50 results about "Auditory masking" patented technology

The embedded audio coder (EAC) is a fully scalable psychoacoustic audio coder which uses a novel perceptual audio coding approach termed “implicit auditory masking” which is intermixed with a scalable entropy coding process. When encoding and decoding an audio file using the EAC, auditory masking thresholds are not sent to a decoder. Instead, the masking thresholds are automatically derived from already coded coefficients. Furthermore, in one embodiment, rather than quantizing the audio coefficients according to the auditory masking thresholds, the masking thresholds are used to control the order that the coefficients are encoded. In particular, in this embodiment, during the scalable coding, larger audio coefficients are encoded first, as the larger components are the coefficients that contribute most to the audio energy level and lead to a higher auditory masking threshold.

The invention relates to sparse parallel signal coding using a neural network which parameters are adaptively determined in dependence on a pre-determined signal shaping characteristic. A signal is provides to a neural network encoder implementing a locally competitive algorithm for sparsely representing the signal. A plurality of interconnected nodes receive projections of the input signal, and each node generates an output once an internal potential thereof exceeds a node-dependent threshold value. The node-dependent threshold value for each of the nodes is set based upon the pre-determined shaping characteristic. In one embodiment, the invention enables to incorporate perceptual auditory masking in the sparse parallel coding of audio signals.

A method for compensating frequency response of digital deaf-aid based on masked curve includes techniques of time frequency domain switch-over, critical band division, masking threshold calculation and frequency response compensation to improve hearing threshold rising phenomenon caused by hearing-masking effect.

The invention discloses a microphone array postfiltering sound enhancement method based on multi-models and hearing characteristic, aiming at two important factors influencing the postfiltering sound enhancement performance of a microphone array, i.e. accurate estimation for signal parameters and suitable compromise between increasing noise reduction performance and reducing voice distortion. Thescheme of the invention comprises the following steps of carrying out time domain alignment on signals collected by the microphone array, and carrying out short-time Fourier transform and characteristic value analysis based of power spectrum; determining the dimensionality of a signal subspace through the existence probability of target voice signal in maximation noise-carried voice signals; self-adaptively selecting a distribution model of a noise power spectrum in the noise-carried voice signals; estimating noise power spectrum by utilizing a conditional probability; estimating an auditory masking threshold value based on the signal subspace; and estimating a postfilter by combining Lagrange multipliers according to the auditory sensing characteristics.

The invention provides a speech enhancement method that can restrain noise component in noisy voice and improve voice quality and intelligibility in noise environment. According to the method of the present invention, it specifically includes the following steps: 1) first, estimating the noise spectrum of the speech using endpoint detection technology; 2) removing noise using two-step Wiener filter technology; 3) meanwhile, adjusting the filter parameters according to auditory masking curve. The de-noising method in the invention has undertaken a thorough study and consideration on the signal level and perceptual level. For the estimation of noise, it uses the endpoint detection method, which has good robustness, to accurate estimate the noise spectrum as far as possible. Furthermore, it uses a two-step Wiener filter to solve this problem. Meanwhile, taking into account people's ears, by using auditory masking curve, it decreases the signal distortion. Thus, the eventually enhanced signal has a big increase on the voice quality and intelligibility.

The invention discloses a speech signal time delay estimation method and system used for echo cancellation. The method comprises the following steps of respectively obtaining a far-end signal and a near-end signal of a voice signal received by a microphone in a call and generating a corresponding far-end frequency domain signal and a corresponding near-end frequency domain signal; adopting a critical band based on the auditory masking effect to respectively carry out sub-band decomposition in the frequency band on the far-end frequency domain signal and the near-end frequency domain signal and obtaining a power spectrum of each critical frequency point of the far-end frequency domain signal and a power spectrum of each critical frequency point of the near-end frequency domain signal; respectively extracting local binary features of the power spectrum of each critical frequency point of the far-end frequency domain signal and the power spectrum of each critical frequency point of the near-end frequency domain signal, respectively matching and generating and outputting a time delay estimation result. According to the method and the system, the correlation between the far-end signal and the near-end signal is counted based on the auditory masking effect, the computation is simple, and the accurate time delay estimation can be obtained to bring great convenience for echo cancellation.

