Abstract: A range profile digitization circuit for converting a repeating analog input signal into a time series of digital amplitude values, the converter comprising: a signal quantizer arranged to receive the analog input signal and a threshold input and arranged to output a binary value quantized output signal based on a comparison of the input signal with the threshold signal; a plurality of samplers each arranged to sample and hold its input signal upon receipt of a trigger signal; and for each sampler: a plurality of decoders and a demultiplexer arranged to receive an output from the sampler and pass it to a selected one of said decoders based on a selector input. With a plurality of decoders associated with each of the samplers, each sampler can be re-used during the building up of the range profile.

Abstract: An audio system has an ambient sound enhancement (ASE) function, in which an against-the-ear audio device having a speaker converts a digitally processed version of an input audio signal into amplified sound. The amplification may be in accordance with a stored hearing profile of the user. The audio system also has an acoustic noise cancellation (ANC) function that may be combined in various ways with the ASE function, and that may be responsive to voice activity detection. Other aspects are also described and claimed.

Abstract: An input signal is provided with a low microphone noise in a hearing apparatus. The microphone noise in the input signal of the hearing apparatus is reduced, by the input signal being filtered by a Wiener filter, if a noise power determined at the input signal is smaller than a predetermined limit value. The Wiener filter is however deactivated, if the noise power is greater than the limit value or equal to the limit value.

Abstract: A system and method for remote active noise correction at a remote device includes receiving, at the remote device, an ambient noise signal from a microphone. The remote device is disposed along a processing and transmission path between the microphone and a headphone. The processing and transmission path exhibit non-zero latency. The remote device further analyzes the ambient noise signal to generate an anti-noise signal, performs a first correction of the anti-noise signal for a headphone interface effect, performs a second correction of the anti-noise signal for the non-zero latency of the processing and transmission path between the microphone and the headphone. The remote device then transmits the corrected anti-noise signal to the headphone.

Abstract: Active noise control (ANC) is performed within a vehicle. Suppressed sound is produced by suppressing frequencies of ambient sound above a threshold frequency that enter an interior cavity of a sound-suppressing enclosure disposed within, and spaced from, interior walls of the vehicle. A microphone disposed within the interior cavity of the sound-suppressing enclosure receives feedback comprising a combination of the suppressed sound produced by the sound-suppressing enclosure and antinoise produced by one or more speakers mounted to a headrest disposed within the interior cavity of the sound-suppressing enclosure. The speakers are controlled to produce the antinoise based on the feedback, such that the antinoise destructively interferes with frequencies of the suppressed sound that are above the threshold frequency.

Abstract: Methods and devices for characterization of repetitious noise in cable networks are disclosed. A frequency band of interest is identified, a time trace of a signal parameter within the frequency band is obtained, and an autocorrelation of the time trace is computed to detect repetitious noise. The repetition frequency can serve as an indicator of the noise source type, and thus it can assist in noise segmentation.

Abstract: According to one embodiment, a noise reduction apparatus includes a first control sound source, a first current detection unit, a second control sound source, a second current detection unit, and an adjustment unit. The first control sound source generates a first control sound for reducing noise from a noise source. The first current detection unit detects a first current flowing from the first control sound source upon receiving the noise from the noise source. The second control sound source is provided at a position different from a position of the first control sound source and generates a second control sound for reducing the noise from the noise source. The second current detection unit detects a second current flowing from the second control sound source upon receiving the noise from the noise source. The adjustment unit adjusts the first control sound and the second control sound so as to make the first current and the second current satisfy a predetermined condition.

Abstract: Disclosed is a device (1) for processing an initial audio signal (4), including: a treble detecting block (2) capable of analyzing the initial audio signal to determine a treble level; and a filter block (3) capable of attenuating the initial audio signal in the treble, if the treble level is less than a maximum threshold.

Abstract: A measuring apparatus includes an optical system configured to project light onto a sample and to receive light via the sample, an imaging device configured to take an image of a light source via the optical system, and a processor configured to obtain an optical characteristic of the sample based on an output of the imaging device. The processor is configured to determine a coefficient of a Wiener filter based on one of the image and a Fourier transform thereof and corresponding one of the light source (an aperture in an aperture unit) and a Fourier transform thereof, and obtain the optical characteristic based on the Wiener filter of which the coefficient has been determined, a Fourier transform of the image, and a Fourier transform of the light source.

Abstract: A capacitive sensing system includes a controller, a node connected to one side of a capacitance, the controller configured to measure the capacitance by measuring a time for a voltage across the capacitance to reach a predetermined reference voltage, a noise measurement circuit configured to measure electrical noise on the node, and the controller receiving the measurement of noise from the noise measurement circuit.

