385 results about "Loudness" patented technology

A system may include a storage device, configured to store computer-executable instructions. A system may include an ear-bud configured to be positioned within an ear canal of a user, the ear-bud comprising: a speaker, a microphone; and one or more processors in communication with the storage device, wherein the computer-executable instructions, when executed by the one or more processors, cause the one or more processors to: obtain a user hearing profile, obtain an equal-loudness hearing profile, receive audio data from the microphone, and generate a second audio data based on a first sound-pressure level, a second sound-pressure level, a first frequency; and cause the speaker to emit the second audio data within the ear canal of the user, such that the user perceives the audio data as if the user has normal hearing.

The invention discloses a self-adaptive sound effect management method and system with a scene analysis module, and relates to the field of self-adaptive sound effect management, and the method comprises the following steps: S1, collecting environmental acoustic characteristics and user behavior data in real time through the scene analysis module; s2, dynamically extracting user hearing preference parameters and scene classification labels; s3, constructing a virtual audition sweet zone matched with the physical space; and S4, dynamically adjusting the phase delay and the energy distribution proportion of each frequency band, and generating an optimal loudness equalization curve meeting the multi-scene constraint. According to the method, adaptive matching of sound field characteristics is realized in a complex dynamic scene through a multi-level perception-optimization closed-loop architecture, acoustic feature extraction and physical space modeling are deeply fused, the technical boundary of traditional pure signal processing is broken through, and the problem of multi-target conflict is solved based on cross-modal collaborative learning and a global optimization algorithm.

A loudspeaker system and method can measure a specific user's hearing and implement compensatory or corrective processing to address the user's hearing deficiencies. The user's hearing acuity may be determined by (i) headphone techniques in which frequency-based hearing thresholds are determined; (ii) a loudspeaker system set up as intended for use in an acoustic space to emit tonal stimuli; or (iii) inducing, acquiring, and interpreting otoacoustic emissions. Once the hearing profile is determined, appropriate signal processing parameters can be used to generate a corrected audio output signal adjustment. The administered hearing test results can include a plurality of hearing loudness threshold levels at respective test frequencies. The hearing correction system's corrected audio output can be generated from corrected loudness values at selected correction frequency sub-bands.

The invention discloses an atmosphere lamp control method, electronic equipment and a program product. The method comprises the following steps: acquiring PCM data of an audio signal in real time in the music playing process of a vehicle; performing frequency domain analysis on the PCM data corresponding to each audio frame, and extracting frequency information and loudness information; the method comprises the following steps: calculating frequency information and loudness information of a preset number of continuous audio frames, and respectively calculating a frequency change rate and a loudness change rate within a preset time; respectively comparing the frequency change rate and the loudness change rate with corresponding change rate thresholds; and if at least one of the frequency change rate and the loudness change rate exceeds the corresponding change rate threshold value, a corresponding light updating instruction is sent to the atmosphere lamp control module. The music rhythm function of the automotive interior atmosphere lamp can be enhanced, and the cooperative interaction ability of music and the atmosphere lamp is improved.

The invention discloses a subjective and objective fused air conditioner sound quality analysis and evaluation method, which comprises the following steps of: 1, forming a working condition-gear combination matrix according to an actual use scene and noise characteristics of an air conditioner; 2, arranging sensor measuring points according to a noise source scene of working condition-gear combination, and obtaining a sound sample; 3, carrying out subjective evaluation on the sound sample, and processing an abnormal score by adopting a 3 sigma principle and a box plot method to obtain a final subjective score; 4, five objective psychological acoustic indexes including loudness, roughness, sharpness, jitter degree and highlight ratio of 24 Bark frequency bands of the sound sample are calculated, and an objective index set is established; and 5, constructing an air conditioner subjective and objective fusion evaluation model of a two-stage fusion model of the time sequence convolutional network and the gradient boosting tree. According to the method, the problem of low evaluation precision of a general model caused by noise characteristic difference under different working conditions-gears is solved, an accurate air conditioner sound quality evaluation system is constructed, and support is provided for sound quality optimization design and quality evaluation of the air conditioner.

An audio device is provided. The audio device comprises a controller and an acoustic transducer. The controller is configured to receive an input audio signal. The controller is further configured to generate, via a loudness detector, a loudness level of a first portion of the input audio signal. The controller is further configured to identify a compression curve of a plurality of compression curves corresponding to a user volume setting. The controller is further configured to adjust, via an audio compressor, a second portion of the input audio signal based on the loudness level and the compression curve to generate an output audio signal. The loudness level is determined according to LUFS model.

