40 results about "Human auditory system" patented technology

Intelligibility of a human voice projected by a loudspeaker in an environment of high ambient noise is enhanced by processing a voice signal in accordance with the frequency response characteristics of the human hearing system. Intelligibility of the human voice is derived largely from the pattern of frequency distribution of voice sounds, such as formants, as perceived by the human hearing system. Intelligibility of speech in a voice signal is enhanced by filtering and expanding the voice signal with a transfer function that approximates an inverse of equal loudness contours for tones in a frontal sound field for humans of average hearing acuity.

An efficient audio coding device that quantizes and encodes digital audio signals with a reduced amount of computation. A spatial transform unit subjects samples of a given audio signal to a spatial transform, thus obtaining transform coefficients of the signal. With a representative value selected out of the transform coefficients of each subband, a quantization step size calculator estimates quantization noise and calculates, in an approximative way, a quantization step size of each subband from the estimated quantization noise, as well as from a masking power threshold determined from a psycho-acoustic model of the human auditory system. A quantizer then quantizes the transform coefficients, based on the calculated quantization step sizes, thereby producing quantized values of those coefficients. The quantization step sizes are also used by a scalefactor calculator to calculate common and individual scalefactors. A coder encodes at least one of the quantized values, common scalefactor, and individual scalefactors.

Methods and apparatus are provided for communicating an audio stream. A perceptual mask is estimated for an audio stream, based on the perceptual threshold of the human auditory system. A hidden sub-channel is dynamically allocated substantially below the estimated perceptual mask based on the characteristics of the audio stream, in which additional payload is transmitted. The additional payload can be related to components of the audio stream that would not otherwise be transmitted in a narrowband signal, or to concurrent services that can be accessed while the audio stream is being transmitted. A suitable receiver can recover the additional payload, whereas the audio stream will be virtually unaffected from a human auditory standpoint when received by a traditional receiver. A coding scheme is also provided in which a portion of a codec is used to code an upper-band portion of an audio stream, while the narrowband portion is left uncoded.

An apparatus for reducing noise includes a comparator for generating a noise timing signal corresponding to a noise producing period of noise introduced from a noise source and contained in an audio signal, a gap time generator for generating a gap period in which to remove noise from the audio signal, a selector switch for selectively outputting the audio signal and a noise-removed signal, a level detector for detecting a signal level of the audio signal, and a masking degree determining unit for determining from the signal level detected by the level detector a gap period for which the audio signal is masked by the human auditory system. The selector switch outputs the noise-removed signal in a period corresponding to the gap period within the noise producing period of the noise timing signal, and outputs the audio signal in other than the gap period.

A communication system for sending a sequence of symbols on a communication link. The system includes a transmitter for placing information indicative of the sequence of symbols on the communication link and a receiver for receiving the information placed on the communication link by the transmitter. The transmitter includes a clock for defining successive frames, each of the frames including M time intervals, where M is an integer greater than 1. A modulator modulates each of M carrier signals with a signal related to the value of one of the symbols thereby generating a modulated carrier signal corresponding to each of the carrier signals. The modulated carriers are combined into a sum signal which is transmitted on the communication link. The carrier signals include first and second carriers, the first carrier having a different bandwidth than the second carrier. In one embodiment, the modulator includes a tree-structured array of filter banks having M leaf nodes, each of the values related to the symbols forming an input to a corresponding one of the leaf nodes. Each of the nodes includes one of the filter banks. Similarly, the receiver can be constructed of a tree-structured array of sub-band filter banks for converting M time-domain samples received on the communication link to M symbol values.Signal processing is performed by splitting a signal into subbands using a plurality of filter banks connected to form a tree-structured array. The filter banks are connected so that the signal is split into subbands of different size. The subbands can be designed to approximate the bands of the human auditory system for audio signal processing applications. Reconstruction of signals using a plurality of synthesis filter banks connected to form a tree-structured array is also performed.

A method for individually fitting a hearing instrument to a user, comprising at least one microphone for generating an input audio signal from ambient sound, an audio signal processing unit for processing the input audio signal into a processed output audio signal, and a transducer for stimulation of the human auditory system according to the processed output audio signal as input to said transducer is described, the method comprising: providing the user with the hearing instrument and starting operation of the hearing instrument; pre-defining a desired target loudness function, wherein loudness perception of a stimulus by the user when using the hearing instrument is defined as function of frequency and input sound pressure level at the microphone; measuring for a given measurement parameter set of perceived loudness levels and frequencies or frequency bands the respective transducer input audio signal level to be applied to the transducer input in order to achieve the respective perceived loudness level at the respective frequency or frequency band, said measurement parameter set comprising at least a low loudness level, an intermediate loudness level and a high loudness level, and said intermediate loudness level being measured for a larger number of frequencies or frequency bands and with a finer frequency resolution than said low and high loudness levels; calculating an individual gain function to be implemented in the audio signal processing unit in order to achieve the pre-defined target loudness function by taking into account the measured transducer input audio signal levels; and operating the hearing instrument with the individual gain function.

