61 results about "Vocal intensity" patented technology

The invention relates to a method for testing a participant's ability to accurately detect speech under background noise, comprising: (a) audibly presenting a series of 2-bit numbers, 3-bit numbers, 4-bit numbers and / or 5-bit numbers in a random or fixed order to a participant's first ear at a fixed or varying signal-to-noise ratio (SNR) in a quiet or noisy environment, (b) audibly presenting a series of 2-bit numbers, 3-bit numbers, 4-bit numbers and / or 5-bit numbers in a random or fixed order to the second ear of the participant at a fixed or random signal-to-noise ratio (SNR) in a quiet or noisy environment, where the numbers are presented in the native language of the participant or the fluent language of the participant, (c) recording the numbers or sequence of numbers correctly recognized by the participant, and (c) storing the recorded numbers or sequence of numbers at the second ear of the participant. And (d) calculating a score as a function of the percentage of the number or sequence of numbers correctly replicated by the subject, or as a function of the sound intensity difference between the speech and noise required to achieve the correct recognition of 50% sequence of numbers. The invention also relates to a device and a method for measuring the digital recognition capability related to hearing loss, and to a computer-implemented method for evaluating the speech recognition capability of a participant.

The application discloses an AAC security steganography algorithm and system based on a masking effect. The human ear hearing masking effect can cause some audio signals with low intensity to be masked by signals with high intensity, and there is space for hiding secret information. Therefore, the application analyzes a quantization process of MDCT coefficients in AAC coding, records the masked audio signals as modifiable positions, and combines an STC adaptive steganography framework to realize embedding of secret information. Experimental results show that the algorithm can achieve a maximum embedding capacity of 13.61 kbps, can guarantee auditory concealment of speech, and has good security.

The application discloses a wireless microphone based on intelligent terminal control, which comprises a disc base, an outer cover and a holding handle, the holding handle is threadedly connected with the disc base, a battery is arranged in the holding handle, a mounting column is connected with the disc base through bolts, three embedded grooves are arranged on the outer wall of the mounting column, sound emitting pieces are arranged in the embedded grooves, a mounting hole is arranged in the center of the mounting column, and a sound receiving piece is arranged in the mounting hole, the sound emitting pieces, the sound receiving piece and a control panel are arranged in the microphone, the control panel is connected with an external audio device through a Bluetooth module, the sound receiving piece converts the received sound into an electric signal, and then the electric signal is transmitted to the sound emitting pieces for external playing, so that the sound conduction can be enlarged without external playing sound, and the relative position of the sound guide area and the sound emitting pieces is changed by rotating the second cover body, so that the sound intensity of the sound external playing and the sound conduction direction are changed.

The invention relates to a method and a system for measuring a sound source by measuring sound at a plurality of discrete measurement locations. AUGMENTED REALITY is used in that a camera scans a physical reference mark arranged in space relative to a source; the processing software generates a surface grid forming the measurement surface in the coordinate system and determines a measurement position to be occupied in a real three-dimensional space with reference to the reference mark such that the measurement position lies on the surface grid, the processing software also generates a virtual mark, and the virtual mark is located on the reference mark. And displaying the virtual indicia on a transparent display of a glasses-type presentation device such that the geometric position of the indicia in real space is fixed with reference to a reference mark and is independent of the orientation of the display. Subsequently, during the display of the indicia, sound measurement probes, which are visible through the display, are sequentially manually guided to the location of each virtual indicia, and sound intensity measurements are then performed there, respectively.

The invention discloses an intelligent power amplifier management method based on a vehicle-mounted sound system and a related device, and the method comprises the steps: firstly, obtaining vehicle window opening and closing state data, a current vehicle speed and an external wind speed; then, in-vehicle airflow velocity distribution data, turbulence intensity and in-vehicle airflow mode combination are calculated; then, simulating a sound propagation path by using an acoustic simulation model to obtain sound intensity distribution data of each area in the vehicle; if the sound intensity distribution data of each area in the vehicle indicates that the volume difference between different areas in the vehicle is greater than a preset volume difference threshold value, generating a first optimization parameter set of the sound amplification system based on a target noise source position positioned through a sound source positioning technology and eddy current noise data; and finally, extracting a volume gain value and an equalizer optimization parameter from the first optimization parameter set, and determining an output parameter of the sound system. Therefore, the problem of regional audition experience difference caused by airflow interference is solved, and all users in the vehicle can obtain better audition experience.