This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

The invention belongs to the technical field of robot and human voice interaction, and particularly relates to a human voice interaction method and system. The method comprises the following steps: step 1, an intelligent agent enables radio duration to realize self-adaption according to environment loudness; 2, designing a multi-agent framework with a large model and a small model cooperating with each other; and step 3, based on the framework designed in the step 2, analyzing the radio in the step 1, and using a retrieval enhancement technology based on a tree-shaped document to improve the question and answer reliability of the intelligent agent and realize the voice interaction of the intelligent agent. The method is used for solving the problems that in the prior art, most voice interaction technologies can only achieve single-round interaction and cannot remember context content of interaction, and pure human voice is difficult to obtain when the traditional voice interaction technology is directly transplanted to a robot which can move and makes noise by itself.

The invention discloses a college dormitory noise multi-dimensional evaluation method based on psychoacoustic and sound field simulation, and belongs to the field of building environment and noise control. The objective of the invention is to solve the problem that an existing dormitory noise evaluation method is single in index and neglects subjective perception and spatial distribution heterogeneity. The method comprises the following steps: acquiring physical data and original audio of dormitory noise through multi-point synchronous acquisition; calculating loudness, sharpness, roughness, fluctuation intensity and comprehensive psychoacoustic annoyance (PA) by using a psychoacoustic model, and scientifically quantifying subjective interference of noise; meanwhile, sound field simulation is carried out based on the accurately established three-dimensional dormitory model, a sound pressure level distribution cloud picture is generated, and a standing wave phenomenon and a sound energy accumulation area are visually revealed; and finally, generating a targeted noise reduction optimization strategy by integrating subjective and objective and spatial analysis results. According to the method, full-chain diagnosis of physical characteristics-subjective perception-spatial distribution of dormitory noise is realized, and a scientific basis is provided for accurate and efficient noise treatment.

The utility model belongs to the technical field of electroacoustic conversion, and relates to a sounding monomer and an electronic device, the sounding monomer comprises a housing, and a vibration system and a magnetic circuit system which are fixedly connected to the housing; the magnetic circuit system comprises a magnetic conductive yoke, a common magnetic part, a central magnetic part and four side magnetic parts, and the four side magnetic parts are arranged on the periphery of the common magnetic part and the central magnetic part; the vibration system comprises a vibrating diaphragm assembly, a serial connection part, two voice coils and four centering disks, the vibrating diaphragm assembly is connected to the shell, the two voice coils are arranged side by side, the first end of each voice coil is connected to the vibrating diaphragm assembly, and the second end of each voice coil is inserted into the two magnetic gaps; the serial connection part is arranged on the vibrating diaphragm assembly, and the two voice coils are electrically connected with the serial connection part so as to be arranged in series. Through the above structure, the ineffective copper foil DC resistance can be reduced, the impedance of the sounding monomer can be reduced, the tone quality effect of the sounding monomer can be improved, the driving force factor BL of the sounding monomer can be improved, and the loudness of the sounding monomer can be improved.

The present invention provides a speaker including a first frame, a second frame, a first vibration system, a second vibration system, and a magnetic circuit system. The first vibration system includes a first diaphragm and a first voice coil. The second vibration system includes a second diaphragm and a second voice coil. The magnetic circuit system includes a first main magnet, first auxiliary magnets a second main magnet and second auxiliary magnets. A projection of the second voice coil along a vibration direction of the speaker is located in the first voice coil, and the projection of the first voice coil on the magnetic circuit system along the vibration direction of the speaker is located on the second auxiliary magnet. In present invention, the BL of the first voice coil is increased, the sensitivity of the speaker is improved, and the loudness becomes larger.