A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs. The multi-channel signal processing system further includes peak detecting means detecting signal peaks of the first and second channel input signals for generating first and second channel control signals. Thereafter, gain adjusting means adjusts signal gains of the first and second channel control signals. The first and second channel compressing units then respectively compress the first and second channel input signals to produce the first and second channel compressed outputs in accordance with the adjusted first and second channel control signals, respectively.

The present system proposes a technique called the spectro-temporal varying technique, to compute the suppression gain. This method is motivated by the perceptual properties of human auditory system; specifically, that the human ear has higher frequency resolution in the lower frequencies band and less frequency resolution in the higher frequencies, and also that the important speech information in the high frequencies are consonants which usually have random noise spectral shape. A second property of the human auditory system is that the human ear has lower temporal resolution in the lower frequencies and higher temporal resolution in the higher frequencies. Based on that, the system uses a spectro-temporal varying method which introduces the concept of frequency-smoothing by modifying the estimation of the a posteriori SNR. In addition, the system also makes the a priori SNR time-smoothing factor depend on frequency. As a result, the present method has better performance in reducing the amount of musical noise and preserves the naturalness of speech especially in very noisy conditions than do conventional methods.

A method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.

Objective measurement methods and devices for predicting perceptual quality of speech signals degraded in speech processing/transporting systems have unreliable prediction results in cases where the degraded and reference signals show in between severe timbre differences. Improvement is achieved by applying a partial compensation step within in a signal processing stage using a frequency dependently clipped compensation factor for compensating power differences between the degraded and reference signals in the frequency domain. Preferably clipping values for clipping the compensation factor have larger frequency-dependency in a range of low frequencies with respect to a centre frequency of the human auditory system, than in a range of high frequencies.

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.

The invented method is a new digital watermark method for embedding gray image (or color image) into original digital audio production, belonging to area of information safety and multimedia information processing. first, watermark image is encoded as one dimension binary sequence, and scrambling encryption is added. Next, sectionalized treating is carried out for digital audio signal. Based on features of time domain shelter of human auditory system, integer type upgraded wavelet transform is carried out for sect of audio frequency selected in selfadaptation. Then, based on features of frequency domain, wavelet coefficient is determined in selfadaptation. Watermark information is embedded into hearing important coefficients with wavelet transformed by using quantization treating procedure. Finally, digital audio production with embedded watermark information is obtained by inversed wavelet transform and recombination of audio data sects. Inversed process can test digital watermark.

The present invention relates to a three-dimension surrounding sound effect method and equipment capable of leading double track audio signal to extend to multi-track three-dimension surrounding audio signal. The present invention establishes a novel model according to the subjective impression principle of human auditory. The hearer relative location between the two speakers can be reflected by the following three parameters: (1) center offset parameter: the hearer relative location between two speakers; (2) swing parameter: mapping the virtual direction of the hearer impression low frequency sound signal; (3) delay parameter: mapping the virtual direction of the hearer impression high frequency sound signal. The auditory system has different subjective impressions of sound directions of low frequency and high frequency, before the present invention using three-dimension surrounding sound effect method, dividing the input sound signals into high frequency sound signals and low frequency sound signals firstly, and mixing sound signals according to human auditory system subjective impression of different frequency band sound signals. The three-dimension surrounding sound effect method designed by the present invention can be easily applied and highly-effectively rebuilt double track sound three-dimension surrounding effect.

The invention discloses an audio watermarking method based on the MP3 encoding principle, which belongs to the technical field of multimedia digital watermarks. The method includes two processes, i.e. watermark embedment and watermark extraction; a watermark is embedded into a low-frequency MDCT (Modified Discrete Cosine Transform) coefficient as an encoding process is carried out synchronously; in order to enhance the robustness of the watermark, an appropriate audio segment is chosen in combination with the frequency domain masking effect in a human auditory system to be embedded; and in order to resist desynchronization attacks, a synchronization mechanism is introduced. By processing the encoding process, the method fulfills the injection of the watermark, and the method can effectively resist common audio attacks, and also has good robustness on desynchronization attacks (such as shearing attacks, time scaling and the like). While taking audio watermark robustness into consideration, the method guarantees the auditory invisibility of audio contents, the computational complexity of the algorithm is low, and the method is easy to implement.

A method for a machine or group of machines to watermark an audio signal includes receiving an audio signal and a watermark signal including multiple symbols, and inserting at least some of the multiple symbols in multiple spectral channels of the audio signal, each spectral channel corresponding to a different frequency range. Optimization of the design incorporates minimizing the human auditory system perceiving the watermark channels by taking into account perceptual time-frequency masking, pattern detection of watermarking messages, the statistics of worst case program content such as speech, and speech-like programs.

