factors such as the material properties, shape, vibration mode of the diaphragm, and the surrounding acoustic environment on sound propagation, thereby establishing a bridge between the nonlinear vibration of the diaphragm and the echo signal, enabling the accurate prediction of the characteristics of the echo signal based on the physical behavior of the diaphragm.

[0087] In some possible embodiments, constructing an acoustic model requires comprehensive and accurate measurement of the acoustic characteristics of the speaker diaphragm. This includes measuring acoustic parameters such as the sound pressure distribution and directivity characteristics on the diaphragm surface under different vibration frequencies, amplitudes, and phase conditions. For example, precise acoustic measurement devices such as octave analyzers and sound intensity probes can be used, combined with a professional acoustic laboratory environment to obtain high-quality data. Then, based on linear acoustic theory and nonlinear acoustic principles, combined with the physical structure and material properties of the diaphragm, numerical modeling methods are used to establish an acoustic model. During the model construction process, it is necessary to couple the nonlinear vibration equation of the diaphragm with the acoustic wave propagation equation, and by continuously adjusting the model parameters, make the output of the model match the actually measured acoustic data.

[0088] Based on the above technical solutions, the nonlinear model in the embodiments of the present application may include an electrical model, a mechanical model, and an acoustic model established by focusing on the nonlinear characteristics at different physical levels of the speaker, comprehensively and systematically covering various complex nonlinear factors during the operation of the speaker. Compared with a single model, it can capture each physical link in the generation of echoes more accurately, thereby improving the accuracy of the echo simulation signal.

[0089] Based on the above embodiments, in some possible embodiments, refer to Figure 11 , for Figure 9 a schematic flowchart of an implementation process of step S01 in a method for echo cancellation provided. As Figure 11 shown, the step S01 of using the nonlinear model to determine the echo simulation signal based on the first input signal may specifically include the following steps: S11: Determine the voice coil nonlinear parameters corresponding to the first input signal based on the electrical model.

[0090] In the embodiments of the present application, by using the electrical model to establish a connection between the electrical signal input to the speaker (i.e., the first input signal) and the nonlinear parameters of the speaker voice coil, the corresponding voice coil nonlinear parameters can be calculated based on the first input signal, laying a foundation for subsequent analysis of voice coil movement and diaphragm vibration.

[0091] When the voice coil of the speaker is working, its electrical characteristics are not linearly invariant. For example, when the amplitude of the input signal is large, the inductance and resistance of the voice coil may change nonlinearly due to factors such as temperature change and magnetic saturation. Traditional linear models often cannot accurately capture these changes, resulting in inaccurate prediction of the voice coil behavior. By determining the voice coil nonlinear parameters corresponding to the first input signal based on the electrical model, the state of the voice coil under actual working conditions can be more realistically reflected, and the accuracy of the finally obtained echo simulation signal can be improved.

[0092] In some embodiments, circuit analysis software or mathematical modeling tools can be used to fit the parameters of the electrical model based on the measured speaker parameters, forming a mathematical expression that can describe the relationship between the input signal and the nonlinear parameters of the voice coil. When the first input signal enters the system, by substituting the characteristics of the signal (such as frequency, amplitude, etc.) into the electrical model, the corresponding nonlinear parameters of the voice coil can be calculated. For example, a microprocessor is used to run a pre-established electrical model algorithm to perform real-time processing on the input signal, obtaining parameters such as the nonlinear inductance value and resistance value of the voice coil, providing accurate electrical inputs for subsequent mechanical and acoustic analyses.

[0093] S12: Determine the diaphragm nonlinear parameters corresponding to the voice coil nonlinear parameters based on the mechanical model.

[0094] In the embodiments of the present application, the movement of the voice coil and the vibration of the diaphragm are closely connected through a mechanical mechanism, but this connection is affected by various nonlinear factors. By using a pre-determined mechanical model, the known nonlinear parameters of the voice coil can be converted into the corresponding diaphragm nonlinear parameters, expressing the nonlinear characteristics generated by the diaphragm under the drive of the voice coil, providing a key mechanical basis for accurately simulating the sound signal emitted by the speaker subsequently.

[0095] The vibration of the diaphragm is not only affected by the direct drive of the voice coil, but also exhibits complex nonlinear behaviors due to factors such as its own material properties, shape, and installation method. Traditional simplified models often assume an ideal linear transmission relationship between the voice coil and the diaphragm, ignoring these nonlinear factors, resulting in inaccurate prediction of the vibration state of the diaphragm. By determining the diaphragm nonlinear parameters corresponding to the voice coil nonlinear parameters based on the mechanical model, the defects of the traditional model can be compensated, and the mechanical transmission process inside the speaker can be more realistically reflected.

[0096] S13: Determine the echo simulation signal corresponding to the diaphragm nonlinear parameters based on the acoustic model.

[0097] In the embodiments of the present application, the sound signal is generated based on the vibration of the diaphragm. By using a pre-determined acoustic model, the known nonlinear parameters of the diaphragm can be converted into the corresponding echo simulation signal. This process simulates the influence of the nonlinear characteristics of the diaphragm on the sound signal generated by the speaker, and can more realistically reflect the echo characteristics in the actual acoustic environment.

[0098] Based on the above technical solutions, the embodiments of the present application can gradually and accurately extract the feature parameters related to the echo from the first input signal and finally generate a high-precision echo simulation signal by applying the electrical, mechanical, and acoustic models in sequence. This step-by-step processing method can consider the nonlinear characteristics of each link of the speaker more comprehensively compared with a single model, thereby effectively improving the accuracy of echo simulation and providing a more reliable basis for subsequent echo cancellation.

[0099] Based on the above embodiments, in some possible embodiments, the nonlinear parameters of the voice coil of the speaker may include the voice coil magnetic force factor, and the nonlinear model may further include a first functional relationship, which is used to describe the nonlinear relationship between the voice coil magnetic force factor of the speaker, time, and the voice coil displacement of the speaker.

[0100] See Figure 12 , which is a schematic structural diagram of the first speaker provided by the embodiments of the present application. Among them, is the voice coil displacement, is the stroke of the voice coil movement, is the thickness of the upper magnetic conductive plate.

[0101] As Figure 12 shown, during the sound generation process of the speaker, when a voice coil in a magnetic gap with a magnetic field strength B of the speaker is passed through by a current I, the voice coil will move in the magnetic field due to the electromagnetic force F = BLI (L is the length of the voice coil wire). However, the distribution of the magnetic circuit in the magnetic field of the speaker is not uniform. The closer to the magnetic pole, the denser the magnetic circuit, and the farther from the magnetic pole, the sparser the magnetic circuit, which results in different magnitudes of magnetic forces received by the voice coil at different positions.

