49 results about "Auditory masking" patented technology

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 front voice enhancement method for identifying a speaker. The method, based on conventional spectral subtraction, through a non-stationary noise estimation algorithm, obtains preliminarily enhanced voice through multiband spectral subtraction to calculate an auditory masking threshold so as to adaptively adjust correlation parameters of the spectral subtraction and obtain a front processor applied to a speaker identification system. Compared to a conventional voice enhancement algorithm, the anti-noise performance of the speaker identification system can be improved, voice distortion can also be reduced to the maximum degree through parameter adjustment, voice features of the speaker is maintained, and the identification rate of the speaker identification system in a noisy environment can be effectively improved.

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 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.

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.

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 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 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 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.