Headphone adaptive tone quality enhancement method and system

Abstract

The invention discloses a self-adaptive sound quality enhancement method and system for a headphone. The method comprises the following steps: firstly, collecting hearing characteristic data of a user in real time and monitoring the noise intensity of the current environment in real time; secondly, sound quality enhancement parameters of the current user are calculated according to the hearing characteristic data and the current environment noise intensity; then, according to the sound quality enhancement parameter of the current user, calculating an audio output parameter of the earphone in real time; the earphone processes the input audio signal according to the calculated final frequency response; and finally, converting the processed digital audio signal into an analog signal through a digital-to-analog converter, and outputting the analog signal through an earphone loudspeaker unit. The audio output parameter of the earphone is calculated in real time by adopting the current user tone quality enhancement parameter calculation, the input audio signal is processed according to the final frequency response, and the overall gain is applied to the audio signal after the final frequency response adjustment, so that the high fidelity rate of the tone quality is greatly improved, and the user experience is greatly improved.

Description

Technical Field

[0001] This invention relates to the field of audio processing technology, specifically to an adaptive sound quality enhancement method and system for headphones. Background Technology

[0002] In recent years, with the rapid development of consumer electronics, especially the widespread adoption of smart devices, headphones, as important audio output devices, have been widely used in various scenarios such as music appreciation, gaming, telephone calls, and noise cancellation. Traditional headphones typically use fixed audio output parameters to provide sound quality, and this static audio setting makes it difficult to meet the personalized needs of different users. Especially when facing complex and varied environmental noise, traditional headphones often struggle to simultaneously guarantee high-fidelity sound quality and noise suppression. Furthermore, users' hearing characteristics vary due to factors such as age, health status, and individual differences; a single sound quality setting cannot provide the optimal listening experience for all users. As people's demands for audio quality continue to increase, adaptive sound enhancement technology has gradually become a research hotspot. Adaptive sound enhancement technology aims to dynamically adjust audio output parameters based on the user's individual hearing characteristics and the current ambient noise level to provide a superior audio experience. This technology not only improves audio clarity and detail but also effectively suppresses noise in noisy environments while maintaining high fidelity in speech or music.

[0003] Most existing adaptive sound enhancement technologies on the market are based on simple noise suppression algorithms or manual user adjustments. However, these methods have certain limitations. For example, simple noise suppression algorithms may lead to sound quality degradation, especially in the low-frequency range; while manual adjustments allow for personalized sound settings, their lack of real-time and intelligent features makes them difficult to adapt to rapid environmental changes and diverse user needs. Furthermore, existing sound enhancement systems often lack a comprehensive analysis of the user's hearing characteristics and fail to fully consider the impact of ear canal structure on audio transmission, resulting in less than ideal sound enhancement effects. Moreover, existing headphone noise removal and sound fidelity considerations are limited, failing to comprehensively consider ear canal structure and environmental noise for real-time adjustments, leading to significantly reduced noise reduction efficiency. To address the limitations of existing headphone noise reduction, a new solution is urgently needed for high-fidelity automated real-time processing to improve processing efficiency and accuracy, thereby enhancing user satisfaction. Summary of the Invention

[0004] To address the aforementioned problems in the prior art, this invention provides an adaptive sound quality enhancement method for headphones. This method first collects the user's hearing characteristic data in real time using sensors built into the headphones and monitors the current ambient noise intensity in real time using an external microphone. Second, it calculates the current user's sound quality enhancement parameters based on the hearing characteristic data and the current ambient noise intensity. Then, based on these parameters, it calculates the headphones' audio output parameters in real time, including volume and frequency response. Third, the headphones process the input audio signal according to the calculated final frequency response, applying an overall gain to the adjusted audio signal. Finally, the processed digital audio signal is converted to an analog signal via a digital-to-analog converter and output through the headphone's speaker unit. This application, by using the current user's sound quality enhancement parameters to calculate the headphones' audio output parameters in real time and processing the input audio signal according to the final frequency response, and applying an overall gain to the adjusted audio signal, significantly improves the high fidelity of the sound quality and greatly enhances the user experience.

[0005] This application provides an adaptive sound quality enhancement method for headphones, including the following steps: S1: The user's hearing characteristic data is collected in real time through the built-in sensor of the earphone. The hearing characteristic data includes the user's ear canal impedance, the user's ear canal sound wave reflectivity, the user's ear canal sound wave absorption rate, the user's ear canal resonant frequency, the air pressure in the ear canal, and the temperature in the ear canal. S2: Real-time monitoring of ambient noise intensity via an external microphone; S3: Calculates the current user's sound quality enhancement parameters based on hearing characteristic data and current ambient noise intensity. ;

[0006] in, Based on the base frequency gain value, The impedance value of the user's ear canal at frequency f. Maximum ear canal impedance For the user's ear canal at frequency acoustic emission rate at that location For the user's ear canal at frequency Sound absorption rate at that location The intensity of the current ambient noise at frequency f. This represents the maximum environmental noise intensity. This represents the current air pressure inside the ear canal. This is the reference air pressure inside the ear canal. The current temperature inside the ear canal. This is the reference temperature inside the ear canal; S4: Based on the current user's sound quality enhancement parameters, calculate the headphone's audio output parameters in real time, including volume and frequency response, to achieve personalized high-fidelity sound effects for the user;

[0007]

[0008] It is the final volume output at frequency f. It is the adjusted final frequency response; This is the initial frequency response curve; S5: The headphones process the input audio signal according to the calculated final frequency response, gain or attenuate different frequency components, and then adjust the overall gain of the signal according to the calculated final volume, and apply the overall gain to the audio signal after the final frequency response adjustment. S6: Converts the processed digital audio signal into an analog signal via a digital-to-analog converter and outputs it through the headphone speaker unit.

[0009] Preferably, the sensors built into the earphone include an acoustic sensor, a barometric pressure sensor, and a temperature sensor; the acoustic sensor is used to measure the sound wave reflectivity, sound wave absorptivity, and resonant frequency of the user's ear canal in real time; the barometric pressure sensor is used to detect the current air pressure in the ear canal; the temperature sensor is used to monitor the current temperature in the ear canal; and the user's ear canal impedance is obtained by measuring an acoustic impedance meter.

[0010] Preferably, the real-time monitoring of the current ambient noise intensity via an external microphone includes: capturing the original sound signal in the current environment using an external microphone; performing a fast Fourier transform on the captured original sound signal; and obtaining the intensity of the ambient noise at each frequency f. The power spectral density of the frequency domain signal obtained through Fast Fourier Transform is calculated, and the maximum power spectral density value is recorded as the maximum value of the ambient noise intensity. .

[0011] Preferably, the earphone processes the input audio signal based on the calculated final frequency response, including: based on the final frequency response. A digital filter is constructed, including an FIR filter or an IIR filter, configured to apply specific gain or attenuation to the audio signal components at different frequencies f; the input audio signal is frequency-decomposed by a digital signal processing module, and the decomposed frequency components are input to the corresponding filters for processing; the frequency components after filter processing are recombined into a complete audio signal.