[0102] The voice coil magnetic force factor is the product of the magnetic field strength and the length of the voice coil wire, which represents the magnitude of the electromagnetic force received by the voice coil in the magnetic field under a unit current, that is, the voice coil magnetic force factor is BL. Due to the uneven distribution of the magnetic circuit in the magnetic field of the speaker, when the voice coil is in different positions, the voice coil magnetic force factor BL is not a fixed value, but will change nonlinearly as the voice coil moves, resulting in nonlinear distortion of the sound emitted by the speaker, which leads to a difference between the echo of the speaker picked up by the pickup and the simulated echo, and further leads to an unsatisfactory echo cancellation effect.

[0103] To solve this problem, the embodiments of the present application incorporate the first functional relationship used to describe the nonlinear relationship between the voice coil magnetic force factor, the voice coil displacement, and time into the nonlinear model, so that the nonlinear model can reflect the nonlinear characteristics of the voice coil magnetic force factor. Among them, the first functional relationship is a mathematical expression used to quantitatively describe how the voice coil magnetic force factor changes nonlinearly with the voice coil displacement and time. Specifically, when the voice coil vibrates in the speaker, the change in its position (displacement) and the passage of time will affect the force exerted by the magnetic field on the voice coil, thereby changing the voice coil magnetic force factor. For example, as the voice coil displacement increases, the magnetic force factor may show a non - linear increasing or decreasing trend due to the change in the magnetic field strength, and at the same time, the time factor may also introduce some dynamic nonlinear characteristics, such as delays or changes caused by the magnetic hysteresis effect of the material. By establishing the first functional relationship, the complex behavior of the voice coil during actual operation can be more accurately simulated.

[0104] In some possible embodiments, the process of establishing the first functional relationship may include the following steps: First, it is necessary to conduct detailed experimental tests on the voice coil of the speaker. Under different voice coil displacements and time conditions, measure the magnetic force factor of the voice coil. For example, a laser displacement sensor can be used to measure the displacement of the voice coil, while a fluxmeter or Hall effect sensor can be used to measure the magnetic field strength, and then calculate the magnetic force factor.

[0105] Then, based on the experimental data, use mathematical methods such as nonlinear regression analysis, neural network modeling, or polynomial fitting to construct a function model that can describe the relationship between the magnetic force factor of the voice coil, displacement, and time. During the model construction process, it is necessary to continuously adjust the parameters of the model to minimize the error between the output of the model and the actual measurement data.

[0106] For example, through polynomial fitting, the first functional relationship can be determined as:

[0107] where is the magnetic force factor of the voice coil, is the displacement of the voice coil, is the time, is the polynomial order.

[0108] In practical applications, when the displacement and time of the voice coil are known, the magnetic force factor of the voice coil can be calculated through this functional relationship, providing accurate input for subsequent non-linear model calculations. For example, in a real-time audio processing system, a microprocessor runs this function model, and based on the real-time displacement and time data of the voice coil, quickly calculates the change in the magnetic force factor, thereby achieving an accurate simulation of the speaker's behavior.

[0109] Based on the above technical solutions, the embodiments of the present application incorporate the non-linear relationship between the magnetic force factor of the voice coil, the displacement of the voice coil, and time into the model, enabling the model to reflect the non-linear characteristics of the magnetic force factor of the voice coil, and thus being able to more accurately capture the actual behavior of the voice coil during the dynamic working process, which helps to improve the accuracy of echo simulation, thereby enhancing the effect of echo cancellation and ensuring the clarity and naturalness of voice communication.

[0110] Based on the above embodiments, in some possible embodiments, the non-linear parameters of the speaker diaphragm may include the diaphragm vibration stiffness, and the non-linear model may further include a second functional relationship, which is used to describe the non-linear relationship between the diaphragm vibration stiffness of the speaker, time, the diaphragm vibration angular frequency of the speaker, and the voice coil displacement.

[0111] The vibration stiffness of the diaphragm refers to the ability of the diaphragm to resist deformation. Its physical essence is the comprehensive manifestation of the material elastic modulus and structural design, which directly affects the acoustic performance.

[0112] See Figure 13 , which is a schematic diagram of the structural relationship between a speaker diaphragm and a voice coil provided by an embodiment of the present application. Among them, x is the voice coil displacement, and F is the voice coil force.

[0113] As Figure 13 shown, during the sound generation process of the speaker, when the voice coil is energized and moves back and forth under the force in the magnetic field, it drives the diaphragm to produce corresponding vibrations. However, when the diaphragm vibrates to the maximum displacement, it generates elastic deformation, and the vibration becomes more difficult. At this time, the vibration stiffness of the diaphragm reaches the maximum. When the voice coil displacement is zero, the diaphragm does not produce elastic deformation, and the vibration is relatively easy. At this time, the vibration stiffness of the diaphragm is the smallest.

[0114] The vibration stiffness of the speaker diaphragm will change non-linearly with the change of the vibration position, resulting in non-linear distortion of the sound emitted by the speaker. This leads to a difference between the echo of the speaker picked up by the pickup and the simulated echo, and further results in an unsatisfactory echo cancellation effect.

[0115] To solve this problem, an embodiment of the present application introduces a second functional relationship in the non-linear model to describe the non-linear relationship between the vibration stiffness of the diaphragm, the angular frequency of the diaphragm vibration, the voice coil displacement, and time. The second functional relationship is a mathematical model used to quantitatively describe this dynamic change relationship. For example, when the angular frequency of the diaphragm vibration increases, the vibration stiffness of the diaphragm may increase or decrease due to the non-linear elastic characteristics of the material; the increase in the voice coil displacement may also cause a change in the vibration stiffness of the diaphragm because the voice coil displacement affects the tension and force state of the diaphragm. The time factor includes some dynamic changes, such as the relaxation effect or fatigue effect of the material, which further affects the vibration stiffness of the diaphragm. By establishing the second functional relationship, the stiffness change characteristics of the diaphragm during the actual working process can be more accurately simulated.

[0116] In some possible embodiments, the establishment process of the second functional relationship may include the following steps: First, conduct systematic experimental tests on the diaphragm of the speaker. Under different conditions of the angular frequency of the diaphragm vibration, the voice coil displacement, and time, measure the vibration stiffness of the diaphragm. For example, a laser vibrometer can be used to measure the vibration displacement of the diaphragm, and a dynamic mechanical analyzer can be combined to measure the stiffness change of the diaphragm. At the same time, record the displacement and time information of the voice coil to ensure the integrity and accuracy of the data.

