144 results about "Silence" patented technology

The invention provides a voice enhancement and high-precision recognition method and system in a complex environment, and relates to the technical field of voice processing, and the method comprises the steps: collecting a time domain signal in an off-road parking sentry box environment for preprocessing, detecting a mute segment signal in a standard time domain signal for noise power spectral density estimation, and obtaining a noise power spectral density value; a reverberation parameter is obtained by combining voice onset information and noise spatial correlation estimation, prediction is performed by using a deep neural network model, voice masking is applied to microphone array signals to perform enhancement processing, adaptive feature extraction is performed on time domain enhanced voice signals, and a voice signal is obtained. And performing high-precision recognition on the voice adaptive feature sequence based on an acoustic model and a language model, and outputting a target recognition text. The technical problems of poor voice signal quality and low recognition accuracy in a complex noise environment in the prior art are solved. The technical effects of improving the voice signal quality and the recognition accuracy and realizing clear, accurate and real-time voice interaction are achieved.

The invention relates to the technical field of elderly care, solves the problems that in the prior art, due to the fact that noisy and intermittent elderly voice is lack of robust sentence bound recognition and interruption repair, misinformation and missing information of calling-for-help triggering are prone to occurring, and time delay is high, and provides an audio processing method and device for elderly calling-for-help, equipment and a medium. The method comprises the following steps: preprocessing a voice signal of the elderly to obtain a preprocessed voice signal; performing statement boundary recognition and labeling on the preprocessed voice signal to obtain a voice fragment set; carrying out interruption segment and silent interval detection on the voice segment set to obtain an initial voice segment sequence, and carrying out intelligent recombination and time sequence repair on the initial voice segment sequence to obtain a target voice segment sequence; and carrying out emergency call recognition on the target voice segment sequence to obtain an emergency call judgment result. According to the invention, the accuracy of distress call identification of the elderly is improved.

The invention relates to the technical field of voice interaction, in particular to a dialogue type AI low-delay response control method and system based on semantic analysis. The method comprises the following steps: firstly, generating a driving intensity correlation factor of a current time window according to the change of CAN bus data; further monitoring the output of the NLU, analyzing the acoustic intonation raising characteristics, and obtaining an interaction suppression coefficient in combination with the driving intensity correlation factor; further acquiring a time pressure coefficient based on the mute duration after the user stops sounding; further comparing the interaction suppression coefficient and the time pressure coefficient of the current time window to obtain a response judgment value; further analyzing the change trend of the interaction inhibition coefficient, and updating an environment improvement flag bit; and finally, according to the driving intensity correlation factor, the environment improvement flag bit and the response decision value of the current time window, carrying out multi-level response decision control, and carrying out selective reset of an interaction state, thereby solving the problems of wrong truncation and response delay of voice interaction under dynamic driving.

The present application relates to the technical field of speech processing, in particular to a sales speech dialogue speaker segmentation and marking method based on role separation, comprising the following steps: generating a time-stamped speech segment sequence, synchronously extracting a voiceprint feature vector of each speech segment, starting a silence intention analysis engine, predicting the attribution role of the current silence according to the semantic content of the speech segments before and after the silence, generating a virtual voiceprint feature vector and inserting it into the speech segment sequence; performing role clustering on the voiceprint feature vector, splitting to generate a new speaker cluster; when detecting that there is speech overlap or audio loss in the speech segment, triggering mixed speech separation and generation compensation. The present application improves the abnormal recognition capability in the scene of sales role transformation and sound line disguise, reduces the risk of sales and customer speech confusion, and ensures the coherent and accurate role marking of each speech segment.

