271 results about "Silence" patented technology

The utility model discloses a mute floor which comprises a surface treatment layer, a protection layer, a decoration layer, a first carrier, a first buffer layer and a second carrier. The surface treatment layer is arranged on one side of the protection layer, the decoration layer is arranged on the other side of the protection layer, the first carrier is arranged between the decoration layer and the first buffer layer, the first buffer layer is arranged between the first carrier and the second carrier, and the first buffer layer is of a porous structure and / or a net structure. A buffer groove is formed in at least one side face of the first buffer layer and extends to the two ends of the side face. By arranging the floor with the first buffer layer with the porous and / or honeycomb structure, noise is reflected and refracted for multiple times in the porous and / or honeycomb structure, so that sound is attenuated, a sound wave propagation path is prolonged, and the effect of more effectively reducing noise propagation is achieved.

The invention relates to the technical field of audio control, and discloses a distributed audio group mute and recovery method and system, a storage medium and a device, and the method comprises the steps: determining a master device in a distributed audio network, and enabling the master device to find and register a plurality of slave devices to establish an audio group; receiving a group mute or recovery instruction sent by a user through the master device, and performing validity verification on the instruction; the master device allocates a uniform timestamp to the verified instruction, and sends a synchronous control command to all slave devices in the group based on the timestamp; the slave equipment executes mute or recovery operation at the moment corresponding to the timestamp, synchronous response of the equipment in the group is realized, all the equipment is ensured to accurately and synchronously execute the mute / recovery operation through unified management of the master equipment and the adoption of global timestamp synchronization and adaptive delay prediction technologies, and an intelligent state monitoring and exception recovery mechanism is provided, so that the service life of the slave equipment is prolonged. Finally, efficient and reliable multi-room audio group control with excellent user experience is realized.

The embodiment of the invention discloses an audio processing method and device, and the method comprises the steps: carrying out the traversal of a segmented sentence list which is determined through the interception of an audio stream based on a mute detection result, and determining a current segmented sentence and an endpoint detection result of the current segmented sentence, determining a start frame of the current segmented sentence as a start frame of a latest to-be-identified sentence when the current segmented sentence has a sentence start point, determining an end frame of the current segmented sentence as an end frame of the latest to-be-identified sentence when the current segmented sentence has a sentence end point, and determining the latest to-be-identified sentence according to the start frame and the end frame of the latest to-be-identified sentence. According to the method and the device, the segmented sentences in the audio stream can be continuously updated, the to-be-recognized sentences can be determined according to the endpoint detection results of the segmented sentences, continuous long speech recognition is realized, and the complete content of the to-be-recognized sentences from the start frame to the end frame is recognized, so that the speech recognition efficiency is improved. The accuracy of long speech recognition can be improved.

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 wireless communication, in particular to an OTA upgrading method and system for Bluetooth sound equipment. The method comprises the following steps: acquiring to-be-upgraded firmware data and currently played audio data, and acquiring power amplifier current feedback data of a sound speaker by sending a double-frequency detection signal containing 21kHz and 22kHz; acquiring equipment transmission capability confirmation data according to a silent response mode of the sound equipment to the preset handshake signal; and calculating a nonlinear distortion coefficient of the loudspeaker according to the intensity of the 1kHz difference frequency component in the power amplifier current feedback data, and obtaining real-time channel evaluation data based on current waveform deviation analysis. According to the method, the nonlinear distortion degree of the loudspeaker is accurately evaluated through the 1kHz difference frequency component generated by the 21kHz and 22kHz dual-frequency detection signals, so that the reliability of data transmission is more accurately predicted.

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 invention relates to an audio recognition method and device based on marking and backtracking correction, and the method comprises the steps: segmenting a target audio, and enabling adjacent segments after segmentation to have a partial overlapping region; if a slice point exists in a non-mute segment of the target audio and the acoustic feature similarity of a preset number of frames before and after the slice point is smaller than a set similarity threshold value, the slice point is marked as a cut-off risk point, and the cut-off risk point is used for indicating that continuous semantics before and after the slice point has a cut-off risk; after a text corresponding to the audio of each slice is recognized through the speech recognition model, the language confidence degree of an overlapping area where the truncation risk point is located is determined, and the language confidence degree is used for indicating context language logic of the overlapping area; and if it is determined that the language confidence is lower than a set confidence threshold, correcting the recognition texts of the slices before and after the truncation risk point. According to the invention, the accuracy of audio recognition is improved.

An apparatus comprising means configured to: obtain, from a further apparatus, at least one bitstream comprising at least one component or stream of an immersive audio bitstream; identify at least one of the at least one component or stream of the immersive audio bitstream to be muted; transmit, to a further apparatus, a mute request indicating the identified at least one component or stream to be muted; obtain, from the further apparatus, the at least one bitstream, wherein the at least one bitstream is modified with respect to the identified at least one component or stream of the at least one immersive audio bitstream to be muted, wherein the modification in the at least one bitstream, is at least one of: the at least one component or stream having reduced energy; the at least one component or stream not being received; the at least one component or stream comprises a silence descriptor indicator for indicating the at least one component or stream is to be muted; the at least one component or stream having a reduced bit rate; and at least one other component or stream having an increased bit rate.

