1026 results about "Speech enhancement" patented technology

Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone. The ILD is utilized by a noise reduction system to enhance the speech of the primary acoustic signal.

The method provides a spectral speech description to be used for synthesis of a speech utterance, where at least one spectral envelope input representation is received. In one solution the improvement is made by manipulation an extremum, i.e. a peak or a valley, in the rapidly varying component of the spectral envelope representation. The rapidly varying component of the spectral envelope representation is manipulated to sharpen and / or accentuate extrema after which it is merged back with the slowly varying component or the spectral envelope input representation to create an enhanced spectral envelope final representation. In other solutions a complex spectrum envelope final representation is created with phase information derived from one of the group delay representation of a real spectral envelope input representation corresponding to a short-time speech signal and a transformed phase component of the discrete complex frequency domain input representation corresponding to the speech utterance.

A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes an adaptive speech enhancement filter for receiving an audio signal that includes a first component indicative of the spoken voice, a second component indicative of a feedback echo of the spoken voice and a third component indicative of the ambient noise, the speech enhancement filter filtering the audio signal by removing the third component to provide a filtered audio signal, the speech enhancement filter adapting to the audio signal at a first adaptation rate, and an adaptive acoustic echo cancellation system for receiving the filtered audio signal and removing the second component in the filtered audio signal to provide an echo-cancelled audio signal, the echo cancellation signal adapting to the filtered audio signal at a second adaption rate, wherein the first adaptation rate and the second adaptation rate are different from each other so that the speech enhancement filter does not adapt in response to operation of the echo-cancellation system and the echo-cancellation system does not adapt in response to operation of the speech enhancement filter.

An adaptive noise canceling microphone system for extracting a desired signal, in particular a desired speech signal, comprising two microphones being arranged at a predefined distance from each other; a signal forming system (SFS) being adapted to receive a first and second input signals resulting from sounds received by the two microphones wherein an acoustical signal component in the first input signal is determined, wherein an acoustical signal component in the second input signal is determined, wherein the acoustical signal component in the first input signal is enhanced to generate a speech enhanced signal, and wherein the acoustical signal component in the second input signal is suppressed to generate a speech nulled signal; an adaptive noise cancellation filtering circuit being adapted to receive the speech enhanced signal and the speech nulled signal, wherein the noise in the speech enhanced signal is cancelled using the speech nulled signal as reference, thereby generating an output filtered signal representing the desired signal.

Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

A method and apparatus for using network traffic logs for search enhancement is disclosed. According to one embodiment, network usage is tracked by generating log files. These log files among other things indicate the frequency web pages are referenced and modified. These log files or information from these log files can then be used to improve document ranking, improve web crawling, determine tiers in a multi-tiered index, determine where to insert a document in a multi-tiered index, determine link weights, and update a search engine index.

A voice enhancement logic improves the perceptual quality of a processed signal. The voice enhancement system includes a noise detector and a noise attenuator. The noise detector detects and models the noise associated with rain. The noise attenuator dampens or reduces the rain noise from a signal to improve the intelligibility of an unvoiced, a fully voiced, or a mixed voice segment.

The system and method comprises enhancement of results for a search engine, wherein the results from the search engine are refined or reorganized, based upon information from an identified secondary source. The results obtained using a conventional search are compared against the identified secondary source, e.g. a ratings service, and are filtered and / or sorted appropriately. In some embodiments, identification of the secondary source, such as a ratings service comprising information which may supplement the subject of a search query, is based upon information entered by the user. In alternate embodiments, the secondary source is associated with a user, as part of general user-specified search parameters, wherein one or more parameters are consulted automatically for searches for appropriate subject matter.

A mobile device includes an air conduction microphone and an alternative sensor that provides an alternative sensor signal indicative of speech. A communication interface permits the mobile device to communicate directly with other mobile devices.

In one aspect thereof the invention provides a method for noise suppression of a speech signal that includes, for a speech signal having a frequency domain representation dividable into a plurality of frequency bins, determining a value of a scaling gain for at least some of said frequency bins and calculating smoothed scaling gain values. Calculating smoothed scaling gain values includes, for the at least some of the frequency bins, combining a currently determined value of the scaling gain and a previously determined value of the smoothed scaling gain. In another aspect a method partitions the plurality of frequency bins into a first set of contiguous frequency bins and a second set of contiguous frequency bins having a boundary frequency there between, where the boundary frequency differentiates between noise suppression techniques, and changes a value of the boundary frequency as a function of the spectral content of the speech signal.

A mobile device includes an air conduction microphone and an alternative sensor that provides an alternative sensor signal indicative of speech. A communication interface permits the mobile device to communicate directly with other mobile devices.

A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes a microphone for receiving the spoken voice and the ambient noise and for converting the spoken voice and the ambient noise into an audio signal, the audio signal having a first component corresponding to the spoken voice and a second component corresponding to the ambient noise, a speech enhancement filter for removing the second component from the audio signal to provide a filtered audio signal, the speech enhancement filter removing the second component by processing the audio signal by a method taking into account elements of psycho-acoustics of a human ear, and a loudspeaker for outputting a clarified voice in response to the filtered audio signal.

