158 results about "Reverberation" patented technology

The invention relates to the technical field of underwater acoustic engineering and ocean exploration, in particular to a multi-channel underwater acoustic signal transmitting and measuring system and a signal processing method, which are used for solving the problems that the existing multi-channel underwater acoustic signal processing generally depends on fixed threshold detection or simple time sequence alignment, and the signal processing efficiency is low. The problem that residual vibration, local noise abrupt change or multipath reverberation is easily misjudged as an effective first arrival signal in a complex underwater environment due to the fact that the influence of the physical response characteristic of a transducer on emission time calibration is not considered and effective verification on whether a candidate arrival structure accords with a sound wave propagation physical rule is lacked is solved. According to the invention, a multi-channel energy time sequence and hydrophone space geometrical relationship is fused through an underwater sound propagation initial physical judgment module, a judgment mechanism is constructed based on distance-time consistency of sound propagation, and self-consistent verification is realized by combining a cross-channel arrival sequence and sound velocity constraint. And the accuracy, the robustness and the physical interpretability of the judgment of the effective launching time of the underwater sound are obviously improved.

The invention discloses an intelligent noise reduction method and system based on a mine earphone, and the method comprises the following steps: S1, obtaining an original noise signal in a mine operation environment, and carrying out the preprocessing of the original noise signal; s2, designing an adaptive compensation mechanism according to the reflection characteristics of the closed space of the mine, calculating a sound wave propagation attenuation coefficient and reflection delay, constructing a roadway reverberation model, and generating advanced compensation sound waves; and S3, carrying out collaborative fusion on the advanced compensation sound wave and a main noise reduction signal to generate a composite noise reduction signal. The invention relates to the technical field of intelligent noise reduction. According to the intelligent noise reduction method and system based on the mine earphone, the noise level in a mine operation environment is effectively reduced, the communication efficiency and the working comfort of operators are improved, potential damage of noise to hearing of the operators is reduced, and therefore the overall safety and the production efficiency of mine operation are improved.

An apparatus for rendering spatial audio with reverberation receive a spatial audio stream, the spatial audio stream comprising at least one audio signal and spatial metadata associated with the at least one audio signal The apparatus further obtains at least one room effect parameter comprising at least one reverberation time, adjusts at least one of the at least one room effect parameter when the at least one reverberation time is less than a determined threshold value and generates a spatial audio signal using the spatial audio stream and the adjusted at least one room effect parameter, the generated spatial audio signal comprising a first spatial audio portion and processed dependent on the spatial metadata and a second spatial audio portion generated based on the at least one audio signal and processed dependent on the adjusted at least one room effect parameter.

The present disclosure relates to a method and an apparatus for adding reverberation, a device, and a product. The method includes: separating reverberant audio based on wet audio, where the wet audio includes dry audio and the reverberant audio corresponding to the dry audio. The method further includes: determining, based on the reverberant audio, reverberation parameters for indicating reverberation effects of the reverberant audio. In addition, the method further includes: adding reverberation to synthesized dry audio based on the reverberation parameters.

For example, reverberation processing in consideration of sound quality can be performed.Provided is an information processing device including a training processing unit that generates training data by convolving a measurement signal representing acoustic characteristics collected by a same sound collection unit as a sound collection unit used for collecting an observation signal with a reference signal having sound quality and reverberation characteristics different from the observation signal, generates teaching data by adapting, to the reference signal, an average level and a delay value of a convolution signal generated by convolving a direct sound component of the measurement signal with the reference signal, and trains a learning model for performing reverberation processing of the observation signal collected by the sound collection unit by using the training data and the teaching data as input data.

The invention relates to the technical field of audio sound calibration, and discloses a room sound calibration method based on intermediate frequency reverberation time and related equipment, which are used for improving the accuracy of a sound calibration result. The room sound calibration method based on the intermediate frequency reverberation time comprises the following steps: acquiring a room impulse response of a target room and calculating the intermediate frequency reverberation time; according to the intermediate frequency reverberation time, the recommended window length of the frequency correlation window is determined by querying a predefined mapping relation, and the mapping relation is established based on acoustic association between the room reverberation time and the analysis window length; based on the recommended window length, determining the actual window length of the actual application in combination with the user setting mode; performing time domain windowing and frequency-related transfer function estimation processing on the room impulse response according to the actual window length to obtain window function weighted transfer function estimation; filter calibration coefficients are generated based on the transfer function estimates and applied for audio calibration.