The invention relates to a speech enhancement method based on a multi-head self-attention mechanism and relates to the technical field of speech enhancement. According to the method, for the problem that noises cannot be clearly suppressed in an attention computing process through adoption of the speech enhancement method based on an attention mechanism, on the basis of research and utilization ofmasking effect existing in an auditory perception process of people, the invention provides the speech enhancement method based on the multi-head self-attention mechanism. According to the method, the part of suppressing noises in the attention mechanism computing process is realized, and speech enhancement performance is improved.

The invention discloses a security alert method and system in an external auditory masking state. The method comprises the following steps: A, collecting external audio information of an environment where a mobile device is located currently; B, carrying out match in a vehicle audio matching database according to the external audio information to obtain a corresponding vehicle model number, and carrying out detection according to the vehicle model number and the external audio information to obtain an impatience degree and a strength value; C, carrying out match in a matching rule database of audio perceptual intensity and distance values by using parameters of the vehicle model number, the impatience degree and the strength value to obtain a danger level; and D, providing corresponding security alert for a user according to the danger level.

An encoder / decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

The invention provides a self-adaptive audio blind watermark method based on an auditory model. Arnold transformation, dimension reduction processing and chaotic transformation are performed on a watermark signal so that watermark information is obtained; a human ear auditory masking threshold is calculated; a watermark embedding segment and an embedding position are self-adaptively selected; embedding of a digital watermark is performed; audio segment recombination is performed, and all the audio segments without embedding of the watermark and the audio segments with embedding of the watermark are recombined so that audio signals containing all the watermarks are formed; and the digital watermark is extracted. The watermark information of large energy is embedded under the premise that the watermark signal does not influence the audio quality so that self-adaptive selection of the watermark embedding segment, the watermark embedding position and the watermark embedding strength can be realized, the imperceptibility of the algorithm can be guaranteed and the watermark robustness and the watermark embedding capacity can be enhanced. The human ear auditory characteristic can be utilized in a more detailed way so that the method has better hiding performance, the security of the watermark can be further guaranteed and the attack of common signal processing can be effectively resisted.

The invention discloses an echo delay estimating and tracking method. According to the method, the echo delay is estimated based on embedding / detection of marking tone in an SVM classifier, thereby simplifying the audio feature extraction process, meanwhile the position and the amplitude of the embedded marking tone are controlled by using a psychoacoustic model based on human ear auditory masking effect so as to avoid auditory distortion after embedding and maximally guarantee the integrity and accuracy of original remote signals, in addition, the embedding program and the detection program of the marking tone are alternately executed, for each embedded and detected marking tone, range verification is executed, and the echo delay is updated according to the detection result, thereby realizing dynamic tracking of the echo delay, compared with the traditional cross-correlation algorithm, the echo delay estimating and tracking method has the advantages of high real-time performance, strong robustness and low computational complexity, and the method does not depend on the background noise or the independence assumption of the remote signal, and has higher reliability.

A priming signal processing technique modifies digital audio recordings to reduce the stress experienced by a listener's auditory system. A priming signal may reduce the instantaneous stress experienced by the auditory system during sudden changes in signal energy. The priming signal may leverage the temporal auditory masking, such that pre-signal priming additions may not result in obvious differences in perceived sounds.

The invention discloses a car driver fatigue alleviating and waking system and method based on a masking effect. The system includes a fatigue detection module, a masking sound memory module, a masking sound play and severe-fatigue waking module and an automatic control module; the method includes a system parameter test method and a fatigue grade judgment method, wherein the system parameter test method includes the steps of testing lanes and setting test status, and the fatigue grade judgment method includes the steps of using a single-chip microcomputer to conduct a comprehensive analysis by combining a real-time speed according to the weight of each set sensor test item, comparing a set threshold value through the system parameter test method to determine the current fatigue grade of a driver, and controlling the play of a masking sound for severe fatigue waking. The system can automatically play the masking sound to alleviate mild fatigue or give a severe fatigue warning according to the driver's fatigue grade. The method can set the parameters of the driver individually to make the fatigue grade more accurate.