This invention discloses an adjustable frequency response microphone integrating XLR and USB outputs, comprising a microphone body, a dynamic microphone driver, a passive filter, a pass-through / preamp circuit, an XLR XLR interface, a USB module, and a function control circuit. The passive filter employs an LC circuit structure, supporting three frequency modes: low-frequency cutoff, mid-frequency boost, and flat response, effectively filtering out ambient noise and enhancing vocal quality. The pass-through / preamp circuit offers both pass-through and preamp modes, with adjustable preamp gain to accommodate dynamic microphone drivers of varying sensitivities, and can be externally or internally powered by phantom power. The USB module integrates a Type-C interface and an audio chip, supporting analog-to-digital / digital-to-analog conversion, real-time headphone monitoring, and external power supply. The function control circuit integrates an encoder and a touch panel, enabling adjustments such as volume, mixing, and mute, and enhances the interactive experience with an RGB lighting module. This invention combines the advantages of professional XLR output with convenient USB output, making it suitable for professional recording, live streaming, gaming, and other scenarios.

This invention relates to the fields of audio signal processing and artificial intelligence, specifically to a lyrics alignment method and electronic device based on automatic speech recognition. The lyrics alignment method based on automatic speech recognition in this application uses a pre-trained automatic speech recognition model to recognize audio samples and generate recognized text. The edit distance sequence between the recognized text and the lyrics text is calculated, and spoken segments are determined by analyzing the distance change pattern through a sliding window. These spoken segments are then replaced with silent segments, effectively eliminating spoken segments and avoiding interference from non-lyric content in the alignment process, thus laying the foundation for subsequent alignment.

Embodiments of the present application provide a soft mute control method and device for a radio device, a medium and a product. By extracting the spectral deviation feature and the time domain energy feature of the digital audio signal of the radio device, the spectral deviation feature can reflect the frequency spectrum similarity between the signal and the noise, and the time domain energy feature can reflect the signal strength. Combined with multi-state decision and energy decision, multi-dimensional decision of the signal state is performed to realize accurate identification of the signal state, and then the output gain is dynamically adjusted. Compared with the prior art, not only the signal receiving strength index is relied on for adjustment, but also the problem of large and small sound caused by signal strength fluctuation due to position change is avoided. The spectral deviation feature is used to effectively distinguish between effective signals and noise, improve the accuracy of soft mute decision, and the time domain energy feature is used as auxiliary decision to enhance the reliability of the system in a low energy scene, so that the volume is smooth and continuous, and the user's radio experience is improved as a whole.

1. Name of the designed product: Double-person silent cabin. 2. Use of the designed product: To provide a space for isolating noise. 3. Design points of the designed product: In shape. 4. Picture or photo best indicating the design points: Perspective view. 5. Other circumstances or explanations: Mark A, B, C parts as transparent glass material.

The utility model relates to silence room material receiving equipment suitable for sound production device processing, which comprises a mounting base, a guide driving device mounted at the rear end of the mounting base, a guide mechanism mounted on the mounting base, a carrier plate device movably connected to the guide mechanism, and a material receiving device connected to the carrier plate device. A main induction device is installed on the installation base, and an auxiliary induction device is installed on the front side of the installation base. Therefore, through cooperation of the guide driving device and the plate carrying device, efficient switching of the sound production device between the feeding station and the material taking station is achieved, the coordinated butt joint requirement for taking materials from a mute room and feeding the materials into a follow-up assembly station is effectively met, and the production efficiency is improved. The carrier plate device adopts the design of limiting grooves and limiting columns and is matched with the appearance of the bottom of the sound production device, accurate placement and positioning are achieved, the limiting columns are inserted into mounting guide holes in the bottom of the sound production device for auxiliary locking, and improper displacement of the sound production device in the transportation process is effectively prevented.

