471 results about "Reverberation" patented technology

A system configured to improve audio processing by performing dereverberation and noise reduction during a communication session. In some examples, the system may include a deep neural network (DNN) configured to perform speech enhancement, which is located after an Acoustic Echo Cancellation (AEC) component. For example, the DNN may process isolated audio data output by the AEC component to jointly mitigate additive noise and reverberation. In other examples, the system may include a DNN configured to perform acoustic interference cancellation, which may jointly mitigate additive noise, reverberation, and residual echo, removing the need to perform residual echo suppression processing. The DNN is configured to process complex-valued spectrograms corresponding to the isolated audio data and / or estimated echo data generated by the AEC component.

The invention provides a manifold constraint-based sonar imaging reverberation elimination method and system, and is applied to the field of imaging data processing. According to the method, the time domain waveform of the echo signal is collected in real time, whether the irregular reflection peak value is detected or not is judged, potential interference and reflection in the underwater environment can be rapidly recognized, and particularly, in the complex underwater environment, non-target objects (such as seabed obstacles or sundries) possibly cause irregular reflection signals, so that the interference and reflection in the underwater environment can be rapidly recognized. Meanwhile, the underwater propagation characteristics of sound waves can be accurately modeled by collecting the reflection path of the echo signal and predicting the propagation parameters of the reverberation signal. And accurate background data can be provided for the propagation modeling of the reverberation signal through the pre-recorded information of submarine topography, geological characteristics, sound wave propagation characteristics and the like.

The invention relates to the technical field of audio signal processing, and discloses a room sound correction method, device and equipment for audio equipment and a storage medium, which are used for improving the accuracy and correction effect of correction parameters and effectively improving the acoustic effect of the audio equipment in a room. The room sound correction method of the audio equipment comprises the following steps: acquiring response data of a room to a multi-band preset test signal, and extracting room acoustic data by adopting a time-frequency conjoint analysis method, the room acoustic data comprising frequency attenuation, reflection paths, phase shifts and reverberation time information of different bands; based on the room acoustic data, performing compensation processing by using preset microphone calibration data to obtain a calibrated frequency response curve; performing fitting calculation on a preset ideal frequency curve and the calibrated frequency response curve according to an optimization algorithm to obtain specific correction parameters of the room; and correspondingly adjusting the output of the audio equipment according to the correction parameter.

A method that includes determining a set of one or more reverberation parameters of an extended reality (XR) environment in which a user is to participate, determining whether an audio source device is wirelessly communicatively coupled to send audio to an audio output device, in response to determining that the audio source device is not wirelessly communicatively coupled to the audio output device obtaining a microphone signal produced by a microphone of the audio source device, producing a reverberant audio signal from the microphone signal according to the set of one or more reverberation parameters, and sending the reverberant audio signal to drive a speaker driver of the audio source device.

The embodiment of the invention provides an audio processing method and device, electronic equipment and a storage medium. The method comprises the following steps: determining a target pulse response of a first audio signal, wherein the target pulse response is a stereo pulse response used for simulating sound reflection and attenuation effects of the first audio signal in an acoustic space; performing convolution processing on the first audio signal and the target pulse response to obtain a second audio signal; and performing audio mixing on the first audio signal and the second audio signal to output a third audio signal. According to the scheme provided by the invention, the stereo pulse response data capable of simulating the reflection and attenuation effects of the sound in the space can be synchronously determined when reverberation is carried out on the audio signal, and reverberation is carried out on the audio signal by using the stereo pulse response data, so that the sound is heard to be generated in different space environments; the reverberation effect of the audio signal is more vivid, the sense of space and the sense of layering of the sound are increased, and the sound is fuller and stereoscopic.

An audio system is provided that includes a loudspeaker, at least one microphone, and at least one controller. The loudspeaker transmits an audio signal into a listening environment defined by at least one wall in a room. The at least one microphone is positioned on the loudspeaker and is configured to capture a reverberated audio signal including a plurality of reverberations and a plurality of peaks. The reverberated audio signal is indicative of the audio signal being reflected from the at least one wall. The at least one controller is programmed to apply a confidence score to the plurality of peaks to obtain a maximum score which is indicative of a maximum peak of the audio signal that is reflected from the at least one wall and determine a distance between the loudspeaker and the wall based at least on the maximum score.

