Patent application number | Description | Published |
20080262849 | VOICE CONTROL SYSTEM - A voice control system allows a user to control a device through voice commands. The voice control system includes a speech recognition unit that receives a control signal from a mobile device and a speech signal from a user. The speech recognition unit configures speech recognition settings in response to the control signal to improve speech recognition. | 10-23-2008 |
20080285772 | ACOUSTIC LOCALIZATION OF A SPEAKER - A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone signal. A processor determines the speaker's direction relative to the microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals. | 11-20-2008 |
20080292108 | DEREVERBERATION SYSTEM FOR USE IN A SIGNAL PROCESSING APPARATUS - A system used in a loudspeaker-room-microphone environment includes a microphone signal partitioner that divides a signal from a microphone into one or more divided portions. A reverberation energy estimator estimates reverberation energy in some of the divided portions of the microphone signal based on a loudspeaker signal. The estimated reverberation energy is processed to generate a dereverberated output signal. | 11-27-2008 |
20080298602 | SYSTEM FOR PROCESSING MICROPHONE SIGNALS TO PROVIDE AN OUTPUT SIGNAL WITH REDUCED INTERFERENCE - A system reduces noise or other external signals that may affect communication. A device converts sound from two or more microphones into an operational signal. Based on one or both signals, a beamformer generates an intermediate signal. Reflected or other undesired signals may be estimated or measured by an echo canceller. Interference may be measured or estimated by processing the echo-reduced signal or estimate by a blocking matrix. An interference canceller may reduce the interference that may modify or disrupt a signal based on the output of the blocking matrix and the intermediate signal. | 12-04-2008 |
20080304679 | SYSTEM FOR PROCESSING AN ACOUSTIC INPUT SIGNAL TO PROVIDE AN OUTPUT SIGNAL WITH REDUCED NOISE - An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below. | 12-11-2008 |
20090063143 | SYSTEM FOR SPEECH SIGNAL ENHANCEMENT IN A NOISY ENVIRONMENT THROUGH CORRECTIVE ADJUSTMENT OF SPECTRAL NOISE POWER DENSITY ESTIMATIONS - A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal. | 03-05-2009 |
20090067642 | NOISE REDUCTION THROUGH SPATIAL SELECTIVITY AND FILTERING - A signal processor uses input devices to detect speech or aural signals. Through a programmable set of weights and/or time delays (or phasing) the output of the input devices may be processed to yield a combined signal. The noise contributions of some or each of the outputs of the input devices may be estimated by a circuit element or a controller that processes the outputs of the respective input devices to yield power densities. A short-term measure or estimate of the noise contribution of the respective outputs of the input devices may be obtained by processing the power densities of some or each of the outputs of the respective input devices. Based on the short-term measure or estimate, the noise contribution of the combined signal may be estimated to enhance the combined signal when processed further. An enhancement device or post-filter may reduce noise more effectively and yield robust speech based on the estimated noise contribution of the combined signal. | 03-12-2009 |
20090089065 | ADJUSTING OR SETTING VEHICLE ELEMENTS THROUGH SPEECH CONTROL - A speech processing device includes an automotive device that filters data that is sent and received across an in-vehicle bus. The device selectively acquires vehicle data related to a user settings or adjustments of an in-vehicle system. An interface acquires the selected vehicle data from one or more in-vehicle sensors in response to a user's articulation of a first code phrase. A memory stores the selected vehicle data with unique identifying data associated with a user. The unique identifying data establishes a connection between the selected vehicle data and the user when a second code phrase is articulated by the user. A data interface provides access to the selected vehicle data and relationship data retained in the memory and enables the processing of the data to customize the in-vehicle system. The data interface is responsive to a user's articulation of a third code phrase to process the selected vehicle data that enables the setting or adjustment of the in-vehicle system. | 04-02-2009 |
20090117948 | METHOD FOR DEREVERBERATION OF AN ACOUSTIC SIGNAL - A method is provided for estimating a reverberation signal component of an acoustic signal detected by a microphone where the acoustic signal is comprised of a direct sound component and a reverberation signal component. A method for dereverberation of an acoustic signal is further provided. | 05-07-2009 |
20090125311 | VEHICULAR VOICE CONTROL SYSTEM - A vehicular voice control system includes a first and a second microphone located on a vehicle external to a vehicle cabin. The microphones receive audio signals from an audio source external to the vehicle and generate microphone output signals. A signal processor processes the microphone output signals, generates a processed signal, and determines a location of the audio source. A speech recognition system receives the processed signal and obtains a recognition result. A controller controls one or more vehicular elements based on the recognition result and the determined location of the audio source. | 05-14-2009 |
20090192796 | FILTERING OF BEAMFORMED SPEECH SIGNALS - The invention relates to speech signal processing that detects a speech signal from more than one microphone and obtains microphone signals that are processed by a beamformer to obtain a beamformed signal that is post-filtered signal with a filter that employs adaptable filter weights to obtain an enhanced beamformed signal with the post-filter adapting the filter weights with previously learned filter weights. | 07-30-2009 |