A mobile device is provided that includes a digit input that can be manipulated by a user's fingers or thumb, an air conduction microphone and an alternative sensor that provides an alternative sensor signal indicative of speech. Under some embodiments, the mobile device also includes a proximity sensor that provides a proximity signal indicative of the distance from the mobile device to an object. Under some embodiments, the signal from the air conduction microphone, the alternative sensor signal, and the proximity signal are used to form an estimate of a clean speech value. In further embodiments, a sound is produced through a speaker in the mobile device based on the amount of noise in the clean speech value. In other embodiments, the sound produced through the speaker is based on the proximity sensor signal.

A service includes verbally editing at least one parameter of a subscriber's advanced intelligent network (AIN) service using speech recognition functionality of an intelligent peripheral. A call is received from the subscriber at a switch, including a dialed number for editing the AIN service. A connection is established with the intelligent peripheral in response to the dialed number. Call service data, including the parameter, is retrieved from a database. The intelligent peripheral plays a voice announcement identifying the parameter to the subscriber, receives a voice instruction from the subscriber in response to the voice announcement and translates the voice instruction into digital command data relating to a change to the parameter. The digital command data and the change to the parameter are stored in the database for near real-time implementation. The intelligent peripheral and the database are likewise accessible by the subscriber through a packet switched data network.

The method provides a spectral speech description to be used for synthesis of a speech utterance, where at least one spectral envelope input representation is received. In one solution the improvement is made by manipulation an extremum, i.e. a peak or a valley, in the rapidly varying component of the spectral envelope representation. The rapidly varying component of the spectral envelope representation is manipulated to sharpen and / or accentuate extrema after which it is merged back with the slowly varying component or the spectral envelope input representation to create an enhanced spectral envelope final representation. In other solutions a complex spectrum envelope final representation is created with phase information derived from one of the group delay representation of a real spectral envelope input representation corresponding to a short-time speech signal and a transformed phase component of the discrete complex frequency domain input representation corresponding to the speech utterance.

The embodiment of the invention discloses lost frame recovering method and equipment as well as speech enhancing method, equipment and system, wherein the lost frame recovering method comprises the following steps of: determining the type of a lost frame according to the type of an adjacent frame adjacent to the lost frame; obtaining a characteristic parameter obtained after unlocking the parameter of the adjacent frame, wherein the characteristic parameter is used for determining the characteristic information of coded data in the frame; constructing the characteristic parameter conforming to the type of the lost frame according to the correlation between the characteristic parameter of the adjacent frame and the characteristic information of the coded data in each frame; and recovering the lost frame according to the constructed characteristic parameter of the lost frame. Through the method, the lost frame recovery of a parameter domain can be realized, and system resources consumed in the recovering process are reduced.

Provided herein are orthodontic devices and methods for patients whose orthodontic devices are causing a lisp. The device can comprise an aligner configured to fit over a patient's dental arch and comprising an occlusal surface section positioned over an occlusal surface of the patient's teeth. The aligner can comprise a barrier portion extending laterally and adjacent to a region of the dental arch, the barrier portion allowing the patient's tongue to form a seal against the barrier portion when the patient is speaking while wearing the device.

An image processing method for executing edge enhancement for an original image includes: extracting edge components based upon the original image; correcting the extracted edge components by attenuating the individual edge components so that a frequency distribution related to intensity of the edge components approximates a Gaussian distribution assuming a specific width; and executing edge enhancement for the original image based upon the corrected edge components.

Techniques to enhance voice signals in a dual microphone system are disclosed. According to one aspect of the present invention, there are at least two microphones that are positioned in a pre-configured array. Two audio signals x1(k) and x2(k) are received and coupled to an adjusting module that is provided to control the gain of each of the audio signals x1(k) and x2(k) to minimize signal differences between the two signals. A separation module is provided to receive matched audio signals x′1(k) and x′2(k) from the adjusting module. The separation module separates the audio signals x′1(k) and x′2(k) to obtain a first audio signal s(k) containing mainly the voice and a second audio signal n(k) containing mainly the noise. An adaptive filtering module is provided to eliminate the noise component in the audio signal s(k) to obtain an estimated voice signal e_s(k) with a higher S / N ratio. Furthermore, the adaptive filtering module can be also configured to suppress echo in the audio signal s(k) at same time. The voice signal e_s(k) may be further coupled to a single-channel voice enhancement module that is configured to eliminate any residual of the noise component in the voice signal e_s(k) according to the differences between the voice signal and the noise signal in time domain and frequency domain, whereby, the S / N ratio is further enhanced.

