Single-line digital audio output control system of high-frequency surgical equipment

Abstract

The invention belongs to the technical field of medical instrument control, and particularly discloses a single-line digital audio output control system of high-frequency surgical equipment, which comprises the steps of constructing a multi-dimensional parameter fused state perception and semantic audio feedback mechanism, accurately identifying the working mode, the power level and the output stability of the equipment through auditory sense, and controlling the output of the high-frequency surgical equipment according to the state perception and semantic audio feedback mechanism. The operation burden and the misjudgment risk in a complex operation environment are effectively reduced; transmission of audio, data and a power supply is integrated by adopting a single-line digital audio bus, so that the internal structure and the electromagnetic compatibility of equipment are optimized fundamentally; by establishing a set of self-adaptive safety closed loop from communication quality monitoring to power and audio cooperative control, the health condition of a communication link can be actively monitored, and when it is detected that the communication quality is reduced, protective measures such as power reduction and safety alarm are automatically triggered, and the potential control failure risk is eliminated in the bud.

Description

Technical Field

[0001] This invention belongs to the field of medical device control technology and relates to a single-line digital audio output control system for high-frequency surgical equipment. Background Technology

[0002] High-frequency surgical devices, also known as electrosurgical devices or high-frequency electrosurgical units, are a class of medical devices widely used in modern surgery. They utilize high-frequency current applied to biological tissues, achieving operations such as cutting, coagulation, and vaporization by precisely controlling the waveform, voltage, and power of the current. The core of these devices lies in their high-frequency energy generator and the accompanying surgical instruments; the stability of their performance and the precision of their control directly affect the safety and outcome of the surgery. During the procedure, the device typically provides feedback to the surgeon on its current status through a screen display and simple audio-visual prompts.

[0003] In existing technologies, high-frequency surgical devices typically employ independent audio prompt modules for status feedback. This module usually consists of a simple buzzer or piezoelectric ceramic plate, driven by the main control unit under specific conditions to emit a fixed tone or frequency of beeping sound to indicate device start-up, shutdown, mode switching, or fault alarms. For example, a continuous beep might be emitted in cutting mode, while an intermittent beep might be emitted in coagulation mode. Meanwhile, operations such as power adjustment and mode selection are primarily performed via panel buttons, resulting in a unidirectional, open-loop control relationship between the device and the audio feedback system.

[0004] However, the aforementioned existing technical solutions have significant technical drawbacks. Their audio feedback information is limited in scope and cannot accurately reflect subtle changes in output power or dynamic shifts in tissue impedance, making it difficult for surgeons to judge the actual effect of energy output solely by hearing. Furthermore, the separation of the independent audio module from the power control system and the multi-cable internal layout increase the risk of electromagnetic interference within the device, hindering miniaturization and integration. Simultaneously, this open-loop feedback method lacks monitoring of the communication link itself; if the transmission paths of the control and feedback signals are interfered with, the system cannot detect it, posing a safety hazard. Summary of the Invention

[0005] In order to overcome the above-mentioned defects of the prior art and to achieve the above objectives, the present invention proposes the following technical solution: a single-line digital audio output control system for a high-frequency surgical device, comprising: a signal acquisition module, which acquires the voltage signal, current signal and impedance signal of the high-frequency output, and identifies the device's operating mode parameters and the output power level set by the user to obtain multi-source output parameters.

[0006] The encoding generation module calculates the output stability coefficient based on the multi-source output parameters, and then encodes and maps the operating mode parameters, output power level and output stability coefficient to generate the device status code.

[0007] The audio synthesis module generates audio control parameters by querying a preset audio mapping rule library based on the device status code.

[0008] The signal modulation module synthesizes semantic audio signals based on audio control parameters, and modulates the semantic audio signals using a combined pulse width modulation and Manchester encoding protocol to generate single-line digital audio signals.

[0009] The quality monitoring module transmits single-line digital audio signals to the audio output terminal via a single-line digital audio bus and monitors the signal error rate of the single-line digital audio bus in real time, generating transmission quality indicators.

[0010] Adjust the prompt module so that when the transmission quality index exceeds the preset threshold, adjust the output power and trigger the safety audio prompt mode.

[0011] Compared with the prior art, the beneficial effects of the present invention are as follows: (1) The present invention greatly improves the intuitiveness and safety of high-frequency surgical equipment by constructing a state perception and semantic audio feedback mechanism with multi-dimensional parameter fusion. The surgeon can accurately identify the working mode, power level and output stability of the equipment by hearing without relying on visual observation, which effectively reduces the operational burden and misjudgment risk in complex surgical environments and enhances the immersiveness and efficiency of human-computer interaction.

[0012] (2) This invention integrates the transmission of audio, data and power by using a single-wire digital audio bus, which significantly reduces internal physical wiring, lowers manufacturing costs and potential failure points, and effectively suppresses coupling interference between signals. It fundamentally optimizes the internal structure and electromagnetic compatibility of the device, providing a solid technical foundation for realizing the miniaturization, portability and stable operation of the device in a high-frequency strong electromagnetic environment.

