185 results about "Digital audio signals" patented technology

The invention relates to the technical field of audio signal processing, and discloses a high-sampling-rate audio analysis optimization method and system, a storage medium and equipment, and the method comprises the steps: upgrading an FFTW (Fast Fourier Transform) library, and building an optimized memory management mechanism; carrying out adaptive framing processing on the input audio data, and optimizing the data access efficiency by adopting a multi-stage cache system; the calculation precision of the FFT fast Fourier transform is dynamically adjusted, and the calculation of the FFT fast Fourier transform is optimized by pre-calculating and caching an FFTW plan; performing clock synchronization optimization and jitter elimination on the COAX digital audio signal; establishing a multi-dimensional result cache matrix and an intelligent multiplexing decision tree, and reducing repeated operation through intelligent cache prediction and similarity calculation; system performance indexes are monitored in real time, adaptive parameter adjustment is realized based on an optimization objective function, audio quality and system stability are monitored in combination with subjective and objective evaluation mechanisms, repeated operation is remarkably reduced, and calculation resource allocation is optimized on the premise of ensuring sound quality.

The invention relates to the technical field of artificial intelligence and digital audio signal processing, and particularly discloses a Karaoke all-in-one machine intelligent tuning method based on AI sound effect optimization, and the method comprises the steps: obtaining an original mixed audio signal sung by a user, and carrying out the preprocessing; processing the preprocessed audio by using a multi-source audio separation model based on a deep neural network, and extracting a human voice track and accompaniment components; extracting time-frequency domain features from the human voice track, and performing quantitative evaluation on the voice tension degree of the user in combination with the trained voice state recognition model; dynamically adjusting an equalizer and a dynamic range compression parameter according to an evaluation result to realize adaptive sound effect optimization; synthesizing the optimized voice and accompaniment and outputting the synthesized voice and accompaniment to monitoring equipment; and finally, establishing a personalized user behavior feature vector by collecting user singing performance and historical tuning data, and performing online fine tuning on a system strategy based on a deep Q network reinforcement learning mechanism to continuously optimize a tuning effect.

The invention relates to the technical field of digital audio signal processing, and discloses an audio synchronization multipoint PTS correction method and system, a storage medium and a device, and the method comprises the steps: setting PTS calibration points at a plurality of key nodes on an audio processing link of a master device and a slave device, and carrying out the systematic initialization of an initial PTS value of each calibration point; a clock synchronization mechanism between the master device and the slave device is established, and dynamic calibration of a slave device clock is achieved by periodically interacting time synchronization signals and calculating clock offset; performing multi-stage dynamic adjustment and compensation on the PTS value of the corresponding calibration point according to the network transmission delay fluctuation of the audio frame and the received actual coding duration and time deviation; the filling state and the PTS deviation value of the playing buffer area are monitored in real time, audio output synchronization between the master device and the slave device is maintained through a playing speed fine tuning strategy, optimization is carried out aiming at a lossy coding scene, and the limitation of a traditional single-point synchronization method is effectively solved.

The invention relates to the technical field of digital playing equipment, and provides a lossless control method for the volume of a digital player and the digital player. The method comprises the following steps of: after receiving a target volume parameter, executing dynamic multiplying power oversampling and high-order noise shaping on an input digital audio signal, and injecting preset jitter to suppress in-band quantization distortion so as to generate an optimized digital audio stream; meanwhile, a high-resolution digital reference current control instruction corresponding to the target volume is independently generated, output of a reference current source is adjusted according to the high-resolution digital reference current control instruction, and a high-precision differential current signal is obtained by combining nonlinearity, temperature excursion and quantization errors of a closed-loop feedback compensation device through a multi-bit current segmentation network; performing bias stabilization processing on the signal, and converting the signal into a differential voltage signal for suppressing common-mode noise through a precision differential resistance network; and finally, low-distortion differential analog audio signals capable of directly driving a load are output through low-distortion gain, impedance conversion and power driving, so that low-distortion, high-signal-to-noise-ratio and wide-dynamic-range output in a full-volume range is realized.

