Audio processing methods for compensating for human hearing signals

Abstract

This invention discloses an audio processing method for compensating hearing signals in the human ear, belonging to the field of hearing aids. Its key technical points include: preparing an audio acquisition device to acquire sound signals within the audible range of the human ear; preprocessing the acquired raw audio signals; performing frequency analysis on the audio signals to identify and extract different frequency components; and adjusting the gain of signals at different frequencies according to the hearing curve of the target audience to ensure appropriate compensation for the audio signals in each frequency band. Through frequency analysis and gain adjustment, this method can provide personalized audio compensation for the hearing curves of different hearing-impaired individuals, effectively improving their hearing experience and language comprehension. By combining multiple advanced technologies, it achieves efficient and personalized hearing compensation, providing hearing-impaired individuals with a better auditory experience and quality of life.

Description

Technical Field

[0001] This invention relates to the field of hearing aid technology, and more particularly to an audio processing method for compensating for human hearing signals. Background Technology

[0002] The background technologies for audio processing methods in the field of hearing aids, particularly those related to human hearing signal compensation, mainly include digital signal processing, artificial intelligence, and machine learning. These technologies analyze the characteristics and causes of hearing loss, as well as the sound characteristics in different environments, and use algorithms and computational techniques to process and manipulate sound signals to compensate for hearing loss, improve speech clarity and intelligibility, and enhance the hearing experience and quality of life for people with hearing impairments.

[0003] With the increasing demand for hearing aids, high-fidelity, low-latency audio processing technology has a broad market prospect. It is suitable for various applications requiring high-quality real-time audio transmission, thus creating an urgent need for a high-fidelity, low-latency audio processing technology that can provide a better user experience. Summary of the Invention

[0004] The purpose of this invention is to address the problem of generally poor audio transmission quality in existing hearing aids, and to propose an audio processing method for compensating for human hearing signals.

[0005] To achieve the above objectives, the present invention adopts the following technical solution:

[0006] An audio processing method for compensating for human hearing signals includes the following steps;

[0007] S1. Prepare audio acquisition equipment to collect sound signals within the range of human hearing.

[0008] S2. Preprocess the acquired raw audio signal;

[0009] S3. Perform frequency analysis on the audio signal to identify and extract different frequency components;

[0010] S4. Adjust the gain of signals at different frequencies according to the hearing curve of the target audience to ensure that the audio signal is properly compensated in each frequency band.

[0011] S5. Compress the processed audio signal while maintaining a low latency;

[0012] S6. Transmit the encoded audio data via the RTP transmission protocol;

[0013] S7. Use the corresponding decoder to decode the audio data into the original format and play it through the audio playback device;

[0014] S8. In real-time communication scenarios, a feedback mechanism is introduced to allow users to adjust the audio quality in real time to meet their personal preferences and listening needs.

[0015] Preferably, the audio acquisition device includes a microphone, and the microphone has a microphone board installed inside.

[0016] Preferably, during the sound acquisition process, the equipment is kept in optimal condition to avoid noise and distortion.

[0017] Preferably, the preprocessing in S2 includes noise reduction, equalization, and dynamic range compression.

[0018] Preferably, the encoded audio data can be input into a USB flash drive or stored in a mobile phone in MP3 or WAV format.

[0019] Compared with the prior art, the present invention provides an audio processing method for compensating for human hearing signals, which has the following beneficial effects:

[0020] 1. Improve hearing compensation effect: Through frequency analysis and gain adjustment, this method can provide personalized audio compensation for the hearing curves of different hearing-impaired individuals, effectively improving their hearing experience and language comprehension ability.

[0021] 2. Reduce noise and distortion: During the audio acquisition process, by adjusting the device status and preprocessing steps (such as noise reduction, equalization, and dynamic range compression), environmental noise and distortion generated by the device itself can be reduced, thereby improving audio quality and clarity.

[0022] 3. High-efficiency compression and low-latency transmission: By compressing audio signals, data volume can be reduced while maintaining audio quality, thus improving transmission efficiency. Simultaneously, through appropriate compression algorithms and transmission protocols, audio transmission latency can be reduced, providing users with a smoother listening experience.

[0023] 4. Convenient storage and portability: The encoded audio data can be stored in MP3 or WAV format on portable devices such as USB flash drives or mobile phones, allowing users to use hearing aids for hearing compensation anytime and anywhere, improving the convenience and flexibility of use.

[0024] 5. Real-time Adjustment and Personalized Settings: With the introduction of a feedback mechanism, users can adjust the audio quality in real time according to their personal preferences and hearing needs, achieving personalized hearing compensation. This helps improve user satisfaction and adaptability.

[0025] In summary, the application of audio processing methods for human hearing signal compensation in the field of hearing aids, by combining multiple advanced technologies, has achieved efficient and personalized hearing compensation, providing hearing-impaired individuals with a better auditory experience and quality of life. Detailed Implementation

[0026] The technical solutions in the embodiments of the present invention will be clearly and completely described below. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments.

