77 results about "Vocal intensity" patented technology

To provide an information processing system, a control method for the information processing system, and a program that allow a user to distinguish a sound from a specific sound source while leaving the sounds from each sound source as they are, even when a plurality of sound sources are present in a direction of interest that is of interest to the user.SOLUTION: An information processing system 1000 comprises: display means (a display unit 107) capable of displaying an image of a mixed reality space; playback means (an audio output unit 105) capable of playing back sounds from a plurality of sound sources to a user; vibration applying means (a vibration unit 117) for applying vibration to the user; control means (a control unit 111) for controlling displaying a virtual object that can be linked to each sound source on the display means; and selection means for selecting the virtual object (an object selection instruction unit 116). The vibration applying means applies vibration to the user in accordance with the intensity of the sound from the sound source associated with the virtual object if the virtual object is selected.SELECTED DRAWING: Figure 2

The invention relates to the technical field of voice processing, in particular to a sound source direction-finding positioning method and system for a complex environment, and the method comprises the steps: serializing the sound intensity of each microphone, and calculating a maximum receiving interval according to the distance between adjacent microphones, the sound transmission speed and the sampling frequency; the method comprises the following steps: determining a short-time audio similarity through moving a sound source signal sequence, extracting a sound source synchronization sequence and a sound source same-frequency sequence, and further evaluating a tone quality deviation degree; optionally selecting a reference microphone, and comparing the reference microphone with a sound source synchronization sequence of other microphones to obtain a time delay data length and a sound source similar sequence; based on the arrangement values of the same position elements of the sound source similar sequence, a sound intensity distribution sequence is obtained, and the sound intensity interference degree of each microphone is obtained; and determining a fixed beam direction and calculating the distance between the reference microphone and the sound source. The invention aims to improve the accuracy of sound source positioning.

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 invention relates to the technical field of data processing, in particular to a human-computer interaction spoken language evaluation method and system based on speech recognition, and the method comprises the steps: obtaining a phoneme sequence of a speech signal of a target examinee, calculating the initial segmentation probability of a target moment in a time window of a target phoneme, and obtaining a speech signal of the target examinee; taking a time window between the target moment and the ending moment of the target phoneme as a first time period, taking a time window between the starting moment of the next phoneme and any moment of the next phoneme as a second time period, and calculating the similarity overlapping degree of the sound intensity of the first time period and the sound intensity of the second time period, and calculating a final segmentation probability of the target moment based on the initial segmentation probability and the similarity overlapping degree, and screening segmentation points to segment the phoneme sequence by using the final segmentation probability, so as to obtain a matching score of the segmented phoneme sequence and a pre-stored standard phoneme sequence based on a GOP algorithm. According to the invention, the accuracy of the obtained matching score can be improved, so that the accuracy of the spoken language evaluation result is ensured.

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 present invention provides a motion control method and system for an interactive teaching robot, which collects sound information around the environment in which the teaching robot is located, controls the teaching robot to switch working states and wake it up according to voiceprint feature information and semantic information in the sound information, and then determines the location of the user according to the sound intensity distribution information in the sound information so as to capture an environmental image; then, identifies obstacles in the environmental image to determine the moving path of the teaching robot, and instructs the teaching robot to play corresponding voice reply messages during the movement, so that it can be ensured that the teaching robot can follow the user in a timely and accurate manner under the user's voice control, and quickly interact with the user by voice, thereby improving the movement flexibility of the teaching robot in the process of interaction with students and improving the intelligent and automated degree of interaction between the teaching robot and students.

The present application discloses a video call voice processing method, a communication terminal, and a readable storage medium. The video call voice processing method includes: identifying a target person in a current video image; obtaining an answering parameter of the target person, where the answering parameter is used to identify the sound intensity heard by the target person and includes at least one of the position of the target person in the current video image, the action of the target person, and the voice of the target person; adjusting the gain of a sound collection device according to the answering parameter, and transmitting a voice signal collected by the adjusted sound collection device to a target terminal. The present application can adjust the input gain of its own microphone according to the situation of the other party in a call scenario, thereby facilitating the provision of high-quality voice call services for users.