[0013] (3) By establishing an adaptive safety closed loop from communication quality monitoring to power and audio coordinated control, this invention can proactively monitor the health status of the communication link and automatically trigger protective measures such as power reduction and safety alarms when a decline in communication quality is detected, thus eliminating potential control failure risks at the outset. This shift from passive alarm to proactive prevention in safety strategy comprehensively improves the robustness and inherent safety of the equipment in unpredictable environments. Attached Figure Description

[0014] To more clearly illustrate the technical solutions of the embodiments of the present invention, the accompanying drawings used in the description of the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0015] Figure 1 This is a schematic diagram of the system module connections of the present invention. Detailed Implementation

[0016] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0017] Please see Figure 1 As shown, the present invention proposes a single-line digital audio output control system for a high-frequency surgical device, comprising: a signal acquisition module, an encoding generation module, an audio synthesis module, a signal modulation module, a quality monitoring module, and an adjustment prompt module.

[0018] The signal acquisition module, encoding generation module, audio synthesis module, signal modulation module, quality monitoring module, and adjustment prompt module are connected in sequence.

[0019] The signal acquisition module acquires high-frequency output voltage, current and impedance signals, identifies the device's operating mode parameters and the user-set output power level, and obtains multi-source output parameters.

[0020] The encoding generation module calculates the output stability coefficient based on the multi-source output parameters, and then encodes and maps the operating mode parameters, output power level, and output stability coefficient to generate the device status code.

[0021] In a preferred embodiment, the step of calculating the output stability coefficient based on multi-source output parameters includes: continuously acquiring current and voltage signals through a high-frequency analog-to-digital converter within a preset time window to obtain a series of discrete current and voltage sample values.

[0022] The high-frequency analog-to-digital converter (ADC) is a key electronic device used to convert continuously changing analog signals into discrete digital signals. Analog signals are continuously changing physical quantities, such as current and voltage, which are continuous in both time and amplitude; while digital signals are discrete, usually represented by binary code. The main function of the high-frequency ADC is to sample continuous analog current and voltage signals at a certain sampling frequency and convert the amplitude of each sample point into a corresponding digital quantity, thereby realizing the conversion from analog to digital signals.

[0023] The average current value is obtained by taking the arithmetic mean of all current samples within a time window, and the current fluctuation value is quantified by calculating the standard deviation of all current samples within the same window relative to the average current value. This reflects the degree of dispersion of the current output.

[0024] The voltage fluctuation rate is calculated by simultaneously acquiring the ratio of the standard deviation of all voltage samples within the time window to their average voltage, and then used to correct power calculation errors.

[0025] The real-time impedance value is obtained by calculating the ratio of voltage to current, and the impedance change rate is obtained by comparing the change in impedance value within a unit time window.

[0026] By weighting and integrating current fluctuations, voltage fluctuations, and impedance changes, a comprehensive instantaneous stability factor is calculated. The magnitude of this factor objectively reflects the relative stability of the output signal under the current average power level, effectively avoiding the interpretation deviation of absolute fluctuation values ​​caused by different power levels.

[0027] The formula for calculating the comprehensive instantaneous stability factor is: In the formula, This represents the characteristic scale of current fluctuation obtained by normalizing the current fluctuation value. This represents the voltage fluctuation characteristic scale obtained by normalizing the voltage fluctuation rate. The impedance fluctuation characteristic scale is represented by the impedance change rate obtained through normalization. The normalization process, such as the minimum-maximum normalization method, makes the current fluctuation value, voltage fluctuation rate, and impedance change rate have the same unit or are essentially dimensionless physical quantities. This indicates the weighting of current fluctuation, voltage fluctuation rate, and impedance change rate in the output stability coefficient. This weighting can be adjusted based on clinical needs or equipment characteristics; for example, to achieve a more sensitive response to changes in tissue impedance, the weighting can be increased. If power stability is a concern, then increase... or .

[0028] The output stability coefficient is generated by weighted averaging of the combined instantaneous stability factors of multiple consecutive time windows.

[0029] In order to smooth transient noise and capture the trend of stability changes, the system does not directly use the instantaneous stability factor, but uses a weighted average algorithm to iteratively smooth the instantaneous stability factor calculated over multiple consecutive time windows.

[0030] Specifically, the output stability coefficient is calculated using the exponential moving average method. The calculation is as follows: In the formula, The output stability coefficient for the current time window. The instantaneous stability factor calculated for the current window. The output stability coefficient for the previous time window. The weighting factor is a preset value, ranging from 0 to 1, and is determined through a large number of experiments and tests.

[0031] In this way, the system combines historical stability data with current data to generate a more stable and reliable output stability coefficient that can quickly respond to significant changes and effectively filter out random spikes. This coefficient will serve as an important input for subsequent state coding and audio synthesis.

[0032] This method achieves dynamic and in-depth quantification of high-frequency energy output quality by calculating the output stability coefficient. Compared to traditional methods that only monitor current amplitude, the output stability coefficient provided by this method can more accurately reflect the impact of subtle changes in tissue impedance and electrode contact state during surgery on energy output stability. This evaluation mechanism based on relative fluctuations ensures consistency and comparability of stability assessments across different power levels, improving the accuracy of the evaluation. This, in turn, upgrades the feedback control of the entire system from passive response to active prediction, significantly enhancing the safety of the surgical equipment and its adaptability to complex operating conditions.

