49 results about "Interaural time difference" patented technology

An auditory scene is synthesized from a mono audio signal by modifying, for each critical band, an auditory scene parameter (e.g., an inter-aural level difference (ILD) and/or an inter-aural time difference (ITD)) for each sub-band within the critical band, where the modification is based on an average estimated coherence for the critical band. The coherence-based modification produces auditory scenes having objects whose widths more accurately match the widths of the objects in the original input auditory scene.

An auditory scene is synthesized by applying two or more different sets of one or more spatial parameters (e.g., an inter-ear level difference (ILD), inter-ear time difference (ITD), and/or head-related transfer function (HRTF)) to two or more different frequency bands of a combined audio signal, where each different frequency band is treated as if it corresponded to a single audio source in the auditory scene. In one embodiment, the combined audio signal corresponds to the combination of two or more different source signals, where each different frequency band corresponds to a region of the combined audio signal in which one of the source signals dominates the others. In this embodiment, the different sets of spatial parameters are applied to synthesize an auditory scene comprising the different source signals. In another embodiment, the combined audio signal corresponds to the combination of the left and right audio signals of a binaural signal corresponding to an input auditory scene. In this embodiment, the different sets of spatial parameters are applied to reconstruct the input auditory scene. In either case, transmission bandwidth requirements are reduced by reducing to one the number of different audio signals that need to be transmitted to a receiver configured to synthesize/reconstruct the auditory scene.

This invention addresses sound recording and mixing methods for 3-D audio rendering of multiple sound sources over headphones or loudspeaker playback systems. Economical techniques are provided, whereby directional panning and mixing of sounds are performed in a multi-channel encoding format which preserves interaural time difference information and does not contain head-related spectral information. Decoders are provided for converting the multi-channel encoded signal into signals for playback over headphones or various loudspeaker arrangements. These decoders ensure faithful reproduction of directional auditory information at the eardrums of the listener and can be adapted to the number and geometrical layout of the loudspeakers and the individual characteristics of the listener. A particular multi-channel encoding format is disclosed, which, in addition to the above advantages, is associated with a practical microphone technique for producing 3-D audio recordings compliant with the decoders described.

A method of enhancing binaural representation for a subject includes receiving a first signal and a second signal in response to a plurality of sound sources, generating a number of estimated interaural time differences using the first signal and the second signal, converting each of the number of estimated interaural time differences to a corresponding interaural level difference, using one or more of the corresponding interaural level differences to generate an adjusted first signal, and using the adjusted first signal to generate a number of signals delivered to the subject for enhancing the hearing of the subject.

A binaural sound source localization method based on sub-band signal to noise ratio estimation is an improved sound source localization method, wherein the mean value of the ITD (Interaural Time Difference) of various orientations is used as the localization characteristic clue for the sound source orientation to build an orientation mapping model; during the actual sound source localization, a dual-channel acoustic signal is input, the input acoustic signal is firstly subjected to frequency domain transformation, a frequency domain is divided into a plurality of sub-bands, signal to noise ratio estimation is carried out in each sub-band, according to the sub-band signal to noise ratio, the power spectrum of the corresponding sub-band is selected to calculate the ITD parameters of each frame, one-by-one match is performed according to the orientation characteristic model built by the ITD characteristic parameters and a training module, and based on the Euclidean distance measurement, the orientation is output. With the binaural sound source localization method, the performance of sound source localization in noisy environments can be improved.

A method and system for reducing head related transfer function (HRTF) storage requirements for 3-D sound processing of an input sound having a specified source angle increment is provided. Interaural time difference (ITD) values are selected based directly on the source angle increment; and HRTFs for processing the input sound are stored in angle increments larger than the source angle increment.

A method of estimating an individualized head-related transfer function and an individualized interaural time difference function of a particular person, comprises the steps of: a) obtaining a plurality of data sets comprising a left and a right audio sample from in-ear microphones, and orientation information from an orientation unit, measured in a test-arrangement where an acoustic test signal is rendered via a loudspeaker and the person is moving the head; b) extracting interaural time difference values and/or spectral values, and corresponding orientation values; c) estimating a direction of the loudspeaker relative to the head using a predefined quality criterion; d) estimating an orientation of the orientation unit relative to the head; e) estimating the individualized ITDF and the individualized HRTF. A computer program product may be provided for performing the method, and a data carrier may contain the computer program.

