Multifunctional interface control method of digital player

Abstract

The invention relates to the technical field of digital playing equipment, electric signal processing and the like, and provides a multifunctional interface control method of a digital player, which comprises the following steps of: detecting each interface input signal in a multifunctional interface in real time, automatically classifying the input signals to generate a signal type classification result, and judging the priority of the signals to generate a priority result; determining an input interface switching sequence based on a classification and priority result, and automatically selecting a scene matching input source; audio signals of the input source are input into an interface control unit, and an adaptive output interface combination is determined from lotus output and cannon output interfaces according to output equipment type information, historical matching data and a current load state; and the gain level, the output impedance and the frequency response equalization strategy are automatically configured based on the electrical characteristics corresponding to the output equipment type information. According to the method, the problems of low efficiency of multi-interface conflict processing, output resource static configuration, scene adaptation deficiency and the like can be solved, and multi-interface intelligent collaboration and sound quality and energy efficiency dual optimization are realized.

Description

Technical Field

[0001] The present invention relates to the technical fields of digital playback equipment, electrical signal processing, and the like, and in particular to a multifunctional interface control method for a digital player. Background Art

[0002] Existing digital player interface control technologies use a polling detection mechanism for heterogeneous interfaces such as power, USB, and analog audio. This makes it impossible to distinguish signal types in real time, resulting in an increased error rate in input source selection. When switching input sources, the user must manually switch the input source. When a USB audio stream and a 3.5mm analog signal are connected simultaneously, the system lacks the ability to dynamically make decisions based on signal priority, causing playback interruptions or signal crosstalk. Furthermore, output interface parameters (such as gain and impedance) use a fixed preset mode and cannot automatically adjust based on dynamic changes in load impedance, resulting in power waste or increased distortion. Furthermore, the Lotus and XLR output interfaces operate independently, with no combined output strategy. When multiple devices are connected, a master-slave collaborative mechanism cannot be established, resulting in incomplete frequency response coverage and phase mismatch.

[0003] In summary, existing digital player interface control technologies have technical problems such as being unable to distinguish signal types in real time, unable to make dynamic decisions based on signal priority, unable to automatically adjust according to dynamic changes in load impedance, unable to establish a master-slave collaboration mechanism, incomplete frequency response coverage, and phase mismatch. Summary of the Invention

[0004] In response to the shortcomings of the above-mentioned existing technologies, the present invention provides a multi-functional interface control method for a digital player to solve problems such as inefficient multi-interface conflict processing, static configuration of output resources and lack of scene adaptation, and realize multi-interface intelligent collaboration and dual optimization of sound quality and energy efficiency.

[0005] The multifunctional interface control method of a digital player provided by the present invention includes: Real-time detection of input signals of each interface in the multi-function interface of the digital player; the multi-function interface includes a power interface, a USB interface, a 3.5 stereo audio input interface, a Lotus audio input interface, a Lotus audio output interface and a XLR audio output interface; Automatically classify the input signals of each interface detected to generate a signal type classification result, and perform signal priority determination on the input signals of each interface detected to generate a signal priority result; Determining the switching order of input interfaces in the multi-function interface according to the signal type classification result and the signal priority result, and automatically selecting a scene-matching input source adapted to the current scene, wherein the scene-matching input source is an interface in the multi-function interface having a valid input signal; The audio signal carried by the scene matching input source is input as a reference signal to the interface control unit; after obtaining the audio signal, the interface control unit detects the output device type information, historical matching data and current load status, and determines an adapted output interface combination from the Lotus audio output interface and the XLR audio output interface included in the multi-function interface for outputting the audio signal, and automatically configures output parameters according to the electrical characteristics corresponding to the output device type information, wherein the output parameters include gain level, output impedance and frequency response equalization strategy.