[0117] Then, based on the experimental data, use mathematical methods such as non-linear regression analysis, neural network modeling, or polynomial fitting to construct a functional model that can describe the relationship between the vibration stiffness of the diaphragm, the angular frequency of the diaphragm vibration, the voice coil displacement, and time.

[0118] For example, the second functional relationship can be determined by polynomial fitting as follows:

[0119] wherein, is the vibration stiffness of the diaphragm, is the angular frequency of the diaphragm vibration, is the voice coil displacement, is the time, is the polynomial order.

[0120] In practical applications, when the angular frequency of the diaphragm vibration, the voice coil displacement, and the time are known, the vibration stiffness of the diaphragm can be calculated through the second functional relationship, providing accurate input for subsequent non-linear model calculations.

[0121] Based on the above technical solution, the embodiment of the present application can more accurately simulate the complex dynamic behavior of the diaphragm during actual operation by introducing the non-linear relationship between the vibration stiffness of the diaphragm, the angular frequency of the diaphragm vibration, the voice coil displacement, and the time into the non-linear model. Furthermore, it can capture the non-linear stiffness changes of the diaphragm under different vibration frequencies, voice coil displacements, and time conditions, thereby improving the accuracy of the echo simulation signal.

[0122] Based on the above embodiment, in some possible embodiments, the non-linear parameters of the diaphragm of the speaker may include the vibration damping of the diaphragm, and the non-linear model may further include a third functional relationship, which is used to describe the non-linear relationship between the vibration damping of the diaphragm of the speaker, the time, the angular frequency of the diaphragm vibration of the speaker, and the voice coil velocity.

[0123] The vibration damping of the diaphragm refers to the ability of the diaphragm to convert mechanical energy into heat energy due to internal friction of the material and structural deformation during vibration. It is a key parameter for energy dissipation of the speaker diaphragm during vibration, and it affects the attenuation characteristics of the diaphragm vibration.

[0124] Refer to Figure 14 , which is the structural schematic diagram of the second speaker provided by the embodiment of the present application. As Figure 14 shown, during the sound generation process of the speaker, when the voice coil is energized and moves back and forth under the action of the magnetic field, it drives the diaphragm to produce corresponding vibrations. However, when the diaphragm vibrates to the maximum displacement and undergoes elastic deformation, the vibration becomes more difficult, and at this time, the vibration stiffness of the diaphragm reaches the maximum. When the voice coil displacement is zero, the diaphragm does not undergo elastic deformation, and the vibration is relatively easy. At this time, the vibration stiffness of the diaphragm is the smallest.

[0125] The vibration damping of the diaphragm of a speaker will exhibit non - linear changes when vibrating up and down due to the asymmetry of the vibrating air environment, resulting in non - linear distortion in the sound emitted by the speaker. This causes a difference between the echo of the speaker picked up by the pickup and the simulated echo, thereby leading to an unsatisfactory echo cancellation effect.

[0126] To solve this problem, the embodiment of this application introduces a third functional relationship into the non - linear model. The third functional relationship is a mathematical model used to quantitatively describe the non - linear relationship between the vibration damping of the diaphragm, the angular frequency of the diaphragm vibration of the speaker, the voice - coil velocity, and time. For example, when the angular frequency of the diaphragm vibration increases, the damping may increase due to the visco - elastic properties of the material; an increase in the voice - coil velocity may also change the damping characteristics because the velocity of the voice - coil affects the air flow and frictional losses around the diaphragm. The time factor may introduce some dynamic changes, such as the relaxation effect of the material or changes in environmental conditions, which further affect the vibration damping of the diaphragm. By establishing the third functional relationship, the energy dissipation behavior of the diaphragm during actual operation can be more accurately simulated.

[0127] In some possible embodiments, the process of establishing the third functional relationship may include the following steps: First, conduct systematic experimental tests on the diaphragm of the speaker. Measure the vibration damping of the diaphragm under different conditions of the angular frequency of the diaphragm vibration, the voice - coil velocity, and time. For example, a laser vibrometer can be used to measure the vibration velocity and displacement of the diaphragm, and a dynamic mechanical analyzer can be combined to measure the damping characteristics of the diaphragm. At the same time, record the velocity and time information of the voice - coil to ensure the integrity and accuracy of the data.

[0128] Then, based on the experimental data, use mathematical methods such as non - linear regression analysis, neural network modeling, or polynomial fitting to construct a functional model that can describe the relationship between the vibration damping of the diaphragm, the angular frequency of the diaphragm vibration, the voice - coil velocity, and time.

[0129] For example, through polynomial fitting, the third functional relationship can be determined as:

[0130] Where, is the vibration damping of the diaphragm, is the angular frequency of the diaphragm vibration, is the voice - coil velocity, is time, is the polynomial order.

[0131] In practical applications, when the angular frequency of the diaphragm vibration, the voice - coil velocity, and time are known, the vibration damping of the diaphragm can be calculated through the third functional relationship, providing accurate input for subsequent non - linear model calculations.

[0132] Based on the above technical solution, in the embodiments of the present application, by introducing the non-linear relationship between the vibration damping of the diaphragm, the angular frequency of the diaphragm vibration, the voice coil velocity, and time into the non-linear model, the energy dissipation characteristics of the diaphragm under different vibration states and time conditions can be captured more accurately, which helps to improve the accuracy of the echo simulation signal.

[0133] Based on the above embodiments, in some possible embodiments, the non-linear parameters of the diaphragm of the speaker may include the vibration area of the diaphragm, and the non-linear model may further include a fourth functional relationship, which is used to describe the non-linear relationship between the vibration area of the diaphragm of the speaker, the angular frequency of the diaphragm vibration, and the voice coil displacement.

[0134] The vibration area of the diaphragm refers to the area of the part on the diaphragm that can vibrate freely and produce sound. This area determines how much sound pressure and frequency response the diaphragm can produce when affected by air flow or other external forces.

[0135] See Figure 15 , which is a schematic structural diagram of a speaker diaphragm provided by the embodiments of the present application. Among them, c1 is the diameter of the effective vibration surface of the diaphragm when the diaphragm vibrates upward, and c2 is the diameter of the effective vibration surface of the diaphragm when the diaphragm vibrates downward.