The invention relates to a method, a device, equipment and a product for adding reverberation. The method includes separating reverberation audio based on wet sound audio, where the wet sound audio includes dry sound audio and reverberation audio corresponding to the dry sound audio. The method further includes determining a reverberation parameter indicative of a reverberation effect of the reverberation audio based on the reverberation audio. Further, the method includes adding reverberation to the synthesized dry sound audio based on the reverberation parameter. Through the method for flexibly adding reverberation, reverberation effects can be adaptively added to different synthetic dry sound audios, the added reverberation effects can be ensured to be close to original wet sound audios, and the user experience is improved.

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

The invention provides a voice recognition method and device and a vehicle, and the method can achieve the fusion of endowing a higher proportion to an early-arriving audio signal and endowing a lower proportion to a late-arriving audio signal through the arrival time of each audio signal, thereby preferentially maintaining voice components from a near-end or a direct path under the condition of not needing sound source positioning, and improving the voice recognition efficiency. Audio signals caused by reflection, reverberation or far-end interference are suppressed, the definition and the signal-to-noise ratio of the first recognition audio are effectively improved, and the reliability of voice wake-up and recognition is enhanced.

A spatial audio processing system operable to enable audio signals to be spatially extracted from, or transmitted to, discrete locations within an acoustic space. Embodiments of the present disclosure enable an array of transducers being installed in an acoustic space to combine their signals via inverting physical and environmental models that are measured, learned, tracked, calculated, or estimated. The models may be combined with a whitening filter to establish a cooperative or non-cooperative information-bearing channel between the array and one or more discrete, targeted physical locations in the acoustic space by applying the inverted models with whitening filter to the received or transmitted acoustical signals. The spatial audio processing system may utilize a model of the combination of direct and indirect reflections in the acoustic space to receive or transmit acoustic information, regardless of ambient noise levels, reverberation, and positioning of physical interferers.

The invention discloses a multi-mode reverberation effect generation method, which comprises the following steps: S1, multi-mode feature extraction: based on an input original audio, generating a plurality of reverberation modes, extracting time domain features and frequency domain features of the original audio and mode features of the reverberation modes, and fusing to form a multi-dimensional feature vector; s2, adaptive weight distribution: inputting the multi-dimensional feature vector into an MLP attention network, and distributing the weight of each reverberation mode based on the content type of the original audio; s3, time domain-frequency domain joint mixing: carrying out weighted stacking on the time domain signals of various reverberation modes to generate a time domain mixed signal; converting the time domain signals of various reverberation modes into frequency domain signals, and realizing cross fusion through complex conjugate multiplication to generate frequency domain mixed signals; fusing the time domain mixed signal and the frequency domain mixed signal to form a preliminary reverberation signal; and S4, spectrum calibration: performing spectrum power analysis on the preliminary reverberation signal, eliminating low-frequency turbidity or high-frequency through balanced filtering, and outputting a final reverberation signal.

To provide a method for rendering a sound source.SOLUTION: The method 1000 comprises: Step s1002 of receiving an input audio signal corresponding to a sound source; Step s1004 of receiving a reverberation parameter indicating a target energy ratio of the audio reverberation component of the sound source; Deriving one or more of: (i) a relative gain associated with a first directivity pattern of the sound source; (ii) a relative gain associated with a first reference distance of the sound source; (iii) a relative gain associated with a first configuration of the reverberation unit; (iv) a relative gain related to a first time limit of the reverberation component; Further Step s1008 of generating an adjusted audio signal using the received input audio signal, the received reverberation parameters, and any one or more of the derived relative gains (I) - (iv).SELECTED DRAWING: Figure 10

The sound signal processing method and apparatus of the present invention suppress the discomfort of sound connection at the junction of the initial reflection and the reverberation. The sound signal processing method generates an initial reflection control signal using analog calculations corresponding to the geometric shape of a virtual space, generates an reverberation control signal using reflection parameters measured in the virtual space, calculates a connection timing based on the geometric shape to make the volumes of the initial reflection control signal and the reverberation control signal the same, and gradually increases the volume of the reverberation control signal in a manner close to the volume of the connection timing during a period earlier than the connection timing.

