1467 results about "Signal enhancement" patented technology

A wireless terminal receives signaling information, pertaining to a reference signal transmission in at least one specifically designated sub frame, the signaling information including a list, the list including base station identities. The terminal determines, from at least one of the base station identities in the list, the time-frequency resources associated with a reference signal transmission intended for observed time difference of arrival (OTDOA) measurements from a transmitting base station associated with said one base station identity. The time of arrival of a transmission from the transmitting base station, relative to reference timing, is measured. The wireless terminal can receive a command from a serving cell to start performing inter-frequency OTDOA measurement on a frequency layer containing reference signals, the frequency layer distinct from the serving frequency layer, the serving frequency layer not containing positioning reference signals. The wireless terminal can perform OTDOA measurements subsequent to the reception of the command on a carrier frequency different from the serving cell carrier frequency. A base station transmitter can jointly schedule a reference signal transmission from a plurality of base station transmitters for the purpose of OTD estimation enhancement, and transmit identical reference signals from the plurality of base station transmitters, the reference signals being identical both in the signal sequence and time-frequency resources used for transmission.

The present invention is embodied in a system and method for performing spectral analysis of a digital signal having a discrete duration by spectrally decomposing the digital signal at predefined frequencies uniformly distributed over a sampling frequency interval into complex frequency coefficients so that magnitude and phase information at each frequency is immediately available to produce a modulated complex lapped transform (MCLT). The present invention includes a MCLT processor, an acoustic echo cancellation device and a noise reducer integrated with an encoder / decoder device.

The invention is a device for multi-band, multi-channel, wireless communications that automatically provides signal amplification when and where necessary, and that automatically avoids harmful interference to base stations and other parts of the communications infrastructure. The device is especially suitable for, but not limited to, cellular and PCS bands and channels. The invention is a unique combination of an adjustable gain, bidirectional amplifier, a GPS receiver, a processor, and one or more removable, non-volatile, updatable memory devices. Alternatively, the memory can be an internal device accessible via an electronic port such as a USB. In either case, the memory stores comprehensive information that determines if amplification is necessary at a particular location sensed by the GPS receiver. Because the memory devices are updatable, they therefore provide protection against piracy and unauthorized and improper use. As an added measure of protection for the wireless networks, the device includes a dedicated apparatus that permits a service technician to remotely deactivate it in the event of a malfunction.

Visual information is used to alter or set an operating parameter of an audio signal processor, other than a beamformer. A digital camera captures visual information about a scene that includes a human speaker and / or a listener. The visual information is analyzed to ascertain information about acoustics of a room. A distance between the speaker and a microphone may be estimated, and this distance estimate may be used to adjust an overall gain of the system. Distances among, and locations of, the speaker, the listener, the microphone, a loudspeaker and / or a sound-reflecting surface may be estimated. These estimates may be used to estimate reverberations within the room and adjust aggressiveness of an anti-reverberation filter, based on an estimated ratio of direct to indirect (reverberated) sound energy expected to reach the microphone. In addition, orientation of the speaker or the listener, relative to the microphone or the loudspeaker, can also be estimated, and this estimate may be used to adjust frequency-dependent filter weights to compensate for uneven frequency propagation of acoustic signals from a mouth, or to a human ear, about a human head.

Input data comprising a video signal is processed using a combination of a known image processing method to deblur, or sharpen, the image and convolution with Pixon™ kernels for smoothing. The smoothing process utilizes a plurality of different size Pixon™ kernels which operate in parallel so that the input data are convolved with each different Pixon™ kernel simultaneously. The smoothed image is convolved with the point response function (PRF) to form data models that are compared against the input data, then the broadest Pixon™ kernel that fits the input data within a predetermined criterion are selected to form a Pixon™ map. The data are smoothed and assembled according to the Pixon™ map, then are deconvolved and output to a video display or other appropriate device, providing a clearer image with less noise.

An enhancement system improves the perceptual quality of a processed speech. The system includes a delay unit that delays a signal received through a discrete input. A spectral modifier linked to the delay unit is programmed to substantially flatten the spectral character of a background noise. An adaptive filter linked to the spectral modifier adapts filter characteristics to match a response of a non-delayed signal. A programmable filter is linked to the delay unit. The programmable filter has a transfer function functionally related to a transfer function of the adaptive filter.

