479 results about "Loudness" patented technology

The invention discloses a sound production device and electronic equipment, and relates to the technical field of electroacoustic transduction, a magnetic circuit system of the sound production device comprises a magnetic conductive yoke, and side magnetic parts, a common magnetic part and two central magnetic parts which are arranged on the magnetic conductive yoke, the side magnetic parts are annularly arranged to form an accommodating space, the common magnetic part is located in the accommodating space and is connected with the side magnetic parts, and the two central magnetic parts are arranged in the accommodating space. The magnetic circuit system is arranged in the containing space to divide the containing space into two installation spaces, each center magnetic part is located in one installation space, the center magnetic parts, the side magnetic parts and the common magnetic part are spaced to define a magnetic gap, the vibration system comprises a vibrating diaphragm assembly and two voice coils, the periphery of the vibrating diaphragm assembly is connected to the shell and is opposite to the magnetic circuit system, and the two voice coils are arranged side by side. One end of each voice coil is connected to the diaphragm assembly, and the other end of each voice coil is suspended in one magnetic gap. According to the sound production device, the size of the voice coil and the magnet volume of the magnetic circuit system are increased, so that the BL value is improved, and the whole machine performance and the sound effect loudness are improved.

A pickleball ball is provided for increasing durability, improving performance, and / or reducing the loudness and / or frequency of sound generated when said ball strikes a paddle. The ball being a hollow ball comprised of a spherical shell defining a hollow cavity. The ball is made of a graphene-infused polymer material and the shell defines a plurality of holes distributed around the surface of the ball.

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.

The invention relates to a self-adaptive frequency optimization system in radio and television microwave transmission, and the system comprises an initial audio collection and processing module which collects an audio signal of a current frequency channel in real time, and generates a first data set; the conversion audio acquisition processing module synchronously acquires a target channel signal according to the frequency switching signal, and calculates the loudness offset between two channels; a feature extraction module extracts frequency domain energy distribution feature vectors of the first data set and the second data set; the model construction module constructs a dynamic gain compensation model and generates a compensation value of a target channel; the tone quality compensation module performs smoothing processing on the compensation value to generate an optimized third data set; and the audio optimization module updates model parameters, dynamically adjusts a frequency domain analysis strategy, and outputs low-delay and high-stability radio and television transmission signals. According to the invention, the problems of high switching delay and degraded tone quality caused by static configuration in a traditional scheme are solved, and the transmission efficiency and user experience in a complex electromagnetic environment are remarkably improved through modular dynamic calibration and compensation.

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.

The invention discloses a sound control method and device for a vehicle cabin and a vehicle. The method comprises the steps that a first sound signal sent by a first passenger in the vehicle is obtained, and position information of a second passenger in the vehicle is obtained; determining a second sound signal based on the first sound signal, the position information of the second passenger and the position information of first sound playing equipment in the vehicle; the first sound playing device is controlled to output a second sound signal, so that interference occurs when the first sound signal and the second sound signal are transmitted to the position where the second passenger is located, and the sound loudness of the position where the second passenger is located is smaller than a preset loudness threshold value. According to the invention, the technical problem of low privacy of conversation in the vehicle cabin is solved.

The invention discloses a song teaching template generation method and system, equipment and a program product. The method comprises the following steps: acquiring song audio information of a target song audio; wherein the song audio information comprises lyrics, lyric timestamps, time points corresponding to paragraphs and audio representation information; performing various intra-sentence teaching feature analysis on the lyric timestamps and the audio representation information of the target song audio, and marking various analyzed intra-sentence teaching features on lyrics; wherein the in-sentence teaching features comprise at least one of accent, treble, long tone, tremolo, gliding and breathing points; the singing difficulty feature comprises at least one of high pitch and long pitch; the skill features comprise at least one of tremolo and glide; on the basis of the time point corresponding to each paragraph and the audio representation information, performing in-segment teaching feature marking on the loudness gradient part of each paragraph in the lyrics of the target song audio; and generating a song teaching template based on the marked lyrics of the target song audio and outputting the song teaching template.

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.

The invention discloses a sound production device and electronic equipment, and relates to the technical field of electroacoustic transduction, a magnetic circuit system of the sound production device comprises a magnetic conductive yoke, and side magnetic parts, a common magnetic part and two central magnetic parts which are arranged on the magnetic conductive yoke, the side magnetic parts are annularly arranged to form an accommodating space, the common magnetic part is located in the accommodating space and is connected with the side magnetic parts, and the two central magnetic parts are arranged in the accommodating space. The central magnetic part is arranged in the accommodating space to divide the accommodating space into two mounting spaces, each central magnetic part is positioned in one mounting space and is separated from the side magnetic part and the common magnetic part to define a magnetic gap, the common magnetic part is provided with a magnetic flux port communicated with the two magnetic gaps, and a voice coil of the vibration system comprises two annular parts and a conduction part connected with the two annular parts. The two annular parts are respectively suspended in the two magnetic gaps, and the conducting part is suspended in the magnetic flux port. According to the sound production device, the size of the voice coil and the magnet volume of the magnetic circuit system are increased, so that the BL value is improved, and the whole machine performance and the sound effect loudness are improved.