[0012] Preferably, adjusting the overall gain of the signal based on the calculated final volume includes: adjusting the final volume at each frequency f based on the calculated final volume. The overall target output volume is calculated by weighted averaging. , target output volume Average volume of the input audio signal In comparison, determining the global gain coefficient .

[0013] Preferably, applying the overall gain to the audio signal after final frequency response adjustment includes, in a digital signal processor, applying the global gain coefficient... As a scaling factor, it is multiplied point-by-point by each frequency component of the audio signal after final frequency response adjustment. The multiplication operation is performed at each sample point or each frequency component of the audio signal to ensure that the overall volume of the signal is proportional to the global gain coefficient. The scale is magnified or reduced, and then converted into an analog signal output to the headphone speaker unit.

[0014] This application also provides an adaptive sound enhancement system for headphones, including: Acquisition module: The earphone's built-in sensors collect the user's hearing characteristic data in real time. The hearing characteristic data includes the user's ear canal impedance, the user's ear canal sound wave reflectivity, the user's ear canal sound wave absorption rate, the user's ear canal resonant frequency, the air pressure inside the ear canal, and the temperature inside the ear canal. Noise acquisition module: Monitors the current ambient noise level in real time via an external microphone; Sound quality enhancement parameter calculation module: Calculates the current user's sound quality enhancement parameters based on hearing characteristic data and the current ambient noise level. ;

[0015] in, Based on the base frequency gain value, This represents the impedance value of the user's ear canal at frequency f. Maximum ear canal impedance For the user's ear canal at frequency acoustic emission rate at that location For the user's ear canal at frequency Sound absorption rate at that location The intensity of the current ambient noise at frequency f. This represents the maximum environmental noise intensity. This represents the current air pressure inside the ear canal. This is the reference air pressure inside the ear canal. The current temperature inside the ear canal. This is the reference temperature inside the ear canal; Audio output parameter calculation module: Based on the current user's sound quality enhancement parameters, calculate the headphone's audio output parameters in real time, including volume and frequency response, to achieve personalized high-fidelity sound effects for the user;

[0016]

[0017] It is the final volume output at frequency f. It is the adjusted final frequency response; This is the initial frequency response curve; Input audio signal processing module: The headphones process the input audio signal according to the calculated final frequency response, gain or attenuate different frequency components, and then adjust the overall gain of the signal according to the calculated final volume, and apply the overall gain to the audio signal after the final frequency response adjustment. Output module: Converts the processed digital audio signal into an analog signal via a digital-to-analog converter and outputs it through the headphone speaker unit.

[0018] Preferably, the earphone processes the input audio signal based on the calculated final frequency response, including: based on the final frequency response. A digital filter is constructed, including an FIR filter or an IIR filter, configured to apply specific gain or attenuation to the audio signal components at different frequencies f; the input audio signal is frequency-decomposed by a digital signal processing module, and the decomposed frequency components are input to the corresponding filters for processing; the frequency components after filter processing are recombined into a complete audio signal.

[0019] Preferably, adjusting the overall gain of the signal based on the calculated final volume includes: adjusting the final volume at each frequency f based on the calculated final volume. The overall target output volume is calculated by weighted averaging. , target output volume Average volume of the input audio signal In comparison, determining the global gain coefficient .

[0020] Preferably, applying the overall gain to the audio signal after final frequency response adjustment includes, in a digital signal processor, applying the global gain coefficient... As a scaling factor, it is multiplied point-by-point by each frequency component of the audio signal after final frequency response adjustment. The multiplication operation is performed at each sample point or each frequency component of the audio signal to ensure that the overall volume of the signal is proportional to the global gain coefficient. The scale is magnified or reduced, and then converted into an analog signal output to the headphone speaker unit.

[0021] This invention provides an adaptive sound quality enhancement method and system for headphones, which achieves the following beneficial technical effects: 1. This invention first collects the user's hearing characteristic data in real time through the built-in sensors of the headphones and monitors the current ambient noise intensity in real time through an external microphone. Second, it calculates the current user's sound quality enhancement parameters based on the hearing characteristic data and the current ambient noise intensity. Then, based on the current user's sound quality enhancement parameters, it calculates the headphones' audio output parameters in real time, including volume and frequency response. Third, the headphones process the input audio signal according to the calculated final frequency response, applying the overall gain to the audio signal adjusted by the final frequency response. Finally, the processed digital audio signal is converted into an analog signal through a digital-to-analog converter and output through the headphone speaker unit. This application significantly improves the high fidelity of sound quality by using the current user's sound quality enhancement parameters to calculate the headphones' audio output parameters in real time and applying the overall gain to the audio signal adjusted by the final frequency response. By considering factors such as ear canal impedance, sound wave reflectivity, and absorption rate, it accurately analyzes the user's hearing characteristics, enabling the audio output to be optimized according to individual differences, providing a sound quality experience that better meets the user's needs and greatly enhancing the user experience.

[0022] 2. This invention calculates the current user's sound quality enhancement parameters based on hearing characteristic data and the current ambient noise intensity. ;

[0023] Taking into account the fundamental frequency gain, the impedance of the user's ear canal at frequency f, and the maximum ear canal impedance, the value of the user's ear canal at frequency f is... The sound wave emissivity at a certain location, and the user's ear canal at a certain frequency. The system comprehensively considers the sound wave absorption rate at a given frequency, the intensity of current ambient noise at frequency f, the maximum intensity of ambient noise, the current air pressure inside the ear canal, the reference air pressure inside the ear canal, the current temperature inside the ear canal, and the reference temperature inside the ear canal. By taking into account the ear canal structure and the influence of ambient noise in real time, the system makes comprehensive adjustments, greatly improving the noise cancellation efficiency of the headphones and significantly enhancing the user experience. This application also incorporates an external microphone to monitor ambient noise in real time, obtains the noise intensity at various frequencies through Fast Fourier Transform (FFT), and dynamically adjusts the frequency response and overall volume based on the user's hearing characteristics. This maintains high-fidelity sound quality in noisy environments while effectively suppressing ambient noise.

[0024] 3. This invention calculates the audio output parameters of the headphones in real time, including volume and frequency response, based on the current user's sound quality enhancement parameters, to achieve personalized high-fidelity sound effects for the user; the headphones process the input audio signal according to the calculated final frequency response, amplify or attenuate different frequency components, and then adjust the overall gain of the signal according to the calculated final volume, applying the overall gain to the audio signal after the final frequency response adjustment, which greatly improves the sound quality, fidelity efficiency, and accuracy. Attached Figure Description

[0025] To more clearly illustrate the technical solutions in the embodiments of the present invention or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, 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.

[0026] Figure 1 This is a schematic diagram illustrating the steps of an adaptive sound quality enhancement method for headphones according to the present invention; Figure 2 This is a schematic diagram of an adaptive sound quality enhancement system for headphones according to the present invention. Detailed Implementation

[0027] 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.