A system for latency-aware orchestration and performance optimization in AI-driven telephone communication, consisting of: a speech capture unit configured to capture an analog audio signal from a telephone interface and convert the analog audio signal into a digital audio signal stream; a feature extraction unit that is operationally coupled with the speech acquisition unit and is configured to generate a feature representation of the digital audio signal stream through spectral decomposition, noise reduction, and temporal segmentation; an AI inference processor communicatively connected to the feature extraction unit, configured to run one or more AI models for automatic speech recognition, natural language understanding, and emotion recognition on the feature representation to generate intermediate results for inference; a latency orchestration controller coupled to the AI ​​inference processor, wherein the latency orchestration controller is configured to monitor latency across multiple processing stages, predict cumulative delay propagation using a hybrid latency estimation model, and orchestrate the execution scheduling of the AI ​​inference processor based on the predicted latency deviation; a performance optimization unit coupled with the latency orchestration controller and configured to dynamically adjust computational accuracy, inference batch size, and feature processing resolution based on latency thresholds and quality constraints set by the latency orchestration controller; and a transmission synchronization array configured to time-align the processed output generated by the AI ​​inference processor and transmit it to a remote communication node, with the transmission synchronization array maintaining deterministic time coordination between successive packets and the orchestrated inference results.

The invention relates to the technical field of audio processing, in particular to an airborne high-reliability intelligent voice call squelch reduction method, which comprises the following steps: performing ADC conversion on an acquired analog audio signal to obtain a digital audio signal; carrying out framing and FFT conversion on the digital audio signal, carrying out phase optimization, calculating a dynamic spectrum entropy weight, combining an auditory masking compensation factor, carrying out spectrum synthesis, and carrying out IFFT conversion and overlap elimination to obtain an audio signal after sound mixing processing; noise reduction processing is carried out on the audio signal after sound mixing processing through multi-mode feature extraction, and an audio signal after noise reduction processing is obtained; performing squelch judgment of Bayesian decision optimization on the audio signal subjected to noise reduction processing through energy feature extraction to obtain an audio signal subjected to squelch processing; performing automatic gain processing on the audio signal after the squelch processing to obtain an audio signal after automatic gain; and performing DAC conversion on the audio signal after automatic gain, and outputting a noise-reduced analog audio signal.

The invention relates to the field of audio signal processing, in particular to a multi-channel audio signal hybrid coding method and device and a storage medium, and the method comprises the steps: collecting digital audio signals through a plurality of audio input channels, carrying out the anti-aliasing filtering of the collected digital audio signals, so as to obtain audio signals in a 48kHz / 16bit format, and outputting the audio signals in the 48kHz / 16bit format. An audio signal in a 48kHz / 16bit format is processed through a loudness weighted average method, the audio signal after loudness weighted average is processed through a frequency domain mixing technology to obtain a mixed signal, and dynamic range compression is performed on the mixed signal based on calculation of an auditory masking curve to obtain a compressed signal. And carrying out hybrid coding on the compressed signal to obtain a compressed data stream, and packaging and outputting the compressed data stream. Through dynamic mixing and compression processing, the problem of signal distortion caused by a traditional linear superposition method is effectively solved, reasonable superposition of all paths of signals is ensured through dynamic mixing, and level overload caused by simple addition is avoided.

The invention provides a data missing compensation method, system and device for a digital audio conference system and a medium, and the method comprises the steps: obtaining a real-time audio data stream in the digital audio conference system, inputting the real-time audio data stream into a dynamic cache region through a dynamic cache technology, and obtaining the data of the dynamic cache region; missing type detection is carried out on sampling point time sequence intervals of the dynamic cache region data, missing type marks are obtained, and the missing type marks comprise single-point missing marks and / or whole-frame missing marks; determining a compensation strategy according to the missing type mark, and compensating the digital audio signal based on the compensation strategy to obtain compensation data, the compensation data comprising a single-point compensation value and / or frame compensation data; and mapping the compensation data to an original audio time sequence position in the real-time audio data stream for interpolation to obtain a continuous audio output stream. By adopting the method, the phase continuity of the audio waveform can be kept, and the call quality and auditory experience in a complex transmission scene can be improved.

The invention discloses a sound console digital audio signal processing method and system, a product and a medium, and relates to the field of voice or sound processing, and the method comprises the steps: obtaining a plurality of paths of audio input signals, carrying out the time-frequency domain conversion, analyzing the frequency distribution rule, extracting the fundamental frequency and overtone structure features, and building a tone feature fingerprint database. Then obtaining a preset standard audio signal, extracting a multi-dimensional feature vector, and matching the multi-dimensional feature vector with the fingerprint database to obtain a feature parameter group; on the basis, a coordination index system including the overall sound coordination degree, the sound level definition and the spatial sound image balance degree is constructed. A coordination index of an input signal is calculated in real time and compared with a standard signal, and an adjustment value is obtained. And finally, in combination with fingerprint database matching features, calculating a target audio processing parameter combination containing balanced, dynamic and spatial parameters. By implementing the method, the sound quality of multiple paths of audio signals can be improved.