The invention is suitable for the field of intelligent sound box control, and provides a broadcast control method and system for an intelligent sound box, and the system comprises a sound collection and direction judgment module, a sound feature extraction and separation module, a voice recognition and content analysis module, a region division and priority judgment module, and a broadcast content generation and region binding module. According to the scheme, voice input in multiple directions can be received at the same time, and the direction of each sound source can be accurately recognized. The capability remarkably improves the operation efficiency of the sound box in a complex environment, and can meet the requirement that multiple persons send out instructions at the same time. Through analysis of the sound intensity and the time difference, the sound box can accurately identify the position of each sound source, and the space is divided into different areas. According to the dynamic space management mode, the sound box can be intelligently adapted according to the actual environment, and the accuracy of sound source identification and the correlation of response are improved.

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.

The present invention provides a vehicle control method and device based on voice, a storage medium, and an electronic device, which belong to the field of vehicle control, wherein the method includes: obtaining a voice signal collected by a vehicle microphone; using a preset voiceprint model to identify the pronunciation object of the voice signal, and identifying the semantic information and sound intensity of the voice signal; searching for a control authority type that matches the sound intensity; and generating a vehicle control instruction of the control authority type based on the pronunciation object and the semantic information. Through the embodiments of the present invention, the technical problem that only authenticated users are allowed to perform voice control on the vehicle in the related art is solved, the personalization of the in-vehicle voice interaction and the recognition accuracy of the voice instructions are improved, thereby providing users with a smarter, more pleasant, and safer driving experience.

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 invention relates to the technical field of speech recognition, and specifically discloses a method and system for recognizing speech emotions. The present invention obtains time-domain feature information, frequency-domain feature information, and channel-domain feature information, and obtains the sound intensity, timbre eigenvalue, and voicelessness rate according to the time-domain feature information, frequency-domain feature information, and channel-domain feature information. Finally, a comprehensive correlation value eigenvalue is obtained according to the sound intensity, timbre eigenvalue, and voicelessness rate. Finally, emotional timbre keywords are extracted according to the comprehensive correlation value eigenvalue, and the emotional timbre keywords are compared with a preset emotional timbre definition library to obtain an emotional timbre definition. In this way, by comparing the preset emotional timbre definition library, the keywords are mapped to specific emotional categories, so as to realize that emotional recognition can know the emotional characteristics of the speech during the speaking process. In this way, the true intention of the speech content can be obtained through the emotional timbre definition.

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 present invention relates to the field of data processing technology, and more specifically, to a human-computer interactive oral language assessment method and system based on speech recognition, the method comprising: obtaining a phoneme sequence of a target candidate's speech signal, calculating an initial segmentation probability of a target moment within a time window of the target phoneme, taking a time window between the target moment and the end moment of the target phoneme as a first time period, taking a time window between a start moment of a next phoneme and any moment of the next phoneme as a second time period, calculating a similarity overlap between the sound intensity of the first time period and the second time period, calculating a final segmentation probability of the target moment based on the initial segmentation probability and the similarity overlap, and using the final segmentation probability to screen a segmentation point to segment the phoneme sequence, so as to obtain a matching score between the segmented phoneme sequence and a pre-stored standard phoneme sequence based on a GOP algorithm. The present invention can improve the accuracy of the obtained matching score, thereby ensuring the accuracy of the oral language assessment result.

The invention discloses an image generation method and device, a storage medium and electronic equipment. The method comprises the following steps: determining a target sound type of a target sound collected by a target terminal, and determining a target basic element corresponding to the target sound type; the target direction where the target sound is located is determined, the target extension direction corresponding to the target direction is determined, the target extension direction is the extension direction of the target texture, the target texture is the texture obtained by arranging a group of target basic elements into a predetermined shape according to the extension direction, and the target extension direction changes along with the change of the target direction; determining the target sound intensity of the target sound, and determining a target number corresponding to the target sound intensity; and displaying the target texture on a preset first image according to the target extension direction and the target number to obtain a second image. According to the invention, the technical problem of low efficiency in the image generation process in the prior art is solved.

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.