Content aware audio processing includes receiving, by a digital signal processor, a frame of audio data. In response to detecting that the frame of audio data is a silent frame, the digital signal processor selects a light graph from a plurality of graphs including the light graph and a full graph. Comfort noise is generated that corresponds to the silent frame. The comfort noise frame is processed through the light graph in place of the silent frame. The light graph is dedicated for processing comfort noise frames.

An input audio signal associated with a communication session is received. A determination is made that a participant associated with the input audio signal is not audibly speaking within the input audio signal. In response to determining that the participant is not audibly speaking, an audio feed corresponding to the input audio signal is muted by rendering the audio feed not audible to at least one other participant device connected to the communication session.

This application discloses an intelligent method and apparatus for detecting active speech with sentence segmentation based on multi-state temporal modeling. The method includes: receiving audio signals from at least one channel; extracting acoustic feature sequences from the audio signals using a target speech recognition model corresponding to the number of channels; determining the probability distribution of each speech frame corresponding to the acoustic feature sequences belonging to different speech activity states, obtaining a state sequence corresponding to each channel, wherein the speech activity state includes at least one of the following: initial silence state, speech state, intra-turn pause silence state, and inter-turn sentence segmentation silence state; and determining the time of sentence segmentation in the audio signal based on the state sequence. This application solves the technical problem of erroneous sentence segmentation in speech activity detection based on a fixed silence threshold in related technologies.

A method includes obtaining initial transcription for input natural speech; performing segmentation of initial transcription into text portions, based on punctuation marks in initial transcription; determining segment-level timestamps for text portions based on the input natural speech; performing audio segmentation on input natural speech, by cutting input natural speech based on segment-level timestamps, to obtain audio chunks; generating transcription portions for each of the audio chunks; merging transcription portions to form re-transcription; determining word-level timestamps for re-transcription, by aligning input natural speech against re-transcription; calculating silence time periods, each corresponding to silence between each two adjacent words of input natural speech, based on word-level timestamps; performing a final segmentation on input natural speech and re-transcription, based on silence time periods, to generate final audio segments and corresponding final transcription portions. The final audio segments and corresponding final transcription portions may be included in training dataset for training a model.

To predict whether there is no sound in a predetermined space based upon whether onomatopoeia indicative of silence that a human perceives in a quiet environment is generated.SOLUTION: A determination device comprises: an input part which inputs predetermined sound data generated by converting a predetermined sound into numeric data through frequency analysis to a determination model outputting whether a specific frequency of 1,800 Hz or more to 2,200 Hz or less indicative of induced otoacoustic emissions is included in addition to the frequency of sound data once the sound data is input; and a control part which displays, on a display unit, the generation of onomatopoeia indicative of silence that a human perceives in a quiet environment when the determination model receiving the predetermined sound data generates an output showing that the specified frequency is detected, but displays, on the display unit, no generation of onomatopoeia when the determination model generates an output showing no detection of the specific frequency.SELECTED DRAWING: Figure 1

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.

The application discloses a speech punctuation detection method, device, equipment, storage medium and program product. The method comprises the following steps: obtaining first speech basic data of a first speech and second speech basic data of a second speech; determining a second silence duration threshold according to the first speech basic data and the second speech basic data; determining a first punctuation detection result of the second speech according to a silence duration of the second speech and the second silence duration threshold, wherein the first punctuation detection result is used for indicating whether the second speech needs to be punctuated. The second speech basic data of the speech that needs to be punctuated and the first speech basic data of the speech that has been punctuated are used to dynamically adjust the silence duration threshold according to the needs, and then the speech punctuation detection is performed according to the second speech basic data and the adjusted silence duration threshold, so that the speech punctuation detection result is obtained, and the accuracy of the speech punctuation detection is improved.

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.

The utility model discloses an adjusting device with safety belt tensioning prompt function relates to children seat regulator technical field, including the casing that is composed of shell and backplate, the shell and backplate are fixedly connected through screw between, the button is slid to be provided with on the shell, be equipped with the sound production module between the button and shell, be equipped with the sliding rod below the button, be equipped with the red green display module for showing the state of adjusting device on the sliding rod, the utility model discloses the sound production module is integrated between the button and shell, and the displacement of linkage sliding rod realizes the sound production / mute switching, reaches the effect whether the safety belt is tensioned of auditory and visual double prompt, has improved the use safety and reliability under the scene of dim light or sight obstruction significantly.

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.