The invention provides an audio processing method, communication equipment, a communication system and a readable storage medium, and the method comprises the steps: obtaining a to-be-processed audio signal; inputting the audio signal and a preset target attenuation coefficient into a first filter to obtain a first output audio signal with an echoing sound effect; inputting the first output audio signal and the target attenuation coefficient into a second filter to obtain a second output audio signal with a sound trailing phenomenon; respectively inputting the second output audio signal into a plurality of third filters to obtain a third output audio signal corresponding to each third filter; and superposing the plurality of third output audio signals to obtain a target output audio signal with a reverberation sound effect. According to the method, the reverberation time of each frequency band can be accurately controlled, the reverberation reality sense of the output audio is effectively improved, the reverberation effect is closer to the convolution reverberation effect, meanwhile, the requirement for the computing power is greatly reduced, the calculation amount of the algorithm is effectively reduced, and the application channel of the algorithm is effectively widened.

A system for detecting speech from reverberant signals is disclosed. The system is programmed to receive spectral temporal amplitude data in the modulation frequency domain. The system is programmed to then enhance the spectral temporal amplitude data by reducing reverberation and other noise as well as smoothing based on certain properties of the spectral temporal spectrogram associated with the spectral temporal amplitude data. Next, the system is programmed to compute various features related to the presence of speech based on the enhanced spectral temporal amplitude data and other data in the modulation frequency domain or in the (acoustic) frequency domain. The system is programmed to then determine an extent of speech present in the audio data corresponding to the received spectral temporal amplitude data based on the various features. The system can be programmed to transmit the extent of speech present to an output device.

The invention provides an audio data processing method, electronic equipment, a storage medium and a computer program product, and the method comprises the steps: extracting audio feature information from target audio data according to a feature extraction strategy corresponding to a specified classification algorithm; classifying the audio feature information according to the specified classification algorithm to obtain a music type of the target audio data; calling a sound effect enhancement strategy matched with the music type, and performing sound effect enhancement processing on the target audio data; wherein the sound effect enhancement strategy refers to an adjustment mode of an equalizer, a reverberation mode and a bass enhancement function. According to the scheme, classification and sound effect enhancement of the audio data are realized under the condition of relatively low hardware requirements and resource consumption.

The invention relates to the technical field of voice signals, in particular to a two-step mixed sound source separation and de-reverberation method, which comprises the following steps of: firstly, carrying out separation network training by taking different types of signals as training targets in various separation networks for separating attention and the like; the invention provides an improved de-reverberation method based on a time convolution network-weight prediction error, multiple improvement strategies such as taking a scale invariance signal-to-noise interference ratio as a network loss function, adopting a transposition mechanism for an input signal and a mechanism for additionally adding a residual value to a network unit are used, and finally, the de-reverberation method based on the time convolution network-weight prediction error is obtained. And cascading the separation attention network with the best separation noise reduction effect with the time convolution-transpose-residual error-WPE network. According to the invention, the problem that the separation effect of the mixed audio signal with reverberation and noise signals is not good under the actual sound field condition is solved, compared with a single one-step or two-step existing separation, de-mixing and noise reduction network, the method is significantly improved, and the quality of the separated voice can be improved for later recognition and discrimination.

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 residual network sound source localization method and system based on a fusion attention mechanism, and the method comprises the steps: adding a 1 * 1 convolution kernel, adding an extra SEAM module to form an SC-SEAM module, adding the SC-SEAM module in the last two stages of a residual network structure, forming a residual network architecture of a fusion channel attention mechanism module, and carrying out the sound source localization of the residual network. An indoor sound source positioning model is trained and generated, and the method comprises the following steps: after a microphone array obtains a to-be-positioned sound signal, STFT is carried out on the to-be-positioned sound signal received by each microphone; extracting a phase component between the STFT coefficients of the sound signals to be positioned received by each microphone, and obtaining a difference to obtain a phase difference diagram between each microphone; and inputting the phase difference image into an indoor sound source positioning model, and outputting an estimated angle value of the sound signal to be positioned. The method is high in sound source positioning accuracy and strong in robustness in indoor reverberation, noise and other multi-scene complex environments.