The invention relates to a multi-source data fusion-based Ningxia cell noise map construction and evaluation method, which comprises the following steps of: establishing a frequency-variable sound absorption parameter matrix of a building surface based on multi-spectral scanning data and an acoustic database, and combining a burst noise voiceprint recognition result and traffic monitoring data, generating a burst noise influence map through the burst noise space-time correlation model; the real-time sound pressure data and the frequency-variable sound absorption parameter matrix are fused, and a sound field is dynamically reconstructed to generate a noise thermodynamic diagram; and based on a Zwicker loudness model, coupling a burst noise influence map and resident electrocardiogram monitoring data, calculating a composite noise dysphoria index, constructing a noise map in combination with a noise thermodynamic diagram, and carrying out comprehensive evaluation by adopting an entropy weight-TOPSIS algorithm to obtain a noise evaluation result. According to the method, the noise map is constructed through multi-source heterogeneous data fusion, dynamic sound field modeling and physiological and psychological coupling calculation, and evaluation is performed in combination with multi-dimensional indexes, so that precise evaluation of a cell noise environment and scientific dynamic optimization of a noise reduction scheme are realized, and objectivity and pertinence of noise treatment are effectively improved.

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 sound production device and electronic equipment, and relates to the technical field of electroacoustic transduction, a magnetic circuit system of the sound production device is provided with a magnetic gap, a diaphragm assembly of a vibration system is opposite to the magnetic circuit system, and a voice coil is suspended in the magnetic gap; wherein the voice coil comprises two first side edges and two second side edges which are connected end to end, the two first side edges are oppositely arranged and are sunken towards the inner side of the voice coil to form a first sunken part, and the two second side edges are oppositely arranged and are sunken towards the inner side of the voice coil to form a second sunken part; the two first concave parts and the two second concave parts define a first area, a second area is formed between the adjacent first concave part and second concave part, and the magnetic circuit system is provided with a first magnetic part located in the first area and a second magnetic part located in the second area. According to the sound production device, the size of the voice coil and the magnet volume of the magnetic circuit system are increased, so that the BL value is improved, and the whole machine performance and the sound effect loudness are improved.

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 invention belongs to the technical field of nuclear power, and particularly relates to a black box transferable voiceprint recognition system attack resisting method and device based on frequency domain disturbance. According to the invention, the perception of the human ear on the voice signal is considered and simultaneously depends on the characteristics of the frequency and the loudness of the voice signal, and the adversarial disturbance is added in the frequency interval insensitive to the human ear, so that the voice adversarial sample generated by the method is hidden to the human ear, the human ear is difficult to distinguish, and the model classification is wrong; in order to solve the problems that most black-box countermeasure attack methods need to query a large amount of attack targets, the attack time is long, and practical application is difficult, the method utilizes the transferability of countermeasure samples, adopts a three-stage iteration method, and improves the concealment and generation speed of the countermeasure samples, so that a large amount of query of the attack targets is avoided, and the security of the countermeasure samples is improved. The method achieves the high-success-rate countermeasure attack for an unknown voiceprint recognition system.

The invention provides a snore recognition method and device, equipment and a storage medium. The snore recognition method comprises the steps that audio signals within a preset time period are obtained; drawing an envelope line based on the audio signal, and obtaining the number of envelopes in the envelope line; wherein each envelope comprises a section of ascending curve and a section of descending curve; acquiring associated data of a starting point, a midpoint and an end point of each envelope; wherein the middle point is the point with the maximum sound loudness value; the associated data comprises sound loudness; performing snore judgment on the audio signals in the preset time period based on the associated data of the starting point, the midpoint and the ending point of each envelope; and if the sound loudness of the starting point, the sound loudness of the middle point and the sound loudness of the end point are different and the sound loudness of the middle point is greater than or equal to second preset loudness, judging that the audio signal in the preset time period has snore. The snore is recognized based on the sound loudness of the audio signal, the recognition algorithm is simple, and the cost is lower than that of an AI algorithm.

The invention provides a vehicle whistling control method and system, and the method comprises the steps: obtaining the environment noise data of a vehicle, and determining a reference whistling parameter according to the environment noise data; in response to a whistling instruction sent by a driver, acquiring the duration of the whistling instruction, and marking the whistling type of the whistling instruction according to the duration; under the condition that the whistling instruction is short whistling, the whistling loudness and the whistling tone are set according to a first rule, and the first rule comprises at least one of reference whistling parameters, the vehicle speed, the vehicle position and a vehicle front pedestrian sign; under the condition that the whistling instruction is long whistling, the whistling loudness is set to be first loudness, the whistling tone is set to be first tone, the first loudness is larger than the reference loudness, and the first tone is higher than the reference tone. According to the method, the whistling loudness and tone can be adjusted according to the vehicle whistling duration, the vehicle speed, whether the vehicle is in the no-whistling area or not and whether pedestrians exist in front of the vehicle or not, and then noise pollution is reduced.

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 relates to the technical field of terminal equipment, and discloses a loudspeaker and electronic equipment. The loudspeaker comprises a frame, a magnetic circuit system and a vibration system, the magnetic circuit system and the vibration system are connected with the frame, and the magnetic circuit system is fixed at one end of the frame. The vibration system comprises a vibrating diaphragm, a dome and a voice coil. One end of the vibrating diaphragm is fixed at the other end of the basin frame, and the vibrating diaphragm protrudes towards the direction of the basin frame; the globe roof comprises a first globe roof and a second globe roof, the first globe roof is located on the side, away from the basin stand, of the vibrating diaphragm, the second globe roof is annularly arranged on the first surface of the first globe roof, and the second globe roof is connected with the other end of the vibrating diaphragm; the voice coil is fixed on the first surface of the first dome and is opposite to the magnetic circuit system; the voice coil is configured to generate a magnetic field effect with the magnetic circuit system to drive the first dome to vibrate and produce sound. Therefore, the dome is divided into two parts, and the diaphragm is used for transversely supporting the dome, so that the shape of the dome can be maximized, the area of the first dome can be increased, the effective sound radiation area of the loudspeaker can be increased, and the loudness of the loudspeaker can be increased.

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.