A mobile communications device contains at least two microphones. One microphone is designated by a selector to provide a voice dominant signal and another microphone is designated to provide a noise or echo dominant signal, for a call or a recording. The selector communicates the designations to a switch that routes the selected microphone signals to the inputs of a processor for voice signal enhancement. The selected voice dominant signal is then enhanced by suppressing ambient noise or canceling echo therein, based on the selected noise or echo dominant signal. The designation of microphones may change at any instant during the call or recording depending on various factors, e.g. based on the quality of the microphone signals. Other embodiments are also described.

Systems and methods for conducting a multi-endpoint video signal conference are provided. Conferencing endpoints are linked by pairs of a reliable and a less reliable communication channel. Conference video signals are scaleable coded in base layer and enhancement layers format. Video signal base layers, which correspond to a minimum picture quality, are communicated over reliable channels. The video signal enhancements layers may be communicated over the less reliable channels. A conference server mediates the switching of video layer information from transmitting endpoints to receiving endpoints without any intermediate coding or re-coding operations. The video conference can be integrated with an audio conference using either scalable coded audio signals or non-scaleable coded audio signals.

An accurate target detection system. The system includes a sensor (22) that receives electromagnetic signals and provides electrical signals in response thereto. A non-uniformity correction circuit (28, 38, 52) corrects non-uniformities in the sensor (22) based on the electrical signals and provides calibrated electrical signals in response thereto. A third circuit (30, 32, 34, 38, 42, 44, 52) determines if a target signal is present within the calibrated electrical signals and provides a target detection signal in response thereto. A fourth circuit (38, 40, 48) selectively activates or deactivates the non-uniformity correction circuit (28, 38, 52) based on the target detection signal. In a specific embodiment, the sensor (22) is an array of electromagnetic energy detectors (22), each detector providing an electrical detector output signal The non-uniformity correction circuit (28, 38, and 52) includes circuit for compensating for gain, background, and noise non-uniformities (28, 38, and 52) in the electromagnetic energy detectors. The non-uniformity correction circuit (28, 38, and 52) includes a detector gain term memory (28) for storing detector gain compensation values. The detector gain compensation values are normalized by noise estimates unique to each of the detectors. The third circuit (30, 32, 34, 38, 42, 44, and 52) includes a signal enhancement circuit for reducing noise (34, 42) in the calibrated electrical signals. The third circuit (30, 32, 34, 38, 42, 44, and 52) includes a noise estimation circuit (32, 38) that estimates noise in each of the detector output signals and provides noise estimates in response thereto. The noise estimation circuit (32, 38) further includes a noise estimator circuit (38) and a recursive background estimator (32). The third circuit (30, 32, 34, 38, 42, 44, 52) further includes a subtractor (30) for subtracting background from the calibrated electrical signals and providing background subtracted signals in response thereto. The signal enhancement circuit (34, 42) includes a frame integrator circuit for adding frames of image data (34), each frame containing data corresponding to the background subtracted signals and providing summed frames in response thereto. The third circuit (30, 32, 34, 38, 42, 44, 52) includes a first threshold circuit (44) for comparing the filtered signal to a first threshold and a second threshold and providing a threshold exceedance signal if the filtered signal is between the first threshold and the second threshold.

A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.

A position indicator is provided for use with a position detecting sensor that carries out position detection by detecting a change in capacitance. The position indicator includes: a first electrode configured to receive an alternating-current (AC) signal from the position detecting sensor; a signal enhancing processing circuit configured to subject the AC signal received via the first electrode to determined signal enhancing processing; and a second electrode different from the first electrode and configured to be supplied with a signal output from the signal enhancing processing circuit. An enhanced signal having a determined correlation with the AC signal received via the first electrode from the position detecting sensor is formed while the enhanced signal is sent out to the position detecting sensor via the second electrode.

