A mobile terminal dual-speaker stereo output system of different frequency sound waves

Abstract

The application discloses a mobile terminal double-speaker different-frequency sound wave stereo output system. The system comprises a master audio processing module arranged in the mobile terminal and a slave audio processing module in communication connection with the master module; the master module outputs a first sound wave of a first frequency in response to an application instruction, and the slave module outputs a second sound wave of a second frequency based on a control instruction, so that synchronous stereo output of two-way different-frequency sound waves is realized to drive at least two loudspeakers. The slave module can be integrated into an external loudspeaker and can be realized by means of an analog modem circuit, a special hardware synchronous link or a highly integrated multi-channel chip scheme. The application has the beneficial effect that the generation and synchronous output of two-way different-frequency sound waves can be realized economically and reliably by flexible software and hardware collaborative design without changing the hardware architecture of the mobile terminal, and a feasible hardware basis is provided for sound wave application based on the binaural audio difference effect.

Description

Technical Field

[0001] This invention relates to the field of audio processing and output technology, and in particular to a mobile terminal dual-speaker heterogeneous sound wave stereo output system. Background Technology

[0002] Currently, mobile terminal devices, such as smartphones and tablets, typically base their audio output systems on a highly integrated audio processing core in terms of hardware architecture. This core is usually an audio codec or a dedicated sound effects integrated circuit, responsible for processing all digital audio signals that need to be played within the device and converting them into analog sound wave signals for output. Whether through the device's built-in stereo speakers or wired or wireless headphones, although the left and right channel signals output by this system can be adjusted in terms of volume and phase to achieve a stereo effect, their fundamental frequency components are synchronized and identical in traditional operating modes. In other words, the standard audio hardware design of existing mobile terminals is intended to reproduce a pre-recorded, identical stereo audio stream, rather than to generate and independently control two original sound wave signals with different fundamental frequencies in real time. Although complex software algorithms can simulate certain acoustic effects in the audio stream, the system is limited by the physical architecture of a single audio processing core, preventing it from generating and outputting two completely independent original sound wave signals with different fundamental frequencies at the same time.

[0003] However, with the continuous expansion of audio application scenarios, especially in emerging fields such as sound wave therapy and immersive interactive experiences based on the principle of "binaural audio difference," there is an urgent need for mobile terminals to be able to drive at least two speakers (such as left and right speakers) to synchronously play two independent sound waves with specific frequency differences. For example, in order to stimulate specific brainwave activity in a user's brain, it may be necessary for the left ear to receive a 210Hz sound wave and the right ear to receive a 200Hz sound wave. The standard audio hardware of existing mobile terminals cannot directly meet this requirement. The fundamental contradiction lies in the fact that a single audio processing core cannot be configured simultaneously to generate two different fundamental frequency signals and allocate them to different output channels. To achieve this function, it is currently often necessary to rely on external professional audio equipment, which not only increases the cost and complexity for users but also completely negates the inherent portability and widespread availability advantages of mobile terminals. Therefore, on existing general-purpose mobile terminal platforms, there is a lack of a technical solution that can effectively overcome the limitations of their built-in audio hardware to achieve economical and convenient synchronous stereo output of dual-channel different frequency sound waves.

[0004] Therefore, given the technical bottleneck that existing mobile terminal audio systems cannot support the synchronous generation and output of dual-channel different frequency sound waves at the hardware level, there is an urgent need to propose an innovative system architecture solution. Summary of the Invention

[0005] To improve upon existing methods and systems, a mobile terminal dual-speaker heterogeneous sound wave stereo output system is provided, comprising: The main audio processing module, located inside the mobile terminal, is configured to respond to application commands and output a first sound wave at a first frequency. The subordinate audio processing module communicates with the main audio processing module and is configured to receive control commands and output a second sound wave at a second frequency based on the control commands. The first frequency is different from the second frequency, and the first sound wave and the second sound wave are configured to be output simultaneously to drive at least two speakers.

[0006] In some embodiments, at least two speakers include a built-in speaker and an external speaker. The built-in speaker is disposed inside the mobile terminal, and the subordinate audio processing module is connected to the external speaker. A first sound wave is output by the built-in speaker of the mobile terminal, and a second sound wave is output by the external speaker.

[0007] In some embodiments, the application instructions include a reference frequency parameter and a frequency difference parameter; the main audio processing module is configured to generate a first sound wave of a first frequency in response to the reference frequency parameter; the control instructions include the reference frequency parameter and the frequency difference parameter, and the subordinate audio processing module is configured to calculate a second frequency based on the reference frequency parameter and the frequency difference parameter.

[0008] In some embodiments, the main audio processing module is configured to ignore the frequency difference parameter in the application instructions and operate only based on the reference frequency parameter.

[0009] In some embodiments, the subordinate audio processing module is an analog modulation and demodulation circuit; the main audio processing module is configured to output a composite signal containing a carrier signal modulated with a target difference frequency to the analog modulation and demodulation circuit; the analog modulation and demodulation circuit is used to demodulate the composite signal to obtain a second sound wave at a second frequency and drive an external speaker.

[0010] In some embodiments, the frequency of the carrier signal is higher than 20kHz, and the target frequency difference is the difference between the first frequency and the second frequency.

[0011] In some embodiments, the system further includes a dedicated hardware synchronization link; the master audio processing module is configured to send a synchronization trigger signal to the slave audio processing module via the hardware synchronization link when it begins to output the first sound wave; the slave audio processing module is configured to start outputting the second sound wave in response to the synchronization trigger signal.

[0012] In some embodiments, the subordinate audio processing module is integrated into the true wireless stereo earphones; the main audio processing module is configured to encode the first sound wave and the second sound wave into a left channel audio stream and a right channel audio stream, respectively, and send them to the true wireless stereo earphones for playback via Bluetooth protocol.

[0013] In some embodiments, the subordinate audio processing module includes a passive demodulation circuit integrated in the microphone circuitry of the wired headphones; the mobile terminal is configured to modulate the second sound wave onto a carrier wave and output it through the microphone pin of the audio interface; the passive demodulation circuit is used to demodulate the signal from the microphone pin to recover the second sound wave and output it through a sound unit in the wired headphones.

[0014] In some embodiments, the main audio processing module and the subordinate audio processing module are integrated into the same hardware chip, which contains at least two independent digital signal processing channels for generating a first sound wave and a second sound wave, respectively.

[0015] The technical solution of this application is a dual-speaker stereo sound wave output system for mobile terminals. It involves setting up a main audio processing module within the mobile terminal and introducing a subordinate audio processing module that communicates with it, working together to output two sound waves of different frequencies. Specifically, the main module outputs a sound wave of the first frequency in response to application commands, while the subordinate module outputs a sound wave of the second frequency based on received control commands, thereby achieving synchronous output of the two different frequency sound waves to drive at least two speakers. Its advantages lie in its innovative breakthrough of the hardware limitation of existing mobile terminals, which cannot simultaneously generate dual different frequency sound waves due to a single audio processing core. It requires no modification to the main hardware architecture of mobile devices such as smartphones. Through flexible and low-cost modular collaborative design, it can economically and reliably achieve high-quality synchronous stereo sound wave output, providing a convenient and efficient hardware implementation foundation for various applications based on binaural audio difference (such as sound wave sleep aids and brainwave synchronization), significantly expanding the audio application scenarios of mobile terminals. Attached Figure Description

[0016] Figure 1 This is a system architecture diagram of the mobile terminal dual-speaker heterogeneous sound wave stereo output system proposed in this invention.

