390 results about "Reverberation" patented technology

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.

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.

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.

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.

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.

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.

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.

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.

The invention relates to the technical field of audio processing, and discloses an intelligent guitar with a karaoke singing function. Comprising the following steps: detecting singing sound of a user in real time, synchronously regulating and controlling audio signal path parameters between a built-in microphone and a loudspeaker, and eliminating delay and synchronization errors caused by transmission; the refresh rate of the display screen is automatically adjusted according to the working frequency of the lyric display module, so that the lyric display time is strictly synchronized with the rolling of the music progress bar to eliminate the display delay; performing multi-stage filtering processing on the acquired audio data sung by the user, and comparing the audio data with a preset standard intonation model to provide high-precision sound acquisition and accurate intonation judgment; and the optimal sound proportion and reverberation effect are ensured to be achieved under various playing conditions. According to the intelligent guitar with the karaoke singing function, real-time and accurate audio playing can be realized, the audio-visual experience of a user in scenes of listening to music, watching videos and the like is improved, and discomfort caused by asynchronism of sound and pictures is avoided.

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 an automatic debugging method, device and equipment for a multimedia sound console and a medium, and the method comprises the steps: obtaining sound field space data, matching a pre-cached basic cloud template according to the sound field space data, obtaining an input audio signal of multiple sound channels, and transmitting the input audio signal to the multimedia sound console; determining a spectrum characteristic and a dynamic range characteristic of the input audio signal; carrying out equalization analysis on the spectrum features, and determining a corresponding equalization parameter set; determining a corresponding gain parameter set according to the dynamic range characteristics; according to the reverberation parameter, the equalization parameter set, the gain parameter set and the filtering parameter set, a corresponding local cache template is established, a cloud template matching technology based on sound field space data is introduced, the limitation that traditional tuning depends on fixed parameters or manual adjustment is overcome, the automation degree of tuning is improved, manual intervention is reduced, and the tuning efficiency is improved. And meanwhile, the stability of audio output quality is ensured.

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.

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 synthesizer (200) for synthesizing two-channel acoustic data from the single-channel acoustic data using the listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from the audio signal and the two-channel acoustic data, where the two-channel synthesizer (200) is configured to separate (210) 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 (220, 230, 230) the two parts. 240) at least two portions to generate two-channel acoustic data for each portion, and wherein the two-channel synthesizer (200) is configured to calculate (230) the two-channel acoustic data for the early reflection portion using a specular portion describing different early reflections and a diffuse reflection portion describing diffuse reflection effects in the early reflection portion.

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 audio device comprises a first receiver (501) that receives audio data of a source comprising a scene of an acoustic environment divided by acoustic attenuation boundaries. A second receiver (501) receives metadata including a set of reverberation parameters indicative of reverberation levels generated by audio sources in the acoustic environments and an energy transfer parameter indicative of energy attenuation between the acoustic environments. For a listening acoustic environment, a selector (513) selects a first set of reverberation parameters in response to a parameter value of the first set of reverberation parameters and an energy transfer parameter indicative of an attenuation of the acoustic environment and the listening acoustic environment from the first set of reverberation parameters. A reverberator of the plurality of reverberators (509, 511) generates a reverberated audio signal based on the first set of reverberation parameters, and a renderer (507) renders an audio signal for listening to the acoustic environment, the audio signal comprising components derived from the reverberated audio signal.