The invention discloses a method and a system for detecting mobile phone noise through sound recorded by a mute box. Belongs to the field of audio signal processing. The method comprises the following steps: S1, placing a to-be-tested mobile phone in a mute box, exciting an audio input device or an audio output device of the mobile phone through a preset audio signal, and respectively collecting corresponding audio data; s2, storing the collected audio data as an audio file with a unique identifier; s3, reading audio signal data in the audio file, and performing data standardization and validity verification; s4, extracting time domain features and frequency domain features based on the standardized audio signal data; s5, according to the time domain features and the frequency domain features, whether the audio signals conform to a preset noise feature mode is judged, and if yes, it is judged that noise exists in the mobile phone to be detected; otherwise, determining that no noise exists. According to the invention, the accuracy, consistency and automation level of mobile phone recovery quality inspection can be improved.

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.

Suppression of noise caused by switching of audio source is performed by receiving, while a speaker outputs audio from a first application, an instruction to output audio from a second application, muting audio output from the speaker in response to receiving the instruction, instructing an amplifier in communication with the speaker to associate a second channel with the second application, instructing the first application to deactivate audio transmission, instructing the amplifier to switch a speaker input from a first channel associated with the first application to the second channel, instructing the second application to activate audio transmission, and unmuting the audio output from the speaker.

The invention provides an audio processing method and system, and the method comprises the steps: obtaining a mute time point set of an audio clip; in response to the fact that the mute time period between the first mute time point and the second mute time point is larger than the preset time period, determining that the first mute time point is the end time point of the previous sentence and the second mute time point is the start time point of the next sentence; and segmenting the audio clip according to the time sequence, the end time point of each sentence and the start time point of each sentence to form a plurality of segmented sentences of the audio clip. According to the method and the device, the audio clip is segmented according to the complete speech of the natural language through mute analysis on the audio according to the habit that people pause for a moment and speak the next sentence when speaking, so that a premise is provided for subsequent business analysis. The dependence on a high-performance server is greatly reduced, the use scene of the server is increased, and the use cost is reduced. And the utterance integrity and accuracy after conversion from voice to text are improved.

The present disclosure discloses a speech data processing method and device, electronic equipment and storage medium. The method comprises: obtaining target speech data to be processed; detecting the target speech data to obtain target phonemes corresponding to each audio frame; determining the target type of the audio frame corresponding to the target phoneme based on the phoneme type corresponding to the target phoneme, wherein the target type includes a silence type or a non-silence type; determining the voice activity boundary in the target speech data based on the target type corresponding to the audio frame, and taking the voice activity boundary as the detection result of the target speech data. The method provided by the present disclosure can more accurately detect the type of audio frame by identifying the phonemes of each audio frame in the speech data and determining the silence type audio frame and the non-silence type audio frame using the phonemes, and can more accurately locate the voice activity boundary in the speech data compared with the existing technology using the two-classification method.

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 container type mute box body silencing structure, and relates to the technical field of containers. Which comprises a container structure, and a cooling structure is fixedly mounted at the bottom of the right side of the container structure, and is characterized in that the cooling structure comprises a cooling box fixedly mounted at the bottom of the right side of the container structure, a mounting cover is fixedly mounted at the top of the cooling box, and a condenser is fixedly mounted at the top of the mounting cover; a sensing pipe is fixedly installed on the outer wall of the bottom of the condenser, the container structure comprises a container body fixedly installed on the left side of the cooling structure, noise reduction treatment is conducted through noise reduction holes and noise reduction cotton, the noise reduction effect is improved accordingly, the material structure is simple and cheap, and the cost is low. And a condenser in the cooling structure can cool the inner cavity of the cooling box, then water can be conveyed into a water guide pipe through a water pump, and the temperature of the inner wall of the container body can be reduced by utilizing the water guide pipe to circulate on the inner wall of the container body.

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 relates to the field of live broadcast silence cabins, and discloses a multifunctional silence cabin which comprises a silence cabin body, a sound insulation layer, a noise reduction layer and a sound absorption layer are sequentially arranged in an interlayer of the silence cabin body from outside to inside, a workbench is installed on the right side of the bottom end in the silence cabin body, and ventilation fans are installed on the two sides of the top end of the silence cabin body. A sound insulation cover is arranged at the top end of the ventilation fan, a main controller is arranged at the top end of the interior of the silence cabin body, and temperature and humidity sensors are fixedly installed on the two sides of the main controller. The lamp tube of the multifunctional mute cabin adopts a multi-color atmosphere lamp, the position of the lamp tube can be adjusted according to direct broadcasting requirements, during adjustment, a damping sliding sleeve slides outside a first lamp holder to adjust the horizontal position of the lamp tube, an adjusting sliding block slides outside a second lamp holder, and the height position of the lamp tube can be adjusted through fixing of a locking bolt; the master controller can flexibly adjust the color and brightness of the lamp tube and the main lamp, the atmosphere feeling of live broadcast can be improved, and the problem that light is inconvenient to flexibly adjust is solved.