Complex acoustic information, such as music, is presented as visual information or as movement of an object in a manner simulating the reception of the complex acoustic information by the human auditory system including a complexity of tempo, rhythms, intensity variation from highs to lows, and silences of the audio, providing a synchronicity with these characteristics. The acoustic information is processed by an acoustic human-like auditory transformation. The transformation may be varied depending on the presentation controlled by the device. The transformed signal is then applied to a tactile or visual presentation. The audience reception of the invention is through light, color, or animation of an image or object complementing the reception of the acoustic information.

The invention discloses an audio processing device comprising reduced artifacts. The audio processing device comprises a forward path and an analysis path. The forward path comprises an input unit for conveying a time-varying electrical input signal representing an audio signal, a signal processing unit for applying a processing algorithm on the electrical input signal and providing a processed signal, and an output unit for transmitting an output signal on the basis of the processed signal; and the electrical input signal comprises a target signal part and a noises signal part. The analysis path comprises a model unit comprising a perceptual model of a human auditory system and providing audibility measurement, an artifact identification unit for identifying the artifacts introduced by the processing algorithm to the processed signal and providing the artifact identification measurement, and a gain control unit for controlling a gain of a signal, which is applied by the processing algorithm to the forward path, on the basis of inputs from the model unit and the artifact identification unit. The audio processing device has the advantage of dynamic optimization of noise reduction aiming at audibility of the artifacts.

An audio coding technique based on modeling spectral dynamics is disclosed. Frequency decomposition of an input audio signal is performed to obtain multiple frequency sub-bands that closely follow critical bands of human auditory system decomposition. Each sub-band is then frequency transformed and linear prediction is applied. This results in a Hilbert envelope and a Hilbert Carrier for each of the sub-bands. Because of application of linear prediction to frequency components, the technique is called Frequency Domain Linear Prediction (FDLP). The Hilbert envelope and the Hilbert Carrier are analogous to spectral envelope and excitation signals in the Time Domain Linear Prediction (TDLP) techniques. Temporal masking is applied to the FDLP sub-bands to improve the compression efficiency. Specifically, forward masking of the sub-band FDLP carrier signal can be employed to improve compression efficiency of an encoded signal.

The invention discloses an improved DCNN music genre classification method. The method comprises the steps of inputting a training set and a verification set, extracting MFCC features of the audio information, generating a frequency spectrum, carrying out frequency spectrum cutting, inputting a network model, training a model, verifying the model, judging whether a specified batch is reached or not, and outputting a model. According to the method, self-adaption of channel dimensions is achieved through a function, so that the coverage range of interaction of local area cross channels is ensured, an ECA module is more effectively integrated into an existing DCNN architecture, obvious performance gain is brought to a network model, and then the working efficiency of music genre classification is improved. Through the Mel-frequency cepstrum coefficient, the perception characteristics of a human auditory system are simulated, and the classification precision is further improved.

The invention discloses a biomechanical research, modeling, verification and application of human ear for railway vehicles to provide thoughts and basis for the design and evaluation of noise comfort of the railway transit equipment. The verification method includes establishing a plane wave source in the external ear as the acoustic excitation and applying pure timbre of a series of different frequency and sound pressure level in the external ear. Based on the acoustic modal analysis of external ear and the domain modal analysis, the ear finite element model and the sound sensitive band of the fluid domain finite element model are obtained and the matching between the sound sensitive band and the actual sensitive band of the human auditory system is compared. And based on the acoustic solid coupling method and the mode superposition method, the tympanic membrane and the acoustic response of stapes footplate are respectively obtained, and compared with the ANSI curve. The reliability of the ear finite element model and the fluid domain finite element model are determined by combining the results of the above two comparisons.

Watermarking of audio signals intends to manipulate the audio signal in a way that the changes in the audio content cannot be recognised by the human auditory system. The watermark data are decoded from the received watermarked audio signal by correlation with corresponding candidate reference sequences. One or more of the sync symbols (S1, S2, S3) are embedded twice in the watermark data frame in the encoder. Thereafter a circular correlation is calculated instead of a standard correlation.

A method for raising the soft and mid-level amplitude of sounds for greater clarity and perceptual benefit, while simultaneously removing the high level amplitude peaks without delay and providing protection for the auditory sense organ. The method does not require a feedback mechanism for the accomplishment of this treatment and exploits the psychoacoustic phenomenon of temporal integration which reduces the audibility of short duration signals, including distortions associated with peak clipping. The human auditory system requires greater time to integrate signal energy for audibility than provided by brief duration waveform peaks.

A method and implementation for detecting and characterizing audible transients in noise includes placing a microphone in a desired location, producing a microphone signal wherein the microphone signal is indicative of the acoustic environment, processing the microphone signal to estimate the acoustic activity that takes place in the human auditory system in response to the acoustic environment, producing an excitation signal indicative of the estimated acoustic activity, processing the excitation signal to identify each impulsive sound frequency-dependent activity as a function of time, producing a detection signal indicative of audible impulse sounds, processing the detection signal to identify an audible impulsive sound, and characterizing each impulsive sound.