221 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.

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 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 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.

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.

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.

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.

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.

This invention discloses a refrigerator and a method for locating food items. By installing multiple sound-collecting devices inside the refrigerator and attaching a sound-emitting unit to an RFID tag, the RFID tag is bound to the food item and placed inside the refrigerator. The sound emitted by the sound-emitting unit can be captured by the sound-collecting devices, and the food item is located based on the loudness of the sound detected by the sound-collecting devices. Using this invention improves the accuracy of food item location, allowing users to quickly determine the storage location of the food item.

The invention discloses a polymorphic tooth sound dynamic perception and self-adaptive suppression method, and relates to the field of voice signal processing. The method specifically comprises the following steps: firstly, inputting a real-time audio or voice signal of a voice stream, and preprocessing according to frames; performing spectral analysis to obtain a high-low frequency tooth sound sensing frequency band corresponding to each frame, and determining a frequency division point of each frequency band in each frame of voice signal; thirdly, the FIR frequency divider carries out frequency division processing on the audio signals of all the frames according to frequency division points of the high-frequency and low-frequency tooth sound sensing frequency bands, and five sub-bands are obtained; performing root mean square RMS calculation on the two sub-bands from Hz to Hz and from Hz to Hz to obtain the instantaneous loudness of each sub-band, judging the instantaneous loudness with a tooth sound adaptive threshold value, and triggering compression processing when the RMS value of a certain sub-band exceeds the threshold value; and finally, performing weighted synthesis on the compressed sub-band audio signals and other sub-band audio signals to obtain balanced output voice of each frame. According to the invention, the harsh feeling of tooth sound is reduced, and the natural brightness and definition of voice are kept.

The application discloses an adaptive sound effect management method and system with a scene analysis module, relates to the field of adaptive sound effect management, and comprises the following steps: S1, collecting environmental acoustic characteristics and user behavior data in real time through a scene analysis module; S2, dynamically extracting user hearing preference parameters and scene classification labels; S3, constructing a virtual listening sweet spot matched with a physical space; and S4, dynamically adjusting the phase delay and energy distribution ratio of each frequency band to generate an optimal loudness equalization curve meeting multi-scene constraints. Through a multi-level perception-optimization closed-loop architecture, the application realizes adaptive matching of sound field characteristics in a complex dynamic scene, deeply integrates acoustic characteristic extraction and physical space modeling, breaks through the technical boundary of traditional pure signal processing, and solves the multi-objective conflict problem based on cross-modal collaborative learning and a global optimization algorithm.

The utility model provides an aluminum basin high pitch loudspeaker, which comprises a loudspeaker body, the loudspeaker body is sequentially provided with T iron, magnetic steel, a washer, a positioning ring and a basin stand from bottom to top, the positioning ring is provided with a vibration assembly, the vibration assembly comprises a supporting plate, a buffer pad, a vibrating diaphragm, a voice coil and a wiring terminal, and the middle part of the supporting plate is provided with a round hole. The bottom surface of the buffer pad is connected with the top surface of the supporting plate, the bottom surface of the vibrating diaphragm is connected with the top surface of the buffer pad, the top of the voice coil penetrates through the circular hole to be connected with the vibrating diaphragm, the two ends of the supporting plate are provided with the wiring terminals, the wiring terminals are connected with the voice coil through leads, the basin frame is in a horn shape, sound generated by vibration of the vibrating diaphragm is amplified through the basin frame, and the effective transmission range of high frequency can be widened. And the phase cone is arranged in the center of the top of the basin stand, so that the sound wave interference can be reduced by the phase cone, the high-frequency directivity can be enhanced, and higher loudness and better definition can be kept at a longer distance.

The invention relates to the technical field of railway vehicle noise analysis and processing, and particularly discloses a railway vehicle noise distinguishing and extracting device and method.The device comprises an acquisition module, a processing module and an output module, and the acquisition module is used for acquiring carriage mixed noise in the running process of a railway vehicle; the processing module is used for performing frequency decomposition, noise classification and independent loudness analysis of various types of classified noise on the collected mixed noise of the carriage; the output module is used for outputting and / or storing the loudness values and the frequency characteristics of various types of noise after classification; according to the method, wheel track noise, passenger voice and train station reporting voice can be accurately separated and subjected to loudness analysis in real time, the problem that the passenger voice and the train station reporting voice are difficult to separate and distinguish is solved, and a scientific noise monitoring and management means is provided for rail transit operation enterprises; the passenger comfort level is improved, the station reporting volume and the operation service quality are optimized, and complaint and transformation cost caused by noise is reduced.

The invention discloses an electric arc welding control method and device, electronic equipment and a storage medium. The method comprises the steps that an electric arc audio target signal of welding operation is obtained; multi-dimensional feature extraction is carried out on the arc audio target signal to obtain a fusion feature vector, and the fusion feature vector is used for representing a timbre feature, a tone feature and a loudness feature; according to the fusion feature vector, the welding state of the welding operation is determined; and based on the welding state, control parameters of the welding operation are adjusted, so that the welding state is kept as the target state.

A method and system for providing a customized configuration of settings for an audio processor. The customized configuration is determined for a user based on results of a hearing test. The test may be provided by the system or another system. The customized configuration includes an increase or decrease to an intensity level of one or more frequency bands of one or more channels (e.g., a left and right) to compensate for a decreased hearing threshold level experienced by the user (e.g., as indicated by the hearing test results). The settings may be further adjusted based on an application of one or more equal-loudness contours representing varying sensitivity of the human ear to different frequencies. When audio is played by an application including or in communication with the audio processor, the audio is adjusted based on the customized configuration and the user is provided with an improved listening experience.