The utility model relates to the technical field of measurement, and discloses a sound toy decibel measuring and positioning device, which comprises a main body assembly, a decibel measuring box, a positioning assembly, a decibel measuring box, a sealing door hinged on one side of the decibel measuring box, a handle fixed on one side of the sealing door, and a positioning piece arranged in the decibel measuring box, and the positioning piece comprises a placing frame. The sound toy box has the beneficial effects that the sound toy can be positioned at the central position of the box, and the distances and angles of the six microphones relative to the sound production part of the toy can be kept relatively fixed and balanced, so that the difference of acoustic parameters such as sound intensity and frequency received by each microphone each time can be greatly reduced; the stability and the accuracy of a sound pressure level result obtained through final comprehensive analysis can be improved, and the real sound production condition of the toy can be reflected more accurately.

The invention discloses an underwater sound wave communication method, which comprises the following steps of: sampling, quantizing and binary coding an original analog signal of information to be transmitted at a transmitting end to obtain a binary bit stream; performing channel coding on the binary bit stream to obtain a coded code stream; sound source coding is carried out on the coded code stream to form a sound source array fuze so as to trigger and control corresponding sound sources to produce sound, and each sound source in a sound source array is uniquely determined by coordinates, sound frequency, sound intensity and a serial number; monitoring the frequency of a sound source at a receiving end, and matching a sounding sound source according to the arrival time, the arrival angle and the arrival intensity of the sound source signal; and decoding the bit code corresponding to the sounding source to extract transmission information. According to the scheme, a traditional complex modulation mode is abandoned, and the transmission efficiency, the transmission distance, the stability and the environmental adaptability of underwater acoustic communication can be improved.

The utility model provides a sound acquisition device and AR glasses. The sound acquisition device comprises at least one microphone; the at least one loudspeaker comprises a vibrating diaphragm, a coil and a magnetic piece; when the coil is electrified, the coil moves relative to the magnetic piece to enable the vibrating diaphragm to vibrate, and when the vibrating diaphragm vibrates, the coil moves relative to the magnetic piece to enable the coil to generate an induction electric signal; the control module is connected with the microphone and the loudspeaker, and the control module is used for obtaining a signal output by the microphone, obtaining an induction electric signal generated by the loudspeaker when the sound intensity of the sound intensity signal collected by the microphone corresponding to the environment sound is greater than or equal to a preset sound intensity threshold value, and sending the induction electric signal to the microphone; and converting the induction electric signal generated by the loudspeaker into the first sound data so as to improve the sound data acquisition effect of the sound acquisition device when the sound intensity of the environment sound is too high.

This invention discloses an FPGA implementation method for visual image processing of sound source identification, comprising: 1. Normalizing the sound intensity values ​​obtained from sound source identification and converting them into grayscale values; 2. Obtaining corresponding color pixel values ​​by performing three independent mappings on the grayscale values; 3. Upsampling the low-resolution color acoustic cloud map using an interpolation algorithm to improve the image resolution and form a visual acoustic cloud map that represents sound intensity in color. This invention enables the visual image processing of sound source identification to be implemented on an FPGA in the form of digital circuits, thereby improving the real-time performance of the entire sound source identification process, reducing the hardware complexity of the sound source identification platform, and saving time and hardware costs.

The utility model belongs to the technical field of electronic buzzer, and discloses a buzzer. The buzzer comprises an outer shell and a sound production assembly, the outer shell comprises a main shell body, an inner cover and an outer cover, the inner cover is connected with the main shell body, the outer cover covers the outer side of the inner cover and is connected with the main shell body, a conical hole is formed in the inner cover, and the small end of the conical hole faces the main shell body; the sounding assembly is arranged in the main shell, and the sounding end of the sounding assembly at least partially penetrates through the small end of the taper hole. The shell with the main shell and the outer cover is arranged, the sound production assembly is arranged in the shell, and the shell can play a role in protecting the sound production assembly; the inner cover is arranged, the conical hole is formed in the inner cover, the sound production end of the sound production assembly at least partially penetrates through the small end of the conical hole, when sound produced by the sound production assembly passes through the conical hole, the conical hole can enable the sound to be transmitted in a concentrated mode, the effect of enhancing the sound intensity is achieved, and sound transmission and diffusion are facilitated.

