124 results about "Masking threshold" patented technology

An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

The apparatus for determining an impulse response in an environment in which a speaker and a microphone are placed works using an audio signal. Means for spectrally coloring a test signal, which preferably is a pseudonoise signal, works using a psychoacoustic masking threshold of the audio signal to obtain a colored test signal, which is embedded in the audio signal to obtain a measuring signal, which can be fed to the speaker. Means for determining the impulse response preferably performs a cross-correlation of a reaction signal received via the microphone from the environment and the test signal or the colored test signal. With this, an impulse response of an environment may also be determined during the presentation of an audio piece to provide an optimal description of environment for a wave-field synthesis.

An method and apparatus to extract an audio signal having an important spectral component (ISC) and a low bit-rate audio signal coding/decoding method using the method and apparatus to extract the ISC. The method of extracting the ISC includes calculating perceptual importance including an SMR (signal-to-mark ratio) value of transformed spectral audio signals by using a psychoacoustic model, selecting spectral signals having a masking threshold value smaller than that of the spectral audio signals using the SMR value as first ISCs, and extracting a spectral peak from the audio signals selected as the ISCs according to a predetermined weighting factor to select second ISCs. Accordingly, the perceptual important spectral components can be efficiently coded so as to obtain high sound quality at a low bit-rate. In addition, it is possible to extract the perceptual important spectral component by using the psychoacoustic model, to perform coding without phase information, and to efficiently represent a spectral signal at a low bit-rate. In addition, the methods and apparatus can be employed in all the applications requiring a low bit-rate audio coding scheme and in a next generation audio scheme.

Prior to embedding a watermark in an audio signal, a spectral representation of the audio signal and a spectral representation of the watermark signal are determined. The spectral representation of the watermark signal is then processed on the basis of a psychoacoustic masking threshold of the audio signal. The processed watermark signal is combined with the audio signal to obtain an audio signal bearing a watermark. The spectral representation of the watermark signal is processed iteratively as follows: first a predetermined watermark initial value is selected, then the interference introduced into the spectral representation of the audio signal after a quantization of the spectral representation of the audio signal is determined and then, if the interference introduced by the watermark initial value exceeds the predetermined interference threshold, the watermark initial value is modified progressively until the resulting interference introduced into the spectral representation of the audio signal after quantization is less than or equal to the predetermined interference threshold. The modified watermark initial value at the end of the iteration is used as the processed watermark signal to be combined with the audio signal. As a result it is no longer possible for a watermark to be quantized out. Instead, full control over the energy of the watermark is achieved. A watermark can therefore be embedded in an audio signal to provide either the best possible degree of watermark detectability or the best possible audio quality.

The invention discloses a voice enhancing method based on a multiresolution auditory cepstrum system and a deep convolutional neural network. The voice enhancing method comprises the following steps:firstly, establishing new characteristic parameters, namely multiresolution auditory cepstrum coefficient (MR-GFCC), capable of distinguishing voice from noise; secondly, establishing a self-adaptivemasking threshold on based on ideal soft masking (IRM) and ideal binary masking (IBM) according to noise variations; further training an established seven-layer neural network by using new extracted characteristic parameters and first/second derivatives thereof and the self-adaptive masking threshold as input and output of the deep convolutional neural network (DCNN); and finally enhancing noise-containing voice by using the self-adaptive masking threshold estimated by the DCNN. By adopting the method, the working mechanism of human ears is sufficiently utilized, voice characteristic parameters simulating a human ear auditory physiological model are disposed, and not only is a relatively great deal of voice information maintained, but also the extraction process is simple and feasible.

A method for processing an audio signal is disclosed. The method for processing an audio signal includes frequency-transforming an audio signal to generate a frequency-spectrum, deciding a weighting per band corresponding to energy per band using the frequency spectrum, receiving a masking threshold based on a psychoacoustic model, applying the weighting to the masking threshold to generate a modified masking threshold, and quantizing the audio signal using the modified masking threshold.