Abstract: A stereo-phonic frequency modulation receiver includes: a frequency modulation demodulation circuit, receiving a reception signal, and generating a demodulated signal according to the reception signal; a frequency-division demultiplexer, generating a sum signal, a difference signal and a pilot amplitude signal according to the demodulated signal; a dual sound channel separation circuit, generating a left-channel output signal and a right-channel output signal according to the sum signal and a weakened difference signal; and a weakening circuit, weakening the difference signal according to the pilot amplitude signal or a signal-to-noise ratio (SNR) to generate the weakened difference signal.

Abstract: An input signal is provided with a low microphone noise in a hearing apparatus. The microphone noise in the input signal of the hearing apparatus is reduced, by the input signal being filtered by a Wiener filter, if a noise power determined at the input signal is smaller than a predetermined limit value. The Wiener filter is however deactivated, if the noise power is greater than the limit value or equal to the limit value.

Abstract: A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is also provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit determines a degree of coupling between the user's ear and the transducer and adjusts the adaptive cancellation of the ambient sounds to prevent erroneous and possibly disruptive generation of the anti-noise signal if the degree of coupling lies either below or above a range of normal operating ear contact pressure.

Abstract: An audio system encodes and decodes audio captured by a microphone array system in the presence of wind noise. The encoder encodes the audio signal in a way that includes beamformed audio signal and a “hidden” representation of a non-beamformed audio signal. The hidden signal is produced by modulating the low frequency signal to a high frequency above the audible range. A decoder can then either output the beamformed audio signal or can use the hidden signal to generate a reduced wind noise audio signal that includes the non-beamformed audio in the low frequency range.

Abstract: An array of microphones placed on a mobile robot provides multiple channels of audio signals. A received set of audio signals is called an audio segment, which is divided into multiple frames. A phase analysis is performed on a frame of the signals from each pair of microphones. If both microphones are in an active state during the frame, a candidate angle is generated for each such pair of microphones. The result is a list of candidate angles for the frame. This list is processed to select a final candidate angle for the frame. The list of candidate angles is tracked over time to assist in the process of selecting the final candidate angle for an audio segment.

Abstract: The present invention provides methods and systems for digitally processing audio signals in two-channel audio systems and/or applications. In particular, the present invention includes a first filter structured to split a two-channel audio input signal into a low frequency signal and a higher frequency signal. A M/S splitter is then structured to split the higher frequency signal into a middle and a side signal. A detection module is then configured to create a detection signal from the middle signal, which is used in a compression module configured to modulate the side signal to create a gain-modulated side signal. A processing module is then structured to combine the low frequency signal, middle signal, and the gain-modulated side signal to form a final output signal.

Abstract: Dynamic adjustment of antenna selection period at a wireless communication device or user equipment having two or more antennas. The described approaches consider an antenna selection performance metric. The antenna selection performance metric may include both measurement overhead (capacity loss due to measurements) and antenna switching capacity gain (gain in capacity through switching antenna subsets). The antenna selection performance metric may further include an estimate of UE speed or an antenna selection change rate. The antenna selection period may be adjusted by increasing or decreasing the antenna selection period based on the antenna selection performance metric. Antenna selection may be enabled or disabled based on the antenna selection performance metric.

Abstract: A system for decoding a stereo multiplex signal, including one or more devices operable to convert the stereo multiplex signal into a sum signal (L+R) and a difference signal (L?R). The sum signal and the difference signal may be derived from a left stereo signal (L) and right stereo signal (R). The system may also be operable to determine a first transfer function HL(f) and a second transfer function (HR(f)) from the sum signal (L+R) and the difference signal (L?R). Further, the system may be operable to filter the sum signal (L+R) according to the first transfer function to provide the left stereo signal (L), and filter the sum signal (L+R) according to the second transfer function to provide the right stereo signal (R).

Abstract: According to an embodiment, the control unit produces a first control signal and a second control signal. The first control signal is to set an intermittent reception period. The second control signal is to control a start-up of reception of a wireless signal. When the reception processing is implemented during the intermittent reception period, the RF unit stops in accordance with an instruction of the control unit after the reception processing signal is outputted, the intermittent reception period is awoken. When the reception processing exceeds the intermittent reception period, the RF unit continues the reception, and stops in accordance with the instruction of the control unit after the reception processing signal is outputted.