The utility model discloses a mute type composite aluminum plate, relates to the composite aluminum plate technology field, the mute type composite aluminum plate comprises a plurality of base plates and a plurality of mute structures, the plurality of mute structures are respectively arranged between two adjacent base plates, the mute structure comprises a sound insulation layer, a reinforcing layer and a mute assembly which are sequentially arranged from the middle base plate to the other base plate, the sound insulation layer comprises a sound insulation felt arranged on the side face of the middle base plate, and the reinforcing layer comprises rock wool arranged on the side, away from the middle base plate, of the sound insulation layer. The mute assembly comprises a sound absorption layer and a weakening layer which are sequentially arranged from the sound insulation assembly to the direction away from the middle base plate, a containing cavity is formed in the sound absorption layer, and the weakening layer is arranged on the side, away from the middle base plate, of the sound absorption layer. The sound intensity is weakened through the mute assembly, the sound intensity is further weakened in cooperation with the upper reinforcing layer, and finally part of sound waves are reflected through the sound insulation layer.

An electronic device is provided. The electronic device analyzes music so as to extract the sound intensity and the rhythm pattern from the music, generates, on the basis of the extracted sound intensity and the extracted rhythm pattern, guide data for an exercise program corresponding to the music, and can determine the exercise content of the exercise program on the basis of the generated guide data and movement information received from a wearable device connected to the electronic device.

The application discloses a strong sound and light repelling system composite material sound basin and a preparation method thereof, and belongs to the technical field of strong sound repelling and driving away equipment. The strong sound and light repelling system composite material sound basin has an integrated multi-element conical structure, is manufactured from high-strength, high-modulus and high-toughness fiber fabric and a high-toughness and high-damping epoxy resin system, and has the advantages of high overall structural strength, high rigidity, good damping property, high stability, outstanding sound intensity, excellent sound directivity and clearness of shouting and the like. The preparation method of the strong sound and light repelling system composite material sound basin comprises the following steps: designing and preparing a mold comprising an integrated multi-element conical cavity, manufacturing by using a vacuum adsorption pouring forming process, impregnating prepared epoxy resin into a preformed body, and finally forming the strong sound and light repelling system composite material sound basin in an integral manner. The strong sound and light repelling system composite material sound basin has the advantages of simple forming process and low cost.

The application discloses a sound perception-based electronic pet behavior triggering method, and relates to the technical field of intelligent interaction.The method comprises the following steps: collecting environmental sound signals through a microphone array and constructing a sound feature vector, determining sound intensity levels, sound source directions, and user voice classes, environmental event sound classes, and third-party biological sound classes; analyzing different sound classes to obtain user instruction intentions, user emotion labels, and sound event types; dynamically updating in combination with current emotion parameters and energy parameters of the electronic pet; generating priority behavior instructions based on the above information and sending the priority behavior instructions to an action execution module; and monitoring the short-term energy of the electronic pet in real time during the execution process and performing self-noise control.The technical problems of single sound processing, rigid state updating, serious self-noise interference, and lack of continuity of behavior response in the life state of the existing electronic pet are solved.

The application discloses a sound source sound intensity estimation method, device and equipment and a computer readable storage medium. The method comprises the following steps: positioning the angle of a vehicle sound source relative to a microphone array based on sound signals collected by the microphone array and video data collected by a camera device; calculating the distance between the vehicle sound source and the microphone array according to the angle of the vehicle sound source relative to the microphone array and the installation position information of the microphone array; and calculating the sound intensity data at the vehicle sound source according to the distance and the intensity data of the sound signals of the vehicle sound source collected by the microphone array. The application improves the accuracy of the estimated noise intensity of the vehicle sound source, thereby improving the monitoring and management accuracy of traffic noise.