[0136] As Figure 15 shown, the surround is an important component connecting the diaphragm and the speaker frame, providing necessary support and elasticity. However, the presence of the surround causes an asymmetric distribution when the diaphragm vibrates up and down, resulting in a change in the effective vibration area of the diaphragm with different vibration directions. When the diaphragm moves downward, the effective vibration area becomes larger, and when the diaphragm moves upward, the effective vibration area becomes smaller.

[0137] During the sound generation process of the speaker, due to the asymmetric distribution of the surround when the diaphragm of the speaker vibrates up and down, the vibration area is inconsistent with the up and down vibration, resulting in non-linear distortion of the sound emitted by the speaker. This causes a difference between the echo picked up by the pick-up and the simulated echo, and further leads to an unsatisfactory echo cancellation effect.

[0138] To solve this problem, the embodiments of the present application introduce a fourth functional relationship into the non-linear model. The fourth functional relationship is a mathematical model used to quantitatively describe the non-linear relationship between the vibration area of the diaphragm of the speaker, the angular frequency of the diaphragm vibration, and the voice coil displacement. For example, when the angular frequency of the diaphragm vibration increases, the vibration mode of the diaphragm may change, resulting in an increase or decrease in the vibration area; an increase in the voice coil displacement may also cause a change in the vibration area of the diaphragm because the voice coil displacement affects the tension and vibration range of the diaphragm. By establishing the fourth functional relationship, the change characteristics of the vibration area of the diaphragm during the actual working process can be simulated more precisely.

[0139] In some possible embodiments, the process of establishing the fourth functional relationship may include the following steps: First, conduct a systematic experimental test on the diaphragm of the speaker. Under different angular frequencies of diaphragm vibration and voice coil displacements, measure the vibration area of the diaphragm. For example, laser scanning can be used to measure the vibration mode and displacement distribution of the diaphragm, and the vibration area of the diaphragm can be calculated by combining image processing techniques. At the same time, record the data of the angular frequency of diaphragm vibration and voice coil displacement to ensure the integrity and accuracy of the data.

[0140] Then, based on the experimental data, use mathematical methods such as nonlinear regression analysis, neural network modeling, or polynomial fitting to construct a functional model that can describe the relationship between the vibration area of the diaphragm, the angular frequency of diaphragm vibration, and the voice coil displacement.

[0141] For example, through polynomial fitting, the fourth functional relationship can be determined as:

[0142] where, is the vibration area of the diaphragm, is the angular frequency of diaphragm vibration, is the voice coil displacement, is the polynomial order.

[0143] In practical applications, when the angular frequency of diaphragm vibration and the voice coil displacement are known, the vibration area of the diaphragm can be calculated through the fourth functional relationship, providing accurate input for subsequent non - linear model calculations.

[0144] Based on the above technical solutions, the embodiments of the present application can more accurately simulate the actual vibration area change of the diaphragm under different vibration states by introducing the non - linear relationship between the vibration area of the diaphragm, the angular frequency of diaphragm vibration, and the voice coil displacement into the non - linear model, which helps to improve the accuracy of the echo simulation signal.

[0145] On the basis of the above embodiments, in some possible embodiments, the non - linear parameters of the diaphragm of the speaker may include the front - cavity stiffness, and the non - linear model may further include a fifth functional relationship, which is used to describe the non - linear relationship between the front - cavity stiffness of the speaker, time, and the voice coil displacement of the speaker.

[0146] The front - cavity stiffness of the speaker is a physical quantity that describes the ability of the closed air cavity in front of the diaphragm to resist volume change, and essentially is the equivalent spring stiffness formed by air elasticity.

[0147] Refer to Figure 16 , which is the structural schematic diagram of the third speaker provided by the embodiments of the present application. As Figure 16As shown, during the sound generation process of the speaker, when the diaphragm moves upward, it compresses the air in the front cavity, and the equivalent compliance decreases (the stiffness increases); when it moves downward, the volume of the front cavity increases, and the compliance increases (the stiffness decreases). That is, the vibration of the diaphragm squeezes the air in the front cavity to generate a reaction force, and its intensity is inversely proportional to the instantaneous volume.

[0148] During the sound generation process of the speaker, most of the space in the front cavity of the speaker is occupied when the speaker diaphragm moves upward, and it recovers when the diaphragm moves downward. This causes the stiffness of the front cavity of the speaker not to be a constant, resulting in nonlinear distortion in the sound emitted by the speaker, which leads to a difference between the echo picked up by the pickup and the simulated echo, and further leads to an unsatisfactory echo cancellation effect.

[0149] To solve this problem, a fifth functional relationship is introduced in the nonlinear model in the embodiments of the present application. The fifth functional relationship is a mathematical model used to quantitatively describe the nonlinear relationship between the stiffness of the front cavity of the speaker, the displacement of the voice coil of the speaker, and time. For example, when the displacement of the voice coil increases, the compression or expansion of the air in the front cavity will cause a change in the stiffness of the front cavity; the time factor can introduce some dynamic changes, such as the viscous effect or thermal effect of the air, which further affects the stiffness of the front cavity. By establishing the fifth functional relationship, the stiffness change characteristics of the air in the front cavity during the actual working process can be more accurately simulated.

[0150] In some possible embodiments, the process of establishing the fifth functional relationship may include the following steps: First, conduct systematic experimental tests on the front cavity of the speaker. Measure the stiffness of the front cavity under different voice coil displacements and time conditions. For example, a pressure sensor can be used to measure the pressure change in the front cavity, and the stiffness of the front cavity can be calculated in combination with the displacement data of the diaphragm. At the same time, record the voice coil displacement and time information to ensure the integrity and accuracy of the data.

[0151] Then, based on the experimental data, use mathematical methods such as nonlinear regression analysis, neural network modeling, or polynomial fitting to construct a function model that can describe the relationship between the stiffness of the front cavity, the displacement of the voice coil, and time.

[0152] For example, through polynomial fitting, the fifth functional relationship can be determined as:

[0153] Among them, is the stiffness of the front cavity, is the displacement of the voice coil, is time, is the polynomial order.

[0154] In practical applications, when the voice coil displacement and time are known, the front cavity stiffness can be calculated through the fifth functional relationship, providing accurate input for subsequent non-linear model calculations.

[0155] Based on the above technical solution, the embodiment of the present application can more accurately simulate the dynamic characteristics of the front cavity of the speaker by introducing the non-linear relationship between the front cavity stiffness, voice coil displacement, and time in the non-linear model. Furthermore, it can capture the changes in the front cavity stiffness under different voice coil displacements and time conditions, thereby improving the accuracy of the echo simulation signal.