A computer assisted method for classifying digital audio files based on features of a digital audio signal comprised in the file, comprising: storing the audio file in a digital memory; determining a portion (p) of drop of the audio file, for example the portion with the highest subjective loudness; classifying said audio file based on features of said drop.

The invention relates to the field of Morse telegraph communication, and discloses an AI-driven Morse telegraph automatic reading and sending method and system, which realize automatic sending and automatic reading in the Morse telegraph communication process in the field of short-wave communication. The method is realized through a full-automatic message reading and sending process: automatically sending a message, converting a message text into a Morse code sequence, and controlling a radio station to send the Morse code sequence; automatically copying and reporting, sampling a Morse audio signal output by a short-wave radio station, and decoding, reasoning and predicting a message text based on an AI model; in communication control, a half-duplex communication protocol is adopted to coordinate a full-automatic transceiving process. The invention also designs an AI-driven Morse telegraph automatic reading and transmitting system, which comprises a central processing module for executing half-duplex communication protocol logic; the telegraph sending module is used for controlling the PTT interface of the radio station to generate a pulse for sending a telegraph; the telegraph receiving module is used for converting the Morse analog audio signal into a digital audio signal; and the AI decoding module is used for reasoning and predicting a message text through a training model.

The invention discloses an audio power amplifier circuit, an audio driving method and electronic equipment, and belongs to the technical field of audio driving. Wherein the audio power amplifier circuit comprises an audio driving circuit and a loudspeaker, the audio driving circuit is connected with the loudspeaker, the audio driving circuit comprises a power amplification module, and the power amplification module is connected with the loudspeaker; the voltage regulation module is connected with the power amplification module; and the envelope tracking module is connected with the voltage regulating module, and the envelope tracking module is used for receiving the digital audio signal and acquiring a signal parameter of the digital audio signal, and controlling the voltage regulating module to increase or decrease a voltage value of a power supply signal for supplying power to the power amplification module based on the signal parameter.

The invention relates to a wireless broadcast connection system and method and a method for automatically configuring an emergency alarm area. The system comprises a wireless broadcast receiving device and an intelligent terminal, the wireless broadcast receiving device is configured to be used for receiving broadcast signals and wirelessly transmitting broadcast audio streams to the intelligent terminal, and the wireless broadcast receiving device comprises a broadcast receiving processing unit, a wireless communication module and a digital signal processing unit. The digital signal processing unit is connected with the broadcast receiving and processing unit and the wireless communication module, and is used for encoding the digital audio signal into an audio stream in a compressed format and carrying out transmission buffer management on the audio stream; wherein the digital signal processing unit is provided with a self-adaptive transmission control module, and the self-adaptive transmission control module has a bidirectional communication capability and is used for receiving a playing buffer area state feedback signal from the intelligent terminal and dynamically adjusting the transmission rate of sending the compressed audio stream to the intelligent terminal according to the feedback signal; therefore, synchronous control of buffer areas of a sending end and a receiving end is realized.

Methods and systems are described for mitigating acoustic feedback via an open ear device. In various examples, systems or methods may receive, via a first microphone and a second microphone positioned on an open ear device, a first audio signal and a second audio signal, respectively. The first audio signal and second audio signal may be converted to a first digital audio signal and a second digital audio signal. The first digital audio signal and second digital audio signal may be independently processed by an AFC to reduce acoustic feedback. A beamformer may adjust the first digital audio signal and second digital audio signal based on a target direction to create a beamformer digital audio signal. The beamformer digital audio signal may be processed via feedforward processing to create a target audio. The target audio may be phase shifted and transmitted to a loudspeaker positioned on the open ear device.