[0028] Example 1: In view of the aforementioned problems mentioned in the prior art, and in order to solve the above technical problems, as shown in the appendix. Figure 1 As shown: This application provides an adaptive sound quality enhancement method for headphones, including the following steps: S1: The user's hearing characteristic data is collected in real time through the built-in sensor of the earphone. The hearing characteristic data includes the user's ear canal impedance, the user's ear canal sound wave reflectivity, the user's ear canal sound wave absorption rate, the user's ear canal resonant frequency, the air pressure in the ear canal, and the temperature in the ear canal. Ear canal impedance refers to the degree to which the ear canal impedes the propagation of sound waves. By measuring ear canal impedance, we can understand the sound wave reflection and absorption characteristics within the ear canal, which is crucial for optimizing audio signal transmission. Ear canal impedance measurement is typically achieved through a built-in acoustic impedance measurement device in headphones. This device includes a small speaker that emits sound waves of known frequencies and a microphone for detecting reflected sound waves. In one embodiment, the built-in acoustic impedance measurement device in headphones is used to acquire the user's ear canal impedance data in real time. The system emits a series of pure tones of known frequencies into the ear canal through the small speaker and detects the reflected sound waves through the microphone. By analyzing the amplitude and phase of the reflected waves, an ear canal impedance curve is calculated. Based on this curve, the system can identify the impedance characteristics of the ear canal and dynamically adjust the gain of the audio signal to optimize the sound quality in the low and high frequency ranges, making the audio signal more suitable for the user's auditory needs.

[0029] Acoustic reflectivity refers to the proportion of sound waves reflected back when they encounter the walls of the ear canal. High reflectivity can cause audio signals to reflect multiple times within the ear canal, affecting sound clarity. By measuring the acoustic reflectivity of the ear canal in real time, the system can adjust the frequency response of the audio signal to reduce unnecessary reflection interference. In one embodiment, the acoustic sensors built into the headphones measure the acoustic reflectivity and absorptivity of the user's ear canal in real time. The system calculates the reflectivity by emitting sound waves and detecting the intensity of their reflections, and calculates the absorptivity by detecting the difference between the reflected wave and the original emitted wave. Based on this data, the system can adjust the frequency response of the audio signal, reducing gain in high-reflectivity frequency bands to avoid sound distortion, and increasing gain in high-absorption frequency bands to compensate for absorbed energy, thereby achieving comprehensive sound quality optimization.

[0030] Sound absorption rate refers to the degree to which the ear canal wall absorbs sound wave energy. Different users have different ear canal structures and surface characteristics, resulting in individual differences in sound absorption rate. By measuring sound absorption rate, the gain of the audio signal can be adjusted to compensate for the absorbed energy and improve sound quality.

[0031] The resonant frequency of the ear canal is the natural frequency of the ear canal structure and has a significant impact on the resonance enhancement effect of sound. Understanding the resonant frequency of a user's ear canal helps to avoid over-amplification of the resonant frequency band, thereby providing a balanced audio output.

[0032] The air pressure inside the ear canal affects the speed of sound propagation and sound pressure level. By monitoring the air pressure inside the ear canal in real time, the dynamic range of the audio signal can be adjusted to adapt to the auditory impact of air pressure changes. Changes in ear canal temperature affect the speed of sound in the air within the ear canal, thus affecting the frequency response of the sound. By monitoring the ear canal temperature in real time, the system can dynamically adjust the frequency response of the audio signal to ensure stable sound quality. The headphones' built-in sensors measure the resonant frequency and air pressure in the user's ear canal. In one embodiment, the system determines the resonant frequency of the ear canal using a frequency scanning method and monitors changes in air pressure within the ear canal. Based on the resonant frequency, the system adjusts the gain of the resonant frequency band in the audio signal to avoid excessive amplification of the resonant frequency band. Simultaneously, the system dynamically adjusts the dynamic range and frequency response of the audio signal based on the real-time measured air pressure value to adapt to the auditory impact of air pressure changes and ensure stable sound quality.

[0033] S2: Real-time monitoring of ambient noise intensity via an external microphone; In one embodiment, ambient noise monitoring involves capturing sound signals from the surrounding environment using an external microphone, then performing frequency analysis on these signals to determine the intensity of the ambient noise at various frequencies. Based on this noise intensity information, the system can adjust the frequency response and overall volume of the audio signal to suppress noise interference while ensuring clarity and detail in the sound quality. (External microphone working principle) An external microphone is used to capture all sounds in the current environment, including background noise, wind, and human voices. These sound signals are recorded in real time and input as raw data into the system for processing. The captured sound signals are first preprocessed, such as removing high-frequency sharp noise or low-frequency vibration noise. Then, the time-domain signal is converted into a frequency-domain signal using a Fast Fourier Transform (FFT). In this way, the system can identify the intensity N(f) of the noise signal at different frequencies.

[0034] For each frequency f, the system calculates the corresponding noise intensity N(f) and records the maximum noise intensity value across all frequency ranges. This data will be used to adjust the headphone's audio output parameters to ensure clear and high-quality sound even in noisy environments.

[0035] The Fast Fourier Transform (FFT) is a fast algorithm that converts a captured time-domain signal into a frequency-domain signal. Through FFT, the originally complex sound signal is decomposed into the amplitude and phase of different frequency components. The square of the amplitude is typically used to represent the power or intensity at that frequency. By analyzing the FFT results, the system can identify which frequency components have higher noise intensity, thus identifying the frequency location of the main noise source. This information is crucial for subsequent audio processing, especially in active noise suppression (ANC) and frequency response adjustment.

[0036] Based on the obtained noise intensity N(f), the system can reduce the gain of the audio signal in high-noise frequency bands to avoid interference from background noise on the target audio (such as speech or music). Conversely, in lower-noise frequency bands, the system can increase the gain to enhance sound quality details. The system dynamically adjusts the overall volume according to the ambient noise intensity. If a high noise level is detected, the system increases the volume to ensure the user can hear the audio content clearly; conversely, if the ambient noise is low, the volume is appropriately reduced to avoid discomfort to the user due to excessive volume.

[0037] S3: Calculates the current user's sound quality enhancement parameters based on hearing characteristic data and current ambient noise intensity. ;

[0038] in, Based on the base frequency gain value, The impedance value of the user's ear canal at frequency f. Maximum ear canal impedance For the user's ear canal at frequency acoustic emission rate at that location For the user's ear canal at frequency Sound absorption rate at that location The intensity of the current ambient noise at frequency f. This represents the maximum environmental noise intensity. This represents the current air pressure inside the ear canal. This is the reference air pressure inside the ear canal. The current temperature inside the ear canal. This is the reference temperature inside the ear canal; The system calculates the sound quality enhancement parameter E(f) by comprehensively analyzing the user's hearing characteristics (including ear canal impedance, sound wave reflectivity, sound wave absorption rate, resonant frequency, ear canal air pressure, ear canal temperature, etc.) and the current ambient noise intensity. This parameter is used to dynamically adjust the headphone's audio output to achieve personalized, high-fidelity sound effects.

[0039] The fundamental frequency gain is the gain value set by the system based on the ideal state of the sound source under conditions of no other interference. This is the ideal volume setting of the system at different frequencies f, and is used as a reference for subsequent adjustments.