The present disclosure relates to a speech determination model training method, an object speech extraction method, a device, an electronic device and a storage medium. The method comprises: obtaining a second speech set, the second speech set comprising a plurality of second sample speeches and a second object identifier corresponding to each second sample speech, each second sample speech comprising a fifth speech and a sixth speech, the fifth speech comprising at least speech corresponding to a fifth speech non-associated object identifier, and the sixth speech being silent audio; performing speech extraction on the fifth speech in each second sample speech based on a speech extraction model to obtain a seventh speech corresponding to each second sample speech; and training a candidate speech determination model based on the seventh speech and the sixth speech to obtain a target speech determination model. The present application increases the training samples by adding the speech corresponding to the fifth speech non-associated object identifier and the silence, trains the candidate speech determination model, obtains a model suitable for richer scenarios, and reduces the error rate of audio recognition.

The utility model discloses a lock body convenient to assemble and capable of preventing abnormal sound, which comprises a mounting plate, a first mounting component is arranged at one end of the mounting plate, a second mounting component is arranged on the first mounting component, and by arranging the first mounting component and the second mounting component, a plastic block is adopted to replace a traditional metal material to be used for mounting. Compared with the prior art, the assembly process is greatly simplified, the production cost is reduced, material waste is reduced, the spring and limiting column structures are additionally arranged in the lock body, connection among all components is more stable, vibration generated in the operation process is effectively absorbed, mechanical noise is greatly reduced, and the service life of the lock is prolonged. In addition, by adopting buffer rubber and limiting holes, the stability and the mute effect of the device are further enhanced, meanwhile, the error-tolerant rate during assembly is increased, and installation is more convenient and faster.

The invention discloses a display chip and a POP sound processing method, and the method comprises the steps: receiving an input digital signal through a data receiving unit in the display chip, and obtaining audio input data; the data detection unit performs anomaly detection on multiple pieces of feature information of the audio input data based on a preset detection rule to obtain a first indication signal; wherein the preset detection rule comprises a plurality of detection rules corresponding to the plurality of pieces of feature information, and the plurality of pieces of feature information are in one-to-one correspondence with the plurality of detection rules; the multiple pieces of feature information comprise pin level data, image data, HDMI data, DP data and clock signals; the first indication signal is used for indicating that feature information which does not meet a preset detection rule appears in the multiple pieces of feature information; and when the data processing unit receives the first indication signal, the audio input data is subjected to mute processing, and the audio input data subjected to the mute processing is output, so that the POP sound generated in the audio channel is eliminated, and the display effect is optimized.

The utility model discloses a lift door device of mute room relates to acoustics equipment technical field. Include: door frame, lifting mechanism, door body, counterweight mechanism. The door frame includes a pair of opposite vertical installation beam, lifting mechanism is connected on a pair of installation beam, the door body is slidably arranged between a pair of installation beam, and the door body is connected lifting mechanism, and the counterweight mechanism includes at least a pair, and a pair of counterweight mechanism is arranged respectively on the outside of a pair of installation beam, and a pair of counterweight mechanism is connected respectively two sides of door body, to control the door body and keep balance in the process of lifting. The utility model discloses the lift door device of mute room can realize the stable operation of door body in the process of lifting.

This allows users to play other parts during silent sections in a song where their own part is silent. [Solution] The CPU of the information processing device searches for silent sections in a song consisting of multiple parts where the melody part played by the user is silent, and determines the most suitable part among the parts other than the melody part that is presumed to be most suitable for the user to play during the silent section. Then, when the silent section is reached during the progress of the song, the CPU transitions the part played by the user from the melody part to the suitable part, and when the silent section ends, it transitions the part played by the user back to the melody part.

The invention discloses an intelligent audio mute segment detection and skip playing method, system and device and a medium, a mute segment detection algorithm adopted by the method can dynamically adjust a threshold according to audio content characteristics, high-precision mute segment identification is realized in combination with an overlapped sliding window, the false detection rate and the omission rate are reduced, and the detection efficiency is improved. The robustness and universality of the algorithm are improved; according to the method, a lightweight real-time detection algorithm is adopted, efficient operation can be achieved in the JavaScript execution environment of the Web browser, good real-time performance and user experience are guaranteed, and meanwhile the CPU occupancy rate and memory consumption are controlled; the method supports streaming processing, mute detection and skip control can be carried out in real time in the audio playing process, dynamic interaction operation of a user is supported, and the whole audio file does not need to be pre-processed; besides, according to the method, an intelligent caching strategy and a resource management mechanism are adopted, so that repeated calculation is avoided, the overall operation efficiency of the system is improved, and occupation of system resources is reduced.