Embodiments of the present application provide a soft mute control method and device for a radio device, a medium and a product. By extracting the spectral deviation feature and the time domain energy feature of the digital audio signal of the radio device, the spectral deviation feature can reflect the frequency spectrum similarity between the signal and the noise, and the time domain energy feature can reflect the signal strength. Combined with multi-state decision and energy decision, multi-dimensional decision of the signal state is performed to realize accurate identification of the signal state, and then the output gain is dynamically adjusted. Compared with the prior art, not only the signal receiving strength index is relied on for adjustment, but also the problem of large and small sound caused by signal strength fluctuation due to position change is avoided. The spectral deviation feature is used to effectively distinguish between effective signals and noise, improve the accuracy of soft mute decision, and the time domain energy feature is used as auxiliary decision to enhance the reliability of the system in a low energy scene, so that the volume is smooth and continuous, and the user's radio experience is improved as a whole.

The utility model relates to a vehicle-mounted sound equipment testing device, which belongs to the field of vehicle-mounted audio testing, and comprises an MCU (Microprogrammed Control Unit) module, which is used for coordinating the operation of each module in equipment and ensuring the ordered transmission of data streams, communicates with an A2B module through an I2C (Inter-Integrated Circuit) protocol, and performs audio data transmission and equipment state control with a PC (Personal Computer) through a USB (Universal Serial Bus) interface; the DSP module is used for receiving input signals from the optical fiber interface and the audio analog interface, processing the received audio signals and transmitting the processed signals to the A2B module and the MCU module; the FLASH memory is used for storing data required by equipment operation; and the CAN module is used for enabling the MCU module to communicate with a CAN / CAN-FD bus of the vehicle, receiving a vehicle control signal or sending debugging data to the vehicle. The system is suitable for the fields of vehicle-mounted audio debugging, audio testing and data communication, and can simultaneously acquire and synchronously process control signals and digital audio signals, thereby improving the debugging efficiency and reducing the development period.

A watermark-based audio processing method for use in the audio player including a receiver, an analog-to-digital converter, a processor, a digital-to-analog converter and a transmitter includes: the receiver receiving an audio input signal, the analog-to-digital converter converting the audio input signal into a digital input signal including a keyword, the processor detecting an energy of the digital input signal, if the energy exceeds a threshold, the processor determining whether a watermark is included in the digital input signal, if not, the processor embedding the watermark into a predefined digital audio signal to generate a digital output signal, then the digital-to-analog converter converting the digital output signal into an audio output signal, and finally, the transmitter outputting the audio output signal for playback.

There is described a computer-implemented method for obtaining M filters of a digital audio signal defined in the time domain, where M is an integer greater than or equal to 1, each filter being associated with a respective sound source, referred to as a secondary source, emitting a sound pressure field, the invention relates to a method for operating a sound pressure field that minimizes a reflected sound pressure field of a set of sound sources, referred to as primary sources, located in a room, comprising:-for each primary sound source and secondary sound source, obtaining (S601) an impulse response of reflected sound pressure at N respective different positions in the room, where all primary and secondary sound sources are placed in an operating position, the N positions are the same for all sound sources; determining (S602) a set of M filters by solving a regularization optimization problem based on said set of impulse responses and formulated in the time domain, the optimization problem being defined as (a) minimizing the norm of the sum of impulse responses at N positions; and (b) introducing a non-zero action delay for each filter; each secondary source is combined with a primary source referred to as an associated primary source.

The application discloses a Bluetooth earphone system, which comprises a charging bin and at least one Bluetooth earphone body accommodated in the charging bin, and a separable electric connection structure is arranged between the charging bin and the Bluetooth earphone body; the charging bin comprises a Bluetooth audio module, an analog audio interface module and a charging interface module; the Bluetooth audio module is integrated with an analog-digital conversion module and a wireless communication module; the charging interface module is integrated with a digital audio interface module connected with the Bluetooth audio module; the analog audio interface module and the digital audio interface module are connected with a non-Bluetooth audio device through signal lines; analog audio signals output by the non-Bluetooth audio device are converted into digital signals, and / or digital audio signals output by the non-Bluetooth audio device are processed by the digital audio interface module, and then the analog audio signals and the digital audio signals are wirelessly transmitted to the Bluetooth earphone body by the Bluetooth audio module. Thus, a user wearing the Bluetooth earphone body can realize wireless monitoring of the non-Bluetooth audio device, the operation is simple, and user experience is improved.