[0040] Ear canal impedance reflects the degree to which the ear canal impedes sound waves of different frequencies. If a user's ear canal has high impedance at a certain frequency f, it means that the sound wave transmission efficiency in this frequency band is low, and the system needs to increase the gain in this frequency band to compensate for the volume loss caused by the impedance. Conversely, if the impedance is low, the gain can be appropriately reduced.

[0041] Using the maximum ear canal impedance as a reference, the system compares the impedance values ​​at each frequency point with the maximum impedance value to determine whether the gain of the corresponding frequency band needs to be adjusted.

[0042] Sound reflectivity is the proportion of sound waves reflected by the ear canal at a specific frequency. High reflectivity can cause multiple reflections and resonances in certain frequency bands, so the system reduces the gain in these bands to avoid sound quality distortion. Sound absorption is the degree to which the ear canal absorbs sound waves. If the absorption rate is high in a certain frequency band, the sound may sound less clear or degraded. The system increases the gain in these frequency bands to compensate for the loss caused by absorption.

[0043] Ambient noise intensity is the current ambient noise level monitored by an external microphone. If the ambient noise is high at a certain frequency band f, the system needs to reduce the gain for that frequency band accordingly to avoid noise interference. If the noise is low, the system can moderately increase the gain in these frequency bands to enhance sound quality. The maximum ambient noise intensity is used as a comparison standard to help the system decide whether to make a larger adjustment to the gain in certain frequency bands.

[0044] Changes in air pressure within the ear canal affect the speed of sound propagation and sound pressure level. If the air pressure is low, the system may need to increase the dynamic range of the audio signal and enhance low-frequency performance to offset the negative impact of air pressure changes on sound quality. Ear canal temperature affects air density and the speed of sound, thus influencing sound quality. The system monitors the ear canal temperature in real time and adjusts the gain in the high-frequency range to ensure that temperature changes do not lead to sound quality degradation.

[0045] The sound quality enhancement parameter E(f) is a comprehensive calculation based on the above parameters. At each frequency point f, the system determines the final gain value for that frequency by combining data such as the fundamental frequency gain, ear canal impedance, sound wave reflectivity, absorptivity, ambient noise intensity, and air pressure and temperature within the ear canal. This gain value will be used to adjust the input audio signal to achieve optimal output sound quality.

[0046] S4: Based on the current user's sound quality enhancement parameters, calculate the headphone's audio output parameters in real time, including volume and frequency response, to achieve personalized high-fidelity sound effects for the user;

[0047]

[0048] It is the final volume output at frequency f. It is the adjusted final frequency response; This is the initial frequency response curve; S5: The headphones process the input audio signal based on the calculated final frequency response, gaining or attenuating different frequency components. Then, based on the calculated final volume, the overall gain of the signal is adjusted and applied to the audio signal after the final frequency response adjustment. In the headphone system, the process of gaining or attenuating different frequency components involves the input audio signal first undergoing frequency decomposition by a digital signal processor (DSP). This step is typically implemented using a Fast Fourier Transform (FFT) or a filter bank to decompose the audio signal into multiple frequency bands (such as low, mid, and high frequencies). Each frequency band is processed individually, and the DSP applies specific gain or attenuation to each band based on the previously calculated final frequency response curve. If a frequency band needs enhancement (e.g., the user has weak hearing in that band or needs to emphasize certain details in the music), the DSP increases the amplitude of the signal in that band. This is usually achieved by multiplying by a gain factor greater than 1. Conversely, if a frequency band needs attenuation (e.g., there is significant noise interference in that band or to avoid resonance), the DSP reduces the amplitude of the signal in that band, i.e., by multiplying by an attenuation factor less than 1. The processed frequency components retain their original phase information to ensure that the overall sound quality of the signal is not distorted. Processing of each frequency band is performed in real time to ensure that the final output audio signal can dynamically adapt to the user's hearing characteristics and environmental noise conditions.

[0049] After processing the gain or attenuation of each frequency band, the system needs to adjust the overall gain of the entire signal based on the calculated final volume. This process ensures the balance and adaptability of the audio signal in terms of volume. The overall gain coefficient is calculated, typically a scaling factor derived from the difference between the target output volume and the input volume. This factor determines the overall amplification or attenuation of all frequency bands. The DSP applies this overall gain coefficient to the audio signal whose frequency response has been adjusted. Specifically, each processed frequency band signal is multiplied by this overall gain coefficient. This operation ensures that while maintaining their relative gain or attenuation relationships, the overall volume reaches the expected target value. After applying the overall gain, the signals from each frequency band are recombined into a complete time-domain audio signal. This signal is then converted to an analog signal by a digital-to-analog converter (DAC) and output through the headphone's speaker unit.

[0050] The headphones first process the input audio signal based on the calculated final frequency response, that is, by applying gain or attenuation to different frequency components individually. Then, the system adjusts the overall gain of the entire signal based on the calculated final volume. Finally, this overall gain is applied to the frequency-response-adjusted audio signal to ensure that the audio output not only meets the user's needs in terms of frequency but also achieves optimal volume.

[0051] The input audio signal is first frequency-decomposed by a digital signal processor (DSP). The DSP breaks down the entire audio signal into different frequency components, allowing each component at frequency f to be processed individually. Based on the final frequency response calculated by the system, the DSP applies specific gain or attenuation to each frequency component. For example, the system might decide to boost the high-frequency signal to enhance detail, while attenuating the low-frequency band to avoid interference from low-frequency noise. Each frequency component, after gain or attenuation, is combined to form a new frequency response signal. After this processing, the frequency response has been adjusted to best suit the user's auditory characteristics and the current environment. This step ensures that the audio signal in each frequency band achieves ideal sound quality.

[0052] The system calculates an overall target final volume value based on the user's hearing characteristics and ambient noise levels. This target value represents the ideal audio output volume the user expects. The system uses this calculated final volume as an overall gain and applies it to the frequency-response-adjusted audio signal. This step involves uniformly amplifying or reducing the entire audio signal to ensure the overall volume meets expectations. For example, if the final volume is too high, the system will increase the overall signal strength; conversely, it will decrease the signal strength. When applying the overall gain, the system checks for any risk of the signal exceeding safe limits (such as excessive volume potentially causing distortion or hearing damage). If such a risk is detected, the system uses a limiter to control the volume within safe limits.

[0053] After frequency response adjustment and overall gain application, the processed digital signal is converted into an analog signal via a digital-to-analog converter (DAC). This analog signal is the final sound signal emitted by the headphones. The processed and converted analog signal is then transmitted to the headphone's speaker unit. The speaker converts the signal into physical sound waves, and the sound the user hears is optimized based on all the previous processing steps, providing the best listening experience in terms of both volume and sound quality.

[0054] In one embodiment, audio processing in a music playback scenario involves a user listening to music at home using headphones. The system detects that the user's hearing is somewhat attenuated in the high-frequency range, while the ambient noise is low. The system adjusts the frequency response based on the user's hearing characteristics, increasing the gain in the high-frequency range to compensate for hearing loss, while appropriately attenuating the low-frequency range to avoid excessive bass. Next, the system calculates a suitable final volume based on the music's dynamic range and applies this volume as the overall gain to the frequency-response-adjusted signal. Ultimately, the music heard by the user is clearer in high-frequency details, while the overall volume is moderate and layered.