[0027] An audio processing method for compensating for human hearing signals includes the following steps;

[0028] S1. Prepare audio acquisition equipment to collect sound signals within the range of human hearing.

[0029] S2. Preprocess the acquired raw audio signal;

[0030] S3. Perform frequency analysis on the audio signal to identify and extract different frequency components;

[0031] S4. Adjust the gain of signals at different frequencies according to the hearing curve of the target audience to ensure that the audio signal is properly compensated in each frequency band.

[0032] S5. Compress the processed audio signal while maintaining a low latency;

[0033] S6. Transmit the encoded audio data via the RTP transmission protocol;

[0034] S7. Use the corresponding decoder to decode the audio data into the original format and play it through the audio playback device;

[0035] S8. In real-time communication scenarios, a feedback mechanism is introduced to allow users to adjust the audio quality in real time to meet their personal preferences and listening needs.

[0036] In this embodiment, the audio acquisition device includes a microphone, and the microphone has a microphone board installed inside.

[0037] In this embodiment, during the sound acquisition process, the device is ensured to be in optimal condition to avoid noise and distortion.

[0038] In this embodiment, the preprocessing in S2 includes noise reduction, equalization, and dynamic range compression.

[0039] In this embodiment, the encoded audio data can be input into a USB flash drive or stored in a mobile phone using MP3 or WAV format.

[0040] Furthermore, this method, once summarized, can be simplified to the following steps.

[0041] Initialization phase: Configure the necessary structures such as audio acquisition equipment, preprocessing module, and frequency analysis module.

[0042] Data acquisition: Capturing raw audio data using audio acquisition devices.

[0043] Signal preprocessing: Performing preprocessing operations such as noise reduction and equalization on the acquired audio data.

[0044] Frequency analysis: Using techniques such as Fourier transform to perform spectrum analysis on audio data.

[0045] Hearing curve matching: Adjust the gain of signals at different frequencies according to the hearing curves of the target audience.

[0046] Low-latency coding: The processed audio data is compressed and coded with low latency.

[0047] Data transmission and storage: Transmitting or storing encoded audio data using appropriate protocols and media formats.

[0048] Decoding and playback: At the receiving end, the encoded audio data is decoded into its original format and played through a playback device.

[0049] Feedback and Adjustments: Adjust the audio quality based on user feedback to meet user needs.

[0050] End: Complete audio processing and close related structures.

[0051] Example 2,

[0052] The signal processing steps, such as the Fourier transform in Example 1, can be as follows:

[0053] Fourier transform (time domain graph → frequency domain graph) → frequency domain graph sorting → remove signal of specified frequency → restore frequency domain graph in order → inverse Fourier transform (frequency domain graph → time domain graph) → restore time domain graph by taking the modulus.

[0054] After completing the above basic steps, analyze the frequency domain information of the signal and perform operations on it (such as filtering out low or high frequencies), and then perform an inverse Fourier transform back to the original signal, thereby achieving signal processing.

[0055] Example 3,

[0056] The microphone has an internal microphone board, which can be a PCI8822 board: a high-precision data acquisition card designed for testing audio and vibration signals. This board provides two synchronous analog input channels, 24-bit resolution, and a maximum sampling rate of 204.8 kS / s. Each channel integrates an independent IEPE excitation source, which can realize signal conditioning related to accelerometer and microphone. It supports both AC and DC coupling methods, and each channel can be configured independently via software.

[0057] Alternatively, the YCWL-A1 intelligent data acquisition unit can be selected as the radio receiver: it adopts a fully industrialized hardware design, suitable for diverse and complex outdoor working environments. The unit can directly connect to the operator's network signal, helping users save on wiring costs and eliminating the hassle of large-scale construction. Simultaneously, the instrument is designed with GPS positioning and GPRS transmission modes to meet users' needs for real-time, dynamic monitoring of terminal devices. The unit supports massive data upload and analysis, fast and stable; it is scalable and easy to deploy and maintain.

[0058] Example 4

[0059] This method requires regular maintenance and updates to ensure system stability and security. Furthermore, as market and user needs change, the system also needs continuous upgrades and improvements to its functional modules and services.

[0060] Example 5

[0061] The method of using Opus low-latency audio coding is as follows:

[0062] S1. Create an encoder state and allocate memory for it.

[0063] S2. Use the opus_encoder_init() function to initialize the encoder state.

[0064] S3. Use the opus_encoder_ctl() function to change the encoder's parameter settings.

[0065] S4. Use the opus_encode() function to encode the audio data.

[0066] S5. Use the opus_packet_get_samples_per_frame() function to get the number of samples per frame.

[0067] S6. Use the opus_decoder_get_nb_samples() function to get the number of available samples for the decoder.

