10066 results about "Audio signal" patented technology

A headset antenna and a connector for the same are provided. The headset antenna includes an audio signal line, an antenna and a high impedance element in specified application frequency ranges. The audio signal line is adapted for transmitting an audio signal and the antenna is adapted for receiving an RF signal. The high impedance element is disposed on a transmission path of the audio signal and generates a high impedance at a specified frequency band of the RF signal, so that the audio signal line is equivalent to an open circuit and the antenna obtains a better receiving capability.

The invention relates to an intelligent MV generation method, system and device based on AIGC and a medium, and belongs to the technical field of artificial intelligence content generation. The method comprises the following steps: receiving an original music file input by a user and a user instruction; performing audio preprocessing on the original music file to generate a standardized audio signal, and mapping a user instruction into a semantic vector; rhythm features and emotion features are extracted, music feature vectors are generated and subjected to feature alignment with semantic vectors, and an initial key frame sequence is generated based on a diffusion model; performing particle effect density dynamic adjustment and shot switching point generation on the initial key frame sequence, outputting an enhanced key frame sequence, performing optical flow frame insertion and long time sequence prediction, and generating an optimized video frame sequence; and performing block parallel rendering on the optimized video frame sequence according to the target platform parameters to generate an MV file and metadata. According to the invention, deep fusion of audio and visual contents can be realized, and high-quality video contents can be generated according to individual demands of users.

The invention relates to the technical field of audio noise reduction, and discloses a Bluetooth headset environment noise adaptive adjustment method and system, and the method comprises the steps: collecting multi-band sound data of headset environment noise through a preset acoustic sensor array, generating a noise spectrogram, extracting a noise masking threshold through time-frequency joint analysis, and collecting a headset audio signal; next, noise suppression is carried out on the audio signal by using a multi-order digital filter bank to obtain a primary noise reduction signal, the primary noise reduction signal is optimized into an optimized noise reduction signal based on a signal-to-noise ratio, and the frequency response matching degree of the optimized noise reduction signal and the original audio signal is calculated; then, performing phase correction and amplitude-frequency equalization processing according to the frequency response matching degree to obtain an audio equalization parameter; and finally, monitoring residual noise energy based on the parameter, generating a noise regulation and control instruction, driving adaptive sampling, and generating an adaptive noise reduction scheme. According to the invention, the audio noise reduction effect of the Bluetooth earphone can be improved.

The invention relates to the technical field of voice processing, can be applied to business scenes of financial science and technology, medical health and the like, and discloses an audio signal authenticity verification method, device, equipment and medium, and the method comprises the steps: constructing an original audio text data set, and generating an adversarial sample set, inputting the original audio text data set and the adversarial sample set into an audio detection model for joint training to obtain an audio detection model subjected to adversarial training; the method comprises the steps of obtaining a to-be-detected audio signal and extracting an acoustic feature of the to-be-detected audio signal, obtaining a non-acoustic feature associated with the to-be-detected audio signal, constructing a multi-dimensional feature vector according to the acoustic feature and the non-acoustic feature, inputting the multi-dimensional feature vector into an audio detection model to generate an abnormal index, and executing a hierarchical response operation based on the abnormal index. According to the method, the robustness of the model is enhanced by introducing adversarial sample training, and the multi-dimensional feature vector is constructed by fusing the multi-modal features, so that accurate recognition and hierarchical response to the voice cloning attack are realized.

A loudspeaker device includes a speaker component, a first microphone element, and a second microphone element. The speaker component includes an earphone core and a core housing. The first microphone element is disposed on a first branch circuit board of a circuit board. The second microphone element is disposed on a second branch circuit board of the circuit board. The circuit board is configured to electrically connect to an audio signal wire, a first auxiliary signal wire, and a second auxiliary signal wire of an external control circuit. The audio signal wire, the first auxiliary signal wire, and the second auxiliary signal wire are electrically and respectively connected to the earphone core, the first microphone element, and the second microphone element through the circuit board. A board surface of the second branch circuit board is tilt with respect to a board surface of the first branch circuit board.