[0055] In one embodiment, audio processing in a call scenario involves a user conducting a conference call using headphones in an office. The system detects some low-frequency noise in the environment (such as air conditioning noise), while the user's ear canal has a high sound wave reflectivity in the mid-frequency range. The system adjusts the frequency response of the input audio signal, reducing gain in the low-frequency range to suppress ambient noise and moderately attenuating the mid-frequency range to reduce resonance caused by reflections. The system then calculates the appropriate final volume for the call, ensuring the sound is clear without being over-amplified, and applies this gain-adjusted signal. During the call, the user can clearly hear the other party's voice without being disturbed by ambient noise or excessive volume.

[0056] In one embodiment, audio processing in an outdoor sports scenario involves a user listening to voice navigation through headphones while running outdoors. An external microphone detects wind and traffic noise, and the user's hearing profile indicates slight hearing loss in the low-frequency range. The system first adjusts the frequency response of the input signal, reducing the gain in the low-frequency range to minimize the impact of wind noise, while increasing the gain in the mid-to-high frequency range to ensure the clarity of navigation instructions. Then, the system calculates the final volume suitable for the outdoor environment, ensuring the audio remains clearly audible even in noisy conditions, and applies this gain to the signal. Ultimately, the user can clearly hear navigation instructions while running, and surrounding noise is effectively suppressed.

[0057] S6: The processed digital audio signal is converted into an analog signal via a digital-to-analog converter (DAC) and output through the headphone speaker unit. The DAC is a key component in converting digital signals into analog signals. In an audio system, after receiving the processed digital audio signal, the DAC converts discrete digital sample values ​​into a continuous voltage signal through a series of complex electronic processes. This voltage signal corresponds to the waveform of the sound wave. To ensure sound quality, headphones typically use high-precision DACs. The higher the precision, the closer the converted analog signal is to the original digital signal, resulting in better sound quality. A high-quality DAC can process a large amount of data in a very small time unit, ensuring that the output analog signal has a smooth waveform and rich detail. During the conversion process, the DAC uses a high-precision clock signal to ensure that the time interval between the sampling points of the digital signal remains consistent. The stability of the clock directly affects the sound quality; a good clock source can reduce jitter and improve the accuracy and clarity of the sound. After conversion to an analog signal, the signal is transmitted to the headphone's amplifier module through internal circuitry.

[0058] To avoid signal interference during transmission, high-end headphones typically use shielded cables or short-distance transmission methods to maintain signal integrity. The amplifier module amplifies the weak analog signal to a level sufficient to drive the headphone speakers. The amplification process needs to be highly linear to avoid introducing distortion. The quality of the amplifier directly affects the dynamic range and clarity of the final output sound. The headphone speaker unit converts the amplified analog electrical signal into sound waves. The speaker unit design includes components such as a diaphragm, coil, and magnet. The vibration of the diaphragm directly generates air compression and rarefaction, thus forming sound waves. High-quality headphones typically use lightweight yet rigid diaphragm materials, such as titanium and graphene, which can quickly respond to changes in electrical signals and reproduce the details and dynamics of sound without distortion. Finally, after passing through the speaker unit, the processed sound is accurately delivered to the user's ears. Excellent speaker design can provide balanced sound performance across the entire frequency range, retaining the depth of low frequencies while showcasing the brightness and delicacy of high frequencies.

[0059] In some embodiments, the sensors built into the headphones include an acoustic sensor, a barometric pressure sensor, and a temperature sensor. The acoustic sensor measures the acoustic reflectivity, acoustic absorptivity, and resonant frequency of the user's ear canal in real time. The barometric pressure sensor detects the current air pressure within the ear canal. The temperature sensor monitors the current temperature within the ear canal. The user's ear canal impedance is measured using an acoustic impedance meter. The acoustic sensor in the headphones emits sound waves of a known frequency into the user's ear canal and measures the intensity of the reflected sound waves. The ratio of the intensity of the reflected wave to the intensity of the original emitted wave is used to calculate the acoustic reflectivity of the ear canal. A high reflectivity indicates that the ear canal walls reflect a large amount of sound waves, which may lead to resonance or distortion. The system adjusts the gain of the audio signal based on this data to reduce potential sound quality problems. Simultaneously, the sensor can also calculate the acoustic absorptivity by analyzing the energy difference between the reflected wave and the original emitted wave. The absorptivity reflects the degree to which the ear canal walls absorb sound waves. For frequency bands with high absorptivity, the system increases the gain of these bands to compensate for the volume loss caused by absorption, ensuring the clarity and integrity of the audio signal.

[0060] A barometric pressure sensor monitors the current air pressure inside the user's ear canal in real time. Changes in ear canal pressure affect the speed of sound propagation and sound pressure level. When the ear canal pressure is below normal (e.g., at high altitudes), the system appropriately increases the dynamic range of the audio signal, especially in the low-frequency range, to compensate for the sound quality impact caused by pressure changes. Conversely, if the pressure is high, the system adjusts the gain to ensure that sound quality is not affected by pressure fluctuations.

[0061] An acoustic impedance meter is a device used to measure the characteristics of sound wave transmission within the ear canal, primarily for assessing ear canal impedance. Ear canal impedance refers to the degree to which the ear canal impedes sound wave transmission, reflecting the sound wave reflection and absorption characteristics of the ear canal at different frequencies. By measuring ear canal impedance, the acoustic characteristics of the ear canal can be understood, providing important data for personalized sound quality optimization. The acoustic impedance meter's speaker emits a series of sound wave signals of known frequencies into the user's ear canal. These sound wave signals are typically short, pure tones, with each frequency signal emitted at a specific sound pressure level. As the sound waves propagate within the ear canal, some are reflected back by the canal walls, while others continue to propagate forward or are absorbed. A microphone detects the sound wave signals reflected back from the ear canal in real time and records the intensity and phase difference of these signals. The ratio of the intensity of the reflected wave to the intensity of the emitted wave is used to calculate the reflectivity, while the phase difference provides additional information about the propagation path of the sound wave within the ear canal. Based on the detected reflected wave data, the system calculates the ear canal impedance at each frequency. Impedance is a complex value that includes acoustic resistance (reflecting energy loss) and acoustic reactance (reflecting energy storage). Through these calculations, the system can plot the frequency response curve of the ear canal impedance, showing the impedance characteristics of the ear canal at different frequencies. By analyzing the frequency response curve of the ear canal impedance, the system identifies the sound wave transmission characteristics of the user's ear canal at various frequencies. For frequency bands with higher impedance, the system may increase the gain of the audio signal to compensate for the volume loss due to impedance. For frequency bands with lower impedance, the system may reduce the gain to avoid over-amplification and potential distortion.