The invention discloses a method for perceptual weighted filtering, which comprises: A. to make a spectrum slant filtering processing to the inputted speech or audio signal. B. to make a traditional perceptual weighted filtering on the output speech or audio signal processed by the spectrum slant filtering. C. to output the speech or audio signal processed by the traditional perceptual weighted filtering and takes the partial coefficient of the coefficient in transfer function used in the perceptual weighted filtering as the coefficient in transfer function used in the said perceptual weighted filtering. Besides, the invention also discloses a perceptual weighted filter designed according to the said method for perceptual weighted filtering. The invention is to achieve the purpose of simulating the formant structure of the inputted speech or audio signal spectrum of the broadband to improve the coding efficiency and enhance the subjective auditory effect by taking full advantages of the human auditory masking effect.

The invention relates to a sound field generator and a method of generating a sound field using the same. More particularly, the invention relates to a sound field generator and a method of generating the same, which can apply a filter in consideration of a masking effect in a time domain to a room impulse response, remove inaudible data depending on a frequency in a signal obtained by multiplying the room impulse response by an input signal in a frequency domain, and remove a signal block having a lower level than a level of a background noise block among output signal blocks to considerably reduce computational complexity required for performing a convolution, making it possible to generate an accurate sound field by minimizing sound quality distortion while implementing a real-time sound field generating system.

A priming signal processing technique modifies digital audio recordings to reduce the stress experienced by a listener's auditory system. A priming signal may reduce the instantaneous stress experienced by the auditory system during sudden changes in signal energy. The priming signal may leverage the temporal auditory masking, such that pre-signal priming additions may not result in obvious differences in perceived sounds.

A technique for auditory masking for a coherence-controlled calibration system includes generating a first auditory masking pattern based on a frequency spectrum of at least a first frame of a first audio signal. The technique continues by generating a first calibration signal based on the first auditory masking pattern. The technique then includes producing a combined input signal based on the first audio signal and the first calibration signal. The technique also includes performing one or more calibration operations based on the combined input signal.

The invention provides a microphone array speech denoising enhancement method with a low signal-to-noise ratio effect. The microphone array speech enhancement algorithm based on improved perfect binary time-frequency masking of human ears solves the problem of poor performance of a microphone array speech enhancement algorithm in the prior art under the condition of low signal-to-noise ratio, themasking effect of the human ears is skillfully added to the microphone array speech denoising enhancement algorithm. Firstly, a primary-order acquisition enhanced voice signal is acquired, then a pre-estimation noise signal is judged according to a noisy signal and the primary-order acquisition enhanced voice signal, a threshold value is set through a signal-to-noise ratio relationship between primary-order voice acquisition enhancement and judged pre-estimation noise, a perfect binary time-frequency masking algorithm is calculated, and the enhanced voice signal is synthesized. The speech signal enhanced by the algorithm is higher in intelligibility and better in sound quality, and the performance advantage is more obvious under the scene of larger noise intensity and low signal-to-noise ratio.

The invention provides a multi-screen interaction method, device and system based on a cloaking communication technology. Based on auditory masking, interaction information, such as vote, forecast and comment, is integrated into the voice of a program without influencing broadcasting quality of the program, multiple types of multi-media file systems can be generated for being played by television, Internet and various kinds of broadcast devices, audiences can receive the program voice through corresponding intelligent terminals and extract the interaction information, namely can conveniently participate in the interaction.