[0068] S7. Use the opus_decode() function to decode the encoded data.

[0069] Example 6

[0070] The operation steps of transmitting the encoded audio data via the RTP transmission protocol in this invention are as follows:

[0071] Initiate a session: Initiate a session using the SIP protocol.

[0072] Encoding: Encode audio data into RTP packets using hardware or software encoding.

[0073] Transmission: Transmitting RTP packets via the RTP transport protocol.

[0074] Decoding: At the receiving end, RTP packets are decoded into audio data using hardware or software decoding.

[0075] End Session: End the session using the SIP protocol.

[0076] Example 7

[0077] The steps for decoding the encoded audio data into its original format at the receiving end and playing it through a playback device are as follows:

[0078] Receive RTP packets: Receive RTP packets transmitted from the network.

[0079] Decoding RTP packets: Use a decoding library to decode RTP packets and restore the original audio data.

[0080] Play audio data: Send the decoded audio data to the playback device for playback.

[0081] Example 8

[0082] Introducing a feedback mechanism allows users to adjust audio quality in real time to meet their personal preferences and listening needs. The steps are as follows:

[0083] 1. Collect user preferences and hearing needs:

[0084] Provide an interface on the application or device that allows users to input their audio preferences and listening needs. This can include preferences for audio clarity, volume, timbre, and so on.

[0085] A preset option, such as "high clarity" or "moderate volume", can be provided to simplify the user's selection.

[0086] 2. Create an audio quality adjustment interface:

[0087] Design an intuitive interface that allows users to adjust audio quality in real time. This interface should be easy to understand and operate.

[0088] Controls such as sliders, buttons, or drop-down menus can be provided to allow users to quickly adjust audio parameters.

[0089] 3. Adjust audio quality in real time:

[0090] When a user adjusts audio parameters through the interface, the receiving end should receive these parameters in real time.

[0091] Based on the parameters set by the user, the decoder should adjust the decoded audio data accordingly to match the user's preferences and listening needs.

[0092] 4. Feedback Mechanism:

[0093] A feedback mechanism is introduced, allowing users to evaluate and adjust the quality of the currently playing audio. This can be achieved by displaying audio quality metrics on the interface or providing a feedback button.

[0094] If users are not satisfied with the current audio quality, they can adjust the parameters or select other preset options to obtain a better listening experience.

[0095] 5. Continuous optimization:

[0096] We continuously optimize audio processing algorithms and parameters based on user feedback and preference data.

[0097] Software updates can be released regularly to improve audio quality and meet the personalized needs of more users.

[0098] 6. Privacy Protection:

[0099] Given that user preferences and hearing needs may involve personal privacy, ensure that privacy regulations are followed when collecting, processing, and storing this data.

[0100] Provides transparent, configurable privacy settings so users understand how their data is used and protected.

[0101] 7. User Support and Education:

[0102] Provide user support to help resolve issues users encounter while adjusting audio quality and settings.

[0103] Educate users on how to make the most of these features and improve their audio experience through help documentation, tutorial videos, and other means.

[0104] The above are merely preferred embodiments of the present invention, but the scope of protection of the present invention is not limited thereto. Any equivalent substitutions or modifications made by those skilled in the art within the scope of the technology disclosed in the present invention, based on the technical solution and inventive concept of the present invention, should be covered within the scope of protection of the present invention.

Claims

1. An audio processing method for compensating for human hearing signals, characterized in that, This processing method Includes the following steps; S1. Prepare audio acquisition equipment to collect sound signals within the range of human hearing. S2. Preprocess the acquired raw audio signal; S3. Perform frequency analysis on the audio signal to identify and extract different frequency components; S4. Adjust the gain of signals at different frequencies according to the hearing curve of the target audience to ensure that the audio signal is properly compensated in each frequency band. S5. Compress the processed audio signal while maintaining a low latency; S6. Transmit the encoded audio data via the RTP transmission protocol; S7. Use the corresponding decoder to decode the audio data into the original format and play it through the audio playback device; S8. In real-time communication scenarios, a feedback mechanism is introduced to allow users to adjust the audio quality in real time to meet their personal preferences and listening needs.

2. The audio processing method for compensating human hearing signals according to claim 1, characterized in that, The audio acquisition device includes a microphone, and the microphone has a microphone board installed inside.

3. The audio processing method for compensating human hearing signals according to claim 1, characterized in that, During sound acquisition, ensure the equipment is in optimal condition to avoid noise and distortion.

4. The audio processing method for compensating human hearing signals according to claim 2, characterized in that, Preprocessing in S2 includes noise reduction, equalization, and dynamic range compression.

5. The audio processing method for compensating human hearing signals according to claim 3, characterized in that, The encoded audio data can be input into a USB flash drive or stored in a mobile phone using MP3 or WAV format.