[0156] On the basis of the above embodiment, in some possible embodiments, the diaphragm non-linear parameters of the speaker include acoustic damping, and the non-linear model further includes a sixth functional relationship, which is used to describe the non-linear relationship between the acoustic damping of the speaker and the voice coil velocity and time of the speaker.

[0157] Acoustic damping refers to the physical phenomenon in which the speaker system converts mechanical vibration energy into heat energy or other forms of dissipation through internal friction or external resistance during the sound propagation process, thereby suppressing the vibration amplitude and reducing noise.

[0158] See Figure 17 , which is a schematic diagram of the air pressure change during the sound generation process of a speaker provided by the embodiment of the present application. As Figure 17 shown, during the sound generation process of the speaker, when the diaphragm vibrates upward (compressing the air in the front cavity), the air needs to overcome a higher ambient air pressure when discharging outward; when vibrating downward (expanding the front cavity), the air is inhaled inward to form a negative pressure area. This pressure gradient difference causes the air flow resistance to show non-linear differences in the discharge and suction directions.

[0159] During the sound generation process of the speaker, when the speaker diaphragm vibrates up and down, the pressures encountered by the air in the front cavity of the speaker when discharging outward and inhaling inward are not the same, resulting in non-linear changes in the front cavity acoustic damping, causing non-linear distortion in the sound emitted by the speaker. This leads to a difference between the echo picked up by the pickup and the simulated echo, and further results in an unsatisfactory echo cancellation effect.

[0160] To solve this problem, the embodiment of the present application introduces a sixth functional relationship in the non-linear model. The sixth functional relationship is a mathematical model used to quantitatively describe the non-linear relationship between the acoustic damping of the speaker and the voice coil velocity and time of the speaker. For example, when the voice coil velocity increases, the flow resistance of the air around the diaphragm will increase, resulting in an increase in acoustic damping; the time factor can introduce some dynamic changes, such as the viscous effect or thermal effect of the air, which further affects the acoustic damping. By establishing the sixth functional relationship, the energy dissipation behavior of the diaphragm during actual operation can be more accurately simulated.

[0161] In some possible embodiments, the process of establishing the sixth functional relationship may include the following steps: First, conduct systematic experimental tests on the front cavity of the speaker. Measure the changes in acoustic damping under different voice coil displacements and time conditions. For example, a laser vibrometer can be used to measure the vibration velocity and displacement of the diaphragm, and in combination with a pressure sensor to measure the pressure changes of the air around the diaphragm, and then calculate the acoustic damping. At the same time, record the voice coil velocity and time information to ensure the integrity and accuracy of the data.

[0162] Then, based on the experimental data, use mathematical methods such as nonlinear regression analysis, neural network modeling, or polynomial fitting to construct a function model that can describe the relationship between acoustic damping, voice coil velocity, and time.

[0163] For example, through polynomial fitting, the sixth functional relationship can be determined as:

[0164] Where, is the acoustic damping, is the voice coil velocity, is the time, is the polynomial order.

[0165] In practical applications, when the voice coil velocity and time are known, the front cavity stiffness can be calculated through the sixth functional relationship, providing accurate input for subsequent nonlinear model calculations.

[0166] Based on the above technical solutions, the embodiments of the present application can more accurately simulate the energy dissipation characteristics of the speaker under different working conditions by introducing the nonlinear relationship between acoustic damping, voice coil velocity, and time into the nonlinear model, thereby helping to improve the accuracy of the echo simulation signal.

[0167] On the basis of the above embodiments, to improve the determination speed of the echo simulation signal, simplify the multi-physics coupling analysis of the speaker, and achieve efficient extraction of key parameters, in some possible embodiments, the electrical model, mechanical model, and acoustic model can be determined based on an equivalent circuit model. This equivalent circuit model can be a multi-physics model of electro-acoustics for the speaker.

[0168] See Figure 18 , which is a schematic diagram of an equivalent circuit model of a speaker provided by the embodiments of the present application.

[0169] As Figure 18 shown, the first input signal may include voltage and current , and the inductor and resistor are used to simulate the voice coil. Among them, the voice coil nonlinear parameters may include the voice coil magnetic force factor 。

[0170] On this basis, the electrical model used to describe the relationship between the first input signal and the characteristic parameters of the voice coil of the speaker can specifically be:

[0171] Among them, is the voltage, is the voice coil simulated inductance, is the diaphragm vibration angular frequency, is the voice coil displacement, is the current, is the time, is the simulated voice coil resistance, is the voice coil magnetic force factor.

[0172] As Figure 18 shown, the diaphragm non-linear parameters of the speaker can include the diaphragm vibration stiffness 、the diaphragm vibration damping and the diaphragm vibration area 。

[0173] On this basis, the mechanical model used to describe the relationship between the voice coil non-linear parameters and the diaphragm non-linear parameters of the speaker can specifically be:

[0174] Among them, is the vibration mass, is the diaphragm vibration damping, is the diaphragm vibration stiffness, is the diaphragm vibration angular frequency, is the voice coil displacement, is the voice coil velocity, is the sound pressure, is the current, is the time, is the simulated voice coil resistance, is the voice coil magnetic force factor, is the diaphragm vibration area.

[0175] On the basis of the above embodiments, in some possible embodiments, the diaphragm non-linear parameters of the speaker can also include the front cavity stiffness and the sound damping 。

[0176] On this basis, the acoustic model used to describe the relationship between the diaphragm non-linear parameters and the echo simulation signal can specifically be:

[0177] Among them, is the volume velocity, is the acoustic mass value, is the angular frequency of the diaphragm vibration, is the voice coil displacement, is the acoustic damping, is the front cavity stiffness, is the time, is the diaphragm vibration area, is the sound pressure.

[0178] Based on the above embodiments, in some possible embodiments, for the determination of the voice coil nonlinear parameters corresponding to the first input signal based on the electrical model mentioned in step S11, the process may specifically include: Link the electrical model and the first functional relationship , and determine the voice coil displacement considering the voice coil magnetic force factor.

[0179] For the determination of the diaphragm nonlinear parameters corresponding to the voice coil nonlinear parameters based on the mechanical model mentioned in step S12, the process may specifically include: Link the mechanical model and the second functional relationship , the third functional relationship , the fourth functional relationship , and determine the diaphragm vibration area considering the diaphragm nonlinear parameters.