A stereophonic sound output apparatus and an early reflection generation method thereof. The stereophonic sound output apparatus includes an early reflection generator to implement an early reflection when a 5.1 channel audio signal is down-mixed to a 2-channel audio signal to play back a 5.1 channel audio signal through a 2-channel headphone. The early reflection generator generates early reflections in pairs in which there is an appropriate time difference between the left side reflections and the right side reflections by generating an interaural time difference between two input audio signals and filtering. It is possible to copy the characteristics of early reflections in a real listening room. It is also possible to implement an early reflection similar to a real reflection measured in an apparatus for playing back the 5.1 channel audio signal through 2-channel headphone. A natural 5.1 channel effect may also be obtained using little computation.

A device and method for generation of virtual surround sound with a two-way approach is provided. The device and method employs a first order head-related model designed to resemble interaural time difference localization and inter-aural level difference localization cues in the respective frequency bands while avoiding phantom imaging and excessive coloration.

The invention discloses a recording data training method, a multi-track audio surrounding method and a recording data training device, and belongs to the field of audio processing. The recording data training method comprises the following steps: according to an hearing environment of a user, modeling to obtain a virtual hearing model; according to physiological data of the head of the user, modeling to obtain a virtual dummy head model; calculating to obtain interaural time difference (ITD) corresponding to direct waves of each sound track; calculating to obtain interaural latency difference (ILD) corresponding to direct waves of each sound track. The multi-track audio surrounding method comprises the following steps: obtaining original multi-track audio content; according to the ITD and the ILD obtained by the recording data training method, performing conversion on sound data of each sound track in the multi-track audio content to obtain left channel data and right channel data which correspond to each sound track; mixing the left sound channel data corresponding to each sound track into target left channel data and mixing right sound channel data corresponding to each sound track into target right sound channel data to generate stereo audio content.

The space parameter selection method for parameter stereo coding is disclosed. Three space parameters of interaural time difference, interaural intensity difference and interaural correlation degree are used for describing the space information in stereo signals. Signals are classified into three frequency ranges of low, middle and high frequency. According to the coding rate of the space parameters, different space parameter combinations are selected in each frequency range to describe space information. Advantage: raised effect of space information expression and coding efficiency of parameter stereo.

The invention discloses a deep learning-based binaural sound source positioning method in a digital hearing aid. The method comprises the following steps: firstly, decomposing a binaural sound sourcesignal into a plurality of channels through a gammatone filter, extracting a high-energy channel through a weighting coefficient, then extracting a first type of features by using a head-related-transform function (HRTF), namely an Interaural Time Difference (ITD) and an Interaural Intensity Difference (IID) which are used as inputs of deep learning, and dividing a horizontal plane into four quadrants to narrow a positioning range; secondly, extracting a second type of features of head-related transform, namely an Interaural Level Difference (ILD) and an Interaural Phase Difference (IPD); andfinally, in order to realize more precise positioning, taking the first and second types of four features as inputs of next deep learning, thereby obtaining an azimuth angle of sound source positioning. Precise positioning of 72 azimuth angles is realized on the horizontal plane from 0 degree to 360 degrees at a step length of 5 degrees.

A sound source localization system includes a plurality of microphones for receiving a signal as an input from a sound source; a time-difference extraction unit for decomposing the signal inputted through the plurality of microphones into time, frequency and amplitude using a sparse coding and then extracting a sparse interaural time difference (SITD) inputted through the plurality of microphones for each frequency; and a sound source localization unit for localizing the sound source using the SITDs. A sound source localization method includes receiving a signal as an input from a sound source; decomposing the signal into time, frequency and amplitude using a sparse coding; extracting an SITD for each frequency; and localizing the sound source using the SITDs.

The invention discloses a binaural speech separation method based on an LSTM (Long Short Term Memory) network. The ITD (Inter-aural Time Difference), the IID (Interaural Intensity Difference) and theCCF (Cross Correlation Function) of each time frequency unit of a training binaural speech signal are extracted to be used as separation space features; space features of a current frame and front andback 5 frames of the time frequency units in the same subband are used as input parameters of a two-way LSTM network to be trained; and a separation model based on the LSTM is obtained. At the test stage, the space features of the current frame and the front and back 5 frames of the time frequency units of a test binaural speech signal are used as input parameters, obtained through training, of the two-way LSTM network, and are used for estimating shielding values of the target speech of the current time frequency unit so as to perform speech separation according to a shielding value. The separation result shows that compared with a method based on a deep neural network, the binaural speech separation method based on the LSTM network provided by the invention has the advantages that the subjective evaluation index is obviously improved, and the algorithm generalization performance is good.