1123 results about "Pickup" patented technology

The invention relates to an intelligent microphone pickup and speech enhancement method, system and device, and the method comprises the steps: obtaining an audio signal collected by a dual-silicon microphone array, and carrying out the cross-correlation analysis of the audio signal, and obtaining a corresponding human voice signal correlation feature; performing sound source positioning analysis on the audio signal based on the human voice signal correlation feature to obtain a corresponding target human voice signal; performing frequency characteristic analysis on the target human voice signal, and performing segmented dynamic gain processing on the signal according to a preset frequency band range to obtain a corresponding voice enhancement signal; performing noise component adaptive filtering processing on the target human voice signal to obtain a corresponding noise suppression parameter; and inputting the speech enhancement signal and the noise suppression parameter into a preset echo cancellation model for joint optimization to obtain a corresponding output human voice signal. According to the invention, the coupling problem of multiple acoustic interferences can be effectively solved.

The invention relates to the technical field of atmosphere lamp control, and discloses an audio analysis control method of a pickup atmosphere lamp and an atmosphere lamp thereof.The audio analysis control method of the pickup atmosphere lamp comprises the steps that the atmosphere lamp collects sound through a built-in sound receiving part, and the sound collected by the sound receiving part comprises music, music, music, music, music and the like; the sound is analyzed and processed through audio analysis, the audio analysis comprises sound preprocessing, the preprocessed sound is subjected to time domain feature extraction and frequency domain feature extraction to obtain the energy ratio of each frequency band of the sound, and each frequency band comprises a high-energy frequency band and a low-energy frequency band; and performing moving average filtering on the low-frequency-band energy to generate an energy envelope line. According to the invention, collected music signals are processed in real time through audio analysis, and information such as rhythm, frequency and intensity of music is extracted. The audio analysis technology ensures that light can be highly synchronized with music, and the user experience is improved.

The invention relates to a simulation digital human real-time intelligent voice interaction system and a simulation digital human real-time intelligent voice interaction method based on vision and a large model, and aims to solve the problems of inaccurate target speaker recognition, high response delay and the like in digital human voice interaction in a complex scene. The system circles an effective recognition range through a camera, triggers audio collection in combination with face detection, locks a target speaker and reduces noise by using lip movement recognition and sound image fusion technologies, converts the target speaker into a text through voice wake-up, generates an answer by means of a large language model (LLM) and knowledge retrieval enhancement (RAG) technologies, generates low-delay voice through a voice synthesis technology accelerated by the vLLM, and performs voice recognition on the target speaker. And driving the preloaded digital human image to synthesize a video stream and pushing the video stream to a front end for rendering in real time. Accurate pickup, low-delay interaction and rapid digital human image switching in a complex environment are realized, the accuracy and real-time performance of intelligent voice question answering are improved, and the method is suitable for government affair halls, exhibition halls and other scenes.

The invention provides a single-microphone multi-array pickup method and system based on sound attenuation simulation, and the method comprises the steps: obtaining an original audio signal recorded by a single microphone and sound source direction information, and carrying out the sound attenuation simulation based on the sound source direction information and a preset virtual microphone array orientation parameter, calculating the included angle between the sound source direction and each virtual microphone; generating an audio intensity attenuation coefficient of each virtual microphone according to the included angle; and acting the audio intensity attenuation coefficient on the original audio signal to generate multi-channel audio data simulating the multi-microphone array. By adopting the method, the response difference of different microphone positions to the sound source can be simulated, the pickup result of the multi-microphone array is obtained, and high hardware cost and complex installation flow caused by actual deployment of a complex microphone array are avoided; and low-cost and high-diversity training data sources are provided for sound source localization, noise suppression and other models based on deep learning.

The embodiments of the present application are applied to the field of audio signal processing. Disclosed are a noise cancellation method, a noise cancellation apparatus and a noise cancellation system. The method in the embodiments of the present application is applied to a first device, which comprises a noise source. The method comprises: acquiring target noise cancellation parameters, which are obtained by performing calculation on the basis of at least one noise signal, wherein the at least one noise signal is noise from a noise source that is collected at a target location by at least one pickup device, and the target noise cancellation parameters are used for indicating ambient information of the target location; and on the basis of the target noise cancellation parameters, processing the noise from the noise source to generate an anti-noise signal, wherein the anti-noise signal and the noise from the noise source have the same frequency but opposite phases. The embodiments of the present application can improve the targeted active noise cancellation effect in open spaces.