The embodiment of the invention provides a method and a device for generating quantification control parameter and an audio coding device. The method concretely comprises: firstly, calculating the energy of a frequency domain signal; according to the energy of the frequency domain signal, calculating the total masking value of at least one frequency point of the frequency domain signal; then, according to the obtained total masking value of the frequency, obtaining a quantification control parameter used for controlling quantization steps. Thus, the invention only needs to calculate the masking effect of one frequency point instead of calculating frequency domain signal noise and pitch masking characteristics, so that the invention reduces the calculated amount and the complexity of a psychoacoustic model under the condition of ensuring tone quality and lowers the working burden of the audio coding device.

The invention discloses a speech enhancement algorithm based on a speech existence probability and an auditory masking effect, and the algorithm comprises the steps: carrying out the preprocessing of an input time-domain speech signal, obtaining a frequency-domain speech signal, and keeping a phase angle; carrying out voice existence probability calculation on the obtained frequency domain signal, and obtaining an estimated noise power spectrum; performing noise masking threshold calculation on the obtained frequency domain signal to obtain a spectral subtraction coefficient value of each frequency point; and performing spectral subtraction in combination with the estimated noise power spectrum and the spectral subtraction coefficient to obtain a pure voice spectrum, and then performing inverse Fourier transform in combination with the reserved phase angle to obtain a pure time domain voice signal. According to the invention, the hearing masking effect of the human ear is utilized, the masking threshold value of the noise signal entering the human ear is calculated, noise estimation is combined, the noise can be eliminated, meanwhile, the perception quality of the voice can be guaranteed as much as possible, and more suddenly-changed peak values are not prone to occurring in the voice signal.

The invention relates to a robust speech recognition method and system based on speech enhancement and improved PNSC. Speech signals are obtained through a microphone, speech enhancement and feature extraction are carried out on the speech signals after the speech signals are preprocessed, and in a signal space, enhancement is carried out by means of a speech enhancement algorithm combining the auditory masking effect and a priori signal-to-noise ratio; in a feature space, feature values of the speech signals are extracted more accurately by means of a feature extraction algorithm improving perception non-uniform spectral compression, the speech signals are trained and recognized through a dynamic time wrapping algorithm, a speech recognition result is converted into a bottom layer drive instruction of an intelligent wheelchair, a corresponding control function is called, and the intelligent wheelchair is driven to move according to the speech signals. By means of the method and system, the recognition rate of the system in a noise environment is increased, precise control over the intelligent wheelchair is achieved, the aim of achieving speech interaction between a user and the intelligent wheelchair is achieved, and the method has good universality in a specific human speech recognition system.

The invention provides an array type auditory masking device, comprising: the main control platform (10) of an accessing interface platform (20) and a plurality of user platforms (30). In the main control platform, a storage device (14) outputs voice interfering signals based on the language choice instruction transmitted by an operation unit (12). A first signal processing unit (16) arbitrarily combines the voice interfering signals and generates a first masking signal. A control unit starts the user platforms according to the user platform choosing instruction outputted by the operation unit. Each user platform comprises a voice acquiring unit (34), a second signal processing unit (36), a sound receiving unit (32) and an audio playing unit (38). The second signal processing unit is in signal connection with the other devices of the user platform as well as the interface platform and segments the voice signals outputted by the voice acquiring unit and arbitrarily combines them into a second masking signal. Based on the first masking signal and the second masking signal, the audio playing unit plays the first masking sound and the second masking sound.

The invention is suitable for the field of noise processing technology improvement, and provides a method for evaluating transmission squeal based on a masking effect, and the method comprises the steps: S1, carrying out the data collection of the squeal working conditions of a plurality of different vehicle types; S2, arranging the collected data and designing fine whistling sound with different orders at different rotating speeds; S3, subjectively evaluating the sound samples and extracting sound quality parameters; S4, screening variables through Pearson correlation analysis of subjective scores and objective parameters; S5, performing Z-score standardization processing on the screened subjective and objective parameters, and further fusing the objective parameters through principal component analysis (PCA); S6, after fusion is completed, carrying out SVM optimization to form an evaluation system of transmission squeal; and S7, after the evaluation system is formed, inputting the values of the three objective psychological parameters obtained by testing to obtain a subjective evaluation score, and judging the severity of the howling according to the subjective evaluation score. Through the three parameters, the prominence degree of howling in one critical frequency band, the prominence degree of howling sound in two adjacent critical frequency bands and the masking effect of slope excitation influence of loudness are considered.

