278 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.

The invention discloses a sonar target detection method, particularly relates to the field of underwater foresight sonar target detection, and is used for solving the problems of weak echo loss and false detection caused by multi-target adjacent and reverberation noise. An adjacent target is accurately stripped by means of a morphological context graph structure, noise and effective echoes are dynamically distinguished by a retention priority coefficient driven by an echo stationary redundancy and a shadow fall dissimilarity degree, and a final target contour and a flow direction tensor are deeply coupled for navigation guidance. The high retention rate and the false detection rate of weak target signals in a complex sound field are remarkably reduced from the overall processing link level, so that the multi-target detection result is highly consistent with the actual underwater topographic change, and real-time accurate positioning can be completed without a fixed threshold value.

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.

The invention belongs to the technical field of building design, and particularly relates to a BIM-based building complex special-shaped component processing method and system, and the method comprises the steps: directly converting a three-dimensional structure model into a topological structure adaptive to acoustic analysis, and integrating a sound field simulation and feedback optimization mechanism into a unified BIM platform. Accurate prediction and dynamic optimization of the acoustic performance of the special-shaped component from the initial stage of design are realized; according to the method, the design efficiency and accuracy are greatly improved, human errors are reduced, potential acoustic problems such as a reverberation abnormal area can be accurately recognized and relieved, and the component form is adjusted through iteration until the preset acoustic standard is met; besides, cooperation among multiple disciplines is promoted, decision support capability based on quantitative data is enhanced, so that overall quality and user satisfaction of a project are improved, cost and time consumption of later modification are remarkably reduced, and the method has important engineering application value.

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 provides an intelligent voice control system and a voice enhancement method in a reverberation environment. The method comprises the following steps: acquiring an initial voice signal in the reverberation environment; determining reverberation energy distribution in each frequency band according to the time-frequency spectrogram of the initial voice signal and the frequency response attenuation curve in each frequency band, and determining reverberation suppression gain of the initial voice signal in a time-frequency domain through all reverberation energy distribution; performing reverberation suppression on the initial voice signal according to the reverberation suppression gain to obtain a voice signal after reverberation suppression; performing time-frequency mask estimation on the initial voice signal to obtain a time-frequency mask of the initial voice signal, and determining residual interference energy of each time-frequency unit of the initial voice signal in a time-frequency domain according to the time-frequency mask and an interference covariance matrix of the target sound source; and performing harmonic enhancement on the voice signal after reverberation suppression through all residual interference energy to obtain a voice enhanced signal. By adopting the scheme of the invention, the acoustic crosstalk caused by reverberation can be suppressed.

The invention provides a method and system for measuring the sound source radiation sound power level in a reverberation pool, and the method comprises the steps: constructing a reverberation test system according to a preset standard, laying a hydrophone array and a to-be-measured sound source in the reverberation pool, and enabling the array element spacing of the hydrophone array to be smaller than or equal to the minimum half-wavelength; enabling the to-be-detected sound source to emit a to-be-detected signal, and collecting complete time domain sound pressure data of the to-be-detected signal; calculating the mean square sound pressure level of the measuring point in the reverberation control area based on the steady-state time domain sound pressure data of the sound signal; calculating complete time domain sound pressure data of all array elements based on Fourier transform to obtain a sound energy density change curve, and fitting the sound energy density change curve to obtain reverberation time; sound field correction is obtained through calculation according to the reverberation time, and the radiation sound power level of the sound source to be measured is obtained according to the sound field correction. According to the invention, the experiment process is simplified, the measurement efficiency is obviously improved, the experiment cost is reduced, and a new technical approach is provided for the rapid measurement of the water pool of the sound source radiation sound power level.

A method performed by an audio renderer. The method includes obtaining metadata for an extended reality scene and obtaining from the metadata, or deriving from the metadata, a first reverberation parameter. The first reverberation parameter is a reverberation time parameter, an acoustical absorption parameter, or a reverberation level parameter. The method further includes, after obtaining the first reverberation parameter from the metadata or deriving the first reverberation parameter from the metadata, using the first reverberation parameter to derive a reflection parameter. The method further includes using the reflection parameter to render audio for a listener.