[0180] For the determination of the echo simulation signal corresponding to the diaphragm nonlinear parameters based on the acoustic model mentioned in step S13, the process may specifically include: Link the acoustic model and the fifth functional relationship , the sixth functional relationship , and determine the output sound pressure considering the diaphragm nonlinear parameters.

[0181] Based on the above technical solutions, the embodiments of the present application use an equivalent circuit model to abstract the acoustic path (such as speaker - microphone coupling) as a circuit network (such as impedance, transmission delay) in echo cancellation, which helps to quantify the frequency response characteristics and phase delay of the echo path, thereby optimizing the parameter design of the adaptive filter and achieving precise echo cancellation. For example, by equating the speaker impedance and room reverberation to RLC components, the echo attenuation effects in different frequency bands can be quickly simulated, improving the algorithm convergence speed and system stability.

[0182] Based on the above embodiments, in some possible embodiments, to achieve more precise echo simulation signal processing and improve the echo cancellation effect, echo cancellation can be achieved through adaptive filtering and inverse processing. That is, as mentioned in step S02, in response to an operation behavior of the user exceeding a preset number of times, a quick task prompt interface is displayed, which can specifically include the following steps: S21: Perform adaptive filtering processing on the echo simulation signal to obtain a filtered signal.

[0183] In a voice communication and audio processing system, the characteristics of the echo signal and environmental conditions may change over time. For example, noise in the call environment, changes in the acoustic path, and the movement of the caller, etc. will all affect the characteristics of the echo signal. Since the parameters of traditional fixed filters cannot be adjusted in real time, it is difficult to cope with these changes, resulting in poor echo cancellation effects.

[0184] In the embodiments of the present application, performing adaptive filtering processing on the echo simulation signal is to dynamically adjust the parameters of the filter through an adaptive filtering algorithm, so that the output of the filter is as close as possible to the actual echo signal. The core of adaptive filtering lies in its ability to adjust the weight coefficients of the filter in real time to minimize the error between the filtered signal and the actual echo signal. In echo cancellation applications, the adaptive filter can effectively adapt to the time-varying characteristics of the signal and environmental changes by continuously learning the characteristics of the input signal, thereby improving the accuracy and efficiency of echo cancellation.

[0185] S22: Perform inverse processing on the filtered signal to obtain a first inverse signal.

[0186] In the embodiments of the present application, performing inverse processing on the filtered signal is to reverse the phase of the filtered signal by 180 degrees to obtain a first inverse signal. This step is achieved by multiplying each sample value of the signal by -1. The signal after inverse processing is symmetric with the original filtered signal in waveform but has the opposite phase. For example, if the amplitude of the original filtered signal is positive at a certain moment, the amplitude of the inverse signal will be negative at that moment, and the magnitude is the same.

[0187] The purpose of inverse processing is to generate a signal with a phase opposite to that of the actual echo signal, so as to achieve echo cancellation in subsequent addition processing. When the inverse signal is added to the actual echo signal, since they have opposite phases and the same amplitude, they can cancel each other out, thereby achieving the effect of echo cancellation.

[0188] S23: Perform addition processing on the first inverse signal and the second input signal to complete the echo cancellation processing.

[0189] In the embodiments of the present application, echo cancellation is achieved by adding the first anti-phase signal and the second input signal. The first anti-phase signal is a filtered signal after inverting processing, and its phase is opposite to that of the original echo signal. The second input signal is the signal picked up by the pickup, which contains the original voice signal and the echo signal. When these two signals are added, the anti-phase signal and the echo signal have opposite phases and the same amplitude, so they cancel each other out, and finally a pure signal containing only the original voice signal is obtained.

[0190] Based on the above technical solution, in the embodiments of the present application, the adaptive filter can dynamically adjust the filter parameters according to the characteristics of the actual echo signal to achieve more accurate processing of the echo simulation signal, and improve the accuracy and adaptability of the echo cancellation effect. The inverting processing can generate a signal with a phase opposite to that of the echo signal. When added to the original echo signal, it can effectively cancel the echo and enhance the efficiency and accuracy of echo cancellation.

[0191] On the basis of the above embodiments, in some possible embodiments, to improve the processing efficiency of echo cancellation, reduce the system load and latency, the second anti-phase signal can be obtained by directly performing inverting processing on the echo simulation signal, and then added to the second input signal to achieve echo cancellation. That is, as mentioned in step S02, echo cancellation processing is performed on the second input signal according to the echo simulation signal, and specifically it can be: S31: Perform inverting processing on the echo simulation signal to obtain the second anti-phase signal.

[0192] In the embodiments of the present application, performing inverting processing on the echo simulation signal is to reverse the phase of the echo simulation signal by 180 degrees to generate a signal with a phase opposite to that of the original echo signal, that is, the second anti-phase signal. This process is achieved by multiplying each sample value of the echo simulation signal by -1, so that the newly generated signal is symmetric with the original echo signal in waveform but has a completely opposite phase. The embodiments of the present application cancel the process of performing adaptive filtering processing on the echo simulation signal, which can further improve the processing efficiency of echo cancellation, reduce the system load and latency.

[0193] S32: Perform addition processing on the second anti-phase signal and the second input signal to complete the echo cancellation processing.

[0194] In the embodiments of the present application, the second anti-phase signal is obtained by performing inverting processing on the echo simulation signal, and its phase is opposite to that of the echo signal. The second input signal contains the user's voice and echo. When these two signals are added, the echo part cancels each other out due to the opposite phases, and the finally output signal mainly contains the user's voice.

[0195] Based on the above technical solution, the second anti-phase signal generated through anti-phase processing can be directly opposite to the echo signal in phase. When added, it can effectively cancel the echo, improve the echo cancellation effect, further improve the processing efficiency of echo cancellation, and reduce the system load and latency.

[0196] See Figure 19 and Figure 20 , Figure 19 which is a comparative effect diagram of a residual time-frequency spectrogram provided by an embodiment of the present application; Figure 20 which is a comparative effect diagram of a residual spectrogram provided by an embodiment of the present application. As Figure 19 shown in (1) therein, when taking the first input signal as a reference and not including the speaker nonlinear factor, the residual after linear echo cancellation is relatively large; as Figure 19 shown in (2) therein, when taking the echo simulation signal considering the speaker nonlinearity as a reference, the residual after linear echo cancellation is relatively small.