The invention relates to the technical field of smart home, and provides a voiceprint recognition method based on a smart home scene and related equipment, and the method comprises the steps: obtaining a to-be-recognized voice segment; determining a pickup device for collecting the to-be-recognized voice segment; determining the position movement frequency of the pickup equipment, and identifying the equipment type of the pickup equipment according to the position movement frequency; and performing voiceprint recognition on the to-be-recognized voice segment by adopting a voiceprint recognition strategy corresponding to the equipment category of the pickup equipment. According to the invention, based on the movement characteristics of the pickup device, the voiceprint recognition strategy is automatically matched, differentiated voiceprint recognition is carried out on the voice segments, and the voiceprint recognition accuracy of the smart home scene is effectively improved.

The invention provides a box body and a signal transmission method. The box body comprises an upper shell, a lower shell, a first microphone, keys, a rotating shaft and an antenna assembly, the lower shell and the upper shell are installed in a matched mode to form a containing cavity, the containing cavity comprises a first surface, opposite to the upper shell, in the lower shell, the lower shell comprises an annular side face connected with the first surface, and the annular side face comprises a first side face; the second side face and the third side face are connected with the first side face, the second side face and the third side face are oppositely arranged, a first microphone pickup hole is formed in the first side face, the first microphone is arranged in the lower shell, communicated with the first microphone pickup hole and used for collecting a first audio signal outside the box body, and the key is arranged on the second side face and used for collecting a second audio signal outside the box body. The rotating shaft is arranged corresponding to the third side surface and is connected between the lower shell and the upper shell, the upper shell and the lower shell are rotatably connected through the rotating shaft, and the antenna assembly is arranged on the first surface and is used for at least transmitting at least part of the first audio signal to the mobile terminal.

This disclosure relates to methods, systems, and techniques for visualizing external audio captured by a microphone of the vehicle. Using the techniques described herein, external audio signals may be interpreted and converted into clear visual indicium that can be understood by individuals associated with the vehicle, such as passengers inside the vehicle or individuals awaiting pickup.

The invention provides electronic equipment and a pickup device, the electronic equipment is provided with an appearance surface, the appearance surface is provided with a first pickup port and a plurality of second pickup ports distributed at intervals, and each second pickup port and the first pickup port are arranged at intervals; a microphone, a first sound channel and a plurality of second sound channels are arranged in the electronic equipment, the first sound channel is communicated with the first pickup port and a first cavity of the microphone, the second sound channels are communicated with the second pickup ports in a one-to-one correspondence mode, and the second sound channels are communicated with a second cavity of the microphone. And the first cavity of the microphone and the second cavity of the microphone are respectively positioned on two sides of a vibrating diaphragm of the microphone. A plurality of pickup vectors can be formed between the plurality of second pickup ports and the first pickup port. A vector obtained by adding the plurality of pickup vectors is consistent with the sensitivity direction of the microphone. The microphone of the electronic equipment can better receive sound from the sensitivity direction of the microphone, and the microphone can realize directional pickup.

The invention discloses a directional pickup method for a microphone. The microphone comprises a left microphone core, a middle microphone core and a right microphone core. Wherein the middle microphone core is located on the central axis of the microphone, and the left microphone core and the right microphone core are symmetrically distributed on the left side and the right side of the middle microphone core; sound of a sound source is synchronously collected through a left microphone core, a middle microphone core and a right microphone core of the microphone, and three paths of sound signals are obtained; respectively filtering the three paths of sound signals to obtain left, middle and right microphone core signals; calculating an included angle between a connecting line of a sound source and the middle microphone core and the central axis of the microphone based on the time difference and the distance difference of the sound reaching the left, middle and right microphone cores to obtain a sound source direction; based on the left microphone core signal, the middle microphone core signal, the right microphone core signal and the sound source direction, through time delay compensation and gain adjustment, the sound signal in the target direction is enhanced, the sound signal in the non-target direction is suppressed, the final output signal is obtained, and the directionality and the sound quality of pickup are improved.