[0017] In the diagram: 1. Mobile terminal; 2. Main audio processing module; 3. Subordinate audio processing module; 4. Built-in speaker; 5. External speaker. Detailed Implementation

[0018] The following description is intended to disclose the invention and enable those skilled in the art to implement it. The preferred embodiments described below are merely examples, and other obvious variations will occur to those skilled in the art.

[0019] Firstly, this application provides a mobile terminal dual-speaker heterogeneous sound wave stereo output system, such as... Figure 1 As shown, it includes: The main audio processing module 2, located inside the mobile terminal 1, is configured to output a first sound wave of a first frequency in response to an application command. The subordinate audio processing module 3 is communicatively connected to the main audio processing module 2 and is configured to receive control commands and output a second sound wave at a second frequency based on the control commands. The first frequency is different from the second frequency, and the first sound wave and the second sound wave are configured to be output simultaneously to drive at least two speakers.

[0020] This embodiment provides a specific implementation of a dual-speaker stereo sound wave output system for a mobile terminal. This system aims to overcome the technical limitation of a general-purpose mobile terminal 1, which, due to its single built-in audio processing chip, cannot drive the left and right speakers to simultaneously play different fundamental frequency sound waves. Its core architecture consists of two cooperating modules: a main audio processing module 2 and a subordinate audio processing module 3. The main audio processing module 2 is the original audio hardware core inside the mobile terminal 1, typically referring to the audio codec or sound effect integrated circuit on the phone's motherboard. In this solution, this module is configured as the system's main controller and one sound wave generator. This is to fully utilize the existing, mature audio processing capabilities of the mobile terminal 1, avoiding modifications to the phone's core architecture. Its specific function is to respond to instructions from the upper-layer application, parse the first frequency parameter contained within, and drive the mobile terminal 1's built-in main speaker or earpiece to output the first sound wave of the corresponding frequency. For example, when the application instruction requests the left channel to output a 210Hz sound wave, the main audio processing module 2 is responsible for accurately generating and outputting this frequency signal.

[0021] The slave audio processing module 3 is an innovative hardware addition introduced in this solution to achieve cross-frequency output. This module is designed as an independently operating audio signal generation unit, typically centered around an additional sound effect integrated circuit, and can be integrated into a specially designed external speaker 5, wired headphones, or true wireless stereo headphones. It establishes a communication connection with the main audio processing module 2 through a hardware interface provided by the mobile terminal 1, such as a Bluetooth wireless link, a USB data channel, or a microphone line in the audio interface. The fundamental reason for setting up this module is to overcome the physical limitation of the main audio processing module 2 in simultaneously generating two different fundamental frequency signals, allowing it to take on the task of generating a second frequency sound wave. Its specific function is to receive control commands from the main module or directly from the application, which contain parameter information for the second frequency, and then drive the integrated speaker to output a second sound wave at the second frequency. For example, while the main module outputs a 210Hz sound wave, the slave module can output a 200Hz sound wave according to the command, thus creating a 10Hz frequency difference.

[0022] The collaborative workflow of the two modules constitutes the complete implementation method of the system. A typical technical path is as follows: The mobile application first generates a set of parameter instructions containing a reference frequency and a frequency difference. This set of instructions is simultaneously sent to the main audio processing module 2 and the slave audio processing module 3. The main module directly reads the reference frequency parameters and outputs the corresponding sound wave. Upon receiving the instruction, the slave module's integrated microprocessor immediately performs real-time calculations, subtracting the frequency difference from the reference frequency to obtain the target frequency, and then controls its integrated sound effect IC to generate a sound wave of that frequency. To ensure that the two sound waves can be played in strict synchronization, the system can use software timestamp synchronization or hardware trigger signals. For example, the application carries a precise start timestamp when sending instructions, and the high-precision clocks inside the two modules align with the playback start point accordingly; or, at the moment playback begins, the main module sends a hardware level trigger signal to the slave module through an additional general-purpose input / output pin to achieve microsecond-level precise synchronization.

[0023] Overall, the system described in this embodiment constructs a master-slave collaborative, distributed audio output architecture. It creatively bypasses the hardware bottleneck of a single audio chip by introducing a flexibly configurable and easily integrated external slave processing unit while retaining the original core audio functions of the mobile terminal 1. The entire system's working logic is clear: commands are uniformly initiated by the application layer, the master module is responsible for one output, the slave module calculates and generates the other output, and finally, through a precise synchronization mechanism, it ensures that two sound waves of different frequencies can be emitted simultaneously and stably from at least two physically separated speakers.

[0024] This technical solution brings several beneficial effects. Its most direct and significant effect is the successful output of dual-channel heterogeneous sound waves on a standard mobile terminal, providing a portable hardware platform for advanced acoustic applications such as binaural audio difference. The derivation of these beneficial effects stems from solving a core problem: the obstacle of traditional solutions lies in the inability of a single chip to handle two independent fundamental frequencies simultaneously. This solution fundamentally resolves this contradiction by adding a dedicated slave chip to share this task. This design avoids the need for custom-designed, expensive, and complex new multi-core audio chips for mobile phones, significantly reducing the cost and barrier to entry for technical implementation, allowing ordinary smartphone users to experience related functions without specialized equipment. Secondly, the architecture has good scalability and adaptability. The slave module can be externally mounted or integrated into accessories such as headphones, giving the solution flexible product forms. Finally, through synchronized control combining hardware and software, the time consistency of the dual-channel sound wave output is ensured. This is crucial for applications that rely on precise frequency differences to produce specific physiological or psychological effects, ultimately achieving efficient, reliable, and low-cost mobile heterogeneous sound wave stereo output.

[0025] In some embodiments, at least two speakers include a built-in speaker 4 and an external speaker 5. The built-in speaker is disposed inside the mobile terminal 1, and the subordinate audio processing module 3 is connected to the external speaker 5. A first sound wave is output by the built-in speaker 4 of the mobile terminal 1, and a second sound wave is output by the external speaker 5.

[0026] This embodiment details a dual-speaker stereo sound output system for a mobile terminal 1. Its core lies in introducing an independent external speaker 5, integrated with a subordinate audio processing module 3, which works in conjunction with the mobile terminal 1's built-in speaker 4 to form a complete dual-channel stereo output hardware system. In this implementation, the main audio processing module 2 inside the mobile terminal 1, i.e., the phone's own audio chip, is responsible for responding to application commands, generating and driving the built-in speaker 4 to output a sound wave of the first frequency. Simultaneously, an independent subordinate audio processing module 3 is pre-integrated within the external speaker 5. This module establishes a communication connection with the phone via wireless or wired means. When the application on the phone needs to output a pair of sound waves with a specific frequency difference, it sends control commands simultaneously or sequentially to the main audio processing module 2 and the subordinate module within the external speaker 5. The main module directly interprets the portion of the command related to the first frequency and drives the phone's built-in speaker to emit sound. Upon receiving an instruction, the slave module integrated within the external speaker 5 uses its internal microprocessor to analyze the parameters required to generate the second frequency sound wave. It then controls its integrated dedicated audio chip to generate a sound wave of the corresponding frequency, which is ultimately output through the diaphragm of the external speaker 5. In this way, the first sound wave is emitted from the phone itself, and the second sound wave is emitted from the external speaker 5. The two are spatially separated and have different frequencies, thus achieving the output of stereo, heterogeneous sound waves.