A system is provided.SOLUTION: A system, comprising: means for converting a user's speech into a digital signal; means for transmitting the digital signal to a server; means for converting the speech digital signal into character string text on the server; means for translating the converted text into a specified language in real time; means for converting the translated text into speech data that retains characteristics of the original speech; means for transmitting the speech data to a terminal; and means for playing the speech data on the terminal.SELECTED DRAWING: Figure 1

The utility model provides an audio interaction host, and relates to the technical field of audio equipment, the audio interaction host of one embodiment comprises a processor, and an analog audio signal input device, a digital audio signal input device and an audio output device which are in communication connection with the processor; the analog audio signal input equipment is used for acquiring analog audio signals, converting the acquired analog audio signals into first digital audio signals and outputting the first digital audio signals to the processor; the digital audio signal input device is used for obtaining a second digital audio signal and outputting the obtained second digital audio signal to the processor, and the processor is used for processing the accessed digital audio signal, obtaining a processed digital audio signal and outputting the obtained processed digital audio signal to the audio output device; and the audio output equipment is used for converting the received processed digital audio signal into an analog audio signal and outputting the analog audio signal. According to the scheme of the embodiment, the interaction performance of the audio interaction equipment is improved.

There is provided an audio signal signature module (500) comprising: an input (502) configured to receive a digital audio signal (121); and a block divider (130) configured to divide the received digital audio signal (121) into a sequence of audio blocks (131) having a block length. A signature unit (140) generates a signature (144) associated with the current audio block (131) by applying a private key (142) to the extracted audio feature (151). The combined information is provided as signed authentication information (141) and stored in a predetermined storage location in a memory (600). The predetermined storage location may be used as authentication access information (162) and a current audio block (131) embedded by an information embedder. An audio output (190) provides an authenticable audio signal (501) comprising a sequence of audio blocks with embedded authentication access information (162). A block divider (130) adjusts the block length of the audio block in accordance with the size of authentication access information to be embedded in the audio block.

The present invention belongs to the technical field of electroacoustics in electronics, and relates to an audio system formed by a speaker-enclosure planar array, and a preparation method for the audio system. Different from the prior art, the present invention uses a plurality of small / -micro-aperture speakers to form a special planar array audio device, such that by using the plurality of small- / micro-aperture speakers, audio signals with frequencies of 20-120 Hz or even higher, which are typically cut off in small-aperture speaker systems, are caused, by means of synchronous superposition operation, to output powerful, clear and pure bass sound waves. Moreover, an algorithm for a linear digital filter is used to perform frequency division and surround sound processing on digital audio signals of the speakers without generating a phase shift. Therefore, a series of functional defects in conventional large-aperture subwoofers are overcome, thereby enabling the planar-array audio system to achieve a surround sound effect in a three-dimensional or two-dimensional space.

An apparatus, data structure and computer-implemented method for digital content processing, where the method comprises providing (202) a first data set, where the first data set comprises elements, providing (204) a second data set, where the second data set comprises elements, where the digital content of a respective one of the elements comprises a digital image, the method includes generating (206) a first set of descriptions, such as video images, radar images, LiDAR images, ultrasound images, motion images or thermal images, or wherein the content of respective ones of the elements includes digital audio signals, in particular using a data-to-text model, where the first set includes element-aspect descriptions of elements of a first data set, where the first set includes element-aspect descriptions of the elements of the first data set, and where the first set includes element-aspect descriptions of the elements of the second data set. Wherein a description of a respective element of the first dataset is determined from the content of the respective element of the first dataset, in particular generated (208) with a data-to-text model, a second set of descriptions wherein the second set comprises an element-aspect description of an element of the second dataset.

The invention discloses an electric lute audio processing system and processing method and an intelligent electric lute, and relates to the technical field of electric lutes, the electric lute audio processing system comprises a pickup unit, a main control unit, a DSP sound effect processing module used for sound effect processing and an output module used for outputting digital audio signals, an analog electric signal acquired by the pickup unit is amplified and then sent to the main control unit; the analog electric signal is converted into a digital signal through the analog-to-digital converter and then is sent to the DSP sound effect processing module; the DSP sound effect processing module is used for performing timbre processing and switching on the collected original sound signals; the processed digital audio signal is output through the output unit; multiple functions of Bluetooth accompaniment, multi-timbre simulation, a tuner, an equalizer, power amplification and charging management are integrated into one body, the integration degree is high, and the functions are rich. Various digital audio processing units are combined with a traditional lute structure, and the application scene and expressive force of the musical instrument are expanded.

