510results about How to "Improve intelligibility" patented technology

A hearing aid system adapted for establishing a conversation group with other hearing aid systems used by different users, comprises a hearing aid (1, 2, 3, 4, 5) with an associated relay device (11, 12, 13, 14, 15). The relay device is (11, 12, 13, 14, 15) adapted for wireless communication (21, 22, 23, 24, 25) with said hearing aid and for wireless communication (31, 32, 33, 34, 35) with a second hearing aid system. The relay device (11, 12, 13, 14, 15) is also adapted for receiving and displaying information about said second hearing aid systems being available for participation in said conversation group, and it comprises means for selection of said other hearing aid systems for inclusion into the conversation group. The invention further provides a method for establishing a conversation group among hearing aid users.

A noise suppression device receives data representative of a noise-corrupted signal which contains a speech signal and a noise signal, divides the received data into data frames, and then passes the data frames through a pre-filter to remove a dc-component and the minimum phase aspect of the noise-corrupted signal. The noise suppression device appends adjacent data frames to eliminate boundary discontinuities, and applies fast Fourier transform to the appended data frames. A voice activity detector of the noise suppression device determines if the noise-corrupted signal contains the speech signal based on components in the time domain and the frequency domain. A smoothed Wiener filter of the noise suppression device filters the data frames in the frequency domain using different sizes of a window based on the existence of the speech signal. Filter coefficients used for Wiener filter are smoothed before filtering. The noise suppression device modifies magnitude of the time domain data based on the voicing information outputted from the voice activity detector.

A system and method for producing a binaural and localized audio signal to a user is provided. A signal processing method is provided for delivering spatialized sound in various ways using highly optimized inverse filters to deliver narrow localized beams of sound from the included speaker array. The inventive method can be used to provide private listening areas in a public space and provide spatialization of source material for a single user to create a virtual surround sound effect. In a binaural mode, a speaker array provides two targeted beams aimed towards the primary user's ears—one discrete beam for the left ear and one discrete beam for the right ear. In a privacy mode, a privacy zone could be created in which a primary audio beam would deliver a signal of interest to the user while secondary beams would be aimed at different angles to provide a masking noise.

A system and method of improving signal intelligibility over an interference signal is provided. The system includes a psychoacoustic professor having a psychoacoustic model wherein the level of a signal-of-interest is improved so as to be audible above noise and so as not to exceed a predetermined maximum output level. The system can be combined with active noise cancellation.

The present invention discloses a prediction model-building method for computer-aided medical diagnosis. It utilizes the medical symptom detection equipment to obtain the symptom of object to be diagnosed to form symptomatic vector, then utilizes the prediction model to make treatment so as to can obtain prediction result. It also provides its concrete steps for implementing said invented prediction model-building method.

A method of operating a hearing instrument for processing an input sound and to provide an output stimulus according to a user's particular needs, and related system, computer readable medium and data processing system. An object is to provide an improved customization of a hearing instrument. The method includes the steps a) providing an estimate of the present cognitive load of the user; b) providing processing of an input signal originating from the input sound according to a user's particular needs; and c) adapting the processing in dependence of the estimate the present cognitive load of the user. The estimate of the present cognitive load of a user is produced by in-situ direct measures of cognitive load (e.g. based on EEG-measurements, body temperature, etc.) or by an on-line cognitive model in the hearing aid system whose parameters have been preferably adjusted to fit to the individual user.

Methods and apparatus for modifying a user interface as a function of the detected emotional state of a system user are described. In one embodiment, stress analysis is performed on received speech to generate an emotional state indicator value, e.g., a stress level indicator value. The stress level indicator value is compared to one or more thresholds. If a first threshold is exceeded the user interface is modified, e.g., the presentation rate of speech is slowed. If a second threshold is not exceeded, another modification to the user interface is made, e.g., the speech presentation rate is accelerated. If the stress level indicator value is between first and second thresholds, user interface operation continues unchanged. The user interface modification techniques of the present invention may be used in combination with known knowledge or expertise based user interface adaptation features.

The present method incorporates audio and visual cues from human gesticulation for automatic recognition. The methodology articulates a framework for co-analyzing gestures and prosodic elements of a person's speech. The methodology can be applied to a wide range of algorithms involving analysis of gesticulating individuals. The examples of interactive technology applications can range from information kiosks to personal computers. The video analysis of human activity provides a basis for the development of automated surveillance technologies in public places such as airports, shopping malls, and sporting events.