In one aspect, opposing microphones may be mounted within a synthetic human ear. The synthetic human ear may form part of a head and torso simulator (HATS) system used for measuring output of any sound source, such as those embodied in headphones or loudspeakers. The opposing microphones may be used to calculate sound intensity and / or sound power, rather than just sound pressure level. The determined sound intensity / power can then be used to research and / or develop audio products, and / or quantify how a sound source is perceived regardless of acoustic loading on the HATS.

The present application provides an information processing method, apparatus, device and storage medium, the information processing method comprising: in response to a request to watch a target game match, displaying a first game screen from a first perspective on the viewing interface of the target game match; in response to detecting the sound of a first virtual object in a first participating team, determining a first sound diffusion range of the first virtual object in the game scene according to a first position of the first virtual object in the game scene of the target game match and a sound intensity parameter of the sound of the first virtual object; in response to the intersection of the first sound diffusion range and the first sound collection range of the second virtual object in the second participating team, outputting target prompt information. In this way, the present application can provide corresponding prompts to the audience when detecting sound interactions between virtual objects in different participating teams, effectively improving the audience's event experience dimension and event participation.

The application provides a three-dimensional virtual surround sound generation method and generation system, and relates to the field of surround sound. The application adopts an excellent structure of a howling elimination technology to eliminate howling, enhance spatial positioning and speech clarity. Reflection sound is constructed by infinite N-level echo simulation to construct a surround sound channel, so that the generated surround sound has more distant background sound, thereby making the virtual surround sound have strong spatial sense, distance sense, diffusivity and surrounding sense, rather than just positioning sense. A post sound channel is constructed by inversely superimposing the delayed signal on the direct signal, so that the spatial sense of the surround sound is obviously enhanced, and the sound seems to come from all directions. The original sound bass is preserved by separation technology, and the virtual sound in each direction is freely controlled, thereby controlling the sound intensity, tone brightness and tone softness.

The invention discloses an equipment control method and device, a storage medium and electronic equipment, and relates to the technical field of smart home, the equipment control method comprises the steps that environment information and sound signals of a target area and object information of a target object are acquired, and the target area is the area where the target object is located; determining equipment noise and non-equipment noise in the sound signal, and determining first sound intensity of the equipment noise and second sound intensity of the non-equipment noise; determining whether to generate a control instruction according to the environment information, the object information, the first sound intensity and the second sound intensity; under the condition that the control instruction is determined to be generated, the equipment in the target area is controlled according to the control instruction, and by adopting the technical scheme, the problem that in the prior art, under the condition that different types and sources of noise cannot be recognized, the operation frequency of the smart home is adjusted at one time, and the noise control effect is poor is solved.

The embodiment of the invention belongs to the display technology, and provides a display device, a sound adjustment method, a storage medium and a program product. The display equipment comprises an audio acquisition device, an image acquisition device and a controller connected with the audio acquisition device and the image acquisition device, acquiring position information of a spokesman through an image acquisition device, and acquiring a sound signal of the spokesman through an audio acquisition device; gains of the sound signals are obtained according to the position information; the gains of the sound signals comprise sound intensity gains and / or sound channel balance gains; and performing stereo adjustment on the sound signal based on the gain of the sound signal. The stereo adjustment is performed on the collected sound signals of the spokesman based on the positions of the different spokesman, so that the stereo adjustment can be performed in a targeted manner, the stereo adjustment effect can be improved, and the user experience is improved.

Some embodiments of the present application show an intelligent device and a spatial sound field calibration method, the method comprising: in response to a user input instruction to start spatial sound field calibration, controlling the loudspeaker to play a preset standard sound at a first sound intensity; receiving a first audio file; calculating a first energy value of a preset frequency point based on the first audio file, and determining a first number of target size audio data in the first audio file; determining a first sound gain of the preset frequency point according to the first energy value, the first number, a preset target gain, a bandwidth and a manual adjustment offset; and calibrating a frequency response curve corresponding to the first sound intensity based on the first sound gain. The embodiments of the present application adjust the sound of the intelligent device according to the room acoustic correction algorithm by collecting the sound attenuated by the room, so that the user can enjoy better sound effect in any differently decorated room and improve the user experience.