Abstract: A processing unit for processing a multi-channel audio signal has a delay element (40) for delaying an FM sum signal (sum) and a converter arrangement (T) for converting an FM difference signal (diff) and a noise signal (diffnoise) to the frequency domain. Frequency-based noise suppression is used to derive a de-noised frequency-domain difference signal using a gain function which is limited to a maximal and a minimal value. This is then converted to the time domain, and the first and second audio signals are calculated from a delayed sum signal (sum2) and the de-noised time domain difference signal (diff2).

Abstract: Spatially selective augmentation of a multichannel audio signal is described.

Abstract: A system and method for switching a mode of a receiver between a monophonic and stereophonic mode is disclosed. The method can include generating a residual signal indicative of a noise level associated with a pilot energy signal, calculating a residual block energy level of the residual signal over an observation interval, generating a monitor signal based on a number of times the pilot energy signal is less than a pilot energy threshold during the observation interval and switching the mode of the receiver based on the residual block energy level and the monitor signal.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also either adjusts the frequency response of the anti-noise signal with respect to the reference microphone signal, and/or by adjusting the response of the adaptive filter independent of the adaptation provided by the reference microphone signal.

Abstract: A receiver may estimate a channel condition based on a difference between a pilot energy signal and a filtered pilot energy signal. The receiver may further switch to a stereophonic mode when both the difference and an energy level of the pilot energy signal satisfy a predetermined requirement of the receiver, and switch to a monophonic mode when either the difference or the energy level of the pilot energy signal fail to satisfy the predetermined requirement of the receiver.

Abstract: Several circuits and methods implemented to perform signal quality estimation and control are disclosed. In an embodiment, a method of signal quality estimation includes generating a demodulated signal associated with a radio signal. Information associated with a quality of the demodulated signal is accessed. Further, a value of radio frequency signal-to-noise ratio (RF-SNR) for the radio signal based on the information is estimated. Estimating the value of RF-SNR facilitates in signal quality estimation of the radio signal and control of the demodulated signal.

Abstract: An audio receiver may include a mono/stereo detector that causes the audio receiver to output either a monophonic or a stereophonic signal based on a difference between a pilot energy signal and a filtered pilot energy signal. The audio receiver includes a filter that filters the pilot energy signal to generate a filtered pilot energy signal, wherein a variable leakage factor associated with the filter is used to minimize a noise level of the filtered pilot energy signal and to reduce a response time of the audio receiver.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: A signal processing apparatus (100) comprising a noise detector (102) configured to: receive a stream of information representative of a stream of audio signal samples (112); and detect samples in the received stream of information (112) that are distorted by impulse noise in order to generate a noise detection signal (114), wherein the noise detection signal (114) also identifies preceding and succeeding samples that are undistorted. The signal processing apparatus (100) also comprises a processor (104) configured to replace distorted samples in the received stream of information with composite predicted values (426; 526) to provide a reconstructed stream of audio signal samples (116).

Abstract: A method for determining leakage factors or adaptation rates, or both, for adaptive filters in an active noise reduction system. The leakage factor or adaptation rate, or both, may vary depending on a parameter of an input reference signal. The parameter may include one or more of reference signal input frequency, rate of change of reference input signal frequency, if a predetermined triggering condition exits, or if a predetermined event has occurred.

Abstract: The invention relates to a noise canceling audio transmitting/receiving device; a stereo headset with an integrated array of microphones utilizing an adaptive beam forming algorithm. The invention also relates to a method of using an adaptive beam forming algorithm that may be incorporated into a stereo headset. The sensor array used herein has adaptive filtering capabilities.

Abstract: Method for reducing noise included in a stereo reproduction signal derived from a stereo input signal characterized by the steps of: *varying stereo channel separation of said stereo reproduction signal with frequency within the frequency range of the stereo input signal in accordance with the frequency response of a filter selectivity located around a center frequency to obtain a stereo channel separation peak value at said center frequency; *at a continuing increase of said noise, decreasing the bandwidth of said filter selectivity to a pre-determined non-zero bandwidth while maintaining the channel separation within the bandwidth of said filter selectivity substantially at said channel separation peak value.

Abstract: The invention provides a noise removal device and method capable of more proper interpolation on an input signal. The noise removal device is for removing noise in an input signal and includes: a noise detector detecting noise in an IF signal and outputting a noise detection signal; an interpolation controller determining a period and amount of interpolation for noise correction processing, based on the IF signal and the noise detection signal; and a noise gate processor performing the noise correction processing on the IF signal, based on the interpolation period and amount supplied from the interpolation controller. The interpolation controller sets a predetermined first interpolation period, based on a first noise detection signal inputted from the noise detector, and redefines a second interpolation period longer than the first interpolation period when a second noise detection signal is detected within the first interpolation period.