The embodiments of the present disclosure provide an electronic apparatus and a method performed by the same, a storage medium, and a computer program product which relates to the field of artificial intelligence. The method includes: acquiring a plurality of first audio signals recorded at a plurality of different locations in a space where a sound source is located; rendering, for each first audio signal, using a transmission response corresponding to at least one other first audio signal to obtain a rendered audio signal; estimating parameters for reflecting acoustic characteristics of the space based on the plurality of rendered audio signals; and performing, based on the estimated parameters, audio rendering of a virtual audio to obtain a reverberation audio. Alternatively, the above method executed by electronic apparatus may be executed by using artificial intelligence models.

The utility model discloses a reinforced acoustic diffusion loudspeaker housing, which relates to the technical field of loudspeakers, and comprises a housing and an installation backboard, the housing is arranged on the front side wall of the installation backboard, the middle of the front side wall of the installation backboard is fixedly connected with a loudspeaker, and the loudspeaker is located at the inner side of the housing. Diffusion holes are evenly formed in the outer side of the shell, and wood miniature diffusion plates are arranged in inner cavities of the diffusion holes. When a loudspeaker is used, the loudspeaker shell capable of enhancing acoustic diffusion scatters incident sound in multiple directions through the wood miniature diffusion plate, reduces reflection and focusing of the sound, reduces echo and reverberation, enables the sound to be clearer, enables the sound to be more uniformly distributed in the space through the diffusion holes, prevents the sound from being concentrated in certain specific directions, and improves the sound quality. And the sound is intensively guided through the cover body when the sound is intensively diffused, so that the diffusion effect is improved.

An electronic device includes a vibration sensor; a microphone; memory storing at least one instruction; and at least one processor, wherein the at least one processor is configured to execute the at least one instruction to: receive, via the microphone, a speech signal including reverberation uttered by a user, receive, via the vibration sensor, a vibration signal related to the speech signal, transmitted through at least a portion of a body of the user, predict reverberation information based on the speech signal and the vibration signal, and eliminate the reverberation included in the speech signal based on the predicted reverberation information.

According to the room acoustic parameter determination method and device, the electronic equipment, the wearable equipment and the storage medium provided by the invention, the reverberation time in the acoustic scene is determined, and the acoustic scene comprises the listener position, the sound source position and the room property; the reflectivity of each reflecting surface in a room with room properties is determined through random light tracing and reverberation time, the random light tracing is used for determining a light propagation path reaching a listener, and the light propagation path reaching the listener represents a propagation path of light emitted from a sound source position of the room and received by the listener at the listener position; the descent curve of the reverberation energy can be determined through the reverberation time, and the random light tracing result is corrected based on the descent curve of the reverberation energy, so that the reflectivity of each reflecting surface in a room is corrected, and a more real spatial audio signal can be generated based on the estimated reflectivity of each reflecting surface.

The utility model discloses a holographic sound absorption module. The holographic sound absorption module comprises a sound absorption substrate and a coating, the sound absorption base plate is made of a silicon dioxide sound absorption material and comprises a sound absorption bottom plate, gridding cloth, a first surface layer and a second surface layer, the first surface layer and the second surface layer are sequentially arranged and are both made of silicon crystal mortar, a plurality of first sound absorption holes are formed in the sound absorption bottom plate, the first surface layer and the second surface layer, and a plurality of second sound absorption holes are formed in the second surface layer. The gridding cloth is arranged between the first surface layer and the sound absorption bottom plate, and the two sides of the gridding cloth are attached to the first surface layer and the sound absorption bottom plate respectively; and the coating is arranged on the outer surface of the second surface layer, and the coating is attached to the second surface layer to shield a gap between the sound absorption base plates, so that the building sound absorption layer is seamless and attractive. The sound absorption base plate made of the silicon dioxide sound absorption material is convenient to splice and adapt to the environment, the construction difficulty can be reduced, meanwhile, the first sound absorption holes and the second sound absorption holes are matched for sound absorption and noise reduction, and reverberation and echo of the indoor space of the special-shaped building are reduced.

The invention discloses an immersive exhibition hall sound field control system, and relates to the field of sound field control. The system comprises an acoustic sensing module which is used for collecting all-domain audio signals of an exhibition hall in real time; the global analysis module is in signal connection with the acoustic sensing module and is used for processing the collected audio signals; the sound field execution module is used for receiving the sound field control strategy generated by the global analysis module; and the central control platform is used for coordinating the work of each module, managing the audio content and providing a man-machine interaction interface. According to the system, crossing from indifference broadcast to partition directional sound field is realized, immersive personalized auditory experience is created, and the system has the capabilities of dynamically sensing and optimizing acoustic environment: the system can position a noise source in real time and evaluate reverberation, so that the problems of mutual crosstalk and loud noise of different explanation contents in a traditional system are avoided, and the system is suitable for popularization and application. And a plurality of exhibition areas are allowed to synchronously carry out clear and non-interfering explanation.