The application relates to the field of diesel generating set equipment, in particular to a super-silence diesel generating set which comprises a machine box, air inlets and air outlets are arranged at two ends of the machine box respectively, sound-absorbing assemblies are additionally arranged at the air inlets and the air outlets, the sound-absorbing assemblies are composed of multiple sound-absorbing plates, the multiple sound-absorbing plates are staggered and spaced in a horizontal direction, sound-absorbing channels are formed between two adjacent sound-absorbing plates, the sound-absorbing plates comprise first plate bodies and second plate bodies, the first plate bodies are used for guiding air flow between the inside and the outside of the machine box, and the second plate bodies are used for prolonging noise propagation paths. The application aims to optimize the air inlet pipeline of the generating set without increasing the overall volume of the generating set, to achieve noise reduction treatment, and to consider daily inspection, maintenance or emergency treatment of the generating set.

The invention discloses a mute detection method and device, audio equipment and a computer readable storage medium, and the method is applied to the audio equipment, and comprises the steps: carrying out the framing processing of an audio input signal, and calculating the energy characteristic quantity of each analysis frame in a preset time window; intercepting a plurality of time periods in a preset time window, and determining a background noise estimation value corresponding to each time period based on the energy characteristic quantity of the plurality of analysis frames in each time period; obtaining statistical characteristic parameters according to the plurality of background noise estimated values; dynamically generating a mute detection threshold according to the background noise estimation value and the statistical characteristic parameters; and in a preset time condition, comparing the energy characteristic quantity corresponding to the real-time analysis frame with a mute detection threshold value, and judging whether the audio channel to be detected is in a mute state or not. Therefore, the silence detection threshold adaptively reflects the stability or fluctuation of the noise level, and the accuracy and anti-jitter capability of silence detection in the non-stationary noise environment are effectively improved in combination with the preset time condition.

The silencing device for the generator set container comprises a frame and a silencing piece assembly installed on the frame, and the silencing piece assembly comprises a plurality of fixed silencing grid plates and at least one movable silencing grid plate. And the movable silencing grid plate is detachably mounted on the frame. When the silencing device is applied, a maintainer can obtain a passage only by disassembling the movable silencing grid plate, and the maintainer enters and exits through the passage to complete corresponding maintenance work. The movable silencing grid plate is simple and convenient to disassemble and assemble, the working efficiency of maintenance personnel can be improved, the maintenance time is saved, and the physical output of the maintenance personnel is reduced. The utility model also provides a box-type generator set device.

The utility model discloses a household music practice room which comprises a music practice room structure, an auxiliary sound insulation structure is detachably installed on the music practice room structure, a musical instrument assembly is placed in the music practice room structure, the music practice room structure comprises a household music room, and a sound insulation assembly is fixed on the household music room. The sound insulation assembly comprises a sound insulation felt layer, a sound absorption cotton layer is detachably installed in the sound insulation felt layer, a sound insulation plate layer is detachably installed in the sound absorption cotton layer, a sound absorption plate layer is detachably installed in the sound insulation plate layer, a sound absorption carpet is placed on the lower side of the interior of the sound absorption plate layer, and an illuminating lamp and a mute ventilating fan are detachably installed on a sound absorption ceiling plate. According to the household music practice room, the sound absorption effect is guaranteed through the arrangement of the sound insulation felt layer, the sound absorption cotton layer, the sound insulation plate layer, the sound absorption plate layer and the sound absorption carpet, internal stuffiness is avoided through the arrangement of the mute ventilator, and the ventilation effect is achieved.