[0006] Compared with the prior art, the present invention has the following beneficial effects: The present invention provides a multifunctional interface control method for a digital player, the method comprising: real-time detection of input signals of each interface in the multifunctional interface of the digital player; the multifunctional interface comprising a power interface, a USB interface, a 3.5 stereo audio input interface, a Lotus audio input interface, a Lotus audio output interface and a XLR audio output interface; automatically classifying the input signals of each interface detected to generate a signal type classification result, and performing signal priority determination on the input signals of each interface detected to generate a signal priority result; determining the switching order of the input interfaces in the multifunctional interface according to the signal type classification result and the signal priority result, and automatically selecting the interface that is suitable for the current scene; The scene matching input source of the scene, the scene matching input source is the interface with a valid input signal in the multi-function interface; the audio signal carried by the scene matching input source is input into the interface control unit as a reference signal; after the interface control unit obtains the audio signal, it detects the output device type information, historical matching data and current load status, and determines the adapted output interface combination for the output of the audio signal from the Lotus audio output interface and the Canon audio output interface contained in the multi-function interface, and automatically configures the output parameters according to the electrical characteristics corresponding to the output device type information, the output parameters including gain level, output impedance and frequency response equalization strategy. This method can solve the problems of inefficient multi-interface conflict processing, static configuration of output resources and lack of scene adaptation, and realize the dual optimization of multi-interface intelligent collaboration and sound quality and energy efficiency. BRIEF DESCRIPTION OF THE DRAWINGS

[0007] The drawings described herein are used to provide a further understanding of the present invention and constitute a part of the present invention. The exemplary embodiments of the present invention and their description are used to explain the present invention and do not constitute an undue limitation of the present invention. Some specific embodiments of the present invention will be described in detail in an illustrative and non-restrictive manner with reference to the drawings. The same reference numerals in the drawings indicate the same or similar components or parts. It should be understood by those skilled in the art that these drawings are not necessarily drawn to scale. In the drawings: Figure 1This is a flow chart of a multifunctional interface control method for a digital player according to an embodiment of the present invention; Figure 2 This is a schematic diagram of a system architecture of a digital player according to an embodiment of the present invention. DETAILED DESCRIPTION

[0008] In order to enable those skilled in the art to better understand the solutions of the present invention, the technical solutions in the embodiments of the present invention will be clearly and completely described below in conjunction with the accompanying drawings in the embodiments of the present invention. Obviously, the embodiments described are only part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without making creative efforts should fall within the scope of protection of the present invention.

[0009] See also Figure 1-Figure 2 The embodiment of the present invention provides a method for controlling a multifunctional interface of a digital player, comprising the following steps: S101, real-time detection of input signals of each interface of the multifunctional interface of the digital player; the multifunctional interface includes a power interface, a USB interface, a 3.5 stereo audio input interface, a Lotus audio input interface, a Lotus audio output interface, and a XLR audio output interface; S102, automatically classifying the input signals of each detected interface to generate a signal type classification result, and performing signal priority determination on the input signals of each detected interface to generate a signal priority result; S103: Determine, based on the signal type classification result and the signal priority result, a switching order of input interfaces in the multi-function interface, and automatically select a scene-matching input source that is adapted to the current scene, where the scene-matching input source is an interface in the multi-function interface that has a valid input signal; S104. Input the audio signal carried by the scene matching input source as a reference signal to the interface control unit; after obtaining the audio signal, the interface control unit detects the output device type information, historical matching data, and current load status, and determines an adapted output interface combination from the Lotus audio output interface and the XLR audio output interface included in the multi-function interface for outputting the audio signal, and automatically configures output parameters based on the electrical characteristics corresponding to the output device type information, the output parameters including gain level, output impedance, and frequency response equalization strategy.