Interfering signals that may be present in a listening environment are masked by reproducing a desired signal in a listening environment, determining a masking threshold associated with the desired signal, identifying an interfering signal that may be present in the environment, comparing the interfering signal to the masking threshold, and adjusting the desired signal over time to raise its masking threshold above the level of the interfering signal.

The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, preferably as the area below the square of the magnitude of the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

An encoding device which encodes audio signals comprises a spectrum power calculation unit for calculating the power of each spectrum obtained by analyzing the frequency of an input audio signal, a tonality parameter calculation unit for calculating a tonality parameter indicating the pure tone level of the input audio signal in each sub-band, using the result of the calculation when dividing the frequency range of the spectrum of the input audio signal into a plurality of sub-bands, and a dynamic masking threshold calculation unit for calculating a dynamic masking threshold value of the masking energy of the input audio signal, using the calculated tonality parameter.

The embodiment of the invention discloses an audio bit rate control method and a system thereof. The method comprises the steps as follows: a sub band for initial coding is determined according to an available bit number; an initial value of the active proportion factor of the sub band is calculated according to the masking threshold value of the sub band; the active proportion factor is adjusted according to the initial value of the active proportion factor of the sub band so as to cause the quantization noise of a sub band frequency-domain coefficient to be smaller than the masking threshold value; the sub band is quantized according to the active proportion factor; coding is conducted to the frequency-domain coefficient after quantization and the bit number needed by the coding is determined; when the bit number needed by the coding is larger than the available bit number, the bit number needed by the coding can be adjusted by adjusting the active proportion factor step by step so as to cause the bit number for coding to be smaller than the available bit number. The method and the system thereof can keep good sound quality on the basis of lowering the algorithm complexity.

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 relates to the coding/decoding of a signal into several sub-bands, in which at least a first and a second sub-bands which are adjacent are transform coded (601, 602). In particular, in order to apply a perceptual weighting, in the transformed domain, to at least the second sub-band, the method comprises:—determining at least one frequency masking threshold (606) to be applied on the second sub-band; and normalizing said masking threshold in order to provide a spectral continuity between the above-mentioned first and second sub-bands. An advantageous application of the invention involves a perceptual weighting of the high-frequency band in the TDAC transform coding of a hierarchical encoder according to standard G.729.1.

A method for generating a psychoacoustic model from an audio signal transforms a block of samples of an audio signal into a frequency spectrum comprising frequency components. From this frequency spectrum, it derives group masking energies. These group masking energies each correspond to a group of neighboring frequency components in the frequency spectrum. For a group of frequency components, the method allocates the group masking energy to the frequency components in the group in proportion to energy of the frequency components within the group to provide adapted mask energies for the frequency components within the group, the adapted mask energies providing masking thresholds for the psychoacoustic model of the audio signal.

For improving the directionality of a hearing aid having a microphone system formed by two or more microphones without in the process creating an increase in the microphone noise that the hearing aid wearer finds to be disturbing, the microphone system is adjusted statically or adaptively taking into account the individual rest hearing threshold and/or taking into account the individual masking threshold for the microphone noise that is produced by the microphone system. The greatest possible extent of directionality thus can be allowed, without the hearing aid wearer in the process finding the microphone noise that is produced by the microphone system to be disturbing.

A method, system, apparatus and computer program product are provided for improved stereo coding. In particular, the method, system, apparatus and computer program product provide a technique for performing Mid-Side (M/S) stereo coding, in which an additional step is added to the coding process, whereby a parameter that is used in determining when the mid and side signals will be used instead of the left and right input signals is modified prior to making the selection between the signal pairs. In particular, the masking threshold associated with either the left or the right input signal may be modified based on a relationship between the energies of the two input signals. In addition, once the selection between the signal pairs has been made, the masking thresholds of the selected signals may be further modified, again based on a relationship between the energies of the left and right input signals.

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.

A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.