The invention discloses a light intelligent control method and system based on sound change, and relates to the technical field of light intelligent control, and the method comprises the steps: obtaining an environment sound signal containing a sound pressure level time sequence in a target sound field region; importing the voiceprint data into a voiceprint feature analysis engine, constructing a time pane matrix which stretches and retracts along with the sound intensity, and generating a voiceprint energy streamline; capturing a slope abrupt change event of the voiceprint energy streamline, and extracting an acoustic disturbance feature packet; the feature packet drives a brightness controller to output a light intensity compensation deformation instruction, compensation deformation is applied to a time pane matrix, and new voiceprint energy streamline distribution is generated; and identifying the steady-state voiceprint channel and mapping the steady-state voiceprint channel into a target illumination value instruction and a color temperature offset According to the method, accurate adaptation of lamplight and sound changes can be realized, and the continuity and stability of lamplight control are improved.

This invention relates to the field of active vibration isolation technology, and more particularly to an active vibration isolation method based on noise reduction technology, comprising: acquiring in real time the noise signal of turbine A in a set of turbine units; determining the noise signal type based on the noise characteristic characterization parameters of the noise signal, and determining a strategy for generating a noise cancellation signal according to the noise intensity evaluation value of the corresponding noise signal type; determining the movement strategy of different sound-absorbing layers of the vibration isolation box based on the noise cancellation characteristic evaluation value of the noise cancellation signal to output the noise cancellation signal to turbine B adjacent to turbine A; determining the qualification of the noise cancellation signal based on the sound intensity value of the output noise cancellation signal and the noise signal of turbine B; and, if the noise cancellation signal is determined to be unqualified, determining to increase the gap between different sound-absorbing layers or increase the pores of the same sound-absorbing layer based on the difference between the sound intensity value and the preset sound intensity value. This invention improves the vibration isolation efficiency of turbine noise.

The present application relates to a kind of earphone control method, device, electronic equipment and storage medium, the earphone control method, comprising: the sound intensity and frequency of the audio output by the earphone in the wearing state are acquired;The hearing range of the user wearing earphone is acquired;When determining that the audio component corresponding to frequency and sound intensity is located outside hearing range, adjust the intensity gain of the audio output by earphone, so that the audio component corresponding to frequency and adjusted sound intensity is located in hearing range.The embodiment of the present application can automatically adjust the intensity gain of the earphone when detecting that the audio component corresponding to the sound intensity and frequency of the audio is located outside the hearing range of the user, so that the audio component corresponding to the frequency and the adjusted sound intensity is located in the hearing range, ensure that the earphone plays audio is always in the safe range, avoid damaging the hearing of the user.

The invention belongs to the technical field of measuring equipment, and particularly relates to a water meter data processing system and method and a readable storage medium. The water meter data processing system comprises a collaborative decision center, a plurality of intelligent water meters and a plurality of sound detection devices; the collaborative decision-making center controls the water outlet end of each water pipeline to be closed and obtains sound intensity through corresponding sound detection equipment so as to judge whether the water pipeline leaks water or not and judge the water leakage position; according to the system, the total flow of all the later-stage water pipelines and the flow of the previous-stage water pipelines are detected in real time, the water pipelines can be coarsely positioned, meanwhile, the sound intensity obtained through the sound detection equipment is compared with the water leakage judgment database, the water pipelines leaking water and the water leakage positions are accurately positioned, and the water leakage accuracy is improved. And each water pipeline only needs to use one intelligent water meter, the cost of the sound detection equipment is lower than that of the intelligent water meter, and the accuracy of water leakage detection is improved on the premise that the use cost is reduced.

The invention provides a dynamic audio-visual navigation method based on intensity calibration geometric coding and cross-modal scene aggregation. The dynamic audio-visual navigation method is suitable for application scenes such as service robots, security inspection and virtual reality interaction. According to the method, a depth visual image and a binaural spectrogram serve as input, firstly, a sound intensity map and depth geometric information are explicitly fused through an intensity calibration geometric encoder, and physical-space correlation across sound categories is modeled; then, a cross-modal scene aggregator is utilized to convert the multiple groups of spatial features into a Token sequence, fine-grained and content-aware multi-modal dynamic fusion is realized through Transform, and global state representation rich in reasoning ability is generated; and finally, training a navigation strategy of time sequence perception under a near-end strategy optimization framework based on the representation, and outputting a tracking action sequence aiming at the mobile sound source. According to the method, the navigation success rate and the path efficiency are remarkably improved under the conditions of a mobile sound source and unknown sound, and the generalization ability and the decision robustness of the system in a dynamic environment are enhanced.