A signal processing device includes a bass signal extractor, a harmonic wave generator, a level detector, and an adjustment controller. The bass signal extractor first extracts a bass signal from an input audio signal. Natural-sounding bass enhancement is achieved as a result of the adjustment controller boosting the bass signal level until the level detector detects the bass signal level at a set level. For input bass signal levels higher than the set level, bass is enhanced virtually using a harmonic signal generated from the bass signal by the harmonic wave generator. As a result, the disadvantages of the boost method and the virtual signal enhancement method are mutually compensated for, and synergistic advantages for bass enhancement are obtained.

Adaptive controller apparatus of a robot may be implemented. The controller may be operated in accordance with a reinforcement learning process. A trainer may observe movements of the robot and provide reinforcement signals to the controller via a remote clicker. The reinforcement may comprise one or more degrees of positive and / or negative reinforcement. Based on the reinforcement signal, the controller may adjust instantaneous cost and to modify controller implementation accordingly. Training via reinforcement combined with particular cost evaluations may enable the robot to move more like an animal.

The invention discloses a voice signal enhancement system and a voice signal enhancement method. The method comprises the following steps of: filtering an input signal by an adaptive filter to obtain a voice reference signal and performing noise extraction processing on the input signal by using a block matrix to obtain a noise reference signal; filtering the noise reference signal and the voice reference signal through multi-channel Wiener filtering to obtain a denoised voice signal; performing voice enhancement processing on the denoised voice signal according to the noise reference signal to obtain an enhanced voice signal and a voice existence probability signal; and updating a filter coefficient of the adaptive matching filter, the block matrix and a filter coefficient and a learning step length of a multi-channel Wiener filter according to the voice existence probability signal. By the voice signal enhancement system and the voice signal enhancement method, under the condition that a microphone array is relatively simple, the problems that convergence cannot be guaranteed, the speed is low and the real-time enhancement distortion is high during real-time tracking of a target source in a practical application environment can be solved.

Near-field sensing of wave signals, for example for application in headsets and earsets, is accomplished by placing two or more spaced-apart microphones along a line generally between the headset and the user's mouth. The signals produced at the output of the microphones will disagree in amplitude and time delay for the desired signal—the wearer's voice—but will disagree in a different manner for the ambient noises. Utilization of this difference enables recognizing, and subsequently ignoring, the noise portion of the signals and passing a clean voice signal. A first approach involves a complex vector difference equation applied in the frequency domain that creates a noise-reduced result. A second approach creates an attenuation value that is proportional to the complex vector difference, and applies this attenuation value to the original signal in order to effect a reduction of the noise. The two approaches can be applied separately or combined.

The invention discloses a brain electric features based emotional state recognition method. The method comprises the following steps of: data acquisition stage: under the condition of international emotional picture induction, extracting 64 brain electric data which is tested under the induction of different-happiness-level pictures; data pretreatment stage: carrying out four stages of reference electric potential variation, down sampling, band-pass filtering, electro-oculogram removal on the collected 64 brain electric data; feature extraction stage: extracting time domain features after signals after pretreatment are filtered by a common space model algorithm; and feature recognition: recognizing the features by using a support vector machine classifier, and differentiating different emotional states. According to the method, an OVR (one versus rest) common space model algorithm is used for removing the interference of background signals, and is used for the signal intensification of multiple types of emotion induced brain electricity; after the background signals are removed, the differences among different types of emotional brain electricity are intensified, the recognition accurate ratio of subjects is relatively ideal when the recognition is carried out by the time domain variance features, and the emotions of different happiness can be differentiated accurately.

Methods and apparatus for color signal compensation are disclosed. A quality of one type of signal is enhanced using a second type of signal, wherein the second type of signal has a higher quality than that of the unenhanced first type of signal. This may be used in an image forming apparatus for scanning a color image, and using higher-resolution monochrome signals to enhance the resolution of lower-resolution color signals.

The invention provides the small-scale microphone array speech enhancement system and method. The system in the invention consists of two-part, signal acquisition and signal processing. Signal acquisition part comprises the microphone array and the AD converter component. Noisy voice signal is collected from the microphone as analog signals, through the AD converter, turning into digital signals, getting signal enhancement in the DSP chip, and then through USB port, transmitting to the computer for preservation. Signal processing part comprises the microphone consistency calibration module, the adaptive noise cancellation module, the SNR (Signal Noise Ratio) estimation module, and acoustic source location adaptive module, etc. The system in the invention can remove the introduced environmental noise in collection process, increase the collected acoustic source SNR, and obtain better acoustic source signal. In most cases the sound source signal is the speech signal, speech intelligibility improves a little after de-noising process.