[0197] As Figure 20 shown, the residual after linear echo cancellation with the echo simulation signal considering the speaker nonlinearity as a reference is significantly smaller than the residual after linear echo cancellation with the first input signal as a reference. That is, from the perspective of the residual, the echo cancellation method provided by the embodiment of the present application has a better effect.

[0198] An embodiment of the present application further provides an echo cancellation device, which may include: An analog signal determination module, configured to determine an echo simulation signal based on the first input signal by using a nonlinear model; the nonlinear model is used to describe the nonlinear relationship between the echo simulation signal and the first input signal; the first input signal is the signal input to the speaker; An echo cancellation module, configured to perform echo cancellation processing on the second input signal according to the echo simulation signal; the second input signal is the signal picked up by the pickup.

[0199] Based on the above embodiment, in some possible embodiments, the nonlinear model includes an electrical model, a mechanical model, and an acoustic model; the electrical model is used to describe the relationship between the first input signal and the voice coil nonlinear parameters of the speaker; the mechanical model is used to describe the relationship between the voice coil nonlinear parameters and the diaphragm nonlinear parameters of the speaker; the acoustic model is used to describe the relationship between the diaphragm nonlinear parameters and the echo simulation signal.

[0200] Based on the above embodiment, in some possible embodiments, the analog signal determination module may specifically be configured to: Determine the voice coil nonlinear parameters corresponding to the first input signal based on the electrical model; Determine the diaphragm nonlinear parameters corresponding to the voice coil nonlinear parameters based on the mechanical model; Determine the echo simulation signal corresponding to the diaphragm nonlinear parameter based on the acoustic model.

[0201] Based on the above embodiments, in some possible embodiments, the voice coil nonlinear parameter of the speaker includes the voice coil magnetic force factor, and the nonlinear model further includes a first functional relationship for describing the nonlinear relationship between the voice coil magnetic force factor of the speaker, the voice coil displacement of the speaker, and time.

[0202] Based on the above embodiments, in some possible embodiments, the diaphragm nonlinear parameter of the speaker includes the diaphragm vibration stiffness, and the nonlinear model further includes a second functional relationship for describing the nonlinear relationship between the diaphragm vibration stiffness of the speaker, the diaphragm vibration angular frequency of the speaker, the voice coil displacement, and time.

[0203] Based on the above embodiments, in some possible embodiments, the diaphragm nonlinear parameter of the speaker includes the diaphragm vibration damping, and the nonlinear model further includes a third functional relationship for describing the nonlinear relationship between the diaphragm vibration damping of the speaker, the diaphragm vibration angular frequency of the speaker, the voice coil velocity, and time.

[0204] Based on the above embodiments, in some possible embodiments, the diaphragm nonlinear parameter of the speaker includes the diaphragm vibration area, and the nonlinear model further includes a fourth functional relationship for describing the nonlinear relationship between the diaphragm vibration area of the speaker, the diaphragm vibration angular frequency, and the voice coil displacement.

[0205] Based on the above embodiments, in some possible embodiments, the diaphragm nonlinear parameter of the speaker includes the front cavity stiffness, and the nonlinear model further includes a fifth functional relationship for describing the nonlinear relationship between the front cavity stiffness of the speaker, the voice coil displacement, and time.

[0206] Based on the above embodiments, in some possible embodiments, the diaphragm nonlinear parameter of the speaker includes the acoustic damping, and the nonlinear model further includes a sixth functional relationship for describing the nonlinear relationship between the acoustic damping of the speaker, the voice coil velocity, and time.

[0207] Based on the above embodiments, in some possible embodiments, the echo cancellation module can specifically be used for: Perform adaptive filtering processing on the echo simulation signal to obtain a filtered signal; Perform an inverting process on the filtered signal to obtain a first inverted signal; Perform an adding process on the first inverted signal and the second input signal to complete the echo cancellation process.

[0208] Based on the above embodiments, in some possible embodiments, the echo cancellation module may specifically be used for: Performing an inverting process on the echo simulation signal to obtain a second inverted signal; Performing an adding process on the second inverted signal and the second input signal to complete the echo cancellation process.

[0209] It should be understood that the term "module" in the embodiments of the present application may be implemented in the form of software and / or hardware, and no specific limitation is made thereto. For example, a "module" may be a software program, a hardware circuit, or a combination of the two that implements the above functions. The hardware circuit may include an application specific integrated circuit (ASIC), an electronic circuit, a processor (such as a shared processor, a dedicated processor, or a group of processors, etc.) for executing one or more software or firmware programs, and a memory, a combined logic circuit, and / or other suitable components that support the described functions.

[0210] Therefore, the modules of the various examples described in the embodiments of the present application can be implemented in electronic hardware, or in a combination of computer software and electronic hardware. Whether these functions are executed in a hardware or software manner depends on the specific application and design constraints of the technical solution. A person skilled in the art can use different methods to implement the described functions for each specific application, but such implementation should not be considered to exceed the scope of the present application.

[0211] The embodiments of the present application also provide a computer-readable storage medium, in which instructions are stored, and when the instructions are run on a computer, the computer is caused to execute the respective steps of the method for echo cancellation in the embodiments of the present application.

[0212] The embodiments of the present application also provide a computer program product containing instructions, and when the computer program product is run on a computer or any at least one processor, the computer is caused to execute the respective steps of the method for echo cancellation in the embodiments of the present application.

[0213] The embodiments of the present application also provide a chip, including a processor and a data interface, and the processor reads instructions stored on a memory through the data interface to perform the corresponding operations and / or processes of the method for echo cancellation provided by the present application.

[0214] Optionally, the chip further includes a memory, which is connected to the processor through a circuit or wire. The processor is configured to read and execute a computer program in the memory. Further optionally, the chip further includes a communication interface, and the processor is connected to the communication interface. The communication interface is configured to receive data and / or information to be processed. The processor obtains the data and / or information from the communication interface and processes the data and / or information. The communication interface may be an input / output interface.

[0215] The memory may be a read-only memory (ROM), other types of static storage devices that can store static information and instructions, a random access memory (RAM), or other types of dynamic storage devices that can store information and instructions. It may also be an electrically erasable programmable read-only memory (EEPROM), a compact disc read-only memory (CD-ROM), or other optical disc storage, optical disc storage (including compact discs, laser discs, optical discs, digital versatile discs, Blu-ray discs, etc.), magnetic disk storage media, or other magnetic storage devices. Or it may also be any other medium that can be used to carry or store desired program code in the form of instructions or data structures and can be accessed by a computer.