[0010] It should be noted that in this embodiment, by real-time detection of the input signal of each interface in the multifunctional interface of the digital player, the detected input signals of each interface are automatically classified to generate a signal type classification result. Without relying on traditional polling or user operation, it can instantly determine whether the signal on each interface is power, digital audio stream, or analog signal, thereby avoiding misidentification, improving the accuracy of signal recognition, and realizing intelligent and automated input source selection, effectively solving the problem of the inability to distinguish signal types in real time in the existing technology. In addition, the signal priority of each detected input signal interface is determined to generate a signal priority result. Based on the signal type classification result and the signal priority result, the switching order of the input interfaces in the multifunctional interface is determined, and the scene-matching input source adapted to the current scene is automatically selected, thereby realizing dynamic priority management of different input signals. For example, when a USB audio stream and a 3.5mm analog signal are detected simultaneously, the priority can be calculated based on weights such as quality score and historical usage frequency, and the optimal input source can be automatically selected, effectively preventing playback interruptions and signal conflicts, overcoming the limitation of the existing technology that relies on manual switching. In addition, after the interface control unit obtains the audio signal, it detects the output device type information, historical matching data, and current load status, and determines an adapted output interface combination for the output of the audio signal from the Lotus audio output interface and the XLR audio output interface included in the multi-functional interface, and automatically configures the output parameters based on the electrical characteristics corresponding to the output device type information, thereby improving power transmission efficiency and suppressing audio distortion caused by mismatch, effectively solving the power waste and sound quality degradation problems caused by fixed parameters. In addition, from the Lotus audio output interface and the XLR audio output interface included in the multi-functional interface, an adapted output interface combination is determined for the output of the audio signal, thereby overcoming the traditional single interface output mode and proposing an output interface combination selection mechanism. Based on the load characteristics and historical matching data, it can dynamically select to enable Lotus, XLR, or both in parallel, to achieve collaborative output under multi-device connection (such as master-slave device coordination, redundant output, etc.), solving the problem of the inability to establish a master-slave mechanism in the prior art. In addition, the output parameters are automatically configured based on the electrical characteristics corresponding to the output device type information. The output parameters include gain level, output impedance and frequency response equalization strategy. This not only improves the consistency of frequency response between different output devices, but also effectively alleviates crosstalk or distortion caused by phase mismatch, and compensates for the technical defects of existing player output systems in frequency domain adaptation.

[0011] Preferably, when automatically classifying the input signal, the method includes: detecting the voltage value connected to the power interface , when the voltage value When the first algorithm is satisfied, the input signal is determined to be a stable DC power supply signal; the first algorithm is: ; In the algorithm, is the ripple factor of the power supply input, defined as: ; in is the peak-to-peak voltage of the power signal.

[0012] It should be noted that in existing digital player interface control technologies, the power interface and the analog audio interface often share a physical interface or are both DC coupled inputs, which makes it difficult for the system to determine whether the interface is for charging or a signal source based solely on the voltage level, resulting in signal recognition errors, mistaking the power supply for the analog input or vice versa. In this embodiment, by defining the ripple coefficient And set the threshold , which can effectively distinguish between stable power supplies (such as 5V regulated power supplies) and pseudo-DC signals caused by high-frequency interference sources or noise, overcoming the limitations of traditional judgments based solely on the static voltage range, making classification more robust and accurate. In addition, when the interface voltage is in the 4.5V-5.5V range but the ripple is too high (such as abnormal USB power supply, under-filtered state), the system can reject it as a valid power signal through the ripple indicator, thereby preventing false triggering of the audio path and avoiding crosstalk or power supply interference. In addition, since both the ripple coefficient and voltage criteria can be obtained through fast ADC sampling, this embodiment is suitable for real-time implementation in embedded controllers or DSP platforms, without the need for time-consuming algorithms such as complex Fourier analysis, has strong adaptability, and improves recognition efficiency.

[0013] Preferably, the automatic classification of the input signal includes: performing frame header analysis on the signal data connected to the USB interface, capturing the synchronization header and verifying the frame structure, and determining that the input signal is a PCM digital audio stream when a second algorithm is satisfied; the second algorithm is: ; In the algorithm, Indicates that the frame header is a synchronization code ; Indicates that the CRC check passed.

[0014] It should be noted that in USB audio transmission protocols (such as MPEG-1 / 2 / Layer III or PCM overUSB), 0xFFF is widely used as a frame synchronization byte. In this embodiment, this is used as a necessary criterion to efficiently skip non-audio data streams (such as USB HID, control commands), and the audio processing flow is triggered only after the synchronization start frame is captured, thereby improving recognition accuracy from the source. In addition, this embodiment enhances signal recognition reliability through CRC check results to prevent false switching. On the basis of synchronization header capture, CRC check (i.e. ) to ensure the legitimacy of the frame structure. This combined criterion significantly enhances anti-interference capabilities and significantly reduces the risk of misclassification caused by jitter, packet loss, or frame distortion. Furthermore, in this embodiment, only after both the synchronization header match and CRC check pass does the system determine that the input signal is a valid PCM digital audio stream. This is then used to prioritize signals and determine the input source for scene matching. This achieves highly reliable digital audio recognition, resolving the issues of large errors and unstable input switching associated with traditional recognition. This improves input source recognition accuracy, system response stability, and overall audio quality assurance.