The utility model provides a pickup system with a self-checking function. The sound pick-up system comprises a sound pick-up and a self-checking device configured for the sound pick-up, and the self-checking device comprises a waveform generator used for generating sine waves; the sound production device is used for producing sound under the driving of the sine wave, so that a sound pickup picks up an audio signal, and related information of the audio signal is sent to a far end, so that the far end checks whether the picked-up audio signal is normal or not; and the waveform generator power supply control circuit is used for controlling the on-off of the waveform generator power supply. Therefore, the fault of the sound pickup can be automatically, timely and accurately detected, and the accuracy and reliability of sound collection are ensured.

A wearable sound pickup device includes a wearable sound pickup device body, where the wearable sound pickup device body includes an extension component. A sound pickup unit array is installed inside the wearable sound pickup device body and is configured for long-distance sound pickup. The sound pickup unit array includes at least two sound pickup units, and an arrangement direction of each sound pickup unit is consistent with an extension direction of the extension component.

The utility model relates to the technical field of telecommunication, and particularly discloses an anti-howling microphone, which comprises a capacitance microphone core, a sound pickup hole is arranged on the top surface of the capacitance microphone core, and an air hole is arranged on the bottom surface of the capacitance microphone core; the microphone core bracket comprises a microphone core fixing part used for installing the capacitance microphone core and a sound wave diffusion part located under the capacitance microphone core; wherein the sound wave diffusion part is of a conical structure which is gradually narrowed towards the capacitance microphone core. The anti-howling microphone provided by the utility model can effectively solve the problem that howling is easily caused when a capacitive microphone core is used for pickup.

The invention belongs to the technical field of pickup, and discloses a dynamic pickup method and system based on a microphone array. The method comprises the following steps: acquiring set pickup area information, wherein the pickup area information comprises a pickup area in a conference environment; the method comprises the following steps: receiving at least one original pickup signal of a conference environment, and obtaining audio characteristics of each original pickup signal; based on the audio characteristics of the original pickup signals, determining a target pickup signal with an effective voice signal from the original pickup signals so as to determine a target sound source direction to be picked up; when the target sound source direction is in a pickup area, at least one alternative wave beam is controlled to point to the target sound source direction to pick up the target pickup signal, and the alternative wave beam refers to a pickup wave beam of a microphone array switched to a standby state when no effective voice signal is detected within a preset duration. The state of the pickup wave beam can be dynamically switched, and the sound source tracking accuracy and the pickup sound quality can be improved.

The invention discloses a dehumidifier and a control method and device thereof, a storage medium and a computer program product. The method comprises the steps that at least one set of setting parameters are stored in advance; each set of set parameters in the at least one set of set parameters comprises set noise, corresponding to the set frequency of a compressor and the set rotating speed of a fan, of the dehumidifier and set reverse-phase sound waves, corresponding to the set noise of the dehumidifier, of the dehumidifier; under the condition that the dehumidifier operates after being started, according to at least one of the current frequency of the compressor, the current rotating speed of the fan, the current pickup sound of the microphone and the at least one set of set parameters, after the noise value in the current pickup sound of the microphone is lower than a preset noise value, the voice recognition function of the dehumidifier is controlled to operate. According to the scheme, the noise of the dehumidifier is reduced by recognizing the environment noise and the noise of the dehumidifier and playing the anti-phase sound waves, and the working parameters capable of enabling the dehumidifier to maintain the lowest noise are found, so that the voice recognition rate is increased, and the user experience is improved.

The invention discloses a microphone device and method capable of locating a sound source and a digital human device.The device comprises a plurality of basic units and a signal processing module, and each basic unit is composed of pickup microphones annularly and evenly arranged at intervals and an infrared laser distance measuring sensor located in the center; the plurality of basic units are arranged to form a ranging microphone array matrix. The signal processing module synchronously collects a sound signal of the pickup microphone and a distance signal of the infrared laser distance measuring sensor, and firstly processes the sound signal based on a TDOA algorithm to obtain a two-dimensional plane coordinate of a sound source; and positioning a corresponding basic unit according to the two-dimensional plane coordinate, calling a distance signal of an infrared laser distance measuring sensor in the unit, calculating a vertical distance between the sound source and the array matrix in combination with the two-dimensional coordinate, and finally determining a three-dimensional coordinate of the sound source. According to the invention, high-precision sound source three-dimensional positioning is realized.