[0033] In a further preferred embodiment, the step of encoding and mapping the working mode parameters, output power level, and output stability coefficient to generate device status code includes: mapping the working mode parameters into high-bit code segments, which is achieved through a preset lookup table. For example, the four modes of standby, cutting, coagulation, and abnormality are fixedly mapped to binary 00, 01, 10, and 11, respectively, forming a 2-bit code segment.

[0034] The output power level is mapped to a 4-bit coded segment, which can represent 16 discrete levels. That is, a continuous power setting range, such as 5W to 200W, is mapped to an integer level from 0 to 15 through non-linear or linear segmentation (equal interval division), and then converted into the corresponding 4-bit binary code. The mapped levels are: level 0 = 5W, level 1 = 18.125W, ..., level 15 = 200W.

[0035] The output stability coefficient is mapped to a low-bit coded segment. This index is a continuously changing value. It is divided into four stability intervals by setting multiple thresholds, such as stable, slightly fluctuating, moderately fluctuating and severely fluctuating, and assigned 2-bit binary codes 00, 01, 10 and 11 respectively.

[0036] Combine the high-order code segment, the middle-order code segment, and the low-order code segment to generate an 8-bit binary format device status code.

[0037] The specific operation is as follows: Device status code = (high-order code segment << 6) | (middle-order code segment << 2) | low-order code segment, where << represents a binary left shift operation, and | represents a binary bitwise OR operation. Through this operation, the high-order code segment is placed in the highest 2 bits of the 8-bit byte, the middle code segment occupies the middle 4 bits, and the low-order code segment occupies the lowest 2 bits, thus generating a structured and compact 8-bit device status code, which will serve as the direct input to the subsequent audio synthesis module.

[0038] Example: The current status of a certain high-frequency surgical device is as follows: Operating mode: In coagulation mode (requires output of pulsed high-frequency current to promote hemostasis).

[0039] Output power: set to 100W, level 7 → middle code segment = 0111 (4 bits).

[0040] Output stability coefficient: Tissue impedance fluctuations were detected and determined to be moderate fluctuations (energy output stability needs to be adjusted).

[0041] Combined coding segments generate an 8-bit device status code: 10 0111 10.

[0042] This method, through hierarchical encoding and bit combination techniques, compresses the multi-dimensional and heterogeneous operating states of a device into a single, standardized data byte. This encoding and mapping mechanism not only significantly improves the efficiency and density of information transmission, making it particularly suitable for bandwidth-constrained single-wire bus systems, but also endows the status information with a clear hierarchical structure. The decoding end can efficiently and independently extract the status information of any dimension simply through bitmasking operations, such as focusing only on the operating mode or judging only the output stability. More importantly, it creates a completely new technical feature: a composite status vector that can comprehensively describe the device's operating status. This enables subsequent audio feedback systems to make judgments and responses based on a comprehensive context of operating mode, power level, and stability, rather than reacting to a single parameter in isolation. This achieves semantic expression from simple parameter alarms to complex operating conditions, laying the foundation for truly intelligent and information-rich audio feedback.

[0043] The audio synthesis module generates audio control parameters by querying a preset audio mapping rule library based on the device status code.

[0044] In a preferred embodiment, the audio control parameters are generated by querying a preset audio mapping rule base based on the device status code, wherein the audio mapping rule base defines the basic correspondence between working modes and audio templates.

[0045] The audio mapping rule base details the correspondence between each bit or segment of the binary state code and the audio attributes. For example, it specifies that the first 3 bits of the binary code are used to determine the approximate range of pitch, the middle 4 bits are used to define the type of timbre, and the last 5 bits are used to control the tempo, etc.

[0046] The steps for generating audio control parameters include: retrieving and loading corresponding audio templates from the audio mapping rule library based on the working mode parameters corresponding to the high-order coding segment in the device status code. For example, if the working mode parameter is "cutting", then load an audio template preset as continuous tone; if it is "coagulation", then load an audio template preset as pulse tone.

[0047] Each preset audio template details its parameters. For example, for a continuous tone template, it specifies the frequency range (e.g., 1000–2000 Hz) and volume (e.g., 70 dB); for a pulse tone template, it specifies the pulse width (e.g., 100 ms) and pulse interval (e.g., 500 ms).

[0048] Based on the output power level corresponding to the median coded segment in the device status code, the basic attributes of the selected audio template are dynamically modulated to adjust the pitch slope of the audio template. For example, for continuous tones in the cut mode, the pitch is not fixed, but is linearly or logarithmically mapped according to the power level. The higher the power, the greater the pitch rise slope, making it sound more rapid and powerful.

[0049] The pitch slope modulation formula includes: linear mapping formula: ,in, The preset slope coefficient determines how quickly the pitch slope changes with the power level. The value depends on the requirements of the audio design; To set the natural constant, a reference value used to adjust the pitch slope, it is set according to the initial pitch slope requirements of the audio template.

[0050] Logarithmic mapping formula: ,in, It controls the amplitude of the pitch slope changing logarithmically with power level, through computer simulation of different... The curve showing the change in pitch slope with power level at a given value should be used to select the curve that best meets the design requirements. value; To set a natural constant, the reference offset used to adjust the pitch slope is set according to the overall pitch slope requirements of the audio template, so as to ensure that the pitch slope changes at different power levels meet the expected auditory effect.