An enhancement system improves the perceptual quality of a processed speech. The system includes a delay unit that delays a signal received through a discrete input. A spectral modifier linked to the delay unit is programmed to substantially flatten the spectral character of a background noise. An adaptive filter linked to the spectral modifier adapts filter characteristics to match a response of a non-delayed signal. A programmable filter is linked to the delay unit. The programmable filter has a transfer function functionally related to a transfer function of the adaptive filter.

A rebroadcasting apparatus for mobile media includes a receiving antenna and a transmitting antenna. The signal to be rebroadcast is received with a circularly polarized directional antenna. The transmitting antenna has circular polarization opposite to the polarization of the receiving antenna. Amplifiers and filters may be included between receiving and transmitting antennas to regulate retransmitted signal quality. The transmitting antenna uses one or more bays of quasi-helical dipole radiators excited using manifold feed equipped with tuning paddles. The transmitting antenna may be formed from light-weight sheet metal or other readily mass-produced materials. The transmitting antenna may be housed within an ordinary security camera enclosure or similar radio-transparent housing.

The invention discloses a microphone array multi-target voice enhancement method based on blind source separation and spectral subtraction. The method comprises: a multi-channel multi-target signals are collected through a microphone array; band-pass filter processing is carried out on the collected single-channel signals respectively to shield non-voice noises and interference, and pre-emphasis processing is carried out; voice windowing and framing processing is carried out to obtain frame signals, short-time Fourier transform is carried out to transform all frames into a frequency domain, and amplitude spectrums and phase spectrums of all frames are extracted; a starting end point and an ending end point of a voice signal are detected and a noise power spectrum is estimated; on the basis of spectral subtraction, background noises of a voice frame are reduced; the signal outputted after spectral subtraction is combined with the phase spectrum to carry out short-time Fourier inverse transform, thereby obtaining a voice signal of a time domain; and then blind source separation is carried out to obtain all target signals. The method can be realized simply; the resource requirement is low; the computing complexity is low; and multi-target signal enhancement can be realized.

The present invention is a bar code verification device having an ergonomic form factor characterized by a domed hand held trigger and a viewing window. The verification device may be disposed in a network that includes a host processor system and other bar code reading devices which may include other bar code verification devices. Processing circuitry for processing image signals corresponding to printed bar codes may be partially disposed within the hand held verification device and partially disposed within a host processor system spaced apart from and associated with the hand held verification device. The hand held verification device may be in wireless communication with the host processor system to which it is associated. The bar code verification system may include signal enhancement modules which interpolate constructed pixel values from actual pixel values and which correct for signal degradation resulting from high frequency spatial sampling.

Multi-touch sensitivity is enabled using a touch-sensitive apparatus comprising a panel for conducting signals from a plurality of incoupling points to a plurality of outcoupling points, thereby defining detection lines between pairs of incoupling and outcoupling points. Signal generators coupled to the incoupling points generate the signals, and signal detectors coupled to the outcoupling points generate an output signal indicative of one or more touches on the surface portion. A signal processor obtains the output signal which, if converted into a set of data samples of a given input format, enables a predetermined reconstruction algorithm to determine an interaction pattern on the surface portion. The signal processor generates, based on the output signal, a modified set of data samples in the given input format; and operates the predetermined reconstruction algorithm on the modified set of data samples so as to determine a modified interaction pattern on the surface portion.

Wirelessly linking a portable audio device to an external audio system permits the external audio system to generate audible signals that enhance local playback of audio by the portable device. For example, the portable device wirelessly transmits lower frequency components of an audio signal to a subwoofer system for reproduction of bass frequencies extending below the playback capability of the portable device. In this manner, the external audio system provides bass enhancement for the portable audio device. Wireless transmissions between the portable audio device and the external audio system may be, but are not limited to, optical or radio frequency (RF) transmissions. Where RF signaling is used, the wireless link may be based on wireless network links, such as those supported by Bluetooth and 802.11b standards, or based on, for example, dedicated RF interfaces.