[0062] In some embodiments, the real-time monitoring of the current ambient noise intensity via an external microphone includes: capturing raw sound signals in the current environment using an external microphone; performing a fast Fourier transform on the captured raw sound signals; and obtaining the intensity of the ambient noise at each frequency f. The power spectral density of the frequency domain signal obtained through Fast Fourier Transform is calculated, and the maximum power spectral density value is recorded as the maximum value of the ambient noise intensity. The real-time monitoring of the current ambient noise intensity via an external microphone includes: capturing the original sound signal in the current environment using an external microphone; performing a fast Fourier transform on the captured original sound signal; obtaining the intensity of the ambient noise at each frequency f; and calculating the power spectral density of the frequency domain signal obtained through the fast Fourier transform, recording the maximum power spectral density value as the maximum value of the ambient noise intensity.

[0063] After capturing the raw audio signal, the system typically preprocesses it to remove irrelevant noise or interference. For example, high-pass or low-pass filters might be applied to remove extremely low- or high-frequency noise. Additionally, denoising algorithms might be used to reduce sudden, sharp noises. The raw audio signal is a time-domain signal, meaning it's the change in sound intensity recorded by the microphone over time. To analyze the distribution of ambient noise across different frequencies, the system performs a Fast Fourier Transform (FFT) on the preprocessed time-domain signal. FFT is a mathematical algorithm that converts a time-domain signal into a frequency-domain signal, revealing the amplitude and phase of different frequency components. The result of FFT is a frequency spectrum, where each frequency point corresponds to an amplitude representing the intensity of that frequency component. The frequency resolution of FFT depends on the signal's sampling rate and the number of FFT points. A high sampling rate and a large number of FFT points provide higher frequency resolution, enabling the system to accurately analyze the spectral distribution of noise.

[0064] The system can calculate the noise intensity at each frequency f using the frequency domain signal obtained through FFT. These intensity values ​​are typically obtained by squared the amplitude at the corresponding frequency, representing the power (power spectral density) at that frequency. This step reveals the spectral characteristics of the ambient noise, showing the distribution of noise across different frequencies. The calculated noise intensities at each frequency f form a spectrum. This spectrum shows the intensity distribution of noise across different frequency ranges. For example, in a street environment, the spectrum might show strong noise intensity in the low-frequency range (e.g., 50Hz to 200Hz), mainly generated by car engine sounds; while in an office environment, the mid-to-high frequency range (e.g., 500Hz to 2000Hz) might have strong noise intensity, mainly generated by human voices and electronic devices. Power spectral density (PSD) is the frequency distribution density of noise signal intensity, typically expressed as power per unit frequency bandwidth. The system calculates the power spectral density at each frequency point by analyzing the FFT results. This process involves squared and normalized the amplitude of the frequency domain signal to obtain the noise power within each frequency bandwidth.

[0065] In some embodiments, the headphones process the input audio signal based on the calculated final frequency response, including: based on the final frequency response. A digital filter, including an FIR or IIR filter, is constructed and configured to apply specific gain or attenuation to the audio signal components at different frequencies (f). The input audio signal undergoes frequency decomposition by a digital signal processing module, and the decomposed frequency components are then fed into the corresponding filters for processing. The filtered frequency components are then recombined into a complete audio signal. This process resynthesizes signals from all frequency bands to form a continuous time-domain signal. The recombined audio signal is transmitted to the headphone's output module, undergoes digital-to-analog conversion (DAC), and is then played out through the speaker unit. This signal has been optimized based on the user's hearing characteristics and ambient noise conditions to provide a high-fidelity, personalized sound experience.

[0066] The system can use two types of digital filters to achieve frequency response adjustment: Finite Impulse Response (FIR) filters and Infinite Impulse Response (IIR) filters. FIR filters have a linear phase response, enabling precise frequency response control, but typically require more computational resources. FIR filters are commonly used in applications with high phase requirements, such as music processing. IIR filters are computationally efficient, allowing for complex frequency responses with fewer coefficients, making them suitable for applications with high computational demands. The filter coefficients are designed based on the final frequency response curve. This design process can be implemented using various algorithms, such as the window function method, frequency sampling method, or least squares error method. The designed filter coefficients are then downloaded to the headphone's digital signal processor (DSP) for real-time audio processing.

[0067] In some embodiments, adjusting the overall gain of the signal based on the calculated final volume includes: adjusting the final volume at each frequency f. The overall target output volume is calculated by weighted averaging. , target output volume Average volume of the input audio signal In comparison, determining the global gain coefficient .

[0068] In some embodiments, applying the overall gain to the audio signal after final frequency response adjustment includes, in a digital signal processor, applying the global gain coefficient... As a scaling factor, it is multiplied point-by-point by each frequency component of the audio signal after final frequency response adjustment. The multiplication operation is performed at each sample point or each frequency component of the audio signal to ensure that the overall volume of the signal is proportional to the global gain coefficient. The scale is magnified or reduced, and then converted into an analog signal output to the headphone speaker unit.

[0069] This application also provides an adaptive sound enhancement system for headphones, such as Figure 2 As shown, the system includes: an acoustic sensor module for real-time measurement of the acoustic reflectivity, acoustic absorptivity, and resonant frequency of the user's ear canal. This module acquires the acoustic characteristics of the ear canal by emitting sound waves into it and detecting the reflected sound signals. The acoustic sensor module is directly connected to a digital signal processor (DSP), transmitting the measured acoustic data to the DSP for analysis and processing. A barometric pressure sensor module detects the current air pressure within the ear canal. The barometric pressure sensor monitors changes in ear canal pressure in real time so that the system can adjust the dynamic range and frequency response of the audio signal. The barometric pressure sensor module is connected to the DSP via an analog-to-digital converter (ADC), converting the barometric pressure signal into digital data and transmitting it to the DSP for processing. A temperature sensor module monitors the current temperature within the ear canal. The temperature data is used to adjust the frequency response of the audio signal, particularly gain adjustment in the high-frequency range. The temperature sensor module is also connected to the DSP via an ADC, converting the temperature signal into digital data and transmitting it to the DSP for processing. An external microphone module captures raw sound signals from the current environment for real-time monitoring of ambient noise intensity. The microphone signals are subjected to a Fast Fourier Transform (FFT) to analyze the frequency distribution of the ambient noise. An external microphone module connects directly to the DSP. The audio signal captured by the microphone is transmitted digitally to the DSP for frequency domain analysis and noise intensity calculation. The DSP is the core processing unit of the digital signal processor (DSP) system. It is responsible for real-time processing of all sensor data, including acoustic characteristic analysis, air pressure and temperature data processing, and noise intensity calculation. The DSP also controls digital filters to gain or attenuate the audio signal based on the calculated final frequency response. The DSP connects to all sensor modules and filters, receives input data from the sensors, performs complex signal processing tasks, and sends the processed data to the subsequent audio output module. The digital filter module performs gain or attenuation operations on the input audio signal based on the final frequency response calculated by the DSP. The digital filter can be an FIR filter or an IIR filter, used for different audio processing needs. The digital filter module is tightly coupled to the DSP, receiving control signals from the DSP and adjusting the input audio signal according to the calculated frequency response. The digital-to-analog converter (DAC) converts the digital audio signal processed by the digital filter into an analog signal, ready for output through the headphone's speaker unit. The DAC is directly connected to the digital filter module, receives the processed digital audio signal, and converts it into an analog signal for output. The speaker unit converts the analog signal output from the DAC into sound waves, which are then emitted through headphones. The speaker unit is the final audio output device and directly affects the user's listening experience. The speaker unit is directly connected to the DAC, receiving the analog signal and converting it into physical sound waves.