According to the multi-mode emotion regulation subtitle generation method fused with cognitive intervention, classroom teaching audio is collected through an omnidirectional condenser microphone, multi-dimensional acoustic features are extracted after preprocessing such as mute segment removal, noise suppression and signal normalization, the audio is converted into a text through a DNN-ASR model and a pre-training BERT model, and a semantic embedding vector is generated. An improved CNN-LSTM model is adopted to fuse acoustic and semantic features for emotion recognition, and a primary cognitive tag is generated in combination with a classroom situation knowledge base. Based on an emotion cognition theory, secondary modeling is carried out on emotion signals through an explanatory subtitle generation sub-model and a background annotation adjustment sub-model, an emotion adjustment factor matrix is formed, and weights are dynamically controlled. And finally, according to the adjustment factor, generating a multi-mode subtitle, matching a color code, a dynamic icon and a font style, and presenting the information on a teaching terminal. According to the invention, classroom teaching information processing efficiency can be effectively improved, and deaf students and other groups can be assisted to better understand teaching contents.

The invention provides a voice recognition method and device, electronic equipment, a computer readable storage medium and a computer program product. The method comprises the following steps: carrying out integrity detection on received voice data until complete voice data is detected, and determining whether the mute time length reaches a mute time length threshold value or not; under the condition that the mute duration reaches a mute duration threshold value, determining first voice data; and performing voice recognition on the first voice data to obtain a recognition result corresponding to the first voice data. According to the invention, the accuracy of speech recognition can be improved.

Various embodiments disclose techniques for generating one or more sound zones and a quiet zone in a listening environment. The techniques include configuring one or more filters of a filter bank, generating a respective speaker signal for each of one or more speakers by processing a respective audio zone signal for each of one or more sound zones using the filter bank, and outputting the respective speaker signals using the one or more speakers. Sound output by the one or more speakers propagates through a listening environment to generate respective sound in each of the one or more sound zones corresponding to the respective audio zone signal for that sound zone. The sound output by the one or more speakers propagates through the listening environment to generate a first quiet zone.

The application discloses a method and device for reducing earphone crosstalk, a computer readable storage medium and a terminal. The method comprises the following steps: in response to an earphone crosstalk calibration operation, a test sound source is played, wherein the signal input to a first sound channel is an audio signal, and the signal input to a second sound channel is a mute signal; within a set earphone crosstalk gain interval range, the earphone crosstalk gain is adjusted according to a set adjustment step; the original signal output by the second sound channel is compensated by using the earphone crosstalk gain to obtain a compensated signal of the second sound channel; the sound size of the compensated signal output by the second earphone is collected, and the sound size of the second sound channel is collected when only the second earphone is worn; the minimum value of the sound output by the second earphone is determined, and the optimal crosstalk gain of the inserted earphone is determined based on the earphone crosstalk gain corresponding to the minimum value. The above scheme can reduce earphone crosstalk and improve the stereo sound effect of the earphone.

The invention discloses a directional pickup method and device for a ceiling microphone cascade scene, and the method comprises the steps: collecting a space image based on a camera disposed in a target pickup space, and detecting the image coordinates of a speaker based on the space image; according to a pre-constructed coordinate mapping model, converting the detected image coordinates of the speaker into corresponding sound source DOA coordinates; a target ceiling microphone closest to the speaker is selected from a ceiling microphone cluster deployed in a cascade mode based on the sound source DOA coordinates, and muting processing is conducted on the rest ceiling microphones; and controlling the target ceiling microphone to execute a beam forming operation according to the sound source DOA coordinate, so that a beam main lobe of the target ceiling microphone points to a sound source direction corresponding to the sound source DOA coordinate. Therefore, in complex scenes such as multiple sounding positions, multi-person alternation and speaking position change, the accuracy of the pickup direction and the clearness and stability of the voice signal can be continuously kept, and adaptive focusing and high-fidelity output in the pickup process are realized.