The technology employs a holistic approach to passenger pickups and other wayfinding situations. This includes identifying where passengers are relative to the vehicle and / or the pickup location. One aspect leverages camera imagery from a rider's client device when providing rider support. This enables an agent to receive the imagery to help guide the rider to the vehicle. Another aspect provides audio information to the rider to help them get to the vehicle. This can include selecting or curating various tones or melodies, giving the rider advanced notice of sounds to listen for, and modifying sounds as the rider approaches the vehicle or to address ambient noise in the environment. Different wayfinding tools may be selected for presentation to the rider based on their proximity state to a pickup location or to the vehicle. Gracefully transitioning between different tools can enhance their usefulness and elevate the rider's experience for a trip.

The invention provides a power plant equipment sound abnormity identification and health prediction method based on granular computing and an LSTM network, and the method comprises the steps: employing an array pickup and a high-dynamic-range microphone array in a complex and high-noise background environment of a power plant, and combining an anti-interference filtering and beam forming algorithm, thereby achieving the sound abnormity identification and health prediction of the power plant equipment. According to the method, directional, multi-channel and non-contact sound acquisition is carried out on key parts of equipment, acquired sound signals are preprocessed, converted into time domain, frequency domain and time-frequency domain representations and coded into multi-dimensional numerical vectors, and compared with a rule-based expert system, the method has higher generalization ability and real-time response ability; a large language model is introduced, so that the output is closer to an engineering language and is suitable for operation and maintenance personnel to understand and execute; a self-defined knowledge base or safety semantic filtering is supported, and closed-loop deployment in an industrial field is facilitated; the system can be in butt joint with an intelligent inspection system, and full-link linkage of voice recognition, health assessment and suggestion generation is achieved.

The invention discloses a dry pump operation state audio monitoring system, and relates to the technical field of dry pump operation state monitoring. The invention aims to solve the problem of monitoring the running state of a dry pump which is inconvenient to additionally install a contact sensor. The device specifically comprises a characteristic sound spectrum database used for storing audio characteristics in the operation state of the dry pump, a sound collection unit used for collecting vibration sound signals in the operation state of the dry pump, an alarm module used for sending out an equipment fault alarm and a central processing unit, and the sound collection unit is in communication connection with the characteristic sound spectrum database. The sound acquisition unit, the alarm module and the characteristic sound spectrum database are respectively in communication connection with the central processing unit; a pickup element of the sound acquisition unit is one of a contact sensor and a microphone or a mixture of the contact sensor and the microphone. According to the invention, the appropriate sound acquisition unit is adopted for anti-interference monitoring, so that the operation and maintenance personnel can judge the operation state of the equipment more visually, the working efficiency of the operation and maintenance personnel is improved, and the fault diagnosis time is shortened.

The voice directional enhancement method comprises the steps that a current frame and a staring direction of mixed voice are acquired, and the mixed voice is acquired by a microphone array; constructing a direction feature based on the current frame of the mixed voice and the gazing direction, and constructing an audio feature based on the current frame of the mixed voice; determining a current frame mask according to the direction features and the audio features; and determining a target enhanced voice of the current frame according to the current frame mask, the current frame of the mixed voice and the plurality of historical frames of the mixed voice. In the embodiment of the invention, the mask of the current frame is generated under the guidance of the gazing direction, so that the target voice can be accurately separated from the mixed voice. Besides, based on the current frame mask and the current frame of the mixed voice, the target voice enhancement is performed frame by frame from the overall characteristics of the mixed voice by referring to a plurality of historical frames, so that the accuracy of the target enhanced voice of the current frame can be effectively improved, and thus the accuracy of directional pickup is improved.