The utility model discloses a sounding device and an electronic device, and relates to the technical field of electroacoustic transduction, the sounding device comprises a magnetic circuit system and a vibration system, a central magnetic part of the magnetic circuit system comprises two central magnetic groups, an edge magnetic part comprises an outer magnetic group located at the outer sides of the two central magnetic groups and a common magnetic group located between the two central magnetic groups, the outer magnetic group and the common magnetic group are spaced from each central magnetic group so as to define two magnetic gaps which are arranged at intervals, each central magnetic group comprises a first central magnet, a central magnetic conductive plate and a second central magnet which are arranged in a stacked mode, and the outer magnetic group and the common magnetic group each comprise a first edge magnet, an edge magnetic conductive plate and a second edge magnet which are arranged in a stacked mode. The vibration system comprises a vibrating diaphragm and two voice coils, and each voice coil corresponds to one magnetic gap. According to the sounding device, the utilization rate of the magnetic circuit system is further improved by optimizing the magnetic circuit system and adopting the double voice coils, so that the loudness and the sensitivity of the sounding device are improved.

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 indoor air quality monitoring equipment and a sensor cooperative control system, relates to the technical field of indoor environment monitoring, and is used for solving the problems of response delay, temperature drift accumulation and alarm accessibility reduction in long-term operation of an air monitoring device. The device realizes feed-forward identification and closed-loop suppression of sample injection degradation, filter material pollution and dislocation, heat drift in the machine and acoustic attenuation: a sampling confirmation window is adaptively adjusted to reduce misjudgment, channel-level temperature drift compensation and power consumption derating are cooperatively executed according to priorities to realize steady-state temperature control, and a prompt link keeps subjective loudness consistent based on equivalent gain; meanwhile, the baseline is refreshed after maintenance to maintain long-term consistency; therefore, under the condition that the mechanical appearance is not changed, the monitoring accuracy, stability and alarm accessibility are remarkably improved, and the device-level design value of structural perception driving is highlighted.

The present invention provides a speaker including a first frame, a second frame, a first vibration system, a second vibration system, and a magnetic circuit system. The first vibration system includes a first diaphragm and a first voice coil. The second vibration system includes a second diaphragm and a second voice coil. The magnetic circuit system includes a first main magnet, first auxiliary magnets, a second main magnet and second auxiliary magnets. A projection of the first voice coil along a vibration direction of the speaker is located out of the second voice coil, and the projection of the second voice coil along the vibration direction of the speaker is located on the first main magnet. In present invention, the BL of the second voice coil is increased, the sensitivity of the speaker is improved, and the loudness becomes larger.

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.

A loudspeaker system and method can measure a specific user's hearing and implement compensatory or corrective processing to address the user's hearing deficiencies. The user's hearing acuity may be determined by (i) headphone techniques in which frequency-based hearing thresholds are determined; (ii) a loudspeaker system set up as intended for use in an acoustic space to emit tonal stimuli; or (iii) inducing, acquiring, and interpreting otoacoustic emissions. Once the hearing profile is determined, appropriate signal processing parameters can be used to generate a corrected audio output signal adjustment. The administered hearing test results can include a plurality of hearing loudness threshold levels at respective test frequencies. The hearing correction system's corrected audio output can be generated from corrected loudness values at selected correction frequency sub-bands.

The invention provides a sound effect optimization method based on an SOC architecture, a storage medium and electronic equipment. Comprising the following steps: S1, inputting an audio signal; s2, performing channel identification and switching on the audio signal; s3, when the sound channel number of the audio signal is less than a preset target sound channel number, performing up-mixing processing on the audio signal to obtain a multi-sound-channel signal with the preset target sound channel number; s4, performing global volume control and overall filter processing on the multi-channel signal; s5, performing gain control and intelligent loudness control on the filtered multichannel signal, wherein the gain control comprises application of a preset sound effect mode selected by a user; and S6, the signal after gain and loudness control is routed to a corresponding output channel for output according to the selected output scene mode. The invention provides a complete set of complete and high-quality audio solution from an input source to multichannel output, and is suitable for various audio products such as earphones, cinemas, automobiles and the like.

The invention discloses an earphone defect detection method and device based on voiceprint feature analysis and a storage medium, and relates to the technical field of electroacoustic quality control. According to the method, human ear hearing characteristics are simulated, Mel-cepstrum voiceprint characteristics are extracted from sound response signals, an adaptive weight mechanism based on an equal-loudness curve is introduced, and a voiceprint abnormal deviation index model conforming to human ear subjective perception is constructed; in order to solve the problem that tiny abnormal noise is difficult to recognize, high-order statistics such as kurtosis and skewness are adopted for feature fusion enhancement, and a support vector machine is combined to achieve accurate capture of transient impulse noise. According to the invention, various defect types such as voice coil friction, imbalance of balance degree, diaphragm damage and the like can be fully automatically distinguished, the problems that a traditional test means is disjointed with subjective hearing sense and the omission ratio of hidden defects is high are effectively solved, and the quality inspection efficiency and accuracy of earphone production lines are remarkably improved.