[0216] In the embodiments of the present application, "and / or" describes the association relationship of associated objects, indicating that three relationships may exist. For example, A and / or B may represent the case where A exists alone, A and B exist simultaneously, or B exists alone. Wherein A and B may be singular or plural. The character " / " generally represents an "or" relationship between the associated objects before and after. "At least one of the following" and its similar expressions refer to any combination of these items, including any combination of single items or plural items. For example, at least one of a, b, and c may represent: a, b, c, a - b, a - c, b - c, or a - b - c, where a, b, and c may be single or multiple.

[0217] Those of ordinary skill in the art can realize that the units and algorithm steps described in the embodiments disclosed herein can be implemented by a combination of electronic hardware, computer software, and electronic hardware. Whether these functions are executed in a hardware or software manner depends on the specific application and design constraints of the technical solution. A professional technician can use different methods for each specific application to implement the described functions, but such implementation should not be considered to exceed the scope of the present application.

[0218] Those skilled in the art can clearly understand that for the convenience and conciseness of description, the specific working processes of the systems, devices, and units described above can refer to the corresponding processes in the foregoing method embodiments and will not be elaborated herein.

[0219] In several embodiments provided in the present application, if any function is implemented in the form of a software functional unit and sold or used as an independent product, it can be stored in a computer-readable storage medium. Based on such an understanding, the technical solution of the present application, in essence, or the part that contributes to the prior art or a part of this technical solution can be embodied in the form of a software product. This computer software product is stored in a storage medium and includes several instructions for causing a computer device (which may be a personal computer, a server, or a network device, etc.) to execute all or part of the steps of the methods in various embodiments of the present application. The foregoing storage medium includes: various media such as USB flash drives, mobile hard disks, read-only memory (ROM), random access memory (RAM), magnetic disks, or optical discs that can store program codes.

[0220] The above content is only the specific implementation manner of the present application. Any person skilled in the art can easily think of changes or substitutions within the technical scope disclosed in the present application, and all of them should be covered by the protection scope of the present application. The protection scope of the present application shall be subject to the protection scope of the claims.

Claims

1. A method for echo cancellation, characterized in that, Comprising: Determining an echo simulation signal based on a first input signal by using a non-linear model; The non-linear model is used to describe the non-linear relationship between the echo simulation signal and the first input signal; The first input signal is a signal input to a speaker; Performing echo cancellation processing on a second input signal according to the echo simulation signal; The second input signal is a signal picked up by a pickup.

2. The method according to claim 1, wherein The non-linear model includes an electrical model, a mechanical model and an acoustic model; the electrical model is used to describe the relationship between the first input signal and the non-linear parameters of the voice coil of the speaker; the mechanical model is used to describe the relationship between the non-linear parameters of the voice coil and the non-linear parameters of the diaphragm of the speaker; The acoustic model is used to describe the relationship between the non-linear parameters of the diaphragm and the echo simulation signal.

3. The method according to claim 2, wherein The determining of the echo simulation signal based on the first input signal by using the non-linear model includes: Determining the non-linear parameters of the voice coil corresponding to the first input signal based on the electrical model; Determining the non-linear parameters of the diaphragm corresponding to the non-linear parameters of the voice coil based on the mechanical model; Determining the echo simulation signal corresponding to the non-linear parameters of the diaphragm based on the acoustic model.

4. The method according to claim 2 or 3, characterized in that, The non-linear parameters of the voice coil of the speaker include a voice coil magnetic force factor, and the non-linear model further includes a first functional relationship, and the first functional relationship is used to describe the non-linear relationship between the voice coil magnetic force factor of the speaker, time, and the voice coil displacement of the speaker.

5. The method according to claim 2 or 3, characterized in that, The non-linear parameters of the diaphragm of the speaker include a diaphragm vibration stiffness, and the non-linear model further includes a second functional relationship, and the second functional relationship is used to describe the non-linear relationship between the diaphragm vibration stiffness of the speaker, time, the diaphragm vibration angular frequency of the speaker, and the voice coil displacement.

6. The method according to claim 2 or 3, characterized in that The non-linear parameters of the diaphragm of the speaker include a diaphragm vibration damping, and the non-linear model further includes a third functional relationship, and the third functional relationship is used to describe the non-linear relationship between the diaphragm vibration damping of the speaker, time, the diaphragm vibration angular frequency of the speaker, and the voice coil velocity.

7. The method according to claim 2 or 3, characterized in that, The non-linear parameters of the diaphragm of the speaker include a diaphragm vibration area, and the non-linear model further includes a fourth functional relationship, and the fourth functional relationship is used to describe the non-linear relationship between the diaphragm vibration area of the speaker, the diaphragm vibration angular frequency of the speaker, and the voice coil displacement.

8. The method according to claim 2 or 3, characterized in that, The non-linear parameters of the diaphragm of the speaker include a front cavity stiffness, and the non-linear model further includes a fifth functional relationship, and the fifth functional relationship is used to describe the non-linear relationship between the front cavity stiffness of the speaker, time, and the voice coil displacement of the speaker.

9. The method according to claim 2 or 3, characterized in that, The non-linear parameters of the diaphragm of the speaker include an acoustic damping, and the non-linear model further includes a sixth functional relationship, and the sixth functional relationship is used to describe the non-linear relationship between the acoustic damping of the speaker, time, and the voice coil velocity of the speaker.

10. The method according to any one of claims 1 to 3, characterized in that, The performing of the echo cancellation processing on the second input signal according to the echo simulation signal includes: Performing adaptive filtering processing on the echo simulation signal to obtain a filtered signal; Performing an inverting process on the filtered signal to obtain a first inverted signal; Perform an addition process on the first anti-phase signal and the second input signal to complete the echo cancellation process.

11. The method according to any one of claims 1 to 3, characterized in that, The echo cancellation process for the second input signal according to the echo simulation signal includes: Perform an anti-phase process on the echo simulation signal to obtain a second anti-phase signal; Perform an addition process on the second anti-phase signal and the second input signal to complete the echo cancellation process.

12. An electronic device, characterized in that, Includes: Processor; Memory; A computer program is stored in the memory, and when the computer program is executed, the electronic device is caused to execute the method according to any one of claims 1-11.

13. A computer-readable storage medium, characterized in that, The computer-readable storage medium includes a stored program, wherein when the program runs, it controls the device where the computer-readable storage medium is located to execute the method according to any one of claims 1-11.

14. A program product, characterized in that, The program product stores a program, and when the program is run by an information processing device, the information processing device is caused to execute the method according to any one of claims 1-11.

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