An apparatus, method and computer program is described comprising: obtaining a virtual audio signal (comprising virtual audio samples) from one or more virtual sound signals of a virtual environment for presentation to a user; processing the virtual audio samples based on first reverberation parameters to generate a first reverberation signal (based on a geometry of a real-world space) for samples up to a first threshold time following the respective virtual audio sample being obtained; processing the virtual audio samples based on second reverberation parameters to generate a second reverberation signal (based on a geometry of the virtual environment) for samples beyond a second threshold time following the respective virtual audio sample being obtained; and generating an audio output for presentation to the user based on a sum of the first and second reverberation signals.

Bidirectional signalling for Immersive Voice Audio Services bitstreams (eg. in different languages or from different audio scenes captured in Metadata Assisted Spatial Audio, MASA) is incorporated int

The invention discloses a room geometry inference method based on positioning of a direct sound source and a first-order reflection sound source, which comprises the following steps of: 1) acquiring reverberation signals of a target room by using a microphone array, converting the reverberation signals into FOA signals, and calculating a time-frequency covariance matrix of the FOA signals, estimating the direction of arrival DOAs of the direct sound source s and the direction of arrival DOAs'of the n first-order reflection sound sources s' according to the time-frequency covariance matrix; 2) estimating the distance ds of the direct sound source s according to the DOAs, the DOAs'and the height dh of the microphone array, and determining the position Ps of the direct sound source s; 3) estimating the distance ds'of the first-order reflection sound source s' based on the FOA signal, the DOAs, the DOAs' and ds, and determining the position Ps' of the first-order reflection sound source s'; and 4) obtaining the geometric space structure of the target room according to the position Ps of the direct sound source s and the position Ps'of each first-order reflection sound source s'.

Various embodiments of the present disclosure provide methods, apparatuses, systems, and / or devices that are configured to separate multi-source audio signal samples into discrete source separated channel audio samples using trained multi-modal audio source channelization models. Multi-source audio signal samples are difficult to separate because they can include multiple target audio sources (e.g., individual speakers) that are often inter-mixed and overlaid with other audio sources such as noise, music, reverberations, and other audio artifacts. The multi-modal audio source channelization models discussed herein are trained to generate source separated channel audio samples based on audio signal samples and on video signal samples.

The invention discloses an indoor multi-sound-source positioning method based on DOA estimation and DOA association, and the method comprises the steps: judging single-source time-frequency points through the correlation of time delay inequality vectors of adjacent time-frequency points between microphones, and constructing a variable-size elastic single-source region; applying the FSSZ distribution condition of the sound source in a reference array to all arrays by using the high correlation characteristic of the FSSZ time-frequency distribution of the same sound source at different microphone arrays in a frame; performing middle DOA estimation on the FSSZs of all the arrays in the whole observation time period by using a circular integral cross spectrum method, and constructing a sound source position histogram in combination with FSSZ aggregation degree weighting; 2D positions of the plurality of sound sources are estimated from the sound source position histogram using 2D-CFAR. According to the invention, under the conditions of indoor reverberation and noise, position estimation can be carried out on a plurality of sound sources with an unknown number, the detection performance under the conditions of positioning precision and missed detection is obviously superior to that of an existing method, and the defects of the prior art are overcome.

The invention provides a method and device for carrying out noise detection on a train, which can be applied to the field of railway and urban rail transit, and the method comprises the following steps: determining the initial noise reverberation duration and the sound source position of noise; iteratively executing the following operations to obtain the dynamic reverberation duration: determining the nth reverberation duration according to the sum of the (n-1) th noise reverberation duration and the preset duration; processing the nth noise reverberation duration and the reference reverberation duration according to tunnel shape parameters for the pipe tunnel to obtain an nth room coefficient; processing the nth room coefficient by using a target noise detection model corresponding to the sound source position to obtain an nth sound pressure level increment value of the noise; under the condition that a difference value between the Nth sound pressure level increment value and a preset standard sound pressure level increment value meets a preset convergence condition, determining the Nth noise reverberation duration as a dynamic reverberation duration; and processing the reference reverberation duration and the dynamic reverberation duration according to a reverberation noise prediction algorithm to obtain a noise value of the noise.