[0015] Preferably, when automatically classifying the input signal, the method includes: sampling the signal connected to the 3.5 stereo audio input interface or the Lotus audio input interface, and calculating its bandwidth. and signal-to-noise ratio When the third algorithm is satisfied, it is determined that the signal connected to the 3.5 stereo audio input interface or the Lotus audio input interface is a high-fidelity analog audio signal; the third algorithm is: .

[0016] It should be noted that the bandwidth criterion It just covers the audible frequency range of human ears, which meets the audio industry's requirements for high-fidelity signals. This bandwidth criterion limits the effective signal to cover the complete audio bandwidth, effectively eliminating pseudo audio signals caused by spectrum distortion, poor device contact or excessive compression. In addition, set The threshold value ensures that the input signal has good dynamic range and low noise characteristics; the SNR calculation process can reflect the real-time analysis of the overall signal amplitude and background noise, thereby effectively eliminating false signals in scenarios such as microphone short circuit, line interference, and empty connection. The threshold is combined with the bandwidth criterion to form a dual-determination mechanism, improving the recognition standard for analog signals. Furthermore, based on signal priority sorting and automatic input source switching, this embodiment uses a quality judgment mechanism to further ensure that the input signal selected by the system is of playback value and technical reliability, avoiding problems such as audio quality degradation and playback interruption caused by the mistaken selection of low-quality analog signals.

[0017] Preferably, when performing signal priority determination on the input signal, the method includes: Assign a priority weight to each detected input signal , the calculation formula of the priority weight is: ; Represents the type weight of the i-th signal, satisfying: ; represents the quality score of the i-th signal, satisfying: ; is the total harmonic distortion of the signal, is the dynamic range of the i-th signal; Represents the historical usage frequency of the i-th signal, satisfying: ; represents the number of times the i-th input signal source has been called or activated during the historical operation of the device; α, β, and γ are weighting coefficients that satisfy: ; All input signals are weighted according to their calculated priority Arrange in descending order to determine the order of input interface switching; when any two signals meet are considered to have the same priority. Represents the difference between any two signals.

[0018] It should be noted that, in this embodiment, the formula , introduces the originally subjective or static priority into the three-dimensional evaluation index system, so that it is no longer based on fixed interface priority or user manual specification, but provides a quantifiable and adjustable priority model to achieve objectivity and automation of decision-making. In addition, It provides two core audio parameters, total harmonic distortion (THD) and dynamic range, to truly quantify the characteristics of the signal itself, avoiding the problem that traditional systems only consider whether the signal exists and ignore the signal quality, thus fundamentally improving the consistency of user-side sound quality. In addition, in this embodiment, the exponential growth function is used to map the number of calls to Weight, so that the system can learn user behavior preferences; for example, if a user often uses the 3.5mm interface, the system can gradually increase its priority to achieve implicit memory interface preference judgment. In addition, , thus ensuring the dominance of signal quality weight. The three-factor weights can be flexibly configured according to the actual product design to adapt to different strategy scenarios and improve the wide applicability of the method. Moreover, by setting the threshold , which can avoid frequent switching caused by small weight differences, improve the system's anti-disturbance ability, and enhance the logical coherence of input source switching and the continuity of user experience.

[0019] Preferably, the automatic selection of a scene-matching input source that adapts to the current scene includes: when it is detected that the current of the power interface is greater than 1 ampere and there is no audio input signal, the USB interface is selected as the input source, and other interfaces are not selected as input candidates; when there are multiple analog signals input at the same time, the 3.5 stereo audio input interface is selected as the input source. If the 3.5 stereo audio input interface is not available, the Lotus audio input interface is selected as the secondary input source.

[0020] It should be noted that in this embodiment, when the power interface current is detected to be greater than 1 ampere and there is no audio signal, the corresponding scenario is inferred to be charging without analog audio. Through a combination of logical conditions, it is inferred that USB is the only valid input source, and other analog interfaces are automatically blocked from participating as candidates, avoiding unnecessary misselection and interference, thereby improving the scene relevance and decision accuracy of the recognition. When multiple analog signals are input simultaneously, the 3.5 stereo audio input interface (3.5mm interface) is prioritized. If the 3.5 stereo audio input interface is unavailable (e.g., due to physical damage or plug detection failure), it automatically downgrades to the RCA interface. This demonstrates an orderly, non-interrupted input selection chain with fallback capability, which is superior to the low-robustness strategy of traditional players that selects one or interrupts playback.