[0027] This technological solution can be implemented in various product forms. A typical example is a specially designed smart speaker or sleep aid device. Users simply pair their mobile phone with this external device via Bluetooth or an audio cable and run a specific application on their phone. The application instructs the phone's speaker to play one frequency (e.g., 210Hz) while simultaneously instructing the external smart speaker to play another frequency (e.g., 200Hz). The user, positioned between the two sound sources, can experience the "binaural audio difference" effect. Another example is adapting to the existing architecture of mainstream true wireless stereo headphones. In this case, the function of the subordinate audio processing module 3 can be implemented by the existing chip inside the headphones through a firmware upgrade. The mobile application pre-generates the two different frequency sound wave signals into different audio streams and sends them to the left and right earpieces of the headphones via Bluetooth. At this time, the left earpiece acts as a "built-in speaker" outputting the first frequency, while the right earpiece acts as an "external speaker" outputting the second frequency, thus achieving the goal of receiving different frequency sound waves with both ears. A more integrated form is to design a portable card-shaped device with its own battery, audio processing chip, and speaker. It connects to a mobile phone via Bluetooth Low Energy. The phone's speaker plays one frequency, while the card device plays another. Both can be flexibly placed next to the user's pillow, making it very convenient to use.

[0028] The benefits of this embodiment are significant and multifaceted. Its fundamental value lies in its ingenious and low-cost breakthrough of the technical barrier that prevents general-purpose smartphones from simultaneously playing different fundamental frequency sound waves using their two speakers, a limitation imposed by hardware constraints. The derivation of these beneficial effects stems directly from its architectural design: since the phone's internal single audio chip cannot process two independent frequencies simultaneously, an external speaker with its own processing capabilities is introduced to share half the task. This "internal-external" distributed processing architecture requires no hardware modifications to the phone or the customization of expensive dedicated chips, greatly reducing the difficulty of technical implementation and commercialization costs. Simultaneously, the form of the external device (such as a dedicated speaker, general-purpose headphones, or portable device) provides the solution with high flexibility and user-friendliness. Users do not need to change their phones; simply by equipping an external device and combining it with a specific application, they can achieve sound wave output effects that previously required professional equipment. This allows applications based on principles such as binaural rhythm for sleep aids and concentration training to be easily popularized on mass-market consumer mobile platforms. Finally, the solution ensures the quality and synchronization of the sound output. The two sound sources are driven by independent hardware units, the signals are pure, and through a precise communication protocol and synchronization algorithm, the two sound waves can be highly aligned in time, which is crucial for applications that rely on precise frequency differences to produce the expected physiological or psychological effects.

[0029] In some embodiments, the application instruction includes a reference frequency parameter and a frequency difference parameter; the main audio processing module 2 is configured to generate a first sound wave of a first frequency in response to the reference frequency parameter; the control instruction includes the reference frequency parameter and the frequency difference parameter, and the subordinate audio processing module 3 is configured to calculate a second frequency based on the reference frequency parameter and the frequency difference parameter.

[0030] This embodiment further refines the core instruction logic and control method for the collaborative operation between the main audio processing module 2 and the subordinate audio processing module 3 in the dual-speaker heterogeneous sound wave stereo output system of mobile terminal 1. The key technology lies in encapsulating the complete information required to generate two different frequency sound waves into a set of structured parameter instructions. Specifically, when the application initiates a request, it does not directly specify two independent frequency values, but generates a composite instruction set containing a "reference frequency parameter" and a "frequency difference parameter." This set of instructions is sent synchronously or time-divisionally to the two processing modules within the system.

[0031] The main audio processing module 2, i.e., the audio chip inside the mobile terminal 1, is pre-configured to a simple operating mode: it only needs to focus on and respond to the reference frequency parameter in the instruction. For example, when the instruction includes the parameters "A=210" and "B=10", the main module will ignore "B" which represents the frequency difference and directly read "A=210", and then drive the built-in speaker 4 to stably output the first sound wave of 210Hz. This design allows the main module to avoid performing additional mathematical calculations, maintaining the simplicity and efficiency of its processing path, and making full use of its essential function as a general audio output core.

[0032] Meanwhile, the subordinate audio processing module 3, whether integrated into the external speaker 5, headphones, or other accessories, is responsible for calculation and generation. The control commands it receives also include the reference frequency parameter and the frequency difference parameter. The internal computing unit (such as a microcontroller) actively performs calculations on these two parameters. Continuing with the above parameters as an example, the computing unit performs an "AB" operation, that is, 210 minus 10, resulting in 200. This calculation result "200" is then determined as the second frequency to be generated. Subsequently, the audio generation circuit within the subordinate module drives the connected speaker to output a 200Hz second sound wave based on this calculated frequency. Throughout the process, the subordinate module exhibits a certain degree of "intelligence," autonomously deriving the final execution target based on the instruction intent.

[0033] This parameterized instruction approach can be implemented in several specific application scenarios. In sleep aid applications, the user interface might only offer a "soothing intensity" slider and a "target brainwave frequency" selection. The background application converts the "target brainwave frequency" (e.g., 10Hz alpha waves) into a frequency difference parameter B, combines it with a fixed, comfortable auditory reference frequency A (e.g., 200Hz), and packages it into an instruction for transmission. The phone's speaker plays a 200Hz sound wave, while a dedicated sleep aid lamp's built-in slave module calculates and plays a 190Hz sound wave, thus generating the desired 10Hz difference frequency effect. In focus training games, the application can dynamically adjust parameters based on level difficulty. Higher difficulty levels might set a smaller frequency difference (e.g., 5Hz), requiring more subtle perception from the user; in this case, parameter B in the instruction is set to 5, the main module outputs the reference frequency, and the slave module outputs the sound wave after subtracting 5 from the reference frequency, creating a corresponding auditory challenge. In addition, in professional acoustic wave research tools and apps, researchers can accurately input the reference frequency A and the desired difference frequency B. Through this parameterized instruction, the system can conveniently and repeatedly generate any desired dual-frequency acoustic wave combination for experiments and effect verification.

[0034] The technical solution defined in this embodiment brings multiple clear and profound benefits. Its core advantage lies in achieving great flexibility and efficiency in system control through ingenious division of responsibilities and parametric design. The derivation of these beneficial effects directly stems from its "reference + offset" instruction model. First, it greatly simplifies the development logic of applications and the user interface. Application developers and end users do not need to concern themselves with the specific values ​​of two complex frequencies; they only need to focus on a main sound (reference frequency) and the desired effect intensity (frequency difference). The system automatically generates the frequency pair at the lower level, reducing the barrier to entry. Second, this design ensures the simplicity and uniformity of instructions. Regardless of the frequency combination to be generated, the structure of the uplink instruction is fixed (A, B), which simplifies the communication protocol, reduces the amount of data that needs to be transmitted between modules, and improves the system's reliability and response speed. For the slave audio processing module 3, it does not need to store a large frequency correspondence table; it only needs to have basic subtraction or addition capabilities. This reduces the hardware computing resource requirements of the slave module, contributing to cost control and miniaturization. Finally, this solution lays a solid foundation for dynamic adjustment and personalized customization of functions. Since the final effect is dynamically determined by two parameters, the application can dynamically adjust the value of A or B at any time based on real-time user feedback, environmental factors, or preset programs, thereby changing the characteristics of the output sound wave in real time and providing strong technical support for achieving an adaptive and personalized sound wave experience.