Abstract: Implementations related to echo cancellation are depicted and described herein.

Abstract: A device for processing an audio signal, wherein the device comprises a detection unit adapted for detecting a widening state of the audio signal, and a widening modification unit adapted for modifying a widening characteristic of the audio signal depending on the detected widening state of the audio signal.

Abstract: A method for determining leakage factors or adaptation rates, or both, for adaptive filters in an active noise reduction system. The leakage factor or adaptation rate, or both, may vary depending on a parameter of an input reference signal. The parameter may include one or more of reference signal input frequency, rate of change of reference input signal frequency, if a predetermined triggering condition exits, or if a predetermined event has occurred.

Abstract: An FM audio receiver can include a mono/stereo detector that causes the audio receiver to output either a monophonic or a stereophonic signal based on a received pilot tone energy. An accurate operation of the receiver, including but not limited to correct decoding of monophonic/stereophonic reception, can be based on the receiver operating with the same rated maximum system deviation (RMSD) as the received signal itself. Aspects of the disclosure describe a system and method of detecting and matching a receiver's RMSD to that of a received signal by demodulating a carrier bearing a an input signal over a first bandwidth, extracting a pilot energy signal from the input signal, and demodulating the carrier bearing the input signal over a second bandwidth if the pilot energy signal is within a pilot energy range for a first predetermined amount of time.

Abstract: A radio receiver may determine whether a received frequency modulated radio signal includes interference from an adjacent channel. The receiver may compare a frequency deviation above the center frequency of the received signal to a frequency deviation below the center frequency of the received signal. The receiver determines that the radio signal is affected by adjacent channel noise when the frequency deviation below the center frequency is substantially different from the frequency deviation above the center frequency. Alternatively, the presence of noise from an adjacent channel may be detected by comparing a positive amplitude and a negative amplitude of a demodulated version of the frequency modulated radio signal. In this alternative, the receiver determines that the radio signal is affected by adjacent channel noise when the extent of the positive amplitude is substantially different from the extent of the negative amplitude.

Abstract: Apparatus to reduce noise in a stereo FM broadcast received via an antenna, the apparatus comprising: a frequency translator configured to translate the received stereo FM broadcast to an intermediate carrier frequency, a demodulation unit configured to demodulate the translated FM signal so as to form left-plus-right and left-minus-right AM signals, a filter configured to form a filtered signal by filtering one of the AM signals, so as to suppress a sub-band of that signal containing unwanted tones, a summing unit for summing the filtered signal and the other of the left-plus-right and left-minus-right signals to produce a stereo audio signal.

Abstract: A system and method for switching a mode of a receiver between a monophonic and stereophonic mode is disclosed. The method can include generating a residual signal indicative of a noise level associated with a pilot energy signal, calculating a residual block energy level of the residual signal over an observation interval, generating a monitor signal based on a number of times the pilot energy signal is less than a pilot energy threshold during the observation interval and switching the mode of the receiver based on the residual block energy level and the monitor signal.

Abstract: A receiver may estimate a channel condition based on a difference between a pilot energy signal and a filtered pilot energy signal. The receiver may further switch to a stereophonic mode when both the difference and an energy level of the pilot energy signal satisfy a predetermined requirement of the receiver, and switch to a monophonic mode when either the difference or the energy level of the pilot energy signal fail to satisfy the predetermined requirement of the receiver.

Abstract: A technique for reducing inherent noise in a real time digital audio wireless system. The technique allows for low power stereo analog to digital and digital to analog converters to be used in a battery operated real time digital audio wireless applications. The audio signal is sent to the left and right channels of an analog to digital converter with a preceding audio amp of different gain at the input to each channel. The signals are simultaneously digitized then combined to generate a new digital value resulting in noise reduction at lower amplitudes. This new data word is then transmitted via radio waves to a digital audio receiver. Once the data is received it is output simultaneously to a stereo digital to analog converter. The digital to analog conversion of both left and right channels is then summed using a resistor network and an audio amplifier. The audio amplifier output is then reduced to create further reduction of noise at all amplitudes.

Abstract: An audio receiver may include a mono/stereo detector that causes the audio receiver to output either a monophonic or a stereophonic signal based on a difference between a pilot energy signal and a filtered pilot energy signal. The audio receiver includes a filter that filters the pilot energy signal to generate a filtered pilot energy signal, wherein a variable leakage factor associated with the filter is used to minimize a noise level of the filtered pilot energy signal and to reduce a response time of the audio receiver.