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.

The invention discloses a sound source localization method and device based on an improved generalized cross-correlation algorithm and a medium, and belongs to the field of sound source localization, and the method comprises the steps: collecting a sound source signal through a quaternary cross sensor array; constructing an improved weighting function according to the sound source signal; wherein the improved weighting function is obtained by weighted summation of a phase transformation weighting function and a maximum likelihood weighting function; performing generalized cross-correlation operation according to the sound source signal and the improved weighting function to obtain time delay; and performing geometric operation in the three-dimensional rectangular coordinate system based on the time delay to obtain a sound source position. Therefore, the accuracy and robustness of sound source localization can be improved in a low signal-to-noise ratio environment and a strong reverberation environment by implementing the method and the device.

The invention discloses an audio environment configuration method and device, electronic equipment and a computer readable storage medium, and the method comprises the steps: collecting an audio signal in a current audio environment, and determining the actual reverberation time of the current audio environment according to the audio signal; determining a target audio environment and target reverberation time corresponding to the target audio environment; when the actual reverberation time is greater than the target reverberation time, performing a dereverberation operation on the audio signal to obtain a target audio signal based on the target audio environment; and when the actual reverberation time is smaller than the target reverberation time, performing reverberation adding operation on the audio signal to obtain a target audio signal based on the target audio environment. By applying the technical scheme provided by the invention, the consistency of the audio and the set virtual audio background environment can be effectively ensured, and the user experience is further improved.

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 provides a far-field voiceprint recognition method and device, electronic equipment and a storage medium, and the method comprises the steps: collecting sample near-field audio data, constructing a geometric acoustic propagation model containing a frequency band related reverberation model and a sound source and receiving equipment directionality model based on the data, and generating a room pulse response function; performing convolution on sample near-field voice and the pulse response function to generate simulated far-field audio data; carrying out pre-training based on the sample near-field audio and carrying out fine tuning based on the simulated far-field audio by adopting a speaker representation weight pre-retention training strategy, and constructing a far-field voiceprint recognition model; and inputting the to-be-detected far-field audio after reverberation removal and the reference near-field audio into the model, calculating the similarity and outputting an identification result. The problems of scarcity of far-field voice data, poor model cross-domain adaptability, serious signal reverberation and the like are solved, and the accuracy and robustness of a far-field voiceprint recognition system are remarkably improved.

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

A device for reducing reverberation is described, comprising a profile rail (1) extending in a longitudinal direction (L), which has two retaining legs (3) projecting from a base (2), which include recesses (5) extending section by section in the longitudinal direction (L) and opposing each other in pairs, which form receptacles (6) for inserting acoustic elements (4) and each have several retaining elements (7) for retaining the acoustic elements (4) against an insertion direction (E).In order to enable more resource-efficient manufacturing while still ensuring a secure hold of the acoustic elements (4) and thereby creating the conditions for weight savings, it is proposed that several retention elements (7) be provided for each receptacle (6), which are spaced apart from each other in groups both in the insertion direction (E) and transversely to the insertion direction (E), whereby the distance of the retention elements (7) transversely to the insertion direction (E) varies along the insertion direction (E).

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.

Examples of the present disclosure describe systems and methods for estimating acoustic properties of an environment. In the example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of the wearable head device, wherein the second audio signal is based on the second reverberation time.

The utility model belongs to the technical field of room reverberation simulation, and particularly relates to a room reverberation simulation device suitable for a wired earphone microphone, which comprises an audio processing chip, an earphone assembly, a microphone, a sound effect switch, a sound effect switching key, a power conversion circuit and a connecting assembly. The earphone assembly, the microphone, the sound effect switch and the sound effect switching key are respectively connected to the audio processing chip; the audio processing chip is connected to the host through the connecting assembly; the input end of the power conversion circuit is connected with an input power supply through the connecting assembly, and the output end of the power conversion circuit is connected to the power end of the audio processing chip. The utility model has the advantages of low power consumption, small volume and capability of meeting diversified requirements.

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.