According to an aspect of the present invention, there is provided an electrical audio signal processing system and device, comprising: a computer graphics processing and selective visual display system with a screen; an eye tracking device; a processor; one or more computer memory devices; wherein the processor is arranged for operations comprising: measuring the user's eye movements to ascertain the specific word on which the user is fixated, by the eye tracking device; modifying the display at the user's current fixation point; applying a delay between the presentation of successive graphic elements based on the user's calculated rate to accommodate the user's required time; and presenting elements to the user at a rate based upon the user's required time.

The invention relates to an audio noise reduction method, device and system based on deep learning, and the method comprises the steps: obtaining an input audio signal with noise, and carrying out the multi-scale time-frequency decomposition, and obtaining a mixed time-frequency feature and a noise fingerprint spectrum; performing parameter parallel processing on the noise fingerprint spectrum through a preset dynamic kernel generation network, and performing preliminary noise reduction processing on the mixed time-frequency characteristics to obtain noise-reduced mixed data; performing dual-path processing structure construction on the noise reduction mixed data to obtain amplitude optimization data and phase optimization data; performing dynamic time-frequency domain cross fusion on the amplitude optimization data and the phase optimization data to obtain fused audio data; and carrying out differentiable acoustic equation constraint adversarial training on the fused audio data, and carrying out inverse time-frequency transformation processing to obtain a target noise-reduced audio signal. According to the invention, the overall efficiency and effect of audio signal processing can be effectively improved.

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 voice processing, can be applied to business scenes of financial science and technology, medical health and the like, and discloses a voice feature processing method, device, equipment and medium. Performing time resolution analysis based on the fused Mel band energy to generate a multi-scale Mel spectrum amplitude value, and performing nonlinear transformation on the multi-scale Mel spectrum amplitude value according to the noise intensity parameter to generate a noise suppression Mel component; and generating a perception weighting coefficient according to an auditory perception model, and executing frequency domain energy adjustment on the noise suppression Mel component to generate Mel spectrum representation. On the basis of frequency resolution self-adaption, time resolution dynamic adjustment and auditory perception modeling, nonlinear transformation and perception weighting processing are applied to the multi-scale Mel spectrum amplitude value, the influence of noise interference on voice features can be effectively reduced, and the key information retention capacity of voice signals is enhanced.

The invention relates to an indoor mobile service robot interaction task execution method and device, a storage medium and an indoor mobile service robot system. The method comprises the following steps: acquiring original visual data and original spatial data, respectively processing the original visual data and the original spatial data, acquiring an original audio signal, and converting the original audio signal into a text signal to obtain a dynamic interactive input instruction; analyzing the dynamic interaction input instruction by utilizing a preset large language model to obtain a meta-action sequence corresponding to the to-be-processed task; according to the meta-action sequence, executing each meta-action in the to-be-processed task in sequence, and scheduling the corresponding processed visual data and spatial data when each meta-action is executed; when the execution result of the meta-action is confirmed, the current task execution state is broadcasted through voice, and task interaction is completed. Compared with the prior art, the method has the advantages of complex and wide task processing, high dynamic adaptability, natural interaction, high task execution success rate of an indoor service scene and the like.

The invention relates to the technical field of voice processing, can be applied to business scenes of financial science and technology, medical health and the like, and discloses an audio coding and decoding method, device, equipment and medium, and the method comprises the steps: carrying out the sliding window segmentation processing of an input audio signal, and generating signal segments; processing the signal segments through an encoder containing a multi-layer self-attention mechanism to generate continuous potential representations; performing decomposition vector quantization processing on the continuous potential representation to generate a discrete code; the discrete codes are processed through a decoder comprising a multi-layer self-attention mechanism, and reconstructed signal segments are generated; and splicing the reconstructed signal segments to generate a complete audio signal. According to the method, a traditional convolution structure is replaced by a multi-layer self-attention mechanism, global time sequence dependence modeling is carried out on the audio signals after sliding window segmentation, effective compression of potential representation is realized in combination with decomposition vector quantization, and continuity and fidelity of audio reconstruction are improved on the premise that calculation complexity is not increased.