This invention provides a low-complexity multi-channel denoising and noise reduction method based on Kalman filtering, comprising: acquiring a signal and preprocessing the acquired signal to obtain a signal in the short-time Fourier domain; calculating a multi-channel noise covariance matrix; estimating multi-channel autoregressive parameters using the delayed, noiseless reverberant signal estimated in the previous frame and the acquired signal in the current frame, and determining the variance value of the Kalman state noise based on the sound source change detection result of the previous frame; estimating a noiseless reverberant signal using the estimated autoregressive parameters, the acquired signal in the current frame, and the estimated multi-channel noise covariance matrix; delaying the estimated noiseless reverberant signal and calculating the estimated noiseless late reverberant signal using the autoregressive coefficients; subtracting the noiseless late reverberant signal from the noiseless reverberant signal to obtain the desired direct sound and early reverberant signal. This invention reduces computational complexity and enables real-time applications in embedded products.

To provide a reverberation control device and a program capable of reducing difficulty in hearing the sound of a desired object due to the influence of reverberation sound.SOLUTION: A reverberation control device includes a metadata analysis unit that analyzes metadata and specifies an identifier of an object to be controlled, a reverberation parameter calculation unit that calculates a gain of late reverberation of the object on the basis of the identifier of the object and a longest distance of an area in which a clarity of the object is secured, and a rendering unit that calculates an amplitude gain coefficient for adjusting an amplitude of the late reverberation in accordance with a distance between a user and the object and the gain of the late reverberation and multiplies the late reverberation of the object by the amplitude gain coefficient to add reverberation.SELECTED DRAWING: Figure 1

A biometric authentication device is provided with: a replay unit for reproducing a sound; an ear authentication unit for acquiring a reverberation sound of the sound in an ear of a user to be authenticated, extracting an ear acoustic feature from the reverberation sound, and calculating an ear authentication score by comparing the extracted ear acoustic feature with an ear acoustic feature stored in advance; a voice authentication unit for extracting a talker feature from a voice of the user that has been input, and calculating a voice authentication score by comparing the extracted talker feature with a talker feature stored in advance; and an authentication integration unit for outputting an authentication integration result calculated based on the ear authentication score and the voice authentication score. After the sound is output into the ear, a recording unit inputs the voice of the user.

The application discloses a neuromorphic auditory event-based anti-reverberation processing method and system for security scene abnormal voiceprint recognition, which comprises the following steps: acquiring an asynchronous auditory event sequence in a high-reverberation space; anchoring an initial pulse cluster triggered by a direct sound wave and extracting the same as a reference feature; constructing a channel-level dynamic refractory period mask window based on the reference feature and an estimated reverberation duration; determining a newly-arrived pulse cluster similar to the reference feature as a delayed ghost pulse generated by an echo and intercepting and filtering the same within the mask window action period, so as to generate a pure pulse stream; and inputting the pure pulse stream into a pulse neural network for recognition. The application effectively filters out redundant pulses generated by physical echoes from a data source in a native asynchronous pulse domain, avoids high-cost waveform reconstruction, can effectively prevent the subsequent pulse neural network from making a misjudgment and power saturation due to redundant data, improves the accuracy, robustness and real-time performance of the system in the abnormal voiceprint recognition under a high-reverberation environment, and meanwhile, retains the low-power consumption advantage of event-driven sensing.

An apparatus comprising at least one processor and at least one memory storing instructions that, when executed by the at least one processor, cause the system at least to perform: obtaining the at least one audio signal; obtaining at least three reverberation parameters based on obtaining reverberation configuration information and directional configuration information; controlling a reverberator using the at least three reverberation parameters for providing a reverberant audio signal using the at least one audio signal, based on the at least three reverberation parameters, the reverberant audio signal comprising at least a late reverberation part; processing at least a portion of the reverberant audio signal, the portion of the reverberant audio signal at least partially interfering with a reflection audio signal; and generating a binaural output audio signal, the binaural output audio signal comprising the processed reverberant audio signal.