[0021] Preferably, when automatically selecting a scene-matching input source that adapts to the current scene, it includes: when it is detected that the current time is in the night mode period, selecting a digital signal source as the input source, and if no digital signal is available, selecting an analog signal that has been subjected to gain reduction processing as an alternative input source. It should be noted that in this embodiment, it is possible to determine whether it is in the night mode period by obtaining the built-in clock of the system or module, and construct a time period-aware input strategy decision path to implement different signal preference logics for the system in two typical usage scenarios, daytime and nighttime, so as to make differentiated responses to environmental behaviors. When no digital signal source is available, an analog signal path that has been subjected to gain reduction processing is provided to ensure listening comfort instead of directly returning to the analog signal.

[0022] Preferably, determining the adapted output interface combination includes: When the output device type information indicates a balanced audio device, the interface control unit forcibly enables the XLR audio output interface for signal output, while disabling the RCA audio output interface to avoid signal interference, and automatically configures the differential signal drive mode. The output voltages of the XLR audio output interfaces are set to: , is the reference voltage; It is the positive output voltage of the XLR audio output interface. It is the negative output voltage of the XLR audio output interface; When the historical matching data indicates that the load impedance connected to the output interface satisfy When the overcurrent protection threshold is set, the redundant fault-tolerant mode is entered and the overcurrent protection threshold is set. , the interface control unit controls the parallel driving of the XLR output interface and the RCA output interface, the overcurrent protection threshold satisfy: ; The maximum power rating of the audio amplifier output.

[0023] It should be noted that when the device type information indicates a balanced audio device, the system will forcibly enable the XLR interface and turn off the RCA output to achieve electrical standard adaptation of the differential output, thereby ensuring that the output mode is consistent with the actual device connection method, and avoiding the single-ended interface from mistakenly participating in the differential drive to cause crosstalk, audio degradation and other problems. In this embodiment, the interface control unit uniformly configures the output level structure to ensure output amplitude symmetry and common mode suppression, effectively improving sound quality and signal anti-interference. When it is identified from historical records that the load device impedance is low, the interface control unit controls the XLR output interface and the RCA output interface to drive in parallel to improve the output current carrying capacity and stability. In addition, the overcurrent protection threshold satisfy , it can automatically adjust the protection strategy to adapt to the load characteristics of different devices, ensuring audio thrust while preventing output stage overheating or breakdown, achieving a dynamic balance between protection and performance.

[0024] Preferably, when automatically configuring the output parameters, the gain level is dynamically adjusted, and the gain level is calculated segmentally according to the input signal amplitude and the signal-to-noise ratio; the gain level is calculated segmentally according to the input signal amplitude and the signal-to-noise ratio, including: When the input signal voltage Less than or equal to full-scale input voltage When the target voltage is half, the gain is determined by the logarithmic calculation method of the target voltage ratio, and the formula is: ; In the formula, The rated input voltage reference value determined according to the output device type information; When the input signal voltage Greater than When , the strategy of linear gain correction based on signal-to-noise ratio is adopted, and the calculation formula is: ;in, is the set minimum gain value, Indicates the signal-to-noise ratio of the current input signal, is the gain adjustment factor, used to adjust the Increase or decrease the linear compensation amount; the strategy of performing linear gain correction based on the signal-to-noise ratio is used to ensure that the output signal meets the driving requirements of the target device and enhance the audio fidelity under high signal-to-noise ratio conditions.