[0035] In some embodiments, the main audio processing module 2 is configured to ignore the frequency difference parameter in the application instructions and operate only based on the reference frequency parameter.

[0036] This embodiment details a specific processing strategy for control commands in the main audio processing module 2 within the dual-speaker heterogeneous sound wave stereo output system of mobile terminal 1. The core technology lies in the "selective ignoring" behavior logic employed by the main audio processing module 2 when parsing complex commands from the application. Specifically, when the application issues a set of commands containing both a reference frequency parameter and a frequency difference parameter, the main audio processing module 2, located within mobile terminal 1, actively filters out the frequency difference-related parameters. Its internal logic is designed to extract and respond only to the reference frequency parameter in the command. For example, when the command content is "reference frequency 210Hz, frequency difference 10Hz," the main module's parsing unit will consciously ignore the "10Hz" information as if it doesn't exist, concentrating all configuration and driving resources on the "210Hz" reference parameter, and then accurately generating and outputting a 210Hz first sound wave. This design ensures the main module's function is pure and its objective is singular.

[0037] This "ignore the difference, only respond to the baseline" technical setting can be implemented through several specific product forms and interaction logics. In a consumer-grade sleep aid audio application, a user might select the "Forest Drizzle" scene through a simple interface. The acoustic model parameters set for this scene in the application's backend are "baseline 200Hz, difference frequency 8Hz". When these parameters are sent to the phone's main audio processing chip, the chip's driver or firmware, according to the design of this embodiment, will automatically mask the difference frequency information, only reading 200Hz and driving the phone's speaker to play a stable, pure 200Hz background sound. At this time, no matter how the user adjusts the sleep aid intensity (actually adjusting the difference frequency parameter from 8Hz to 5Hz or 12Hz), the basic tone frequency emitted by the phone's main speaker remains unchanged; the changing instructions are only sent to the external slave module for calculation and execution. In another example, consider an interactive system for cognitive training that includes a specially designed wireless earbud with a built-in slave module. The mobile application generates instructions, such as "baseline frequency 440Hz (standard A tone), difference frequency 15Hz". The phone's main audio processing module 2 ignores the "15Hz" and continuously emits 440Hz sound through the bottom speaker. Meanwhile, the slave module in the wireless earbuds receives the complete parameters, calculates, and plays the 425Hz sound. This design ensures stable and predictable sound output from the phone during interaction, with all dynamic effects based on frequency difference changes implemented entirely by external accessories, creating a clear functional division in terms of user experience. Furthermore, in terms of industrial design, this logic can be embedded in the driver chip of the main audio processing module 2, implementing parameter filtering through hardware description languages ​​or dedicated microcode. This ensures consistency in behavior at the hardware level and prevents the main module from incorrectly attempting frequency difference calculation tasks it is not suited for due to accidental software modifications.

[0038] The explicit stipulation that the main audio processing module 2 ignores frequency difference parameters brings direct and crucial benefits. The derivation of these benefits stems from considerations of system robustness, complexity, and clear responsibilities. The primary effect is a significant simplification of the design complexity and software burden of the main audio processing module 2, i.e., the original core audio hardware of the mobile terminal 1. Since there is no need to parse, verify, or process frequency difference parameters, the module's instruction set can be more streamlined, the firmware more stable, and errors or resource consumption that might occur due to processing irrelevant parameters are completely avoided, enhancing the determinism and reliability of the downstream audio output link. Secondly, this design establishes clear and unambiguous boundaries of responsibility at the system architecture level, achieving specialization. The main module specializes in high-fidelity output of the reference frequency, while the subordinate module is fully responsible for calculating the difference frequency and generating different frequency sound waves. This mandatory functional separation allows the two modules to be independently optimized, upgraded, or even replaced without worrying about compatibility issues caused by logical coupling. For example, when a mobile phone system undergoes a major version upgrade, its audio driver does not need to consider any logic related to the generation of external different frequency sound waves; it only needs to ensure that the reference frequency output function is normal. Finally, from the perspective of the overall system's power consumption and efficiency, this design avoids the waste of computing resources. The main module does not need to allocate any calculation cycles or cache space for difference frequency parameters that it does not use. These valuable resources can be fully concentrated on ensuring the output quality and synchronization timing of the basic sound waves, thereby improving the overall system's energy efficiency ratio. This advantage is particularly important for battery-powered mobile terminals and external portable devices.

[0039] In some embodiments, the subordinate audio processing module 3 is an analog modulation and demodulation circuit; the main audio processing module 2 is configured to output a composite signal containing a carrier signal modulated with a target difference frequency to the analog modulation and demodulation circuit; the analog modulation and demodulation circuit is used to demodulate the composite signal to obtain a second sound wave of a second frequency and drive the external speaker 5.

[0040] This embodiment proposes a clever and low-cost hardware implementation path. Its core lies in the fact that the subordinate audio processing module 3 does not employ a digital computing unit, but rather consists of a purely analog modulation and demodulation circuit. In this scheme, the main audio processing module 2 inside the mobile terminal 1 is responsible for signal preprocessing and synthesis. It is configured to generate a special composite signal, which is not a direct audible sound wave, but rather a product of modulating a high-frequency signal (carrier) with the desired "target difference frequency" (i.e., the difference between the first and second frequencies, for example, 10Hz). For example, to generate a 10Hz difference frequency effect, the main module can generate a 20kHz ultrasonic wave with an amplitude modulated by a 10Hz signal. This composite signal is transmitted through an audio interface or a wireless channel.

[0041] The external device's integrated slave audio processing module 3, namely the analog modulation and demodulation circuit, is the key to this solution. This circuit is typically composed of simple passive components (such as resistors, capacitors, and inductors) or active components (such as operational amplifiers), designed as a demodulator, such as an envelope detector circuit. When it receives the composite signal from the main module, its circuit characteristics naturally perform the demodulation process, much like a radio extracting audio from a broadcast signal. This physical process automatically strips away the high-frequency carrier component and restores the slowly varying target difference frequency signal modulated on the carrier. This restored signal is the desired second frequency sound wave. Subsequently, this sound wave signal is appropriately amplified to directly drive the connected external speaker 5 to produce sound. Throughout the entire process, the external device does not perform any digital calculations or software decoding; it relies entirely on the physical characteristics of the analog circuit to extract frequency information and reconstruct sound.