The invention relates to the technical field of artificial intelligence and emotion calculation, in particular to an audio and video dual-mode emotion recognition method and system based on adapter fusion. The method comprises the following steps: acquiring a video frame sequence and an audio signal, and preprocessing the video frame sequence and the audio signal; constructing an emotion recognition model; based on a bimodal feature extraction module, a space adapter and a global adapter are embedded in sequence, and corresponding modal enhanced space features and global features are obtained in sequence; generating intermediate representations of the corresponding modes based on the global features, and performing feature fusion according to the intermediate representations to obtain fusion features of the corresponding modes; the fusion features are spliced, time sequence features are extracted, and final features are obtained; inputting the final features into a classifier to obtain a predicted emotion category, training an emotion recognition model by adopting a loss function, and determining an optimal emotion recognition model; and inputting a to-be-recognized video frame sequence and an audio signal into the emotion recognition model, and outputting a recognition result.

The invention relates to the technical field of voice processing, can be applied to business scenes of financial science and technology, medical health and the like, and discloses a voice enhancement method, device and equipment based on noise perception and a medium. Environment feature information is extracted and input into an audio enhancement model to generate an enhanced audio signal; obtaining a reference audio sample, extracting a personalized feature vector, and carrying out personalized processing on the enhanced audio signal; and collecting playing feedback data, determining a playing time domain adjustment parameter and a playing frequency domain adjustment parameter, adjusting the personalized enhanced audio signal, and generating an optimized audio signal. According to the method, dynamic adjustment is realized in combination with the feedback parameters in the playing process by fusing the environmental perception information and the personalized speaker characteristics, clear and natural optimized audio output with personalized styles can be generated in a complex environment, and the voice interaction quality and adaptability are improved.

The invention relates to the technical field of voice interaction, particularly discloses an AI intelligent robot voice interaction method and system, and aims to solve the problems of low voice interaction accuracy, insufficient reliability and lack of authority control in a complex noise environment. A dynamic noise feature library containing steady-state noise, impact noise and human voice interference features and a pre-stored gesture instruction library are constructed, audio signals are collected in real time, low-frequency-band, middle-frequency-band and high-frequency-band differential noise reduction is executed, Mel-frequency cepstral coefficient features are extracted, noise scenes are matched, corresponding voice recognition models are switched, and voice recognition is achieved. And calculating a confidence value of the voice instruction, outputting multi-modal verification data in combination with a dynamic confidence threshold, and outputting an authority control signal through voiceprint matching, authority verification and instruction consistency judgment. Through multi-modal fusion, dynamic adaptation and authority control, the voice recognition accuracy and interaction safety in a complex noise environment are remarkably improved, and the method is suitable for scenes such as factory intelligent inspection.

A method for synchronizing audio reproduction in collocated end devices is presented. Each of the devices auto-correlates using noise to determine a threshold prior to the antenna receiving an audio signal. When the devices receive a common audio signal, they provide audio outputs. Each device cross-correlates its audio output with the audio outputs of the other devices. The timing of the audio output of each device is then adjusted such that the audio outputs of all of the devices align temporally with the lagging or leading device.

Methods, systems, and computer programs are presented for audio-visual speech generation with multi-modal latent diffusion models. One method includes encoding raw audio signals and video frames into respective latent spaces using audio and visual autoencoders. A text transcript is processed into phoneme sequences using a text transcript processor. The audio and visual latent spaces are conditioned using the text transcript and a conditioning variable. Joint distributions of the visual and audio latent spaces, text transcript, and conditioning variable are learned using a multi-modal latent diffusion model. The model adds noise to the latent audio-visual representations and predicts the noise through denoising neural networks. An inverted diffusion process is utilized to generate diverse speech content and speaker characteristics, resulting in realistic audio-visual speech. The technology presented provides a novel approach to conditional speech generation with potential applications in speech synthesis, voice conversion, and speech recognition.