The invention relates to the field of laser shock peening, and particularly discloses a laser shock peening black adhesive tape damage monitoring method based on sound signals, which comprises the following steps: collecting laser shock discrete sound signals, obtaining a discrete sound signal sequence, and carrying out Hilbert envelope extraction on the discrete sound signal sequence to obtain envelope signals; performing weighting processing on the extracted envelope signal to obtain a weighted envelope, and constructing a loudness index based on the weighted envelope; short-time window frequency analysis is adopted, a dominant frequency sequence with the dominant frequency changing along with time is extracted, and the dominant frequency stability offset is calculated according to the dominant frequency sequence; and constructing a loudness-frequency coupling damage scoring criterion, and calculating a damage score. According to the method, the damage identification sensitivity is remarkably improved, the real-time online monitoring of the laser shock process is realized, the environmental adaptability is high, the engineering deployment is simple and convenient, the consistency of the laser shock peening quality is remarkably improved, and meanwhile, the method has good expandability and universality.

A device for adjusting the volume of an audio signal in an audio system is proposed, wherein the audio system has an input for receiving the audio signal from an input source and an output for outputting the audio signal. The device comprises a determining unit configured to determine the loudness of the audio signal and an adjustment unit configured to regulate the level of the audio signal to a target value. The device further comprises an attenuation unit downstream of the adjustment unit, which is configured to attenuate the audio signal regulated by the adjustment unit by a predetermined attenuation value based on the determined loudness.

The present invention provides a refrigerator comprising an exciter assembly including a first exciter and a second exciter; a housing including an insulation layer, a first shell, and a second shell; the insulation layer being provided with a first recess and a second recess spaced apart from each other; the first exciter being located within the first recess, and the second exciter being located within the second recess; a portion of the second shell corresponding to the first recess constituting a first vibrating portion, and a portion of the second shell corresponding to the second recess constituting a second vibrating portion; the first exciter being connected to the first vibrating portion, and the second exciter being connected to the second vibrating portion; and the difference in loudness per octave between the sound waves emitted by the first vibrating portion and the sound waves emitted by the second vibrating portion being less than 3dB. The refrigerator provided by the present invention can reduce a user's sensitivity to sound sources.

The present invention proposes a method and system for evaluating the wind vibration performance of an automobile, which specifically include: a wind tunnel test of the wind vibration performance of the whole vehicle, and a road test of the wind vibration performance of the whole vehicle; scoring the wind vibration sound performance of the whole vehicle based on the objective measurement results and subjective scores of the wind tunnel test and the objective measurement results and subjective scores of the road test; establishing a database and performing analysis, setting 10 levels from L1 to L10 and corresponding scores for each level, to form a wind vibration level table; determining the wind vibration performance level of the tested vehicle model by comparing the wind vibration level table; the present invention is not limited to sound pressure level, and objective measurement results such as loudness and speech clarity and their corresponding recommended values ​​can be added as objective evaluation indicators; it is not limited to collecting objective data from the driver's outer ear, and collection from any point in the vehicle can be added; it is not limited to recording test conditions in the existing wind tunnel and road test condition tables, and can be added or modified, such as changing coasting to uniform acceleration.

Aspects of the present disclosure relate to a method and system for enhancing conversation intelligibility in a raw audio signal including a conversation component and a non-conversation component. The method comprises providing a dialogue component of the original audio signal in a first individual audio signal, providing a non-dialogue component of the original audio signal in a second individual audio signal, separately processing the first individual audio signal and the second individual audio signal, wherein processing the first and second individual audio signals comprises processing the loudness of the first individual audio signal and / or the second individual audio signal, and combining the processed first and second individual audio signals to provide a processed audio signal.