[0042] This analog circuit solution can be implemented in several specific forms. A typical example is a specially designed sleep aid pillow or eye mask. It embeds a miniature speaker and a microchip integrating analog demodulation circuitry. The user's phone plays a special audio file containing an ultrasonic carrier wave modulated by a difference frequency. The phone's speaker emits this composite signal, almost inaudible to the human ear. The nearby sleep aid pillow captures this signal through its built-in microphone or induction coil. The internal demodulation circuit then "extracts," for example, a 10Hz difference frequency sound wave and plays it through the miniature speaker. This, combined with another reference frequency sound wave that the phone may be playing, creates a sleep-inducing effect. Another example is an ultra-low-cost portable sound wave generator, which can be made into a device the size of a keychain, connecting to the phone's headphone jack via a 3.5mm audio cable. The mobile app outputs a modulated composite electrical signal, transmitted to this device via the audio cable. The device's internal demodulation circuit, composed of simple transistors and capacitors, restores the signal to the target sound wave, which is then played by a buzzer. In addition, this solution can also be used to upgrade traditional multimedia speakers by using an external, matchbox-sized demodulator accessory to turn ordinary speakers into smart devices that can receive mobile phone modulation signals and play specific difference frequency sound waves.

[0043] The requirement for analog modulation and demodulation circuits in the subordinate modules brings a series of significant advantages over digital solutions. These advantages stem directly from the shift in technological path from "digital domain computing" to "analog domain physical conversion." The most prominent advantage lies in extremely low cost and low power consumption. By completely eliminating expensive digital microprocessors, dedicated audio codec chips, and related firmware development, the core cost of the external device is reduced to a few simple, general-purpose electronic components. This allows for widespread adoption at a very low price, even as a disposable or free accessory. Simultaneously, the power consumption of analog circuits during both static and operational states is far lower than that of digital chips, significantly extending the battery life of the external device. Secondly, this solution offers extremely high reliability and real-time performance. Analog circuit processing is a continuous and parallel physical process, eliminating the delays and uncertainties introduced by software startup, instruction parsing, and calculation. The signal demodulation and output delays are extremely low and constant, which is crucial for applications requiring strict synchronization of two sound waves. Furthermore, the absence of software means there will be no program crashes, freezes, or upgrades required, resulting in strong functional stability and anti-interference capabilities. Finally, this design achieves extreme simplification and miniaturization of hardware functions. Complex frequency calculation tasks are moved to the resource-rich mobile device, and the external device only needs to perform a single physical demodulation function. This allows its circuit board to be designed to be very small and easier to integrate into various forms of products (such as wearable devices and small ornaments), providing great flexibility for product design.

[0044] In some embodiments, the frequency of the carrier signal is higher than 20kHz, and the target frequency difference is the difference between the first frequency and the second frequency.

[0045] This embodiment specifies key parameters for the analog modulation and demodulation circuit scheme, clarifying the frequency attributes of the carrier signal and its relationship with the target difference frequency. Its technical feature is that the carrier signal frequency used in the composite signal generated by the main audio processing module 2 is intentionally set to a frequency band higher than the range of human hearing (usually above 20kHz), such as 30kHz, 40kHz, or higher ultrasonic frequencies. Simultaneously, the frequency of the signal modulated on this carrier is explicitly set to the desired "target difference frequency," which is precisely the arithmetic difference between the first and second sound wave frequencies that ultimately need to be achieved. For example, if the left and right speakers ultimately need to output sound waves of 210Hz and 200Hz respectively to produce a 10Hz beat frequency effect, then the main module will generate an ultrasonic composite signal with a frequency of 30kHz modulated by a 10Hz signal.

[0046] In practical applications, this technology can be implemented through various product forms. Its core principle is to ensure the carrier wave itself is inaudible, retaining only the demodulated difference frequency signal as the effective output. A typical application example is a sleep aid system integrated into the bedroom environment. A specially designed bedside lamp or small speaker with a built-in ultrasonic receiver and analog demodulation circuitry serves as an external device. The user's mobile phone plays an audio file containing a 32kHz carrier wave modulated with a 10Hz difference frequency. Since 32kHz is far beyond the human hearing threshold, the user hears almost no harsh high-frequency noise from the phone itself, avoiding unnecessary interference. After receiving this ultrasonic signal, the bedside device's internal envelope detection circuit successfully demodulates the 10Hz low-frequency sound wave and plays it. This, combined with another audible reference frequency sound wave (such as 200Hz) that the phone may be playing simultaneously, creates the desired binaural beat environment at the user's location, without any unpleasant carrier noise throughout the process. Another example is a portable device for concentration training. A clip-on device with a miniature speaker can be attached to a collar. A mobile app generates a 38kHz carrier signal modulated with a specific difference frequency (e.g., a 7Hz Theta wave difference frequency) according to a training program and plays it through the phone's speaker. The user cannot hear this high-frequency signal, but the device on their collar receives and demodulates the 7Hz sound wave. This, combined with a reference sound wave heard through bone conduction headphones, allows for private training even in public places, without disturbing others and protecting privacy. In professional acoustic research or medical rehabilitation, this approach can be used to construct precise stimulation sources. The research device can emit a fixed-frequency carrier wave (e.g., 40kHz) modulated with a precisely set difference frequency (e.g., from 0.5Hz to 20Hz). The receiving device demodulates this to produce a pure difference frequency sound wave. Because the carrier frequency is fixed and extremely high, the demodulation circuit can be optimally designed for this single carrier frequency, ensuring extremely high purity and stability of the demodulated difference frequency signal. This avoids distortion that may be introduced by carrier frequency variations, providing reliable and repeatable acoustic stimulation for experiments.

[0047] The technical feature of specifying a carrier frequency higher than the audible range of the human ear and defining the target difference frequency as the frequency difference between the two sound waves brings crucial and multi-layered benefits. The derivation of these benefits is rooted in a comprehensive consideration of user experience, signal integrity, and system design simplicity. The primary and direct effect is the complete elimination of audible interference from the carrier signal itself, greatly enhancing the comfort and purity of the overall acoustic experience. If the carrier frequency falls within the audible range, even when used for modulation, it may be perceived by the user as background noise, interfering with or even disrupting application scenarios designed to produce psychological effects such as relaxation and focus through a specific difference frequency. Setting it in the ultrasonic band physically eliminates this source of interference, ensuring that only the two intended audible sound waves or their beat frequency effects ultimately affect the user. Secondly, from a signal processing perspective, modulating the difference frequency signal onto a fixed, high-frequency carrier facilitates the design and optimization of subsequent analog demodulation circuits. The high-frequency carrier allows for greater separation of the modulated signal's sidebands from the carrier in the spectrum. This enables more efficient and cleaner separation of the target difference frequency signal using simpler circuits such as envelope detection, reducing the possibility of low-frequency noise intrusion and thus improving the accuracy and fidelity of the generated second acoustic signal. Finally, by clearly defining the relationship that "the target difference frequency is the difference between the first and second frequencies," a clear and intuitive parameter mapping is established at the system level. This makes the application's logic design very straightforward: developers only need to calculate the difference frequency value corresponding to the desired beat frequency effect and use it as the modulated signal, without needing to worry about the complex conversion between the carrier and the demodulation end. This simplicity reduces the coupling complexity between system modules, allowing the audio generation front-end (mobile app and main module) and the sound restoration back-end (external demodulation device) to collaborate based on a clear and unified physical quantity (difference frequency), enhancing the understandability and feasibility of the entire system design.