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.

The invention provides an ocean reverberation time domain waveform realization method and system based on convolution operation, and the method comprises the steps: generating a one-dimensional random number sequence meeting a random process, and the data size of the random sequence is the same as the data size of a reverberation time domain waveform to be realized; generating a sound source signal time domain waveform; performing convolution operation on the random number sequence and the sound source signal time domain waveform to obtain a stabilized time domain waveform signal subjected to reverberation intensity stabilization processing; generating an ocean reverberation intensity attenuation curve according to the reverberation intensity theoretical model; and performing intensity weighting on the stabilized time-domain waveform signal subjected to reverberation intensity stabilization processing by using the attenuation curve to obtain a reverberation time-domain waveform. The method has the advantages that cross superposition of scattering echoes of different discrete points does not need to be considered; the calculation speed is high, the requirement for the data cache capacity is small, and the amplitude and the phase of each discrete point echo and the broadening of a transmitted signal do not need to be calculated; the method is suitable for realizing reverberation time domain waveforms in various waveguide environments.

A 3D neural radiance field (NeRF) is used to generate an impulse response (IR) characterization, which can then be input to a convolutional reverberation engine to create an audio experience that reflects the in-game world on a 2.0 stereo speaker system. The NeRF recreates a background geometry and the impulse response of a virtual 3D space generated using NeRF is input to the convolutional reverberation engine to transform game sounds / music to appear as though they are occurring inside the 3D space of the game. The same may be done for the player's real-world space in which the virtual IR and real IR are blended together in real-time, and the real-life player tracked as he moves around the room to create audio as it would sound were the player moving through the virtual space while adjusting for the acoustics of the real space.

The embodiment of the invention provides an audio signal processing method and device, and relates to the technical field of audio signal processing. The method comprises the following steps: acquiring a first audio signal, wherein the first audio signal is an audio signal corresponding to a first sampling point; acquiring a delay signal corresponding to the first audio signal; performing delay processing on a reverberation signal corresponding to the second audio signal to obtain a delay feedback signal corresponding to the first sampling point; the second audio signal is an audio signal corresponding to a second sampling point, and the second sampling point is a sampling point before the first sampling point; and generating a reverberation signal corresponding to the first audio signal according to the delay signal corresponding to the first audio signal and the delay feedback signal corresponding to the first audio signal. The embodiment of the invention is used for improving the reverberation sound effect of sound.

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.

An audio reverberation method and system in the technical field of audio reverberation. The method comprises: performing pre-processing on an input audio signal to obtain a reverberation input signal (S110); performing reverberation processing on the reverberation input signal to generate an initial reverberation audio signal of a target scenario (S120); analyzing audio content of the input audio signal to obtain an audio content feature of the input audio signal (S130); determining a content adaptive masking matrix on the basis of the audio content feature (S140); performing weighted mixing on the content adaptive masking matrix and the initial reverberation audio signal to obtain a content adaptive reverberation signal (S150); and performing weighted mixing on the content adaptive reverberation signal and the reverberation input signal according to a preset proportion, to obtain a final reverberation audio signal (S160). Therefore, different reverberation effects are adapted for audio signals having different audio content, thereby avoiding the problem of distortion of a final reverberation audio signal.

Audio signal processor for generating a two-channel audio signal has: an input interface for providing single-channel acoustic data describing an acoustic environment; a two-channel synthesizer for synthesizing two-channel acoustic data from the single-channel acoustic data using a listener position or rotation; and a sound generator for generating the two-channel audio signal from an audio signal and the two-channel acoustic data, wherein the two-channel synthesizer is configured to separate the single-channel acoustic data into at least two parts consisting of a direct sound part and at least one of an early reflection part and a late reverberation part, and to individually process the at least two parts for generating two-channel acoustic data for each part, and wherein the two-channel synthesizer is configured to calculate the two-channel acoustic data for the early reflection part using a specular part describing distinct early reflections and a diffuse part describing a diffuse influence in the early reflection part.