[0070] Acquisition Module: This module collects real-time hearing characteristic data from the user via sensors built into the headphones. This data includes ear canal impedance, ear canal sound wave reflectivity, ear canal sound wave absorption, ear canal resonant frequency, ear canal air pressure, and ear canal temperature. Modern sensors and processors are quite small and can typically be integrated into headphones. Sensors using MEMS (Micro-Electro-Mechanical Systems) technology are well-suited for integration into headphones due to their small size and low power consumption. Advanced System-on-Chip (SoC) integrates multiple functional modules onto a single chip, including DSPs, DACs, ADCs, and wireless communication modules. This SoC design significantly reduces the required physical space and power consumption; it employs MEMS microphones, such as Bosch's IM69D130 from the BMD100, which can not only be used for sound capture but also, through design optimization, for detecting the reflection and absorption characteristics of sound waves in the ear canal; Bosch's BMP388 barometric pressure sensor, measuring 2.0mm x 2.0mm x 0.75mm, is suitable for integration into headphones; and Texas Instruments' TMP117 temperature sensor, measuring 1.5mm x 1.5mm, features high accuracy and low power consumption.

[0071] Noise acquisition module: Monitors the current ambient noise level in real time via an external microphone; Sound quality enhancement parameter calculation module: Calculates the current user's sound quality enhancement parameters based on hearing characteristic data and the current ambient noise level. ;

[0072] in, Based on the base frequency gain value, The impedance value of the user's ear canal at frequency f. Maximum ear canal impedance For the user's ear canal at frequency acoustic emission rate at that location For the user's ear canal at frequency Sound absorption rate at that location The intensity of the current ambient noise at frequency f. This represents the maximum environmental noise intensity. This represents the current air pressure inside the ear canal. This is the reference air pressure inside the ear canal. The current temperature inside the ear canal. This is the reference temperature inside the ear canal; Audio output parameter calculation module: Based on the current user's sound quality enhancement parameters, calculate the headphone's audio output parameters in real time, including volume and frequency response, to achieve personalized high-fidelity sound effects for the user;

[0073]

[0074] It is the final volume output at frequency f. It is the adjusted final frequency response; This is the initial frequency response curve; Input audio signal processing module: The headphones process the input audio signal according to the calculated final frequency response, gain or attenuate different frequency components, and then adjust the overall gain of the signal according to the calculated final volume, and apply the overall gain to the audio signal after the final frequency response adjustment. Output module: Converts the processed digital audio signal into an analog signal via a digital-to-analog converter and outputs it through the headphone speaker unit.

[0075] In some embodiments, the headphones process the input audio signal based on the calculated final frequency response, including: based on the final frequency response. A digital filter is constructed, including an FIR filter or an IIR filter, configured to apply specific gain or attenuation to the audio signal components at different frequencies f; the input audio signal is frequency-decomposed by a digital signal processing module, and the decomposed frequency components are input to the corresponding filters for processing; the frequency components after filter processing are recombined into a complete audio signal.

[0076] In some embodiments, adjusting the overall gain of the signal based on the calculated final volume includes: adjusting the final volume at each frequency f. The overall target output volume is calculated by weighted averaging. , target output volume Average volume of the input audio signal In comparison, determining the global gain coefficient .

[0077] In some embodiments, applying the overall gain to the audio signal after final frequency response adjustment includes, in a digital signal processor, applying the global gain coefficient... As a scaling factor, it is multiplied point-by-point by each frequency component of the audio signal after final frequency response adjustment. The multiplication operation is performed at each sample point or each frequency component of the audio signal to ensure that the overall volume of the signal is proportional to the global gain coefficient. The scale is magnified or reduced, and then converted into an analog signal output to the headphone speaker unit.

[0078] This invention provides an adaptive sound quality enhancement method and system for headphones, which achieves the following beneficial technical effects: 1. This invention first collects the user's hearing characteristic data in real time through the built-in sensors of the headphones and monitors the current ambient noise intensity in real time through an external microphone. Second, it calculates the current user's sound quality enhancement parameters based on the hearing characteristic data and the current ambient noise intensity. Then, based on the current user's sound quality enhancement parameters, it calculates the headphones' audio output parameters in real time, including volume and frequency response. Third, the headphones process the input audio signal according to the calculated final frequency response, applying the overall gain to the audio signal adjusted by the final frequency response. Finally, the processed digital audio signal is converted into an analog signal through a digital-to-analog converter and output through the headphone speaker unit. This application significantly improves the high fidelity of sound quality by using the current user's sound quality enhancement parameters to calculate the headphones' audio output parameters in real time and applying the overall gain to the audio signal adjusted by the final frequency response. By considering factors such as ear canal impedance, sound wave reflectivity, and absorption rate, it accurately analyzes the user's hearing characteristics, enabling the audio output to be optimized according to individual differences, providing a sound quality experience that better meets the user's needs and greatly enhancing the user experience.

[0079] 2. This invention calculates the current user's sound quality enhancement parameters based on hearing characteristic data and the current ambient noise intensity. ;

[0080] Taking into account the fundamental frequency gain, the impedance of the user's ear canal at frequency f, and the maximum ear canal impedance, the value of the user's ear canal at frequency f is... The sound wave emissivity at a certain location, and the user's ear canal at a certain frequency. The system comprehensively considers the sound wave absorption rate at a given frequency, the intensity of current ambient noise at frequency f, the maximum intensity of ambient noise, the current air pressure inside the ear canal, the reference air pressure inside the ear canal, the current temperature inside the ear canal, and the reference temperature inside the ear canal. By taking into account the ear canal structure and the influence of ambient noise in real time, the system makes comprehensive adjustments, greatly improving the noise cancellation efficiency of the headphones and significantly enhancing the user experience. This application also incorporates an external microphone to monitor ambient noise in real time, obtains the noise intensity at various frequencies through Fast Fourier Transform (FFT), and dynamically adjusts the frequency response and overall volume based on the user's hearing characteristics. This maintains high-fidelity sound quality in noisy environments while effectively suppressing ambient noise.

[0081] 3. This invention calculates the audio output parameters of the headphones in real time, including volume and frequency response, based on the current user's sound quality enhancement parameters, to achieve personalized high-fidelity sound effects for the user; the headphones process the input audio signal according to the calculated final frequency response, amplify or attenuate different frequency components, and then adjust the overall gain of the signal according to the calculated final volume, applying the overall gain to the audio signal after the final frequency response adjustment, which greatly improves the sound quality, fidelity efficiency, and accuracy.

[0082] The above provides a detailed description of an adaptive sound quality enhancement method and system for headphones. Specific examples have been used to illustrate the principles and implementation methods of the present invention. The descriptions of the above embodiments are only for the purpose of helping to understand the core ideas of the present invention. At the same time, for those skilled in the art, there will be changes in the specific implementation methods and application scope based on the ideas and methods of the present invention. Therefore, the content of this specification should not be construed as a limitation of the present invention.