[0048] In some embodiments, the system further includes a dedicated hardware synchronization link; the main audio processing module 2 is configured to send a synchronization trigger signal to the subordinate audio processing module 3 via the hardware synchronization link when it begins to output the first sound wave; the subordinate audio processing module 3 is configured to start outputting the second sound wave in response to the synchronization trigger signal.

[0049] This embodiment introduces a high-precision timing control mechanism for the dual-speaker stereo sound output system of mobile terminal 1. Its core lies in adding a dedicated hardware synchronization link at the physical level. This link is independent of the main communication channel (such as Bluetooth or an audio data line) for transmitting audio data or control commands, and is specifically used to transmit a precise trigger signal. Specifically, when the main audio processing module 2 inside mobile terminal 1 starts driving the built-in speaker 4 to output the first sound wave according to the command, it sends a synchronization trigger signal to the slave audio processing module 3 through this additional hardware link, such as a general-purpose input / output pin, a dedicated wire, or a specific radio frequency pulse. This signal is typically a short level transition or digital pulse. The slave audio processing module 3, integrated in the external device, continuously monitors this dedicated link. Once it detects this trigger signal, it immediately activates the playback timing of its own audio generation circuit and begins outputting the pre-calculated or prepared second frequency sound wave. In this way, the start times of the two sound wave playbacks are strictly aligned by a defined hardware event.

[0050] The specific implementation of this hardware synchronization solution can be designed according to different peripheral forms and connection methods. A typical example is a sleep aid speaker with a smart dock or dedicated connection cable. In addition to a general interface cable for transmitting audio data (such as USB-C), an extra pin is defined in the interface as a synchronization signal line. When the phone starts playing a reference frequency sound wave, its audio driver chip simultaneously controls this dedicated pin to output a high-level pulse. After the circuit inside the speaker detects this pulse, it immediately starts playing the demodulated or calculated difference frequency sound wave, achieving microsecond-level synchronization accuracy. Another example is in some professional binaural beat research equipment, where the phone sends audio streams and control parameters via Bluetooth, but simultaneously uses a specific broadcast feature of the Bluetooth Low Energy channel or a moment of contact via near-field communication to send a precise timestamp or trigger command. The receiving device uses this event as an absolute time zero point to align the playback buffer queues of the two sound waves. In addition, in the scheme of using wired headphone microphone lines to transmit modulated signals, synchronization triggering can be achieved by superimposing a specific pattern of guide pulse on the carrier signal. Before demodulation, the analog circuit at the receiving end first uses a simple digital detection circuit to identify the pilot pulse and use it as the trigger signal to start demodulation and playback, thus embedding hardware synchronization logic into the analog transmission system.

[0051] The use of a dedicated hardware synchronization link offers unparalleled advantages over software solutions in addressing the critical timing consistency issue in distributed audio systems. This advantage stems from replacing uncertain software scheduling with deterministic hardware events. The core advantage is achieving ultra-high precision and extremely low jitter synchronization. Software-level synchronization relies on real-time scheduling by the operating system, network protocol stack processing, and the precision of internal clocks within each module. These factors introduce millisecond-level or even larger, unpredictable delays (jitter). In contrast, a simple hardware signal line has a nanosecond-level and highly stable transmission delay. This ensures that the time difference between the start of two sound waves is controlled within a range imperceptible to human hearing. This is crucial for sound wave interference effects that rely on precise phase relationships (such as binaural beats), guaranteeing the stability and repeatability of the effect. Secondly, this solution exhibits strong anti-interference and reliability. The dedicated hardware link transmits only a single synchronization function, unaffected by audio data stream buffering, network congestion, or instruction parsing delays that may occur on the main data channel. Even if there are occasional stutters in the main audio data transmission, the slave module can start playing on time as soon as the trigger signal is issued, ensuring the robustness of the system's core timing functions. Finally, it simplifies the synchronization logic design of the entire system. The application program and the main control chip do not need to run complex network timing protocols or high-precision software timers; they only need to issue a simple hardware signal when playback begins. This reduces the real-time requirements of the main system and allows the design of slave devices to be simpler and more focused. They only need to respond to hardware triggers, rather than maintaining a complex clock system that is completely synchronized with the host. This improves performance while potentially reducing system complexity and power consumption.

[0052] In some embodiments, the subordinate audio processing module 3 is integrated into the true wireless stereo earphones; the main audio processing module 2 is configured to encode the first sound wave and the second sound wave into a left channel audio stream and a right channel audio stream, respectively, and send them to the true wireless stereo earphones for playback via Bluetooth protocol.

[0053] This embodiment reveals a convenient implementation method that fully utilizes existing consumer electronics products. Its core concept is to virtualize the function of the subordinate audio processing module 3 and implement it based on the standard hardware architecture of true wireless stereo headphones. In this solution, the subordinate audio processing module 3 does not exist as a physically independent external chip, but is understood as a set of logical functions jointly carried by the existing audio processing chip, Bluetooth receiver chip, and supporting firmware within the true wireless stereo headphones. The main audio processing module 2 (i.e., the mobile phone audio system) inside the mobile terminal 1 is configured to perform a crucial preprocessing task: it needs to allocate the "space" of the generated first frequency sound wave and second frequency sound wave at the digital audio stream level. Specifically, the main module will encode and fill the data of the first sound wave into the left channel of the stereo audio stream, and simultaneously encode and fill the data of the second sound wave into the right channel, thereby packaging it into a standard stereo audio stream. Subsequently, this audio stream containing different frequency information is sent to the paired true wireless stereo headphones via the conventional Bluetooth audio transmission protocol.

[0054] After receiving the audio stream, the earphones' internal standard workflow naturally separates and plays the different frequency sound waves. The working principle of true wireless stereo earphones is that the main unit (usually the right earphone) receives the complete stereo data stream, and then forwards the left channel data to the secondary unit (left earphone) through the earphone communication link. In this application scenario, this means that the left earphone ultimately plays the first frequency sound wave from the left channel of the audio stream, while the right earphone plays the second frequency sound wave from the right channel. In this way, the user's ears can simultaneously receive two sound waves of different frequencies. Throughout the process, the earphones are simply faithfully performing their original function of reproducing stereo audio, unaware of the frequency difference between the left and right channels. However, this standard playback mechanism perfectly achieves the physical effect of different frequency stereo output.

[0055] This technological approach can be implemented in various specific application scenarios. One of the most direct applications is a binaural beat-based sleep aid or focus enhancement app on a mobile phone. Users simply put on their true wireless stereo headphones, open the app, and select the desired frequency difference mode, just like listening to regular music. The application generates two digital audio signals in real time in the background, such as a 210Hz sine wave in the left channel and a 200Hz sine wave in the right channel, mixes them, and pushes them to the headphones via Bluetooth. Users can immediately experience the binaural beat effect, without any additional dedicated hardware. Another example is in immersive interactive audio content, such as interactive radio dramas or games, where developers can use guiding binaural beat sound effects as background elements. When the plot requires a character to enter a meditative or sleep state, the audio engine can dynamically generate a set of stereo sound effects with relaxing frequency differences, played through the user's everyday headphones, enhancing the emotional impact of the narrative. In addition, in voice communication or online meeting scenarios, a "private focus" mode could be designed. While transmitting the other party's voice, a local application could mix a specific frequency sound wave into another channel, which would be provided to the user through the headphones for listening in only one ear. This would help the user maintain focus in noisy remote communication without the other party being aware of it.