A pickup, a stringed instrument, and a pickup control method are disclosed. The pickup comprises: pickup components (110), multi-touch screens (130) and processing components (140). The multi-touch screens (130) and the processing components (140) are connected. The pickup components (110) and the processing components (140) are connected. The pickup components (110) is configured to pick up sound information emitted by a musical instrument. The multi-touch screen (130) is configured to display function application information that is selected from a group consisting of timbre adjustment function application information, tone adjustment function application information, and equalizer adjustment function application information. The multi-touch screen is configured to receive a function triggering instruction. The multi-touch screen displays information as to functional applications available through the pickup capabilities. Control functions available to a user are expanded beyond control by a single knob as in the prior art, the playing experience of a user on the instrument is enhanced.

The invention relates to a speech recognition method for a tactical AR helmet, and the method comprises the steps: collecting a first audio signal through a near-field pickup disposed near the mouth and nose of a wearer, and collecting a second audio signal through an environment pickup disposed at the outer side of the helmet or away from the mouth and nose; synchronously acquiring a third signal sounding with the wearer, wherein the third signal is at least one of helmet inner side contact type micro vibration, bone conduction vibration pickup or helmet inner sound pressure change; calculating a near-field difference indicator based on the first audio signal and the second audio signal, the near-field difference indicator comprising at least one of an energy difference or an energy ratio; determining sound production starting and ending moments from the first audio signal and the third signal, and calculating a time sequence synchronization index; and when the near-field difference index satisfies a preset near-field condition and the time sequence synchronization index satisfies a preset synchronization condition, determining that the voice is sent by the helmet wearer and a corresponding voice segment is output for subsequent identification, otherwise, shielding the voice segment.

The invention discloses a robot head device which comprises a neck fixing frame, a motor support, a mounting frame, a lower head shell, an upper head shell and a sound pick-up, the motor support is fixed relative to the neck fixing frame, and a rotating flange face of a motor protrudes out of the upper side of the motor support; the mounting frame is connected to the rotating flange face and connected with a camera, a screen, loudspeakers, an audio control panel and a voltage reduction module, the camera and the screen are located in front of the motor support, the loudspeakers are located on the left side and the right side of the motor support, the audio control panel is located above the motor support, and the voltage reduction module is located behind the audio control panel; the lower head shell is connected to the mounting frame, and an opening is formed in the lower part of the lower head shell; the upper head shell is detachably connected to the lower head shell, the front side of the upper head shell is provided with a first installation opening used for installing a camera and a second installation opening used for installing a screen, the upper head shell and the lower head shell define a containing cavity, and the problems that an existing robot is insufficient in head space utilization and inconvenient to maintain can be solved.

The utility model relates to the technical field of pickup equipment, in particular to an intelligent acoustic positioning imager. The sound source positioning device comprises a lower shell, a sealing rubber mat and a microphone main board, the lower shell, the sealing rubber mat and the microphone main board are sequentially connected from outside to inside to form a sound source positioning assembly, and a plurality of digital silicon microphone structures are distributed on the sound source positioning assembly in an array mode. Each digital silicon microphone structure comprises a lower shell pickup port arranged on the lower shell, a rubber mat opening arranged on the sealing rubber mat and a mainboard pickup port arranged on the microphone mainboard, the microphone, the lower shell pickup port, the rubber mat opening and the mainboard pickup port are sequentially butted and communicated, the microphone is positioned on the microphone mainboard and is aligned with the mainboard pickup port, and the lower shell pickup port is positioned on the lower shell and is aligned with the mainboard pickup port. The digital silicon microphone structures are mutually independent. Multiple paths of high-sensitivity digital silicon microphones are arranged in an array mode to form a recording system for cooperative work, and the functions of sound source positioning, noise suppression and the like are achieved. And a stepped pickup channel is adopted, so that the direction and the distance to a sound source can be accurately positioned.

Systems and methods are provided for automatically adjusting lobe gain by receiving, from at least one microphone, sound location information for an audio source detected by the at least one microphone; calculating a distance between the audio source and the at least one microphone based on the sound location information, a first height associated with the at least one microphone, and a second height associated with the audio source; determining, based on the distance, a gain value for an audio pickup lobe directed towards the audio source by the at least one microphone; and applying the gain value to an audio channel of the at least one microphone, the audio channel configured to receive audio signals captured using the audio pickup lobe.