The invention discloses a sound activation detecting method and a sound activation detector, the core of which is: extracting the feature parameters of the current signal frame when the sound activation detection is needed; and determining the sound type of the current signal frame according to the feature parameters and the set parameter threshold. By the invention, the specific coding algorithm is not relied on when the used feature parameters in the classifying process are extracted, thus being convenient for maintenance and updating, and classifying the input signals into more sound types. When being used in the sound coding technical field, the invention can not only be used as new-opened variable rate sound frequency coding algorithm and standard rate selection foundation, but also provide foundation of rate selection for prior variable rate voice or sound frequency coding standard without VAD algorithm. The invention can be applicable to voice boosting, voice recognition, recognition of spoken person and other voice signal processing fields with strong commonality.

The invention relates to a dual-microphone-based speech enhancement device and method. The device comprises a microphone array module and a fixed beam former, the microphone array module comprises a first microphone and a second microphone which are used for receiving time-domain noisy speech signals and outputting received time-domain noisy speech signals, wherein the time-domain noisy speech signals received by the first microphone and the second microphone are respectively x1 and x2; and the fixed beam former is used for receiving the time-domain noisy speech signals output by the microphone array module and overlying the speech part of the received time-domain noisy speech signals to amplify the time-domain noisy speech signals so as to form initially enhanced speech signals, and then outputting the signals. The device and the method have good denoising effect.

The invention provides a microphone array voice enhancement device with a sound source direction tracking function and a method for the microphone array voice enhancement device. The microphone array voice enhancement device relates to voice signal processing. A microphone array, an adjustable parallel beam former group module, a fixed parameter FIR (Finite Impulse Response) filter module, a fixed parameter signal blocking module, an adaptive noise canceller module and a sound source direction update module are aranged in the devide. The method comprises the following steps of: initializing, forming an adjustable beam, fixing parameter filtering, blocking a signal, carrying out adaptive noise cancelling, and updating a sound source direction. The invention provides an adjustable parallel beam former group combined with a sidelobe canceller structure to achieve real-time tracking of a target sound source direction, embeds the sound source direction tracking function directly to a generalized sidelobe canceller structure, can achieve sound source direction tracking and voice enhancement, thereby overcoming sensibility of an algorithm performance on a DOA (Direction of Arrival) estimation error.

By automatically compensating for unwanted lighting effects caused by varying ambient light sources, digital still images and video are made to appear more naturally realistic. A set of correction factors based upon an ascertained ambient lighting conditions, such as color temperature, provides for selectively color correcting only those pixels requiring correction. In this way, only those portions of a digital image having excessive color due to ambient light conditions are improved without affecting those portions of the digital image not requiring correction.

A voice enhancement logic improves the perceptual quality of a processed voice. The voice enhancement system includes a passing tire hiss noise detector and a passing tire hiss noise attenuator. The passing tire hiss noise detector detects a passing tire hiss noise by modeling the passing tire hiss. The passing tire hiss noise attenuator dampens the passing tire hiss noise to improve the intelligibility of a speech signal.

The invention provides a method and apparatus for noise reduction of a signal, for example speech enhancement of a speech signal. The method involves a two-stage algorithm comprising performing a preprocessing first spectral subtraction to remove tonal noise and generate a tonal noise removed signal, and performing a second spectral subtraction to remove noise from the said tonal noise removed signal. In both spectral subtraction stages noise is not removed completely but only to a level below an audible threshold in order to avoid unwanted artifacts.

This invention provides one double microwave sound strength method and device suitable for small mobile communication device to process the input signal x1 and x2 and to adopt wave beam forming technique and use aim sound signal source and noise signal source difference to isolate signals to get the sound signal S(k) and noise signal n(k); using two paths of signals relationship to remove noise part and sound point to get síõ(k) and níõ(k).

A method and apparatus for processing gray level image data. Gray level image data is subjected to plural separate halftone screen processing to form plural separate halftone processed screen gray level image data. The current pixel is also analyzed for contrast index. In response to analysis of the contrast index blending coefficients for processing of the halftone screen modified image data is made. The respective halftone outputs of the screen processings are multiplied by the respective blending coefficient. The resulting blended halftone gray value of the current pixel as well as neighboring pixels are then subjected to a threshold criterion test to determine if this represents a substantially binary image file such as might occur in a saturated text image file. The blended halftone gray value data for the current pixel is subjected to gray level edge enhancement processing to replace certain binary pixels adjacent an edge to reduce anti-aliasing effects. A signal resulting from the threshold criterion test is used to determine whether there is output to the printer or display of an edge enhanced version of the current blended halftone pixel or a pixel value representing the blended halftone pixel.

A speech enhancement system improves the perceptual quality of a processed voice signal. The system improves the perceptual quality of a voice signal by removing unwanted noise components from a voice signal. The system removes undesirable signals that may result in the loss of information. The system receives and analyzes signals to determine whether an undesired random or persistent signal corresponds to one or more modeled noises. When one or more noise components are detected, the noise components are substantially removed or dampened from the signal to provide a less noisy voice signal.