[0056] The decision to use true wireless stereo earbuds as the implementation platform has yielded significant and market-oriented benefits. These benefits stem from maximizing the use of existing general-purpose hardware capabilities and completely eliminating user barriers. The primary and most disruptive advantage lies in achieving widespread adoption with "zero additional hardware costs." Users no longer need to purchase, carry, or charge any specialized external speaker devices; their already ubiquitous TWS earbuds instantly transform into fully functional binaural beat generators. This breaks down the peripheral barriers that have long existed for this type of application, allowing cutting-edge acoustic experiences to penetrate a massive user base at near-zero cost. Secondly, this solution offers excellent convenience and a seamless experience. The user operation process is no different from simply listening to music or watching videos—it's plug-and-play, without complicated device pairing or mode switching, greatly enhancing the smoothness and acceptability of the user experience. From a technical implementation perspective, this solution is reliable and mature. It relies entirely on the standard Bluetooth audio transmission protocol and TWS earphone architecture, which have been proven by billions of devices. There's no need to develop any non-standard communication protocols or hardware interfaces, resulting in high stability and extremely strong compatibility, virtually eliminating the risk of functional failure due to compatibility issues. Finally, this architecture inherently possesses excellent synchronization characteristics. The Bluetooth protocol itself guarantees strict synchronization of the left and right channel data when transmitting stereo audio streams, while the master-slave synchronization technology within modern TWS earphones further ensures micro-level consistency in the playback timing of the left and right units. This provides dual synchronization guarantees at the system level for the output of two different frequency sound waves, ensuring the quality of the final effect.

[0057] In some embodiments, the subordinate audio processing module 3 includes a passive demodulation circuit integrated in the microphone circuit of the wired headphones; the mobile terminal 1 is configured to modulate the second sound wave onto a carrier wave and output it through the microphone pin of the audio interface; the passive demodulation circuit is used to demodulate the signal from the microphone pin to restore the second sound wave and output it through a sound unit in the wired headphones.

[0058] This embodiment proposes a highly sophisticated and integrated hardware implementation scheme. Its core idea lies in redefining and utilizing the existing microphone circuitry on wired headphones, transforming it into a signal transmission and demodulation channel. In this scheme, the subordinate audio processing module 3 is not a separate box or add-on device, but rather a miniature passive demodulation circuit directly integrated into the wired headphones' remote control, plug housing, or splitter. This circuit typically consists of a few inexpensive passive components such as diodes, capacitors, and resistors, for example, a simple envelope detector circuit. The mobile terminal 1, i.e., the phone, is configured to perform a special audio output task: it not only outputs a first-frequency sound wave through the main audio processing module 2 and the left and right channel pins of the audio interface, but also modulates a second sound wave signal onto a carrier frequency, generating a composite electrical signal, and outputs this composite signal through the microphone pins on the audio interface originally used for receiving voice input. The microphone circuitry inside the cable connecting the phone and headphones no longer carries the user's voice, but rather a modulated signal carrying the target sound wave information.

[0059] The passive demodulation circuit integrated into the headphone cable connects to the microphone pin of the headphone plug and a speaker unit inside the headphone (e.g., the right channel speaker). When the modulated signal arrives from the phone, the demodulation circuit automatically operates based on the physical characteristics of its components. For example, diodes rectify the signal, and capacitors filter out high-frequency carrier components, ultimately recovering the initially modulated second sound wave signal from the composite signal. This recovered analog electrical signal with the second frequency is then directly sent to the connected speaker unit, driving it to produce sound. Simultaneously, the left (or right) channel pin of the phone's audio interface outputs an unmodulated first-frequency sound wave, driving the other speaker unit in the headphone. Thus, the left and right earbuds of the same wired headphone can synchronously play two sounds of different frequencies.

[0060] This technological solution can be implemented through specific product forms. A typical example is a specially designed "sleep-aid headphones." It looks no different from ordinary wired headphones, but a tiny passive demodulation circuit, about the size of a grain of rice, is integrated into the end of its 3.5mm plug or inside the in-line control. Users connect these headphones to their phones, open a specific application, and select sleep mode. The phone plays a soothing 210Hz reference sound wave through the left channel of the headphones, while simultaneously sending a composite signal modulated at a 10Hz difference frequency through the microphone. The demodulation circuit inside the headphones instantly activates, driving the right earbud to play a 200Hz sound wave; the beat frequency effect perceived by both ears helps the user fall asleep. Another example is "study headphones" for concentration training. Students use these headphones to connect to tablets for online classes. The application can, while playing the course audio (as the main sound wave), subtly mix in a specific difference frequency sound wave (such as a 15Hz Beta beat) through the microphone, emitted from the other earbud, providing assistance without interfering with the main course content. In addition, this technology can also be used in some simple interactive toys, such as a story-telling doll. When the accompanying headphones play the story audio (left ear), they can simultaneously generate environmental sound effects or emotionally guiding beats related to the plot through a hidden sound wave channel (right ear), enhancing the sense of immersion.

[0061] The technical path defined in this embodiment brings several outstanding benefits, stemming from the creative reuse of existing consumer electronics interface functions and a minimalist hardware design. The most significant advantage lies in achieving extreme low cost and high integration. Since the subordinate audio processing function is implemented using only a few low-value passive components directly embedded in the headphone cable, the production cost of headphones with binaural frequency output capability is only slightly higher than ordinary headphones, yet a huge functional leap is achieved, enabling the technology to be widely adopted with extremely low barriers to entry. Secondly, this solution has true "zero power consumption" characteristics. The passive demodulation circuit itself does not require a power supply; its operating energy comes entirely from the signal output from the microphone pins of the phone's audio interface. This means it does not add any power burden to the headphones and does not require a built-in battery, achieving seamless functional addition. Furthermore, this design achieves perfect hidden utilization of the universal interface. Both the user and the phone system still recognize the device as a standard wired headphone; all special functions are completed in the background through signal modulation and physical demodulation, resulting in extremely high compatibility. No special drivers or system permissions are required; it is plug-and-play. Finally, the response of the analog passive circuit is instantaneous and stable, with almost no delay caused by digital processing. This ensures that the two sound waves can achieve near-perfect synchronization based on the hardware connection, thus ensuring the quality and reliability of the final acoustic effect.

[0062] In some embodiments, the main audio processing module 2 and the subordinate audio processing module 3 are integrated into the same hardware chip, which contains at least two independent digital signal processing channels for generating a first sound wave and a second sound wave, respectively.

[0063] This embodiment proposes a highly integrated hardware implementation path. Its core concept lies in physically merging the main audio processing module 2 and the subordinate audio processing module 3, designing and integrating them into a single dedicated hardware chip. In this scheme, two independent functional units, traditionally distributed within the phone and external devices, are integrated into a single semiconductor package. This chip is specially designed in its architecture, containing at least two independent digital signal processing channels. These two channels are logically and physically isolated, allowing them to work in parallel. One DSP channel is logically mapped and configured as the "main audio processing module," specifically for receiving reference frequency parameters and generating a first sound wave at a first frequency; the other parallel DSP channel is logically mapped and configured as the "subordinate audio processing module," which receives complete parameters including the reference frequency and frequency difference, and internally calculates to generate a second sound wave at a second frequency. Although they coexist on the same chip, they are like two independent lanes on a highway, simultaneously outputting two different digital audio streams without interference.