[0051] Adjust the pulse frequency or waveform asymmetry of the audio template based on the output stability coefficient corresponding to the low-order coding segment in the device status code.

[0052] Specifically, the output stability coefficient is used to adjust the rhythm or timbre characteristics of the audio template. For example, for the pulse tone in the coagulation mode, the output stability coefficient will be used to adjust the pulse repetition frequency; the higher the stability, the more stable the pulse frequency; when the stability decreases, the pulse frequency will decrease accordingly or become irregular to serve as an alert. For alarm tone templates loaded under abnormal conditions, the output stability coefficient may be used to adjust the waveform asymmetry of the asymmetric sawtooth wave; the worse the stability, the sharper and more piercing the waveform.

[0053] The formula for adjusting the pulse frequency is: The fundamental frequency is the initial frequency of the pulse tone without considering the influence of the output stability coefficient, and its value depends on the design requirements of the audio template; the frequency adjustment factor is used to measure the degree of influence of the output stability coefficient on the pulse frequency, and its value is determined according to the expected amplitude of the pulse frequency change with the output stability coefficient.

[0054] Assuming an asymmetric sawtooth wave is used, the waveform asymmetry is represented by the ratio of the rise time to the fall time. The waveform asymmetry adjustment formula is as follows: Let the rise time be tr (the time it takes for the waveform to rise from a certain percentage (10%) of its steady-state value to a certain percentage (90%) of another steady-state value), and the fall time be tf (the time it takes for the waveform to fall from 90% of its steady-state value to 10%). The waveform asymmetry... Adjust the formula: ,in, This indicates the initial asymmetry of the waveform without considering the influence of the output stability coefficient. Its value depends on the design requirements of the audio template and the initial characteristic settings of the waveform. The output stability coefficient determines the degree of influence of waveform asymmetry, and its value is determined based on the expected magnitude of waveform asymmetry change with the stability coefficient.

[0055] By performing composite modulation on multiple audio dimensions of the template, a set of comprehensive audio control parameters containing information such as pitch, volume, waveform, and rhythm is finally generated. These parameters will be passed to the subsequent audio synthesis module.

[0056] This method combines fixed audio templates with dynamic parameter modulation to generate audio control parameters with clear semantics, achieving a highly contextualized audio generation mechanism. This mechanism transforms the real-time operating status of the device into an auditory language with rich layers and semantic information. The operating mode determines the "subject" of the audio, i.e., the basic timbre; the power level adjusts the "tone" of the audio, i.e., the pitch; and the output stability adjusts the "rhythm" of the audio, i.e., the regularity. This multi-dimensional collaborative modulation of audio output allows operators to intuitively and accurately perceive the core status of the surgical device, including the intensity and stability of energy output, solely through hearing, without needing to look at the screen. This greatly enhances the intuitiveness of human-computer interaction and the immersive experience of operation. It solves the problem of traditional audio prompts being singular and vague, elevating audio feedback from a simple alarm tool to a high-level interactive interface that can assist decision-making and enhance situational awareness.

[0057] The signal modulation module synthesizes semantic audio signals based on audio control parameters, and modulates the semantic audio signals using a pulse width modulation and Manchester encoding composite protocol to generate single-line digital audio signals.

[0058] In a preferred embodiment, the step of using a pulse width modulation and Manchester encoding composite protocol to modulate the semantic audio signal to generate a single-line digital audio signal includes: generating a baseband audio signal through a digital signal processor or a dedicated audio codec based on the input audio control parameters. This signal is a digitized waveform data stream, the sampling rate and bit depth of which are preset by the system, and the waveform, frequency and amplitude of which are defined by the audio control parameters, thus constituting the original form of the audio content.

[0059] Specifically, the sampling rate of the baseband audio signal is set to an adjustable range of 8kHz to 192kHz to meet the audio quality requirements of different surgical scenarios. The bit depth offers two options: 16-bit or 24-bit, to ensure the precision and dynamic range of the audio signal.

[0060] Pulse width modulation (PWM) technology is used to perform pulse width modulation on the baseband audio signal to convert the baseband audio signal into a pulse width modulated signal.

[0061] The pulse width modulation (PWM) technique involves linearly mapping the instantaneous amplitude of the baseband audio signal to the duty cycle of the PWM signal. The duty cycle is essentially the proportion of the high-level time to the period, ranging from 0% to 100%. Specifically, a high-frequency triangular or sawtooth wave carrier wave is compared with the baseband audio signal. When the audio signal amplitude is higher than the carrier wave amplitude, the PWM output is high; otherwise, it is low. In this way, the amplitude information of the original audio signal is encoded into the width of a series of rectangular pulses, thus generating a pulse width modulated signal. The advantages of this signal are its constant amplitude, strong noise immunity, and ease of recovery of the original analog audio through simple low-pass filtering.

[0062] Manchester encoding is performed on the pulse width modulation signal, and a synchronization header and parity bit are embedded to generate a single-line digital audio signal.