A method and system for enhancing dialog determined by an audio input signal. In some embodiments the input signal is a stereo signal, and the system includes an analysis subsystem configured to analyze the stereo signal to generate filter control values, and a filtering subsystem including upmixing circuitry configured to upmix the input signal to generate a speech channel and non-speech channels and a peaking filter configured to filter the speech channel to enhance dialog while being steered by at least one of the control values. The filtering subsystem also includes ducking circuitry for attenuating the non-speech channels while being steered by at least some of the control values, and downmixing circuitry configured to combine outputs of the peaking filter and ducking circuitry to generate a filtered stereo output. In some embodiments, the system is configured to downmix a multichannel input signal to generate a downmixed stereo signal, an analysis subsystem is configured to analyze the downmixed stereo signal to generate filter control values, and a filtering subsystem is configured to generate a dialog-enhanced audio signal in response to the input signal while being steered by at least some of the filter control values. Preferably, the filter control values are generated without use of feedback including by generating power ratios (for pairs of speech and non-speech channels) and preferably also shaping in nonlinear fashion and scaling at least one of the power ratios.

A method of converting electrical signals into mechanical vibration by means of a transducer in the human ear, an ear canal transducer and a headset wherein a sensation of hearing is achieved by exciting the tissue of the ear canal directly with said transducer, whereby the vibrations propagate to the tympanic membrane and into the human sound sensing organs.

A system and method are provided for improving the quality and intelligibility of speech signals. The system and method apply frequency compression to the higher frequency components of speech signals while leaving lower frequency components substantially unchanged. This preserves higher frequency information related to consonants which is typically lost to filtering and bandpass constraints. This information is preserved without significantly altering the fundamental pitch of the speech signal so that when the speech signal is reproduced its overall tone qualities are preserved. The system and method further apply frequency expansion to speech signals. Like the compression, only the upper frequencies of a received speech signal are expanded. When the frequency expansion is applied to a speech signal that has been compressed according to the invention, the speech signal is substantially returned to its pre-compressed state. However, frequency compression according to the invention provides improved intelligibility even when the speech signal is not subsequently re-expanded. Likewise, speech signals may be expanded even though the original signal was not compressed, without significant degradation of the speech signal quality. Thus, a transmitter may include the system for applying high frequency compression without regard to whether a receiver will be capable of re-expanding the signal. Likewise, a receiver may expand a received speech signal without regard to whether the signal was previously compressed.

The invention provides a speech enhancing and frequency response compensation fusion method in a digital hearing-aid. The speech enhancing and frequency response compensation fusion method includes the steps of (1) obtaining estimated noise and initial enhanced speech with an MCRA method, respectively carrying out filtering processing on the estimated noise and the initial enhanced speech through a gammatone filter, dividing a signal into M frequency bands through the perception mechanism of the cochleas to the signal, and meanwhile obtaining a time frequency expression mode of the signal, (2) computing masking threshold values of the frequency bands through factors such as the audio masking characteristic of the human ears and the frequency band signal to noise ratios, (3) dynamically computing masking values of noise-contained speech in a time frequency domain through a hearing curve of a hearing disorder patient, and processing speech enhancing and frequency response compensation at the same time, and (4) synthesizing output speech of the heating-aid through the masking values. According to the speech enhancing and frequency response compensation fusion method, the working mechanism of the human ears is sufficiently used, the speech characteristics are kept, music noise introduced through a spectral subtraction method is eliminated, the speech intelligibility of output signals of the hearing-aid is greatly improved, and the low complexity and the low power consumption are achieved.

The invention discloses a computer security access control system and a computer security access control method, and belongs to the technical field of information security. A unified extensible firmware interface (UEFI) chip and a transmission control module (TCM) chip are included; a control system comprises an operating system pre-booting access control subsystem and an operating system running access control subsystem, wherein the operating system pre-booting access control subsystem comprises USBKey equipment, a USBkey equipment initialization module, a user management module, an operating system pre-booting identity authentication module and an encryption storage module; and the operating system running access control subsystem comprises USBKey equipment, an operating system user logon module, and an operating system user running protection module. The control method comprises an initialization method, an operating system pre-booting access control method and an operating system running access control method. The invention solves the problems that in the operating system, the reliability of access control is low, user identity information storage is insecure, and identity protection is insufficient in the running process after a user with legal identity logs on.

A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.

Methods and apparatus for maximizing speech intelligibility use psycho-acoustic variables of a model of speech perception to control the determination of optimal frequency-band specific gain adjustments. Speech signals (or other audio input) whose intelligibility is to be improved are characterized by parameters which are applied to the model. These include measurements or estimates of speech intensity level, average noise spectrum of the incoming audio signal, and/or the current frequency-gain characteristic of the hearing compensation device. Characterizations of listeners based on hearing test results, for example, may also be applied to the model. Frequency-band specific gain adjustments generated by use of the model can be used for hearing aids, assistive listening devices, telephones, cellular telephones, or other speech delivery systems, personal music delivery systems, public-address systems, sound systems, speech generating systems, or other devices or mediums which project, transfer or assist in the detection or recognition of speech.

A system for separation of a user's voice from ambient sound, comprising a device to be worn at the user's ear or at least partly in the user's ear canal comprising a first microphone oriented acoustically outwardly towards the environment and a second microphone oriented acoustically inwardly towards the user's ear canal, and an audio signal processing unit for processing audio signals from the first and second microphone by a blind source separation algorithm adapted to separate the user's voice from ambient sound.