The invention discloses an intelligent pickup and speech recognition system based on multi-modal fusion, and relates to the technical field of artificial intelligence and speech recognition crossing. The system comprises a main control module, a plurality of syllable points and a multi-modal fusion engine, wherein the multi-modal fusion engine comprises four core components, namely a sound source localization and separation component, an environment adaptive noise reduction component, a cross-modal feature fusion component and a dynamic context understanding component. Multi-modal data are acquired through an array microphone and an auxiliary sensor group, and the system realizes sound source localization and separation, dynamic environment noise suppression, multi-modal feature deep fusion and context semantic correction. According to the method, the robustness, the accuracy and the intelligent interaction capability of voice recognition are effectively improved, the voice interaction experience is improved in complex scenes such as a noise environment and accent change, and a more reliable voice processing solution is provided for intelligent voice interaction equipment.

The invention belongs to the technical field of detection, and particularly relates to a spinneret plate detection device and a detection method thereof.The spinneret plate detection device comprises an installation disc, a detection module, a first detection module and a second detection module, the installation disc is provided with an installation groove used for containing a spinneret plate, and the groove bottom of the installation groove is provided with an overflowing hole; the first driver is arranged on the mounting disc and is used for driving the spinneret plate in the mounting groove to rotate, so that a to-be-detected spinneret through hole in the spinneret plate is overlapped with the overflowing hole; the air nozzle is positioned above the mounting disc, is connected with the second driver, and is used for covering the to-be-detected spinneret through hole and blowing air; the detection mechanism is positioned below the mounting disc; according to the spinneret plate detection device and the detection method thereof, the sound pickup is arranged to acquire the audio data at the air hole, and the acquired audio data is compared with the audio data sent by the standard spinneret through hole at the air hole, so that the existence of impurities in the to-be-detected spinneret through hole is judged when the two audio data are different; meanwhile, impurities are cleaned when the impurities are detected through the outer sleeve and the inner pipe.

The invention relates to the technical field of directional pickup, and discloses a directional pickup method and device, computer equipment and a medium, and the method comprises the steps: obtaining time domain voice signals collected by at least two microphones, and carrying out the time-frequency conversion; calculating an observation phase difference of each frequency point, and determining a target theoretical phase difference according to a target pickup direction specified by a user and the geometric parameters of the microphone; encoding the target theoretical phase difference into a query vector, encoding the observation phase difference into a key vector, and generating a space confidence mask through a geometric cross attention mechanism; and filtering the frequency domain signal by using the mask, and outputting enhanced voice in a target direction through inverse time-frequency transformation. According to the invention, through an attention mechanism guided by physical prior, the problem of phase winding is effectively solved, and high-robustness and high-fidelity speech pickup in any direction is realized in a strong reverberation and low signal-to-noise ratio environment.

The invention relates to the technical field of audio signal processing, and discloses a microphone far-field pickup amplitude dynamic range control method, and the method comprises the steps: processing synchronous bone conduction and air conduction signals; respectively extracting feature vectors representing near-field interference and far-field human voice from the bone conduction signal and the air conduction signal; inputting the feature vector into an artificial intelligence decision engine, and carrying out collaborative analysis to generate a gain strategy considering interference suppression and human voice maintenance; and the gain of the air conduction signal is adjusted according to the strategy. The system comprises a signal preprocessing module, a first feature extraction module, a second feature extraction module, an artificial intelligence decision engine and a gain adjustment and reconstruction module. Unambiguous observation is carried out on near-field interference by using the bone conduction signal, the gain control problem when far-field weak human voice and near-field strong interference coexist is solved, and the pickup quality can be ensured while signal clipping can be effectively prevented.

The invention relates to the technical field of microphone pickup, and discloses an intelligent pickup method and system of a microphone, and the method comprises the steps: respectively picking up an audio signal through a plurality of microphone units in a microphone array, carrying out the analysis of the audio signal to obtain a human voice signal and a background signal, and carrying out the collection of a voice signal based on the human voice signal and the background signal picked by each microphone unit. According to the technical scheme, the orientation angle of each microphone unit is adjusted, so that the microphone array is adjusted to the optimal pickup state, key frequency band recognition is performed on the human voice signal collected in the optimal pickup state, and voice enhancement processing is performed on the human voice signal according to the recognition result, so that the problem of poor pickup tracking capability of a microphone in the prior art is solved.