[0063] Specifically, the generated pulse width modulation (PWM) signal is Manchester encoded to embed digital status information and enhance signal synchronization performance. Manchester encoding is a self-synchronizing encoding method that represents each data bit with a level transition, typically "0" from high to low and "1" from low to high. In this method, the system applies Manchester encoding rules bit-by-bit to the complete 8-bit device status code, along with the synchronization header for frame synchronization and the check bit for error checking, superimposing or embedding them into specific time slots of the PWM signal to form the final single-line digital audio signal. This signal not only carries PWM-encoded analog audio information but also digitally carries device status code and frame control information, and possesses self-synchronizing characteristics, requiring no additional clock line.

[0064] Example: For device status code: 11010110. The sequence after adding the synchronization header: 1010 11010110. Calculate the check bit using methods such as Cyclic Redundancy Check (CRC) or simple parity check. Here, we use simple parity check as an example, counting the number of 1s in the data. If the number is odd, the check bit is 1; if it's even, it's 0. The resulting sequence after adding the check bit is: 1010 11010110 1. Perform Manchester encoding on the sequence: 1 -> low to high transition (represented as 01); 0 -> high to low transition (represented as 10). Encoded (hypothetically): 0110 10011001 01 (Note: Actual encoding is performed bit-by-bit according to Manchester rules).

[0065] This method employs a composite protocol of pulse width modulation (PWM) and Manchester encoding to achieve an innovative signal modulation scheme. It successfully and efficiently integrates two different types of information—analog audio content and digital status data—into a single digital signal stream, endowing this stream with excellent anti-interference capabilities and self-synchronization characteristics. PWM modulation ensures the fidelity of audio quality during transmission, while Manchester encoding ensures reliable transmission of digital data and clock recovery. The combination of these two techniques solves the technical challenge of transmitting complex mixed signals over a single cable. This composite modulation method provides the physical layer foundation for building a single-wire multi-functional bus, enabling the co-linear transmission of audio signals, status data, and even low-voltage power supplies. This greatly simplifies system wiring, reduces the risk of internal electromagnetic coupling, and creates favorable conditions for realizing highly integrated and miniaturized portable surgical devices.

[0066] The quality monitoring module transmits single-line digital audio signals to the audio output terminal via a single-line digital audio bus, and monitors the signal error rate of the single-line digital audio bus in real time to generate transmission quality indicators.

[0067] In a preferred embodiment, the transmission of a single-wire digital audio signal to an audio output terminal via a single-wire digital audio bus is wherein: the single-wire digital audio bus simultaneously transmits the single-wire digital audio signal, device status data, and low-voltage DC power supply, and the device status data includes, for example, the device's on / off status, operating mode, output power level, output stability coefficient, resonance parameters, etc.

[0068] The transmission steps include: injecting a single-wire digital audio signal into the power line through a single-wire interface coupler, so that a composite signal is formed on the bus, that is, a stable DC level superimposed with a high-frequency modulated AC signal.

[0069] Specifically, at the transmission initiator, i.e. the main control unit side, the system uses a single-wire interface coupler to combine the single-wire digital audio signal generated by the signal modulation module with an independent low-voltage DC power supply.

[0070] The coupler is essentially a frequency-selective network, comprising an inductor connected in series with a low-voltage DC power supply and a capacitor connected in series with a single-wire digital audio signal source.

[0071] The inductor exhibits extremely low impedance to DC current and high impedance to high-frequency audio signals, thus ensuring that DC power can be smoothly fed into the bus while preventing high-frequency signals from flowing back into the power module. The capacitor exhibits low impedance to high-frequency audio signals and is open-circuit to DC, allowing single-wire digital audio signals to be injected into the bus while isolating the signal modulation module from DC voltage surges.

[0072] On the audio output terminal side, such as at the audio output terminal or handheld device, a bandpass filter performs the opposite separation operation, ensuring that the composite signal formed on the bus passes through this filter before entering the terminal circuit. Specifically, the bandpass filter is designed so that its passband frequency precisely matches the spectrum of the single-line digital audio signal, while providing extremely high attenuation for DC components and out-of-band noise.

[0073] The filter output is split into two paths: one is the clean, single-line digital audio signal recovered after filtering, which is sent to the terminal's signal processing unit for demodulation and decoding; the other is a low-voltage DC power supply extracted through a low-pass filter branch or a simple choke, which powers the terminal's internal circuitry, ensuring the device functions properly. In this way, the system successfully achieves simultaneous, bidirectional, or unidirectional transmission of three different types of signals on a single wire without adding additional physical cabling.

[0074] The low-pass filter branch is a circuit or device that allows signals within a specific frequency range to pass through while suppressing signals outside that range.

[0075] The choke is essentially a high-frequency inductor that presents low impedance to DC (almost no voltage drop) and high impedance to high-frequency AC (blocking signals).

[0076] This method achieves fundamental simplification and optimization at the system architecture level by constructing a single-wire digital audio bus that integrates power and data. First, it significantly reduces the complexity of internal wiring and space occupation, providing crucial support for miniaturization and portability. More importantly, by drastically reducing the number of cables and connectors, electromagnetic coupling paths and potential sources of signal crosstalk within the system are effectively suppressed, thereby significantly improving the electromagnetic compatibility and signal transmission stability of the entire device in high-frequency operating environments. This design, which integrates the power supply network and signal network, not only reduces manufacturing costs and failure rates but also creates a more robust and reliable internal communication architecture, resulting in higher integration between different functional modules and synergistic performance gains.