The application discloses an audio effect evaluation method and device, equipment and a storage medium. The application inputs a preset reference sound source into an audio device to be tested for playing, wherein the preset reference sound source is a sound source determined by statistical analysis of sound sources, in combination with a parameter corrected according to a preset evaluation result and a peak factor corresponding to a music playing platform; collects audio to be tested output by the audio device to be tested when playing the preset reference sound source; determines a relative frequency response and an estimated loudness value according to a spectrum corresponding to the audio to be tested and a spectrum corresponding to the preset reference sound source; and generates an audio effect evaluation result of the audio device to be tested according to the relative frequency response and the estimated loudness value. The application evaluates the audio effect of the audio device to be tested by using the reference sound source constructed in advance, solves the problem of traditional sweep frequency signal tuning deviation, greatly reduces the number of repeated trial and error, and significantly improves the efficiency, accuracy and reliability of audio product tuning work.

A device for adjusting the volume of an audio signal in an audio system is provided, wherein the audio system has an input for receiving the audio signal from an input source and an output for outputting the audio signal. The device includes a determination unit configured to determine a loudness of the audio signal, and an adjustment unit configured to regulate a level of the audio signal to a target value. The device further includes a damping unit, downstream of the adjustment unit, which is configured to attenuate the audio signal regulated by the adjustment unit by a predetermined damping value based on the determined loudness.

The invention relates to the technical field of Bluetooth, and discloses a dynamic bass enhancement method and system for Bluetooth equipment, and the method comprises the steps: separating a complex tone signal component and a pure tone signal component from an audio signal, so as to calculate a harmonic proportion attenuation factor of the audio signal; determining an auditory transmission distance between the Bluetooth device and the audience, and calculating a low-frequency attenuation amount of the audio signal in a transmission process; calculating the bass perception loudness of the audience, and obtaining the target bass enhancement intensity of the audience; determining a harmonic balance equivalent of an output sound field of the Bluetooth device, and calculating an effective bass enhancement amount of the Bluetooth device; and performing bass enhancement processing on the audio signal based on the effective bass enhancement quantity and the target bass enhancement intensity to obtain an enhanced audio signal. According to the invention, the dynamic bass enhancement effect of the Bluetooth device can be improved.

The application relates to the technical field of audio processing, in particular to an audio processing method, an electronic device and a readable storage medium. In the method, a sound signal is received by a device that can be worn on the ear of a user, then the sound signal is divided into a low-frequency sound signal and a high-frequency sound signal by using a filter, and a steady-state sound signal and a transient sound signal in the low-frequency sound signal and the high-frequency sound signal are extracted. The steady-state sound signal and the transient sound signal are processed based on the individual calibration parameters corresponding to the user, so that the calibrated sound signal finally heard by the user conforms to the target psychoacoustic loudness of the user. In this way, the hearing protection of the user can be realized.

The invention relates to the technical field of braking systems, in particular to an acoustic performance prediction method for an electronic parking braking system. A method for predicting the acoustic performance of an electronic parking brake system comprises a first-stage transmission mechanism and is characterized by comprising the following specific steps: (1) collecting the sizes of parts influencing MGU noise in the first-stage transmission mechanism; (2) sample SPL calculation; (3), carrying out the loudness calculation of the Moore-Glasberg-Schlittenache, and carrying out the loudness calculation of the Moore-Glasberg-Schlittenache; (4) BP neural network structure algorithm; (5) genetic algorithm; and (6) designing a GA-BP fusion model. Compared with the prior art, the method for predicting the acoustic performance of the electronic parking brake system has the advantages that a plurality of key sizes, SPL (sound pressure level) and loudness value acoustic parameters of an MGU transmission system are collected, iterative learning is carried out by applying a genetic BP (GA-BP) neural network, and the MGU acoustic performance prediction method is obtained and used for later project development and optimization; the test time is reduced to a great extent, and the overall working efficiency is improved.

The application discloses a control method, device and electronic equipment. The method comprises: obtaining a power supply adjustment instruction sent by a first processor, wherein the power supply adjustment instruction comprises a first demand power supply parameter of the first processor; determining a frequency of voltage jumps satisfying a first setting condition based on the first demand power supply parameter in the received power supply adjustment instruction, wherein the frequency represents a number of times of voltage jumps satisfying the first setting condition within a setting time period; and if the frequency reaches a frequency threshold, controlling the first processor to adjust the first demand power supply parameter to a second demand power supply parameter, wherein the second demand power supply parameter makes a power supply parameter of a power supply controller satisfy a second setting condition, and a loudness of sound caused by the power supply parameter satisfying the second setting condition is lower than a loudness of sound caused by a power supply parameter not satisfying the second setting condition.