Claims

1. A method for adaptive sound quality enhancement in headphones, characterized in that, Including the following steps: S1: The user's hearing characteristic data is collected in real time through the built-in sensor of the earphone. The hearing characteristic data includes the user's ear canal impedance, the user's ear canal sound wave reflectivity, the user's ear canal sound wave absorption rate, the user's ear canal resonant frequency, the air pressure in the ear canal, and the temperature in the ear canal. S2: Real-time monitoring of ambient noise intensity via an external microphone; S3: Calculates the current user's sound quality enhancement parameters based on hearing characteristic data and current ambient noise intensity. ; ； in, Based on the base frequency gain value, The impedance value of the user's ear canal at frequency f. Maximum ear canal impedance For the user's ear canal at frequency acoustic emission rate at that location For the user's ear canal at frequency Sound absorption rate at that location The intensity of the current ambient noise at frequency f. This represents the maximum environmental noise intensity. This represents the current air pressure inside the ear canal. This is the reference air pressure inside the ear canal. The current temperature inside the ear canal. This is the reference temperature inside the ear canal; S4: Based on the current user's sound quality enhancement parameters, calculate the headphone's audio output parameters in real time, including volume and frequency response, to achieve personalized high-fidelity sound effects for the user; ； ； It is the final volume output at frequency f. It is the adjusted final frequency response; This is the initial frequency response curve; S5: The headphones process the input audio signal according to the calculated final frequency response, gain or attenuate different frequency components, and then adjust the overall gain of the signal according to the calculated final volume, and apply the overall gain to the audio signal after the final frequency response adjustment. S6: Converts the processed digital audio signal into an analog signal via a digital-to-analog converter and outputs it through the headphone speaker unit.

2. The adaptive sound quality enhancement method for headphones as described in claim 1, characterized in that, The built-in sensors in the earphone include an acoustic sensor, a barometric pressure sensor, and a temperature sensor. The acoustic sensor is used to measure the sound wave reflectivity, sound wave absorptivity, and resonant frequency of the user's ear canal in real time. The barometric pressure sensor is used to detect the current air pressure in the ear canal. The temperature sensor is used to monitor the current temperature in the ear canal. The user's ear canal impedance is measured by an acoustic impedance meter.

3. The adaptive sound quality enhancement method for headphones as described in claim 1, characterized in that, The real-time monitoring of current ambient noise intensity via an external microphone includes: capturing raw sound signals from the current environment using an external microphone; performing a Fast Fourier Transform on the captured raw sound signals; and obtaining the intensity of the ambient noise at various frequencies f. The power spectral density of the frequency domain signal obtained through Fast Fourier Transform is calculated, and the maximum power spectral density value is recorded as the maximum value of the environmental noise intensity. .

4. The adaptive sound quality enhancement method for headphones as described in claim 1, characterized in that, The headphones process the input audio signal based on the calculated final frequency response, including: based on the final frequency response. A digital filter is constructed, including an FIR filter or an IIR filter, configured to apply specific gain or attenuation to the audio signal components at different frequencies f; the input audio signal is frequency-decomposed by a digital signal processing module, and the decomposed frequency components are input to the corresponding filters for processing; the frequency components after filter processing are recombined into a complete audio signal.

5. The adaptive sound quality enhancement method for headphones as described in claim 1, characterized in that, The overall gain of the calculated final volume adjustment signal includes: the final volume at each frequency f. The overall target output volume is calculated by weighted averaging. , target output volume Average volume of the input audio signal In comparison, determining the global gain coefficient .

6. The adaptive sound quality enhancement method for headphones as described in claim 5, characterized in that, The application of the overall gain to the audio signal after final frequency response adjustment includes, in a digital signal processor, applying the global gain coefficient... As a scaling factor, it is multiplied point-by-point by each frequency component of the audio signal after final frequency response adjustment. The multiplication operation is performed at each sample point or each frequency component of the audio signal to ensure that the overall volume of the signal is proportional to the global gain coefficient. The scale is magnified or reduced, and then converted into an analog signal output to the headphone speaker unit.

7. An adaptive sound enhancement system for headphones, characterized in that, include: Acquisition module: The earphone's built-in sensors collect the user's hearing characteristic data in real time. The hearing characteristic data includes the user's ear canal impedance, the user's ear canal sound wave reflectivity, the user's ear canal sound wave absorption rate, the user's ear canal resonant frequency, the air pressure inside the ear canal, and the temperature inside the ear canal. Noise acquisition module: Monitors the current ambient noise level in real time via an external microphone; Sound quality enhancement parameter calculation module: Calculates the current user's sound quality enhancement parameters based on hearing characteristic data and the current ambient noise level. ; ； in, Based on the base frequency gain value, The impedance value of the user's ear canal at frequency f. Maximum ear canal impedance For the user's ear canal at frequency acoustic emission rate at that location For the user's ear canal at frequency Sound absorption rate at that location The intensity of the current ambient noise at frequency f. This represents the maximum environmental noise intensity. This represents the current air pressure inside the ear canal. This is the reference air pressure inside the ear canal. The current temperature inside the ear canal. This is the reference temperature inside the ear canal; Audio output parameter calculation module: Based on the current user's sound quality enhancement parameters, calculate the headphone's audio output parameters in real time, including volume and frequency response, to achieve personalized high-fidelity sound effects for the user; ； ； It is the final volume output at frequency f. It is the adjusted final frequency response; This is the initial frequency response curve; Input audio signal processing module: The headphones process the input audio signal according to the calculated final frequency response, gain or attenuate different frequency components, and then adjust the overall gain of the signal according to the calculated final volume, and apply the overall gain to the audio signal after the final frequency response adjustment. Output module: Converts the processed digital audio signal into an analog signal via a digital-to-analog converter and outputs it through the headphone speaker unit.

8. The adaptive sound enhancement system for headphones as described in claim 7, characterized in that, The headphones process the input audio signal based on the calculated final frequency response, including: based on the final frequency response. A digital filter is constructed, including an FIR filter or an IIR filter, configured to apply specific gain or attenuation to the audio signal components at different frequencies f; the input audio signal is frequency-decomposed by a digital signal processing module, and the decomposed frequency components are input to the corresponding filters for processing; the frequency components after filter processing are recombined into a complete audio signal.

9. The adaptive sound enhancement system for headphones as described in claim 7, characterized in that, The overall gain of the calculated final volume adjustment signal includes: the final volume at each frequency f. The overall target output volume is calculated by weighted averaging. , target output volume Average volume of the input audio signal In comparison, determining the global gain coefficient .

10. The adaptive sound enhancement system for headphones as described in claim 9, characterized in that, The application of the overall gain to the audio signal after final frequency response adjustment includes, in a digital signal processor, applying the global gain coefficient... As a scaling factor, it is multiplied point-by-point by each frequency component of the audio signal after final frequency response adjustment. The multiplication operation is performed at each sample point or each frequency component of the audio signal to ensure that the overall volume of the signal is proportional to the global gain coefficient. The scale is magnified or reduced, and then converted into an analog signal output to the headphone speaker unit.

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