The invention relates to the technical field of driving scene recognition, in particular to an intelligent driving scene recognition method and system. The method comprises the following steps: acquiring driving sound wave data through a vehicle-mounted microphone array; sound wave feature extraction is carried out according to the driving sound wave data to obtain sound wave feature data; performing time-frequency domain reverberation feature extraction on the sound wave feature data to obtain time-frequency domain reverberation feature data; performing structure type discrimination according to the time-frequency domain reverberation characteristic data to obtain structure type data; and performing scene confidence fusion on the structure type data to obtain driving scene data. According to the invention, through collaborative modeling of the sound wave reflection characteristics, the resonance modal behavior and the vehicle dynamics state, the recognition precision and the system robustness in a complex structure scene are improved. Compared with a traditional single vision or radar sensing method, the method has higher environment adaptability and recognition credibility under the condition of limited illumination or multi-path interference.

The application discloses a building construction site data acquisition system based on Internet of Things and relates to the technical field of building engineering. The system is provided with a sound wave acquisition module, three-dimensional space is laid out with acquisition points in underground pouring structures, and an excitation source position is set in combination with the position of a pre-buried anchoring support, so that sound wave reverberation signals generated by wideband vibration signals are used to realize high-density and high-time-efficiency data acquisition. The MEMS three-axis accelerometer arranged in the acquisition point can collect the sound wave decay time Tdec and the reverberation residual energy peak value Ares of each acquisition point in real time, form a standard vibration sound wave energy data set, and provide a data basis for subsequent structure integrity judgment. Compared with traditional manual detection, image recognition and external detection methods, the system has the advantages of not needing to be damaged and not needing to additionally lay out ultrasonic equipment, and can realize early warning of internal interlayers, cavities and cold joints and real-time visualization of structure states in the construction process.

The application discloses a multi-sound source direction-finding positioning method based on a vector microphone in a strong reverberation environment, which comprises the following steps: converting mixed voice signals collected by the vector microphone from a waveform domain to a time-frequency domain to obtain a spectrogram of an observation signal; removing virtual source single time-frequency points by using a frequency smoothing method and iteratively generating low-reverberation self-term source time-frequency point sets; constructing a steering vector dictionary; finding a steering vector most likely to appear in any time-frequency point in the self-term source time-frequency point sets from the steering vector dictionary, and then calculating the STFT coefficient value of the video point corresponding to the source according to the steering vector, determining whether the time-frequency point is a single source time-frequency point not interfered by reverberation according to the STFT coefficient value, and generating a single source time-frequency point set not interfered by reverberation; estimating the number of sources by using a smoothing histogram method; and obtaining multi-source direction-finding results by using MUSIC. The application can accurately extract single source time-frequency points not interfered by reverberation in the time-frequency domain, and then realize accurate multi-source positioning.

This utility model relates to the field of sound pickup equipment technology, specifically a sound pickup structure. The sound pickup structure includes a housing with an internal cavity and a through-hole first channel; a circuit board disposed within the cavity; and microphones, with multiple microphones arranged in an array on the circuit board, each microphone having a pickup element and a through-hole second channel within the pickup element. The first and second channels are interconnected. By arranging multiple microphones in an array on the circuit board, omnidirectional sound pickup is achieved. Furthermore, the first channel on the housing serves as the primary channel, maximizing the decibel level of sound pickup. The through-hole second channel within the pickup element, with the microphone pickup element tightly fitted so that the bottom of the second channel faces the microphone, reduces multiple reverberations and bounces of audio within the cavity, minimizes the inflow of noise from unintended directions, and results in better sound pickup performance.

An audio signal processor for generating a two-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a two-channel combiner (200) for synthesizing two-channel acoustic data from single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from an audio signal and two-channel acoustic data, wherein the two-channel combiner (200) is configured to separate the single-channel acoustic data into at least two parts consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion (210), and to process at least two parts individually for each part to generate two-channel acoustic data (220, 230, 240), and the two-channel combiner (200) is configured to compute two-channel acoustic data for the early reflection portion using a specular reflection portion describing a separate early reflection and a diffusion portion describing the diffusion effect within the early reflection portion (230).

The embodiment of the invention discloses a playback audio generation method, and the method sequentially carries out low-pass filtering, noise increasing, reverberation processing and clipping processing, and converts original audio data into simulated playback audio. The key physical process in the playback attack is simulated, and the generated audio is highly similar to the real playback attack in acoustic characteristics, so that the high-quality playback audio is provided for the audio living body detection model.