An acoustic output device is provided. The acoustic output device includes a housing; a bone conduction vibrator configured to generate bone conduction sound waves and transmit the bone conduction sound waves to a cochlea through the housing to produce sound; an air conduction vibrator configured to generate air conduction sound waves, the air conduction sound waves being transmitted to an ear through sound guiding holes in the housing; and a processing module configured to provide the bone conduction vibrator and the air conduction vibrator with a first audio signal and a second audio signal, respectively, the first audio signal and the second audio signal having a frequency division point, the first audio signal and the second audio signal including components with frequencies above and below the frequency division point, respectively. The frequency division point is not less than 600 Hz.

The invention relates to the technical field of big data, and discloses an audio and video monitoring and early warning method and system based on multi-modal model driving, and the method comprises the steps: S1, collecting audio and video signals in a monitoring scene, and extracting the energy distribution characteristics of the audio and video signals, wherein the energy distribution characteristics at least comprise the frequency energy distribution of the audio signal and the brightness and color change rate of the video signal; s2, multi-modal feature extraction of energy distribution features is carried out based on audio and video signals, and environment background feature vectors are constructed in combination with environment background information; s3, dynamically adjusting anomaly detection thresholds of the audio signal and the video signal based on the environmental background feature vector; s4, comparing the audio and video signal energy distribution characteristics extracted in real time with an anomaly detection threshold value, and when the audio and video signal energy distribution characteristics exceed the range of the anomaly detection threshold value, determining an abnormal event and obtaining early warning information; and S5, sending the early warning information from the edge equipment to a monitoring center through a low-bandwidth communication protocol.

The invention relates to the technical field of voice processing, can be applied to service system platforms of medical health, financial science and technology, communication and the like, and discloses an audio compression and reconstruction method, device, equipment and medium. Residual vector quantization and inverse quantization are carried out to generate basic spectrum features; and compensating a high-frequency component by adopting a spectrum expansion network, predicting a time domain error in combination with a long-short-term memory network, generating a time domain residual compensation signal, and superposing the time domain residual compensation signal with the reconstructed time domain audio signal to obtain a target audio signal. Through residual vector quantization and neural network modeling, the audio reconstruction quality under low-bit-rate compression is improved, spectrum detail reservation and time domain error compensation are optimized, meanwhile, the calculation complexity is reduced, and the method is suitable for high-sampling-rate and resource-limited scenes.

The invention provides an expectoration auxiliary control method and device. The method comprises the steps that audio signals of breathing sounds and cough sounds at multiple positions of the chest of a patient are collected through an audio sensor array, and a standardized audio feature data set is obtained through noise reduction, segmentation and feature extraction; a dynamic weighted graph model is constructed, and a random edge sampling algorithm and a parallel batch processing dynamic algorithm are combined to analyze and obtain a sputum viscosity index and a sputum distribution position map; based on the result, mapping the vibration parameter space into an unweighted disk diagram, and obtaining a personalized vibration treatment scheme by using a shortest path algorithm; in combination with the real-time breathing cycle of the patient, working parameters of the sound wave vibrator and the negative pressure suction device are synchronously controlled, and a coordinated and consistent multi-mode treatment execution instruction is obtained; and collecting real-time feedback data in treatment, and dynamically adjusting working parameters through reinforcement learning to obtain an optimized expectoration adjuvant therapy scheme. Intelligent adjustment can be achieved based on the real-time breathing state and sputum characteristics of the patient, the expectoration efficiency is improved, and discomfort of the patient is reduced.

The invention relates to the technical field of audio and video control, and discloses an audio and video player control method based on a voice instruction. The method comprises the steps that an original voice instruction stream of a user is collected, the instruction stream comprises a time domain audio signal sequence, an environment noise spectrum and user pronunciation characteristic parameters, and voice information can be comprehensively captured; multi-modal instruction analysis processing is carried out on the original voice instruction stream, a structured control instruction set containing acoustic control intention identification, semantic operation object description and context correlation parameters is generated, and the analysis precision is improved; then executing player state adaptation based on the set, generating a dynamic control response sequence containing an equipment state adjustment command, a media content positioning parameter and an interface interaction logic identifier, driving a player to execute a multi-dimensional control operation and generating real-time play control effect feedback data; and finally, multi-modal analysis parameters are optimized according to feedback data, a self-adaptive instruction analysis strategy is generated, and the control experience of a user on the audio and video player is optimized.