[0077] In a further preferred embodiment, the transmission of a single-wire digital audio signal to an audio output terminal via a single-wire digital audio bus also includes closed-loop feedback control: parsing device status codes from the single-wire digital audio signal.

[0078] The parsed device status code is compared with the current actual status to generate a status deviation value.

[0079] The operating parameters of the high-frequency output module are fine-tuned based on the state deviation value.

[0080] The high-frequency output module's operating parameters are adjusted in a coordinated manner, including power level, operating mode, stability coefficient, resonance parameters, drive signal duty cycle, and safety protection threshold. This enables precise control to adapt to different surgical needs and tissue characteristics, real-time monitoring and optimization of energy output quality, and improved reliability and safety of surgical equipment.

[0081] Specifically, this method establishes a parallel closed-loop feedback control loop while performing unidirectional audio signal transmission, enabling real-time calibration of the device status using the communication link. At the audio output terminal, after demodulating and decoding the received single-line digital audio signal, the system not only sends the audio information to the speaker but also extracts the device status code embedded in each frame of data.

[0082] The receiving end sends the parsed device status code to its internal microcontroller via the local bus. Simultaneously, the microcontroller independently collects its own (i.e., the terminal device's) actual status parameters. For example, if the terminal is a handheld device with buttons, it reads the button states, including on / off; if it has its own sensors, it collects sensor data. The system then compares the status represented by the device status code received from the master device with the terminal device's actual status, generating a status deviation value. This deviation value can be a logical value (such as whether the button states match) or a numerical difference (such as the difference between sensor readings).

[0083] Once a non-zero state deviation value is detected, the terminal device immediately sends the deviation value, or a specific command containing a calibration request, back to the main control unit via the uplink channel of the single-wire digital audio bus (usually using the same cable for bidirectional communication). Upon receiving this feedback, the main control unit instantly fine-tunes the operating parameters of the high-frequency output module based on the specific content of the state deviation value. For example, if the deviation stems from inconsistencies in the button states of the handheld device, the main control unit immediately updates the power output start / stop control; if the deviation originates from subtle differences in impedance measurements, the main control unit fine-tunes the resonant parameters of the power amplifier unit to optimize energy matching. Through this continuous cycle of analysis, comparison, feedback, and fine-tuning, the system constructs a bidirectional information interaction and closed-loop control system based on a single-wire communication link.

[0084] This method creatively constructs a low-latency, high-precision closed-loop control system by multiplexing data channels in the audio transmission link. This allows the main control unit to not only publish its status but also receive real-time "echoes" from the execution terminal, thereby sensing and correcting inconsistencies between the master and slave devices. This capability is crucial for ensuring the accuracy and safety of delicate operations in minimally invasive surgery. For example, it can instantly confirm the triggering of handpiece buttons, avoiding erroneous energy output due to signal delay or loss. At a deeper level, it tightly couples previously independent device components (such as the main unit and handpiece) into a collaborative whole through a dynamic information closed loop. This synergistic effect improves the system's response speed to operational commands and its adaptability to external changes, making the operation of the entire high-frequency surgical equipment more precise and reliable, and achieving a high degree of synchronization and consistency of the system's internal state.

[0085] In a further preferred embodiment, the real-time monitoring of the signal bit error rate of the single-line digital audio bus and the generation of transmission quality indicators include: extracting the parity bit in the single-line digital audio signal for error detection.

[0086] The operating principle is as follows: On the audio output terminal side, after demodulating the single-line digital audio signal, the system first performs Manchester decoding to recover the original binary data stream. Due to the characteristics of Manchester encoding, the system can accurately identify the synchronization header of each frame, thereby achieving frame synchronization and locating the check bit of each frame.

[0087] Error detection is achieved through a checksum bit: At the sending end, when constructing each data frame, the system calculates a checksum bit based on the device status code and other data content within the frame, using a preset checksum algorithm, such as Cyclic Redundancy Check (CRC) or simple parity check, and appends it to the end of the frame. At the receiving end, the system applies the same checksum algorithm to each received data frame (excluding the checksum bit) and recalculates a local checksum bit. Then, the local checksum bit is compared with the checksum bit inherent in the received frame. If they match, the data transmission of that frame is considered correct; if they do not match, the frame is determined to be an erroneous frame.

[0088] Count the number of erroneous frames per unit time and calculate the real-time bit error rate.

[0089] The operating principle is as follows: The system sets up an internal counter specifically for counting the number of erroneous frames detected per unit time (e.g., per second). For example, if 1000 frames are received in 1 second, and 10 of them are erroneous, then the number of erroneous frames is 10. By dividing this number of erroneous frames by the total number of frames received per unit time, an instantaneous real-time bit error rate is obtained. This bit error rate directly reflects the current noise level and interference intensity of the communication link.

[0090] A moving average filter is applied to the real-time bit error rate to generate transmission quality indicators.

[0091] The operating principle is as follows: To avoid misjudgments caused by a single burst of noise and to obtain an indicator that better reflects the long-term stability of the link, the system uses a moving average filtering algorithm to smooth a series of continuous real-time bit error rates. Specifically, the system maintains a fixed-length first-in-first-out queue to store the real-time bit error rate values ​​for the most recent N periods. Whenever a new real-time bit error rate is calculated, it is added to the queue, the oldest value is removed, and then the arithmetic mean of all values ​​in the queue is calculated. This smoothed average value is defined as the final transmission quality indicator, which can stably and continuously quantify the communication quality of the bus and is used as the basis for subsequent power control decisions.