The invention relates to a sound field expansion method, computer equipment and a computer readable storage medium, and the method comprises the following steps: obtaining at least two sound channel signals, wherein at least two sound channel signals comprise a first sound channel signal and a second sound channel signal; obtaining a first gain signal and a third gain signal corresponding to the first sound channel signal, and a second gain signal corresponding to the second sound channel signal; obtaining a first loudness compensation signal according to the first gain signal and the second gain signal; and obtaining an output signal of the first sound channel signal according to the third gain signal and the first loudness compensation signal. Through loudness compensation, strong soundis removed, weak sound which cannot be perceived by human ears due to the masking effect is amplified and enhanced and is perceived by the human ears, the three-dimensional sound effect is improved, and a real sound field is restored and obtained.

The application discloses an echo cancellation method and device. The method comprises the following steps: acquiring an initial multi-channel audio signal to be subjected to echo cancellation; carrying out linear elimination processing on the obtained initial multi-channel audio signal to obtain a processed first residual multi-channel audio signal; carrying out nonlinear suppression processing on the processed first residual multi-channel audio signal by utilizing the masking effect of sound to obtain a second residual multi-channel audio signal. Therefore, the echo cancellation effect can be enhanced, the multi-channel audio signal is high in definition, and the effect of reducing scene echoes is achieved.

The invention provides a speech enhancement method that can restrain noise component in noisy voice and improve voice quality and intelligibility in noise environment. According to the method of the present invention, it specifically includes the following steps: 1) first, estimating the noise spectrum of the speech using endpoint detection technology; 2) removing noise using two-step Wiener filter technology; 3) meanwhile, adjusting the filter parameters according to auditory masking curve. The de-noising method in the invention has undertaken a thorough study and consideration on the signal level and perceptual level. For the estimation of noise, it uses the endpoint detection method, which has good robustness, to accurate estimate the noise spectrum as far as possible. Furthermore, it uses a two-step Wiener filter to solve this problem. Meanwhile, taking into account people's ears, by using auditory masking curve, it decreases the signal distortion. Thus, the eventually enhanced signal has a big increase on the voice quality and intelligibility.

The invention is suitable for the field of noise processing technology improvement, and provides a method for evaluating road noise based on a masking effect. The method comprises the steps: S1, collecting data of howling working conditions of a plurality of vehicles of different types; S2, arranging howling working condition data acquired by a real vehicle, and adding a plurality of virtual samples; S3, subjectively evaluating all sound samples and objectively extracting sound quality objective parameters; S4, obtaining objective parameters with high relevancy with subjective evaluation through correlation analysis; and S5, judging whether the obtained objective parameters meet requirements or not, if YES, establishing a function relation between the subjective score and the objective parameters by using multiple linear regression, calculating the subjective score by inputting the objective parameters to obtain a sound quality evaluation system of the road noise, and if NO, rejecting the data collected this time. Bark1 and total loudness parameters are extracted from the road noise, subjective scores of corresponding samples can be obtained through a calculation formula of subjective and objective scores, and a standard system for evaluating the road noise is formed.

The invention provides a method and device for transmitting hidden information by using music. The method uses the masking effect of human hearing to hide information in music for transmission, which can make the transmitted audio more pleasing to the ear and has higher security . The present invention also provides a device for realizing the above method at the same time, including two parts: a sending end and a receiving end, wherein the sending end is composed of a sending end memory, an input module, a modulator, a sound mixing module and a playback module, and the receiving end is composed of a receiving module, The synchronous module, the demodulator and the memory of the receiving end are jointly formed. The invention can be used in occasions such as mobile data sharing, initial configuration of wireless devices, control of intelligent toys and the like.