The invention provides a water turbine anomaly detection and fault diagnosis method based on multi-sensor data fusion, and the method comprises the steps: collecting stress data, audio signals, vibration signals and working condition data during the operation of a water turbine, and carrying out the preprocessing of the collected data, including data cleaning, alignment and normalization; extracting time-domain, frequency-domain and time-frequency-domain features of the preprocessed stress data, audio signals and vibration signals, forming a high-dimensional feature vector in combination with the preprocessed working condition data, and performing dimensionality reduction on the high-dimensional feature vector by using principal component analysis; performing anomaly detection on the dimensionality-reduced high-dimensional feature vector according to a clustering algorithm, a local outlier factor algorithm and an anomaly detection algorithm constructed by an auto-encoder neural network, and generating a comprehensive anomaly score in combination with a preset weight to obtain an anomaly detection result; and when the water turbine is in an abnormal state, the fault type is diagnosed through the similarity between the dimensionality-reduced high-dimensional feature vector and the historical fault feature template.

The invention relates to an audio feature extraction method and device based on a neural network. The method comprises the following steps: performing framing processing and fast Fourier transform on an original audio signal acquired by a vehicle-mounted microphone to obtain a frequency domain feature sequence; extracting short-time energy, a zero-crossing rate, a frequency spectrum centroid and a Mel-frequency cepstrum coefficient from the frequency domain feature sequence, and combining first-order difference and second-order difference features of each feature to obtain a multi-dimensional acoustic feature point sequence; performing time sequence arrangement on the multi-dimensional acoustic feature point sequence, constructing an original feature matrix, and performing principal component analysis on the original feature matrix to obtain a feature description matrix; inputting the feature description matrix into a dynamic time feature extraction network for deep time-frequency feature analysis to obtain vehicle-mounted voice deep time-frequency features; and performing voice detection classification based on the vehicle-mounted voice deep time-frequency features, and outputting a vehicle-mounted voice detection result. According to the invention, various noise interferences in a vehicle-mounted environment are effectively suppressed, and the anti-noise capability of the system is enhanced.

The invention relates to the technical field of audio transmission systems, and provides a noise interference compensation method applied to an audio transmission system, and the method comprises the steps: firstly obtaining a noise-containing audio signal set of a target audio transmission channel, the noise-containing audio signal set comprises a plurality of noise-containing audio segments and pure audio segments corresponding to the noise-containing audio segments; then noise interference feature extraction is carried out on the noise frequency signal set to obtain a time domain noise feature and a frequency domain noise feature of each noise frequency segment, and then dynamic feature fusion is carried out on the time domain noise feature and the frequency domain noise feature based on a preset noise compensation model to generate a noise interference feature set; the method comprises the following steps: acquiring a noise interference feature set, training a noise compensation model according to a mapping relation between the noise interference feature set and a corresponding pure audio clip, generating a noise compensation parameter set, finally loading the noise compensation parameter set to a real-time processing module, and performing noise interference compensation operation on a noise-containing audio signal transmitted in real time. Therefore, the influence of noise on audio signals is effectively reduced, and the audio transmission quality is improved.

The invention relates to the technical field of speech processing, can be applied to business scenes of medical health, financial science and technology and the like, and discloses a speech enhancement method based on multi-scale feature learning, which comprises the following steps: framing an input audio signal, extracting a Mel-frequency spectrum feature, extracting a frequency domain feature by using a multi-scale convolutional neural network, and carrying out multi-scale feature learning on the frequency domain feature; carrying out coding dimension reduction on the image; noise is suppressed through a deep residual network, enhanced audio features are generated, a non-autoregression generative model is adopted for feature conversion, and finally a generative adversarial network is used for reconstructing a target voice waveform. Voice frequency domain features are extracted through the multi-scale convolutional neural network, and the feature expression ability of different frequency bands is improved; noise suppression is carried out through a deep residual network, and the purity of the voice signals is enhanced; feature conversion is optimized through a non-autoregression generation model, and the modeling efficiency of speech enhancement is improved; the target voice waveform is reconstructed through the generative adversarial network, and the naturalness and definition of the generated voice are improved.

