511results about How to "Improve intelligibility" patented technology

The method and preprocessor enhances the intelligibility of narrowband speech without essentially lengthening the overall time duration of the signal. Both spectral enhancements and variable-rate time-scaling procedures are implemented to improve the salience of initial consonants, particularly the perceptually important formant transitions. Emphasis is transferred from the dominating vowel to the preceding consonant through adaptation of the phoneme timing structure. In a further embodiment, the technique is applied as a preprocessor to a speech coder.

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.

An electronic equipment is provided that includes an audio reproduction circuit for receiving an audio signal and reproducing the audio signal in accordance with a preselected audio frequency gain profile. In addition, the electronic equipment includes a volume gain control circuit for providing user adjustable volume gain to the audio signal reproduced by the audio reproduction circuit. The electronic equipment also includes an audio output for outputting the reproduced audio signal. The audio reproduction circuit alters the preselected audio frequency gain profile as a function of the user adjusted volume gain.

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.

An adaptive “temporal audio scaler” is provided for automatically stretching and compressing frames of audio signals received across a packet-based network. Prior to stretching or compressing segments of a current frame, the temporal audio scaler first computes a pitch period for each frame for sizing signal templates used for matching operations in stretching and compressing segments. Further, the temporal audio scaler also determines the type or types of segments comprising each frame. These segment types include “voiced” segments, “unvoiced” segments, and “mixed” segments which include both voiced and unvoiced portions. The stretching or compression methods applied to segments of each frame are then dependent upon the type of segments comprising each frame. Further, the amount of stretching and compression applied to particular segments is automatically variable for minimizing signal artifacts while still ensuring that an overall target stretching or compression ratio is maintained for each frame.

A system for processing a speech signal to enhance signal intelligibility identifies portions of the speech signal that include sounds that typically present intelligibility problems and modifies those portions in an appropriate manner. First, the speech signal is divided into a plurality of time-based frames. Each of the frames is then analyzed to determine a sound type associated with the frame. Selected frames are then modified based on the sound type associated with the frame or with surrounding frames. For example, the amplitude of frames determined to include unvoiced plosive sounds may be boosted as these sounds are known to be important to intelligibility and are typically harder to hear than other sounds in normal speech. In a similar manner, the amplitudes of frames preceding such unvoiced plosive sounds can be reduced to better accentuate the plosive. Such techniques will make these sounds easier to distinguish upon subsequent playback.

An augmented reality system and an assembling method are provided for assembling a first set-up component to a second set-up component under the assistance of an augmented reality system. The augmented realty system may capture a first part of a marker and a second part of the marker associated with the respective component. The augmented realty system may capture a variable marker associated with one of the components. The augmented realty system may capture a first marker and a second marker associated with the respective component. The augmented reality system can recognize the location and / or status of the markers and thus decide whether the connection between the first and second set-up component is established correctly or not.

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.

The notion of an easily retrofitable passive lighting-safety device designed to be generally installed into the existing rear lighting wiring-circuit of any motor-vehicle from motorcycles to cars and trucks, -so as to significantly boost a following driver's visualacuity cognition threshold of awareness regarding the state of a leading vehicle's sudden breaking procedure. The invention provides an intense quasi-precognitive white-light "flashback" alerting effect, preferably appearing (particularly under nighttime conditions) as a nearly subliminal approximate 100-200 / millisecond flashback of visually non-deacclimating "white-light", emanating from a vehicle's existing back-up lights (or integrated with the DOT high-center brake-light). The effect being to essentially arouse a following motorist's attention to the simultaneous light-up of the preceding vehicle's stoplights. Regardless as to how long braking action continues, the so called STOPWHITES TM only illuminate in flashback once for a brief instant per actuation of the brakes; -having served their sole purpose in attracting a following driver's attention to need for braking. A following driver being thus able to react substantially sooner than viewing conventional stoplights only; -thereby significantly shortening the braking response-time. Additionally, exemplary circuits teach wiring the preferably solid-state electrical-module with directional-signal lights, as well as the 4-Way / Emergency-flasher system.

A method of reducing noise in an audio signal, comprising the steps of: using a furrow filter to select spectral components that are narrow in frequency but relatively broad in time; using a bar filter to select spectral components that are broad in frequency but relatively narrow in time; analyzing the relative energy distribution between the output of the furrow and bar filters to determine the optimal proportion of spectral components for the output signal; and reconstructing the audio signal to generate the output signal. A second pair of time-frequency filters may be used to further improve intelligibility of the output signal. The temporal relationship between the furrow filter output and the bar filter output may be monitored so that the fricative components are allowed primarily at boundaries between intervals with no voiced signal present and intervals with voice components. A noise reduction system for an audio signal.

An apparatus for generating, storing or editing an audio representation of an audio scene includes audio processing means for generating a plurality of speaker signals from a plurality of input channels as well as means for providing an object-oriented description of the audio scene, wherein the object-oriented description of the audio scene includes a plurality of audio objects, wherein an audio object is associated with an audio signal, a starting time instant and an end time instant. The apparatus for generating further distinguishes itself by mapping means for mapping the object-oriented description of the audio scene to the plurality of input channels, wherein an assignment of temporally overlapping audio objects to parallel input channels is performed by the mapping means, whereas temporally sequential audio objects are associated with the same channel. With this, an object-oriented representation is transferred into a channel-oriented representation, whereby on the object-oriented side the optimal representation of a scene may be used, whereas on channel-oriented side the channel-oriented concept users are used to may be maintained.

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 outwardly towards the environment and a second microphone oriented 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.

A canal phone incorporates a transducer within an isolation sound chamber for reproducing sound from an audio source, and a sound-transmission passageway, or ambient sound port, for allowing ambient sounds to enter the isolation sound chamber. The passageway has inner and outer apertures, one oriented towards the ambient noise environment outside of the listener's ear and the other oriented toward the isolated chamber formed by the space between a loudspeaker diaphragm within the canal phone and the tympanum of the listener's ear. The canal phone has a port blocking switch mechanism that selectively opens or closes the aperture of the passageway that is open to the ambient noise. This switch mechanism may also electrically disconnect the transducer and seal either the outer or the inner end of the passageway in different embodiments of the invention.

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.

An adaptive “temporal audio scaler” is provided for automatically stretching and compressing frames of audio signals received across a packet-based network. Prior to stretching or compressing segments of a current frame, the temporal audio scaler first computes a pitch period for each frame for sizing signal templates used for matching operations in stretching and compressing segments. Further, the temporal audio scaler also determines the type or types of segments comprising each frame. These segment types include “voiced” segments, “unvoiced” segments, and “mixed” segments which include both voiced and unvoiced portions. The stretching or compression methods applied to segments of each frame are then dependent upon the type of segments comprising each frame. Further, the amount of stretching and compression applied to particular segments is automatically variable for minimizing signal artifacts while still ensuring that an overall target stretching or compression ratio is maintained for each frame.

A lane change assist information visualization system allows a vehicle driver to change lanes with minimum effort and stress by displaying a lane change option that suits the particular lane changing situation in an intuitively understandable manner, and information on the surrounding vehicles adequate for the vehicle driver to make a reasonable decision in changing lanes. The system includes a lane change program computing device (9) that computes at least (a) a lane change possibility, (b) a time period that has to elapse before a lane change is allowed if a lane change possibility does not currently exists, or a time period in which a lane change should be made if a lane change possibility currently exists, and (c) a need for acceleration or deceleration, and a display unit (10) for displaying this information.