The present application provides a signal processing method and a related apparatus. The method comprises: acquiring N quantized audio features of raw non-speech audio, wherein N is a positive integer greater than or equal to 1; determining a first predicted silence audio feature based on the N quantized audio features and reference noise; and determining, based on the first predicted silence audio feature and a first neural network, first non-speech audio corresponding to the raw non-speech audio.

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 application relates to the technical field of box-type substations, and discloses a mute room structure for a box-type substation, which solves the problems that the box-type substation on the market produces humming noise in the operation process, the noise has a great influence on the surrounding environment and people, and the mute room is complicated to disassemble, and the mute room structure comprises a sound insulation board, a buffer base is fixedly connected to the bottom of the sound insulation board, a top sound insulation baffle is fixedly connected to the top of the sound insulation board, a ventilation opening is fixedly connected to one side of the sound insulation board, a dismounting mechanism is arranged on one side of the sound insulation board, the dismounting mechanism comprises a protective plate, a connecting frame and a bottom supporting plate, the arrangement of the sound insulation mechanism can reduce the noise generated by the ventilation opening, reduces the influence on the surrounding environment, the arrangement of the dismounting mechanism can conveniently dismount the sound insulation mechanism, the portability of the device is improved, and the work load of workers is reduced.

The utility model provides a silence room with an active noise reduction function. The silence room comprises a silence room body and an active noise reduction assembly. The active noise reduction assembly is arranged in the mute room and used for collecting environment sound waves outside the mute room, determining target sound waves according to the environment sound waves and making sound according to the target sound waves, and the target sound waves are sound waves with the sound wave intensity smaller than the preset intensity after being overlapped with the environment sound waves; the controller controls the loudspeaker to emit the target sound wave, that is, after the loudspeaker emits the target sound wave, the target sound wave can be neutralized and counteracted with the sound wave in the environment outside the silence chamber, so that the intensity of the interior of the silence chamber is reduced to be below the preset intensity after noise reduction is carried out on the interior of the silence chamber through the active noise reduction assembly provided by the embodiment of the invention. According to the mute room with the active noise reduction function, sound waves sensed by a user in the mute room in the area where the ears are located are low-strength sound waves, and compared with an existing mute room adopting a sound insulation plate for passive sound insulation, the mute room with the active noise reduction function is better in mute effect.

A method is performed by an endpoint device that includes a microphone and a loudspeaker. The method comprises: muting the loudspeaker; participating in a conference session with a neighbor endpoint device that shares a space with the endpoint device; detecting audio in the space using the microphone to produce detected audio; determining whether audio distortion, originating at a neighbor loudspeaker of the neighbor endpoint device, is present or absent in the detected audio; and taking action to transmit the detected audio to the conference session, or not transmit the detected audio to the conference session to prevent echo, based on a result of the determining.

The invention discloses a voice starting point recognition method and system based on electromyographic signals, and belongs to the technical field of voice activity detection. The method comprises the following steps: carrying out band-pass filtering, power frequency notch and framing windowing preprocessing on an original electromyographic signal acquired by a vocal muscle group; simultaneously acquiring time domain, frequency domain and time-frequency domain features in the same feature extraction layer to form a complementary multi-domain feature set; after a Teager-Kaiser nonlinear operator is utilized to enhance transient jump, a dynamic variance weighting strategy is adopted to fuse the transient jump into a comprehensive feature sequence; estimating background noise in real time based on the absolute deviation of the median of the resting section, dynamically generating an upper threshold value and a lower threshold value, and finishing initial starting and ending point judgment through a state machine; and finally, outputting a final voice start / end point through median filtering, morphological closed operation and constraint of a minimum voice segment and a maximum silence gap. According to the invention, high-precision and low-power-consumption real-time voice activity detection can be realized under the scene of strong noise, multi-person conversation or sound loss.