The invention provides a wind turbine generator voiceprint fault recognition method, and relates to the technical field of wind turbine generator state monitoring and fault diagnosis, and the method comprises the steps: carrying out the noise reduction of an original audio signal through variational mode decomposition, screening a target mode of which the frequency, energy and kurtosis accord with features, and reconstructing the signal; extracting a Mel frequency cepstrum coefficient and a sensing noise robust coefficient, and generating multi-dimensional voiceprint data in combination with statistical characteristics such as a frequency spectrum gravity center, a spectrum entropy, energy, kurtosis and a zero-crossing rate; constructing a support set based on the prototype network, realizing small sample fault classification by calculating the Euclidean distance between the feature vector and the prototype vector, and outputting a preliminary result; judging whether the voiceprint is abnormal according to a preset threshold value, if so, storing the voiceprint into a dynamic abnormal voiceprint knowledge base; frequently occurring abnormal samples are manually labeled and added into a support set, the prototype network is retrained to update the model, and continuous optimization of the fault recognition capability is achieved.

The invention discloses an intelligent security and protection monitoring system and method fusing multi-modal data analysis, and relates to the field of intelligent monitoring, and the method comprises the following steps: collecting a video stream, infrared thermal imaging data and an environment audio signal in real time through a heterogeneous sensor array; operating a lightweight neural network at an edge computing node to carry out space-time alignment and feature level fusion on the multi-source data, and generating an enhanced environment sensing matrix; a dual-channel anomaly detection mechanism is adopted, a first channel identifies short-term sudden abnormal events through a dynamic threshold self-adaptive model, and a second channel early warns potential risk behaviors through a time sequence prediction model; according to the system and the method, a dynamic privacy protection mechanism is adopted, behavior analysis is completed on the premise of ensuring biological feature safety, and transformation from passive monitoring to active early warning is realized in combination with a space-time prediction model; the problem of multi-view target tracking is solved through a distributed consensus algorithm, and the monitoring reliability in a complex scene is remarkably improved.

At least one aspect of the present disclosure provides an emotion recognition method using an audio stream performed by an emotion recognition apparatus including receiving an audio signal having a preset unit length to generate the audio stream corresponding to the audio signal; converting the audio stream into a text stream corresponding to the audio stream; and inputting the audio stream and the converted text stream to a pre-trained emotion recognition model to output a multi-modal emotion corresponding to the audio signal.

The invention provides a noise reduction processing method and system for Bluetooth audio equipment, and the method comprises the steps: obtaining an audio signal, and generating spectrum distribution data based on the audio signal; according to the intensity value of each frequency band in the spectrum distribution data, extracting a frequency characteristic set of a multi-source interference component; for the frequency characteristic set, determining a dynamic fluctuation parameter corresponding to the frequency characteristic; adjusting the configuration of the filter according to the corresponding relation between the dynamic fluctuation parameter and the preset noise width; synchronously updating the time domain response parameter of the filter based on the amplitude change rate of the dynamic fluctuation parameter; calculating dynamic threshold upper and lower limits according to the time domain response parameters, and generating a dynamic threshold sequence; and performing gain compensation processing on the environment audio signal based on the dynamic threshold sequence to generate a noise reduction audio signal. Adaptive time domain response parameters and a dynamic threshold sequence are adopted, so that filter configuration can be adjusted in real time according to noise characteristics.

An ear-worn device is provided that operates to isolate and individually treat the received speech of a target speaker or multiple target speakers from an audio input signal detected in a multi-speaker environment. The ear-worn device uses a machine learning model that receives a voice signature of each of one or more target speakers as input signals, to identify and isolate the component of the audio input signal attributable to the target speaker(s). Once isolated, the target speaker's speech may be enhanced, de-emphasized, or otherwise processed in a manner desired by the wearer of the ear-worn device. The wearer may use an external electronic device, e.g., a phone, to select one or more target speakers in a conversation and / or configure various settings associated with processing the speech on the ear-worn device.