[0025] It should be noted that when the input signal voltage In the low-amplitude region, the gain is determined by the logarithmic calculation method of the target voltage ratio, thereby achieving automatic compensation for weak input signals, so that the output level meets the driving amplitude required by the target device, ensuring output signal clarity, and avoiding problems such as low volume or signal loss. When the input level is high, a linear gain correction strategy based on the signal-to-noise ratio is adopted. The calculation formula is: , linearly correct the gain according to the signal-to-noise ratio SNR to enhance fidelity. When the SNR is low (i.e. high amplitude but noisy), the output gain is automatically reduced to prevent high-level and high-noise signals from causing distortion amplification or noise spikes. When the SNR is high (high-quality signal), increase the gain to maximize the sound quality rather than simply maximize the loudness. In this embodiment. The input voltage is used to determine whether to enter the compensation or correction branch, the input quality (SNR) is used to determine the gain correction amplitude, and the output is matched to the rated level of the target device to ensure physical driving capabilities, effectively avoiding the blind amplification or sound quality sacrifice of traditional AGC, and improving the stability, accuracy and listening consistency of dynamic response. At the same time, The rated input voltage reference value determined based on the output device type information serves as the device type-aware reference voltage to achieve output level matching for different audio load devices. It supports automatic setting of the target gain response strategy based on the connected device type, improving the compatibility and intelligence level of the overall system.

[0026] Preferably, when detecting the current load state, the load impedance connected to the output interface is To perform measurements and identify abnormalities, inject a test signal with a frequency of 10 kHz at the output interface, measure the output voltage and current signals respectively, and calculate the load impedance connected to the current output interface through the phase difference. : ; is the test voltage amplitude measured at the output interface, is the corresponding test current amplitude, is the phase angle between voltage and current; When the current load impedance is detected With nominal impedance When the deviation exceeds 50%, the gain level is automatically reduced to prevent power overload due to load mismatch, and the frequency response equalization strategy is switched to flat mode to ensure that the frequency response remains consistent in each frequency band. The historical matching data is updated synchronously to record the impedance anomaly information of the current device under the selected output interface combination.

[0027] It should be noted that in this embodiment, the real impedance is calculated by injecting the test signal and measuring the voltage, current amplitude and phase difference. , which is different from the rough estimation method based only on amplitude calculation; The introduction of impedance detection enables the ability to distinguish capacitive and inductive load characteristics, which is applicable to more types of audio equipment, industrial loads or dynamic power systems, and significantly improves the robustness of load identification under high-frequency interference, complex load circuits or transient anomalies. With nominal impedance When the deviation exceeds 50%, the output gain is automatically reduced to suppress audio distortion or device damage caused by too light or too heavy loads. The frequency response equalization strategy switches to flat frequency response mode to avoid excessive power push in specific frequency bands and effectively control overload risks. After a load deviation occurs, the historical matching database corresponding to the output interface combination and the device is synchronously updated. When the device or interface combination is detected again in the future, the output strategy or early warning prompt can be adjusted in advance based on the recorded load anomaly, realizing intelligent learning optimization at the device level and building long-term evolvable load control and audio interface management.

[0028] Preferably, the frequency response equalization strategy includes constructing a frequency response compensation filter in a digital signal processor according to the frequency response defect characteristics described in the output device type information to achieve adaptive correction of the frequency response. Constructing the frequency response compensation filter specifically includes the following steps: extracting the center frequency of the frequency response valley value according to the device frequency response defect characteristics recorded in the database of historical matching data; and the center frequency of the frequency response valley A compensation filter function in the form of a second-order parametric equalizer is constructed for the center ; ; in, Indicates the current frequency, represents a complex unit; Q is the quality factor, which is used to control the filter bandwidth and is calculated as: ; 3dB bandwidth; Frequency point The gain compensation factor at , is defined as: ; Indicates frequency point The gain value required for compensation.

[0029] It should be noted that, based on the device frequency response defect characteristics recorded in the historical matching database, the center frequency of the frequency response valley is extracted. Compared with the traditional equalizer that only performs fixed compensation based on the general frequency band, this embodiment can generate personalized filter parameters according to the differences between different devices, improve the system adaptability, and solve the problems of frequency response distortion and sound quality degradation caused by device differences. At the same time, the filter is not preset with fixed parameters, but is adaptively corrected within the DSP according to the device defects. A compensation filter function in the form of a second-order parametric equalizer is constructed for the center , achieving fine-grained correction of the frequency response. When facing different loads or interface changes, the frequency response self-correction process can be completed without manual intervention. The set gain compensation factor adopts logarithmic scale processing ( ), which conforms to the logarithmic characteristics of human hearing and will not cause abrupt volume jumps under the same compensation gain. It is suitable for high-fidelity playback systems. Combined with a second-order equalizer, it can improve the ability to restore the original audio signal while ensuring spectrum balance.