[0092] Example: Assuming the queue length is 5, and the bit error rate for 5 consecutive cycles is [2%, 1%, 3%, 2%, 1%], then .

[0093] This method, based on checksum and statistical bit error rate calculation, provides an objective and quantitative indicator to measure the degree of electromagnetic interference affecting the bus. By employing moving average filtering, it effectively avoids system oversensitivity or sluggish response, enabling the system to accurately detect communication quality degradation trends caused by external environmental changes or internal faults, rather than waiting for catastrophic communication outages. This predictive monitoring capability creates conditions for the system to take preventative measures such as power reduction and switching to safe modes when communication quality deteriorates, thereby eliminating potential safety risks at their inception and significantly improving the operational robustness and reliability of the entire high-frequency surgical system in complex electromagnetic environments.

[0094] The adjustment prompt module adjusts the output power and triggers a safety audio prompt mode when the transmission quality index exceeds a preset threshold.

[0095] In a preferred embodiment, adjusting the output power and triggering a safety audio prompt mode when the transmission quality index exceeds a preset threshold includes: generating a power reduction control command when the transmission quality index exceeds the preset threshold.

[0096] Specifically, the system continuously compares real-time transmission quality metrics with a preset threshold set before the device leaves the factory or configured by the user. This threshold represents the maximum communication error rate the system can tolerate. Once the transmission quality metric exceeds this threshold, it means that the reliability of bus communication has dropped to an unacceptable level, and the security intervention mechanism is immediately activated.

[0097] The high-frequency output power is gradually reduced according to the power reduction control command.

[0098] The power reduction control command does not directly reduce the power to zero, but rather triggers a gradual power reduction procedure. Based on this command, the power control module gradually reduces the high-frequency output power at a preset rate by adjusting the output voltage of the DC-DC power conversion unit or the duty cycle of the drive signal of the high-frequency power amplifier unit. This gradual reduction design aims to avoid sudden organizational changes or impacts on the power grid caused by abrupt power fluctuations, ensuring a smooth operational transition. The power reduction process continues until the transmission quality indicators fall below the safety threshold, or the power is reduced to a preset minimum safe level.

[0099] At the same time, switch the audio control parameters to the preset security alarm audio template.

[0100] While performing power reduction, the system simultaneously adjusts its audio feedback strategy. It immediately interrupts the currently playing regular semantic audio and forcibly switches the audio control parameters to a preset safety alarm audio template. This template is typically designed as a highly recognizable and warning sound, such as a high-volume, intermittent buzzing tone at a fixed frequency. This audio mode switching aims to communicate to the operator in the most direct and explicit way that the system is in an abnormal or risky state and that safety intervention is being implemented. This dual intervention—physical power reduction and explicit warning at the human-machine interface—works synergistically to ensure that the operator can promptly perceive the risk and take appropriate action until system communication returns to normal.

[0101] This method establishes a closed-loop feedback link from communication quality monitoring to power and audio coordinated control, realizing an intelligent proactive safety protection mechanism. It directly links a potential communication problem—high bit error rate—to the device's most critical safety parameter, output power, enabling the system to possess "self-protection" capabilities. When unreliable communication is detected, it automatically enters a safer, lower-risk operating mode, effectively preventing uncontrolled energy output due to erroneous or lost control commands. Through smooth power reduction and synchronized alarm tone switching, the system ensures safety while also maintaining operational continuity and the user's right to know. This adaptive risk avoidance strategy greatly enhances the robustness and inherent safety of high-frequency surgical equipment in the face of unpredictable electromagnetic interference environments, elevating safety protection to a new level of foresight and systematic approach.

[0102] It should be noted that the formulas described above, through the principle of dimensional consistency and mathematical standardization methods (such as normalization, dimensionless parameter conversion, or unit system unification), can translate physical quantities with different properties into unitless standard values ​​or superimposed parameters of the same dimension. This eliminates the interference of different dimensions on the computational logic, allowing the formulas to retain the original data distribution characteristics while possessing mathematical rationality and adaptability to objective laws. The descriptions are merely exemplary embodiments of the present invention and should not be construed as limiting the scope of the invention.

Claims

1. A single-line digital audio output control system for a high-frequency surgical device, characterized in that, include: The signal acquisition module acquires high-frequency output voltage, current, and impedance signals, and identifies the device's operating mode parameters and the user-defined output power level to obtain multi-source output parameters. The encoding generation module calculates the output stability coefficient based on the multi-source output parameters, and then encodes and maps the operating mode parameters, output power level and output stability coefficient to generate the device status code. The audio synthesis module queries a preset audio mapping rule library based on the device status code to generate audio control parameters; The signal modulation module synthesizes semantic audio signals based on audio control parameters, and modulates the semantic audio signals using a combination of pulse width modulation and Manchester encoding protocol to generate single-line digital audio signals. The quality monitoring module transmits single-line digital audio signals to the audio output terminal via a single-line digital audio bus and monitors the signal error rate of the single-line digital audio bus in real time, generating transmission quality indicators. Adjust the prompt module so that when the transmission quality index exceeds the preset threshold, adjust the output power and trigger the safety audio prompt mode.