The invention provides a hearing aid device, method and device with a self-adaptive hearing compensation function, the hearing aid device comprises a wireless communication module, a main control chip and an audio output module, and a pre-training hearing compensation model is built in the main control chip. A full-band continuous personalized frequency response gain curve with gain amplitude positively correlated with hearing loss is output through model reasoning, an audio output module performs full-band frequency-point-by-frequency balanced linear gain compensation on an original digital audio signal played by the device based on the curve, the audio dynamic range is not compressed in the whole process, the original spectrum structure is not changed, and the gain amplitude of the original digital audio signal is not changed. And the high-fidelity tone quality is kept. Different from a traditional hearing aid and a conventional audio playing device, only for audio processing inside the device, pickup and noise reduction are not needed, hearing features of a hearing impaired user can be accurately adapted, clear and balanced listening can be achieved, tone quality distortion can be avoided, and compensation precision, use comfort and scene universality are considered.

The invention relates to an audio link detection device, system and method, related equipment and a vehicle, and relates to the technical field of signal detection, and the method comprises the steps: receiving an analog drive signal sent by a vehicle machine through a vehicle-mounted audio bus; the analog driving signal is converted into a digital audio signal, the digital audio signal is sent to the vehicle machine, and the vehicle machine is used for conducting audio link detection of the vehicle-mounted audio bus according to the analog driving signal and the digital audio signal. According to the technical scheme, the links of actual loudspeaker sounding and microphone acquisition can be omitted, sound interference of the external environment is avoided, the accuracy of the audio link detection result is improved, in addition, multi-path cascade parallel testing is supported, and the testing efficiency is improved.

The present application relates to the technical field of hearing aids, in particular to a method and system for enhancing the audio clarity directivity of a hearing aid in two channels, comprising: sending a detection signal to identify a mobile phone power supply protocol, receiving and identifying a UAC protocol, outputting corresponding safe voltage and current according to the type of mobile phone power supply protocol, triggering three-level protection, identifying the current audio output mode, sampling rate and bit depth of the mobile phone by analyzing the UAC protocol, and obtaining a standard audio stream format; receiving a scene mode instruction, controlling the dual-microphone channel according to the scene mode instruction to collect analog signals, and converting the analog signals into digital audio signals with source identification respectively; generating enhanced digital audio according to the type of digital audio signal; reading the connection state parameters fed back by the mobile phone in real time, and encapsulating the enhanced digital audio into a Bluetooth signal according to the connection state parameters and transmitting the Bluetooth signal to a Bluetooth hearing aid. The present application realizes high-precision audio enhancement by combining dynamic adaptive filtering with multi-frequency band regulation.

The present application belongs to the technical field of audio processing, and particularly relates to an audio reproduction method and system for live sound field data collection and digital signal processing. The method comprises collecting sound field data of a real acoustic space and generating a dedicated sound field parameter model; receiving an original digital audio signal and a user instruction, obtaining a dedicated sound field parameter model corresponding to the user instruction as a target dedicated sound field parameter model; performing convolution processing on the original digital audio signal and the target dedicated sound field parameter model to obtain a processed audio signal with enhanced spatial sense; and outputting the processed audio signal after multi-channel digital amplification, thereby realizing audio reproduction of the acoustic response characteristics of the dedicated sound field. The present application generates a dedicated sound field parameter model based on impulse response data collected in a real acoustic space, and the reproduced reverberation characteristics, spatial sense and sound image positioning are highly consistent with the target space, and the sound field restoration accuracy is significantly improved.

This invention relates to a method for modeling audio response curves in a web environment, comprising the following steps: S1: Generating frequency and gain based on screen click events, and forming control points by user input of Q values; or obtaining three parameters—frequency, gain, and Q value—input by the user to form control points; S2: Converting the frequency, gain, and Q value of the control points into parameters of a Gaussian function and generating connection points, smoothing the connection points with a Bézier curve, and then drawing the audio response curve; S3: Mapping the Gaussian function parameters to a Web Audio API and obtaining a digital audio signal through a DSP algorithm, and then converting it into an analog audio signal. By simulating the peak value and gain characteristics of the audio response curve in local regions using a Gaussian function, a natural transition of the audio response curve in local regions is ensured; by using a Bézier curve to smoothly connect the Gaussian curves, a highly accurate and sufficiently smooth audio response curve is obtained.