Abstract: A method for delivering a message to a recipient in an environment with ambient noise includes the steps of recording the ambient noise in the environment at a certain time interval, analyzing the recorded ambient noise to obtain an average power Pnoise or a RMS amplitude Anoise of the ambient noise, providing a predetermined desired SNRdesired, calculating an average signal power Psignal or a RMS amplitude Asignal of the message to be delivered based on the Pnoise or Anoise and the desired SNRdesired, and adjusting a volume of the message to be delivered according to the Psignal or Asignal. Alternatively, the actual SNRactual will be computed and the message will be repeated if the SNRactual falls below the SNRmin. Systems for delivering a message to a recipient in an environment with ambient noise and computer-readable media having computer-executable instructions for carrying out the methods are also provided.

Abstract: A receiving apparatus 10 includes a stereo blend control unit 21 for carrying out a stereo blend process according to the reception state of a broadcast wave, a pilot detecting unit 18 for detecting a pilot signal included in the broadcast wave, and a high cut control unit 22 for carrying out high cut control according to the reception state of the broadcast wave and for carrying out the high cut control when the pilot signal is not detected by the pilot detecting unit.

Abstract: An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.

Abstract: Disclosed is a stereo signal processing apparatus, in particular for a digital BTSC television decoder, comprising a sub-channel signal processing section which comprises an input for inputting an input sub-channel signal, a DBX expanding means and an output for outputting an output sub-channel signal. The particularity of the present invention is that said sub-channel signal processing section further comprises a phase error compensating means for correcting a phase error of said DBX expanding means so that at said output of said sub-channel signal processing section the phase of the output sub-channel signal is essentially constant or zero over a predetermined frequency range.

Abstract: An FM tuner that has a multipath reception detection unit and an adjacent-channel interference detection unit while reducing expansion of the scale of the circuit is available. A control signal for a stereo separation level is generated on the basis of a signal SM-DC corresponding to a reception field strength and of a signal SMP corresponding to a multipath noise component. The SMP is extracted from a fluctuation component signal SM-AC from an S-meter circuit 92 by a multipath noise extraction circuit 104. The multipath noise extraction circuit 104 used for control of the stereo separation level is shared as a portion of the circuitry for detecting the multipath signal reception state. A multipath reception detection unit 108 compares the SMP with a threshold value to detect a multipath reception state.

Abstract: An exemplary ambient noise filtering method includes: sampling ambient noise signals to obtain multiple groups of sampled signals; analyzing each group of the sampled signals in each sampling period to determine ambient noise frequency band(s) in each sampling period, wherein the ambient noise frequency band is a frequency band where the ambient noise distributes; determining main noise frequency band(s) by statistically analyzing all the ambient noise frequency bands, wherein the main noise frequency band is a frequency band where the ambient noise concentrates and mainly distributes; and filtering the ambient noise out of signals generated during a communication by filtering the determined main noise frequency band(s) if the main noise frequency band(s) does not have a same range as that of a speech frequency band. Related exemplary mobile phone performing the method is also provided.

Abstract: A method of denoise a stereo signal comprising a stereo sum signal and a stereo difference signal, performs a frequency selective stereo to mono blending based on the masking effect of the human auditory system. Therefore, a stereo signal noise reducer, comprising a first filter bank (1) to split the stereo difference signal (l?r) into a plurality of subbands, respective first multipliers (CO, . . . , CN) to weight each of the subbands of the stereo difference signal with a respective corresponding control signal (CO, . . . , CN), and a first adder (3) to sum all weighted subbands of the stereo difference signal (l?r) to build a frequency selective weighted stereo difference signal (diff), within which a number and width of the subbands obtained via the first filter bank (1) are choosen according to the properties of the human auditory system, further comprises a weighting factor determination unit which determines a respective control signal (CO, . . .

Abstract: It is an object of the present invention to provide an improved noise removal device having an improved noise removal precision. A second noise extracting unit extracts a noise of an IF signal, and outputs a second noise signal which is then smoothed by a second noise smoothing unit to produce a second noise-smoothed signal. A noise hold control signal generating unit compares the second noise-smoothed signal with the second noise signal to produce a noise hold control signal indicating an occurring position and an occurring period of the noise of the IF signal. A first noise extracting unit extracts a noise of an FM detection signal, and outputs a first noise signal. A noise hold unit holds the first noise signal in accordance with the noise hold control signal to produce a held noise signal which is then smoothed by a first noise smoothing unit to produce a first noise-smoothed signal.