[0064] This single-chip dual-channel solution can be implemented in several specific product forms. A typical application is a custom audio processing chip for high-end or professional audio mobile phones. Mobile phone manufacturers can use this special chip in their motherboard design to replace the general audio codec. When a user runs a sleep aid application, the application sends instructions to this chip. The chip's two DSP cores generate a 210Hz signal to the phone's upper speaker and simultaneously generate a 200Hz signal to the phone's lower speaker, thus achieving stereo output from the phone's own dual speakers without relying on any external devices. Another example is creating a high-performance external USB sound card or audio adapter. The core of this device is this chip with integrated dual DSP channels, which connects to a mobile phone or computer via a USB interface. The computer software sends instructions, and one channel of the chip outputs one sound wave to the adapter's left output interface, while the other channel outputs another sound wave to the right output interface. Users can connect two independent speakers or headphones to obtain accurate and synchronized dual-frequency sound waves, which is very suitable for professional sound therapy or audio experimentation scenarios. In addition, this chip can also be an enhanced version of the main control chip for future true wireless stereo headphones. The headphone chip itself needs to process the left and right channels. This enhanced model upgrades the processing capability of the two channels to two completely independent DSP channels that can be configured with different frequencies. This allows the headphone chip to generate and play sound waves with preset frequency differences in the left and right earbuds by simply sending simple parameter commands from the mobile phone. This improves efficiency and reduces the amount of data transmitted wirelessly.

[0065] The specification integrates the master and slave modules onto a single chip using independent DSP channels, a technical feature that fundamentally improves system performance and simplifies design. This benefit stems from the fundamental architectural shift from "multi-chip distributed collaboration" to "single-chip integrated design." The core advantage is achieving extreme synchronization accuracy and extremely low output jitter. Since the two sound waves are generated from two DSP channels on the same silicon chip, sharing the same master clock and power supply, their operating rhythms are perfectly synchronized. The path delay from signal generation to output is highly predictable and consistent, enabling nanosecond-level synchronization accuracy—unmatched by any distributed system relying on external communication for synchronization. This provides the ultimate solution for phase-sensitive high-end applications. Secondly, this solution significantly reduces overall system power consumption, cost, and physical complexity. Compared to using two independent chips, the single-chip design reduces the number of packages, peripheral circuits, and connectors, lowering overall power consumption and material costs, while significantly saving board space, facilitating device miniaturization. For mobile terminals or portable peripherals, this translates to longer battery life and a more compact design. Finally, this architecture provides system developers with a simplified design paradigm. On the hardware side, only the power supply, interface, and driver for a single chip need to be handled; on the software side, only a unified control interface is needed to configure the two channels. This completely eliminates the need to deal with complex communication protocols, connection management, and fault coordination issues between two independent devices, significantly improving system reliability and development efficiency. It represents an important direction for the evolution of this technology towards high-performance, high-reliability products.

[0066] It should be noted that the order of the above embodiments of the present invention is merely for descriptive purposes and does not represent the superiority or inferiority of the embodiments. Furthermore, the above description focuses on specific embodiments of this specification. Additionally, the processes depicted in the accompanying drawings do not necessarily require a specific or sequential order to achieve the desired results. In some embodiments, multitasking and parallel processing are possible or may be advantageous.

[0067] The various embodiments in this specification are described in a progressive manner. The same or similar parts between the various embodiments can be referred to each other. Each embodiment focuses on describing the differences from other embodiments.

[0068] The above description is only a preferred embodiment of the present invention and is not intended to limit the present invention. Any modifications, equivalent substitutions, improvements, etc., made within the principles of the present invention should be included within the protection scope of the present invention.

Claims

1. A mobile terminal dual-speaker heterogeneous sound wave stereo output system, characterized in that, include: The main audio processing module (2) is located inside the mobile terminal (1) and is configured to output a first sound wave of a first frequency in response to an application command; The subordinate audio processing module (3) is communicatively connected to the main audio processing module (2) and is configured to receive control commands and output a second sound wave of a second frequency based on the control commands. Wherein, the first frequency is different from the second frequency, and the first sound wave and the second sound wave are configured to be output simultaneously to drive at least two speakers.

2. The system according to claim 1, characterized in that, The at least two speakers include a built-in speaker (4) and an external speaker (5). The built-in speaker is located inside the mobile terminal (1). The subordinate audio processing module (3) is connected to the external speaker (5). The first sound wave is output by the built-in speaker (4) of the mobile terminal (1), and the second sound wave is output by the external speaker (5).

3. The system according to claim 2, characterized in that, The application instruction includes a reference frequency parameter and a frequency difference parameter; the main audio processing module (2) is configured to generate the first sound wave of the first frequency in response to the reference frequency parameter; the control instruction includes the reference frequency parameter and the frequency difference parameter, and the subordinate audio processing module (3) is configured to calculate the second frequency based on the reference frequency parameter and the frequency difference parameter.

4. The system according to claim 3, characterized in that, The main audio processing module (2) is configured to ignore the frequency difference parameter in the application instructions and operate only based on the reference frequency parameter.

5. The system according to claim 2, characterized in that, The subordinate audio processing module (3) is an analog modulation and demodulation circuit; the main audio processing module (2) is configured to output a composite signal containing a carrier signal modulated with a target difference frequency to the analog modulation and demodulation circuit; the analog modulation and demodulation circuit is used to demodulate the composite signal to obtain the second sound wave at the second frequency and drive the external speaker (5).

6. The system according to claim 5, characterized in that, The frequency of the carrier signal is higher than 20kHz, and the target frequency difference is the difference between the first frequency and the second frequency.

7. The system according to claim 2 or 5, characterized in that, The system also includes a dedicated hardware synchronization link; the main audio processing module (2) is configured to send a synchronization trigger signal to the subordinate audio processing module (3) through the hardware synchronization link when it starts outputting the first sound wave; the subordinate audio processing module (3) is configured to start outputting the second sound wave in response to the synchronization trigger signal.

8. The system according to claim 1, characterized in that, The subordinate audio processing module (3) is integrated into the true wireless stereo headphones; the main audio processing module (2) is configured to encode the first sound wave and the second sound wave into a left channel audio stream and a right channel audio stream, respectively, and send them to the true wireless stereo headphones for playback via Bluetooth protocol.

9. The system according to claim 1, characterized in that, The subordinate audio processing module (3) includes a passive demodulation circuit integrated in the microphone circuit of the wired headphones; the mobile terminal (1) is configured to modulate the second sound wave onto a carrier wave and output it through the microphone pin of the audio interface; the passive demodulation circuit is used to demodulate the signal from the microphone pin to restore the second sound wave and output it through a sound unit in the wired headphones.

10. The system according to claim 1, characterized in that, The main audio processing module (2) and the subordinate audio processing module (3) are integrated in the same hardware chip, which contains at least two independent digital signal processing channels for generating the first sound wave and the second sound wave, respectively.

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