[0030] Preferably, when it is detected that there is a DC offset in the Lotus audio output interface, and the DC offset voltage of the Lotus audio output interface satisfy: , it is determined that there is an electrical anomaly in the Lotus audio output interface and the electrical anomaly processing process is executed. The electrical anomaly processing process includes: automatically switching the current output channel to the XLR audio output interface to ensure the audio output quality and equipment safety; activating the DC calibration algorithm of the output signal in the interface control unit to correct the abnormal offset in real time, and the output voltage after correction satisfy: ; in, is the current audio output signal, is the calibration sampling period; The unique identifier of the abnormal interface is recorded in the database of historical matching data, and the priority weight of the abnormal interface is reduced; after the interface enters the abnormal state, a periodic retry mechanism is set to automatically try to reinitialize the faulty interface at intervals to verify whether the faulty interface has returned to normal.

[0031] It should be noted that, in this embodiment, a DC offset of ≥50 mV is used as the threshold for judging electrical anomalies, so as to avoid traditional unresponsive fuzzy judgments, implement quantitative voltage monitoring on the Lotus audio output interface, and provide a verifiable and traceable fault detection mechanism, thereby achieving high-precision, low-false-alarm interface anomaly identification, which is beneficial to enhancing system stability. Once an anomaly is determined, switch to the XLR interface to avoid the continued transmission of distorted signals and damage to downstream equipment. The switching strategy is based on the principles of signal quality and equipment safety, and completes fault isolation without affecting the user experience, thereby ensuring the continuity and safety of the high-fidelity audio system when an offset occurs. Corrected output voltage The system uses an integral averaging model to offset the output signal, automatically approaching the center level (zero-point symmetry). Compared to simple threshold compression or fixed offset correction, this adapts to various input scenarios and improves audio output fidelity and stability. Furthermore, abnormal interfaces not only trigger a switchover but are also uniquely identified and written to a historical matching database. The priority of these abnormal interfaces is dynamically reduced. Combined with periodic retry and reinitialization strategies, this system achieves self-healing of interface status and improves the long-term robustness of the system.

[0032] The above embodiments are only preferred specific implementation methods of the present invention, and the protection scope of the present invention is not limited thereto. Any changes or replacements that can be easily thought of by any technician familiar with this technical field within the technical scope disclosed by the present invention should be covered within the protection scope of the present invention. The protection scope of the present invention should be based on the protection scope of the claims.

Claims

1. A multifunctional interface control method for a digital player, characterized in that: include: Real-time detection of input signals of each interface in the multi-function interface of the digital player; The multifunctional interface includes a power interface, a USB interface, a 3.5 stereo audio input interface, a Lotus audio input interface, a Lotus audio output interface and a XLR audio output interface; Automatically classify the input signals of each interface detected to generate a signal type classification result, and perform signal priority determination on the input signals of each interface detected to generate a signal priority result; Determining the switching order of input interfaces in the multi-function interface according to the signal type classification result and the signal priority result, and automatically selecting a scene-matching input source adapted to the current scene, wherein the scene-matching input source is an interface in the multi-function interface having a valid input signal; The audio signal carried by the scene matching input source is input as a reference signal to the interface control unit; after obtaining the audio signal, the interface control unit detects the output device type information, historical matching data and current load status, and determines an adapted output interface combination from the Lotus audio output interface and the XLR audio output interface included in the multi-function interface for outputting the audio signal, and automatically configures output parameters according to the electrical characteristics corresponding to the output device type information, wherein the output parameters include gain level, output impedance and frequency response equalization strategy.

2. The multifunctional interface control method of a digital player according to claim 1, characterized in that: When automatically classifying the input signal, it includes: detecting the voltage value connected to the power interface , when the voltage value When the first algorithm is satisfied, the input signal is determined to be a stable DC power supply signal; the first algorithm is: ； In the algorithm, is the ripple factor of the power supply input, defined as: ； in is the peak-to-peak voltage of the power signal.

