280 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 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 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 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.

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 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 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.

The utility model belongs to the technical field of earphones, and particularly relates to a sound quality enhancing earphone structure which comprises an earphone body and an ear hook connected with the earphone body. The earphone body comprises an external earphone shell; a mounting cavity is formed in the earphone shell, a sound outlet hole communicated with the mounting cavity is formed in the earphone shell, the position of the sound outlet hole is aligned with the position of an ear canal when the earphone is worn, a sound production unit is arranged in the mounting cavity, the sound production unit comprises a moving coil loudspeaker and a sound guide tube component connected with the moving coil loudspeaker, and a sound outlet hole communicated with the front sound cavity is formed in the earphone shell; the sound guide tube component comprises a sound guide channel. And the sound outlet hole faces the external auditory canal of the human body, so that the medium-high frequency directivity is strong, the loudness is large, and the effect of enhancing the medium-high frequency is achieved. Meanwhile, the sound guide tube part is arranged, the sound transmission direction can be changed through a sound guide channel, the peak resonant frequency of a harmonic peak of air guide sound in a middle-high frequency band or a high frequency band can be weakened conveniently, and the effect of middle and low frequencies can be effectively enhanced.

The application discloses a protection and distortion control method and system during sound operation, and relates to the technical field of audio processing. An audio signal and a reference signal are acquired; the reference signal is preprocessed to obtain a trigger signal of a level effective value; a trigger threshold is acquired according to the trigger signal of the level effective value; the trigger threshold is adjusted to obtain a preset trigger threshold; the preset trigger threshold is dynamically adjusted according to a voltage value corresponding to a frequency of the audio signal to obtain a voltage value corresponding to the preset trigger threshold; and the audio frequency corresponding to the voltage value corresponding to the preset trigger threshold is a protected and distortion-controlled audio signal. The application realizes dynamic change of a high-frequency cutoff frequency of an audio high-frequency unit while maintaining the loudness of a sound field restoration of a sound system, and improves the protection and distortion control accuracy of the sound in a limit operation state of high power and long time.

The application discloses a kind of induced mood pulse sound feature screening method, belong to cognitive neuroscientific technology field, solve the problem of lack of effective screening in prior art mood pulse sound feature.A kind of induced mood pulse sound feature screening method, the method comprises: under the condition of subject within experiment, according to the scene of inducing mood, set multiple pulse sounds;The independent variable of the pulse sound includes sound frequency, repetition frequency and relative loudness;Different sound frequency, repetition frequency and relative loudness are cross designed, form multiple sound stimuli;Using multiple sound stimuli, a plurality of subjects are experimented respectively;And the physiological signal of each subject under each sound stimulus, emotional valence score and arousal degree score are acquired;Based on the physiological signal of all subjects under all sound stimuli, emotional valence score and arousal degree score, the feature screening result of corresponding induced mood pulse sound is obtained.

1. The name of the design product: effecter (1). 2. The use of the design product: for home, dance hall or conference room to increase the loudness of sound performance. 3. The design points of the design product: in shape. 4. The picture or photo that best indicates the design points: perspective view.

The application provides a control method, device and equipment of a clothes processing device and a storage medium. The method comprises: acquiring a voice signal collected by the clothes processing device after a washing program is completed; determining whether a target user is close to the clothes processing device according to whether there is a footstep sound in the voice signal and a change in loudness of the footstep sound; if it is determined that the target user is close to the clothes processing device, acquiring a footstep soundprint feature of the target user at at least one time; determining distances between the target user at the at least one time and the clothes processing device according to the footstep soundprint feature of the target user at the at least one time; and outputting a clothes taking reminder if at least part of the distances between the target user at the at least one time and the clothes processing device are within a preset range of the clothes processing device. The method improves the effectiveness and intelligence of the clothes processing device in reminding a user to take out clothes.

An audio control system includes: an obtainer that obtains sound data indicating a sound picked up by a microphone disposed in a second space located in an area surrounding a first space used by a person; a storage in which sound source data is stored; and a controller that causes a loudspeaker disposed in the first space or the second space to reproduce an audio obtained from the sound source data more loudly in the second space than in the first space. The controller determines a loudness the sound indicated in the sound data obtained, and adjusts a sound volume of the audio outputted from the loudspeaker to be lower with an increase in the loudness of the sound determined.

In at least one embodiment, an audio system is provided. one or more microphones capture an audio input signal including at least one of captured entertainment audio and a speech output from at least one occupant positioned in a vehicle. An at least one audio controller is programmed to transmit first entertainment audio into a cabin of the vehicle via one or more loudspeakers. The at least one audio controller is further programmed to receive the audio input signal from the one or more microphones and to receive a reference signal indicative of the first entertainment audio that is being transmitted into the cabin of the vehicle. The at least one controller is further programmed to apply to the reference signal to the captured entertainment audio to provide the speech output and to compare a loudness of the speech output to a loudness threshold.

A kind of transmission equipment intelligent monitoring system and method based on noise feature analysis, through the method, expert can input the sound evaluation information heard from console, the sound evaluation value of expert corresponds to the state of specific reduction mechanism, the loudness and frequency characteristic of noise signal output by spectrum analyzer as input layer, the sound evaluation value of expert as output layer, mapping relationship between the two is established through data accumulation and neural network algorithm. Through the database of abnormal noise experts as a reference, warning is found by comparing similar abnormal noise data, ensure that noise meets the standard, through the noise evaluation of experts and neural network algorithm, the timbre can be evaluated, and then the specific type of fault can be judged. Moreover, the influence of on-site noise on sound signal collection is excluded, and the accuracy of machine detection is further improved.