The utility model belongs to the field of audio technology and acoustic engineering, and discloses a sound amplifying cover for audio equipment, which comprises a hollow cylinder, a connecting piece and a horn, the hollow cylinder, the connecting piece and the horn are all processed and manufactured by log wood, the hollow cylinder is sleeved outside a loudspeaker, the rotatable connecting piece is arranged above the hollow cylinder, and the horn is arranged above the connecting piece. The connecting piece is provided with a horn which is convenient to disassemble and assemble, the hollow cylinder is formed by splicing and combining four groups of zigzag blocks, each zigzag block is of a zigzag arc-shaped structure, an annular groove is formed in the inner side of the hollow cylinder, the bottom of the connecting column is provided with a male ring corresponding to the annular groove, and the male ring is slidably clamped in the annular groove, so that the connecting piece can rotate relative to the hollow cylinder; the loudspeaker is made of wood materials with different sizes and opening angles. The loudspeaker is compact and reasonable in structural design, can improve the sound intensity in the opening direction, achieves the sound amplification effect, can achieve various timbres, improves the music quality, can adjust the angle, and is convenient to use.

The invention discloses an equipment voice control method, device and system and electrical equipment, and belongs to the field of voice equipment control. And when the electrical equipment receives the wake-up word, acquiring first wake-up information, then determining an arbitration strategy based on the connection condition with the control center, and determining whether to respond to wake-up based on the arbitration strategy and the first wake-up information. During offline arbitration, the electrical equipment determines whether to respond to awakening based on the voiceprint features, the awakening time, the sound source direction and the sound intensity, and during online arbitration, the control center determines the electrical equipment responding to awakening based on the connection time, the voiceprint features, the sound source direction and the sound intensity. According to the scheme of the invention, both offline arbitration and online arbitration can be realized, and quick judgment of the electrical equipment is not affected no matter whether the network is poor or not. In addition, during offline arbitration, control of a central device or a server is not needed, wake-up delay does not need to be introduced, the response speed is high, and response is accurate. And the use experience of the user is greatly improved.

The utility model relates to a straight pen type vascular tinnitus intraoperative sensor. The straight pen type vascular tinnitus intraoperative sensor comprises a panel host, a wire, a handle, a connecting pipe and a sensor body. The panel host is connected with the handle through a wire, the other end of the handle is connected with the connecting pipe, and the sensor is installed at the end of the connecting pipe. The sensor comprises a one-way centralized radio device and a position sensor; the one-way centralized sound receiving device is used for collecting sound at a sinus sigmoideus bone wall defect part and transmitting decibels of sound intensity, sound frequency and sound interval time to the panel host, and the decibels, the sound frequency and the sound interval time are displayed through the panel host; the position sensor is used for detecting the specific position and tinnitus positioning of a sinus sigmoideus bone wall defect part and transmitting vascular tinnitus sound and intraoperative position information to the panel host, and the information is displayed through the panel host. The device has the advantages that whether blood flow sound exists at the defect position of the bone wall or not can be determined, and the possibility of defect of the rest sinus sigmoideus bone walls is found out.

The invention discloses a voice control information acquisition device and a tobacco curing barn control system. The device comprises a first volume sensor, a second volume sensor, a voice interaction unit and a main control unit. The voice interaction unit determines voice control information in the collected sound signals and sends the voice control information and the sound duration time period to the main control unit; the main control unit is used for controlling the sound intensity of a sound signal received by the first volume sensor in the sound duration time period and the sound intensity of a sound signal received by the second volume sensor in the sound duration time period when the sound duration time period is received. And determining whether a sound source sending the sound signal is in a preset area, determining a control instruction corresponding to the voice control information when the sound source is in the preset area, and sending the control instruction to a far-end upper computer. According to the mode, the acquisition accuracy of the voice control information can be improved.