3. The multifunctional interface control method of a digital player according to claim 1, wherein: Automatically classifying the input signal includes: performing frame header analysis on the signal data connected to the USB interface, capturing the synchronization header and verifying the frame structure, and determining that the input signal is a PCM digital audio stream when a second algorithm is satisfied; the second algorithm is: ； In the algorithm, Indicates that the frame header is a synchronization code ; Indicates that the CRC check passed.

4. The multifunctional interface control method of a digital player according to claim 1, wherein: When automatically classifying the input signal, it includes: sampling the signal connected to the 3.5 stereo audio input interface or the lotus audio input interface, calculating its bandwidth and signal-to-noise ratio When the third algorithm is satisfied, it is determined that the signal connected to the 3.5 stereo audio input interface or the Lotus audio input interface is a high-fidelity analog audio signal; the third algorithm is: 。 5. The multifunctional interface control method of a digital player according to claim 1, characterized in that: When performing signal priority determination on the input signal, the method includes: Assign a priority weight to each detected input signal , the calculation formula of the priority weight is: ; Represents the type weight of the i-th signal, satisfying: ； represents the quality score of the i-th signal, satisfying: ; is the total harmonic distortion of the signal, is the dynamic range of the i-th signal; Represents the historical usage frequency of the i-th signal, satisfying: ; Indicates the number of times the i-th input signal source has been called or activated during the historical operation of the device; α, β, γ are weighted coefficients, satisfying: ; All input signals are weighted according to their calculated priority Arrange in descending order to determine the order of input interface switching; when any two signals meet are considered to have the same priority. Represents the difference between any two signals.

6. The multifunctional interface control method of a digital player according to claim 1, characterized in that: Automatically select the scene-matching input source that adapts to the current scene, including: when the current of the power interface is detected to be greater than 1 ampere and there is no audio input signal, the USB interface is selected as the input source, and other interfaces are not selected as input candidates; when there are multiple analog signals input simultaneously, the 3.5 stereo audio input interface is selected as the input source. If the 3.5 stereo audio input interface is unavailable, the Lotus audio input interface is selected as the secondary input source.

7. The multifunctional interface control method of a digital player according to claim 1, wherein: Automatically selecting a scene-matching input source that adapts to the current scene includes: when it is detected that the current time is in the night mode period, selecting a digital signal source as the input source. If no digital signal is available, selecting an analog signal after gain reduction processing as an alternative input source.

8. The multifunctional interface control method of a digital player according to any one of claims 1 to 7, characterized in that: When determining the output interface combination for adaptation, include: When the output device type information indicates a balanced audio device, the interface control unit forcibly enables the XLR audio output interface for signal output, while disabling the RCA audio output interface to avoid signal interference, and automatically configures the differential signal drive mode. The output voltages of the XLR audio output interfaces are set to: , is the reference voltage; It is the positive output voltage of the XLR audio output interface. It is the negative output voltage of the XLR audio output interface; When the historical matching data indicates that the load impedance of the connected device satisfy When the overcurrent protection threshold is set, the redundant fault-tolerant mode is entered and the overcurrent protection threshold is set. , the interface control unit controls the parallel driving of the XLR output interface and the RCA output interface, the overcurrent protection threshold satisfy: ; The maximum power rating of the audio amplifier output.

9. The multifunctional interface control method of a digital player according to any one of claims 1 to 7, characterized in that: When automatically configuring output parameters, the gain level is dynamically adjusted, and the gain level is calculated in sections based on the input signal amplitude and the signal-to-noise ratio.

10. The multifunctional interface control method of a digital player according to claim 9, characterized in that: The gain level is calculated piecewise based on the input signal amplitude and the signal-to-noise ratio, including: When the input signal voltage Less than or equal to full-scale input voltage When the target voltage is half, the gain is determined by the logarithmic calculation method of the target voltage ratio, and the formula is: ； In the formula, The rated input voltage reference value determined according to the output device type information; When the input signal voltage Greater than When , the strategy of linear gain correction based on signal-to-noise ratio is adopted, and the calculation formula is: ;in, is the set minimum gain value, Indicates the signal-to-noise ratio of the current input signal, is the gain adjustment factor, used to adjust the Increase or decrease the linear compensation amount; the strategy of performing linear gain correction based on the signal-to-noise ratio is used to ensure that the output signal meets the driving requirements of the target device and enhance the audio fidelity under high signal-to-noise ratio conditions.

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