2. The single-line digital audio output control system for a high-frequency surgical device according to claim 1, characterized in that, The calculation of the output stability coefficient based on multi-source output parameters includes: Within a preset time window, current and voltage signals are continuously acquired by a high-frequency analog-to-digital converter, thereby obtaining a series of discrete current and voltage sample values. The average current value is obtained by taking the arithmetic mean of all current samples within the time window, and the current fluctuation value is quantified by calculating the standard deviation of all current samples within the same window relative to the average current value. The voltage fluctuation rate is calculated by simultaneously acquiring the ratio of the standard deviation of all voltage samples within the time window to their average voltage. The real-time impedance value is obtained by calculating the ratio of voltage to current, and the impedance change rate is obtained by comparing the change amplitude of impedance value within a unit time window. The comprehensive instantaneous stability factor is calculated by weighting and integrating the current fluctuation value, voltage fluctuation rate, and impedance change rate. The output stability coefficient is generated by weighted averaging of the combined instantaneous stability factors of multiple consecutive time windows.

3. The single-line digital audio output control system for a high-frequency surgical device according to claim 1, characterized in that, The step of encoding and mapping operating mode parameters, output power levels, and output stability coefficients to generate device status codes includes: The working mode parameters are mapped to high-bit code segments to form a 2-bit code segment; The output power level is mapped to a 4-bit coded segment, which can represent 16 discrete levels. The output stability coefficient is mapped to a low-order coded segment and assigned a 2-bit binary code. Combine the high-order code segment, the middle-order code segment, and the low-order code segment to generate an 8-bit binary format device status code.

4. The single-line digital audio output control system for a high-frequency surgical device according to claim 3, characterized in that, The audio control parameters are generated by querying a preset audio mapping rule base based on the device status code, wherein: The audio mapping rule base defines the basic correspondence between working modes and audio templates; The steps for generating audio control parameters include: Based on the working mode parameters corresponding to the high-order coding segment in the device status code, the corresponding audio template is retrieved from the audio mapping rule base and loaded. Based on the output power level corresponding to the median coding segment in the device status code, the basic attributes of the selected audio template are dynamically modulated to adjust the pitch slope of the audio template. Adjust the pulse frequency or waveform asymmetry of the audio template based on the output stability coefficient corresponding to the low-order coding segment in the device status code.

5. The single-line digital audio output control system for a high-frequency surgical device according to claim 1, characterized in that, The process of modulating semantic audio signals using a combined pulse width modulation and Manchester coding protocol to generate single-line digital audio signals includes: Based on the input audio control parameters, a baseband audio signal is generated by a digital signal processor or a dedicated audio codec. Pulse width modulation (PWM) technology is used to perform pulse width modulation on the baseband audio signal to convert the baseband audio signal into a pulse width modulated signal; Manchester encoding is performed on the pulse width modulation signal, and a synchronization header and parity bit are embedded to generate a single-line digital audio signal.

6. The single-line digital audio output control system for a high-frequency surgical device according to claim 1, characterized in that, The transmission of a single-wire digital audio signal to the audio output terminal via a single-wire digital audio bus, wherein: A single-wire digital audio bus simultaneously transmits single-wire digital audio signals, device status data, and low-voltage DC power. The transmission steps include: injecting a single-wire digital audio signal into the power line through a single-wire interface coupler to form a composite signal on the bus; The coupler is essentially a frequency-selective network, comprising an inductor connected in series with a low-voltage DC power supply and a capacitor connected in series with a single-wire digital audio signal source. On the audio output terminal side, a bandpass filter is used to perform the opposite separation operation, so that the composite signal formed on the bus passes through this filter before entering the terminal circuit. The filter output is split into two paths: one is the pure single-line digital audio signal recovered after filtering, which is sent to the terminal's signal processing unit for demodulation and decoding; the other is a low-voltage DC power supply extracted through a low-pass filter branch or a simple choke, which is used to drive the terminal's internal circuitry.

7. The single-line digital audio output control system for a high-frequency surgical device according to claim 6, characterized in that, The single-wire digital audio signal is transmitted to the audio output terminal via a single-wire digital audio bus, and closed-loop feedback control is also performed simultaneously. Parse device status codes from single-line digital audio signals; The parsed device status code is compared with the current actual status to generate a status deviation value; The operating parameters of the high-frequency output module are fine-tuned based on the state deviation value.

8. A single-line digital audio output control system for a high-frequency surgical device according to claim 5, characterized in that, The real-time monitoring of the signal bit error rate of the single-line digital audio bus and the generation of transmission quality indicators include: Extract the parity bit from the single-line digital audio signal for error detection; Count the number of erroneous frames per unit time and calculate the real-time bit error rate; A moving average filter is applied to the real-time bit error rate to generate transmission quality indicators.

9. A single-line digital audio output control system for a high-frequency surgical device according to claim 1, characterized in that, The step of adjusting the output power and triggering a safety audio prompt mode when the transmission quality index exceeds a preset threshold includes: When the transmission quality index exceeds the preset threshold, a power reduction control command is generated; The high-frequency output power is gradually reduced according to the power reduction control command; At the same time, switch the audio control parameters to the preset security alarm audio template.

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