20090254342 | DETECTING BARGE-IN IN A SPEECH DIALOGUE SYSTEM - A method for detecting barge-in in a speech dialogue system comprising determining whether a speech prompt is output by the speech dialogue system, and detecting whether speech activity is present in an input signal based on a time-varying sensitivity threshold of a speech activity detector and/or based on speaker information, where the sensitivity threshold is increased if output of a speech prompt is determined and decreased if no output of a speech prompt is determined. If speech activity is detected in the input signal, the speech prompt may be interrupted or faded out. A speech dialogue system configured to detect barge-in is also disclosed. | 10-08-2009 |
20100014690 | Beamforming Pre-Processing for Speaker Localization - Embodiments of the present invention relate to methods, systems, and computer program products for signal processing. A first plurality of microphone signals is obtained by a first microphone array. A second plurality of microphone signals is obtained by a second microphone array different from the first microphone array. The first plurality of microphone signals is beamformed by a first beamformer comprising beamforming weights to obtain a first beamformed signal. The second plurality of microphone signals is beamformed by a second beamformer comprising the same beamforming weights as the first beamformer to obtain a second beamformed signal. The beamforming weights are adjusted such that the power density of echo components and/or noise components present in the first and second plurality of microphone signals is substantially reduced. | 01-21-2010 |
20100054085 | Method and Device for Locating a Sound Source - A method of locating a sound source based on sound received at an array of microphones comprises the steps of determining a correlation function of signals provided by microphones of the array and establishing a direction in which the sound source is located based on at least one eigenvector of a matrix having matrix elements which are determined based on the correlation function. The correlation function has first and second frequency components associated with a first and second frequency band, respectively. The first frequency component is determined based on signals from microphones having a first distance, and the second frequency component is determined based on signals from microphones having a second distance different from the first distance. | 03-04-2010 |
20100150364 | Method for Determining a Time Delay for Time Delay Compensation - The invention provides a computer-implemented method for determining a time delay for time delay compensation of a microphone signal from a microphone array in a beamformer arrangement. For a given time, an instantaneous estimate of a position of a wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source is determined. The computer system then determines whether the instantaneous estimate deviates from a preset estimate of a position of the wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source according to a predetermined criterion. The predetermined criterion comprises a check whether the instantaneous estimate deviates from the preset estimate by at least a predetermined deviation threshold. If the predetermined criterion is fulfilled, the instantaneous estimate for the given time is set by the computer system as the preset estimate, and the computer system determines the time delay for time delay compensation of the microphone signal based on the instantaneous estimate. | 06-17-2010 |
20100150375 | Determination of the Coherence of Audio Signals - Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals. The first and the second microphone signals are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the first and second filtered signals. | 06-17-2010 |
20100215184 | Method for Determining a Set of Filter Coefficients for an Acoustic Echo Compensator - The invention provides a method for determining a set of filter coefficients for an acoustic echo compensator in a beamformer arrangement. The acoustic echo compensator compensates for echoes within the beamformed signal. A plurality of sets of filter coefficients for the acoustic echo compensator is provided. Each set of filter coefficients corresponds to one of a predetermined number of steering directions of the beamformer arrangement. The predetermined number of steering directions is equal to or greater than the number of microphones in the microphone array. For a current steering direction, a current set of filter coefficients for the acoustic echo compensator is determined based on the provided sets of filter coefficients. | 08-26-2010 |
20100246844 | Method for Determining a Signal Component for Reducing Noise in an Input Signal - The invention provides a method for determining a signal component for reducing noise in an input signal, which comprises a noise component, comprising the steps of: estimating the noise component in the input signal, estimating a reverberation component in the noise component, and removing the estimated reverberation component from the estimated noise component to obtain a modified estimate of the noise component. | 09-30-2010 |
20100246851 | Method for Determining a Noise Reference Signal for Noise Compensation and/or Noise Reduction - The invention provides a method for determining a noise reference signal for noise compensation and/or noise reduction. A first audio signal on a first signal path and a second audio signal on a second signal path are received. The first audio signal is filtered using a first adaptive filter to obtain a first filtered audio signal. The second audio signal is filtered using a second adaptive filter to obtain a second filtered audio signal. The first and the second filtered audio signal are combined to obtain the noise reference signal. The first and the second adaptive filter are adapted such as to minimize a wanted signal component in the noise reference signal. | 09-30-2010 |
20110019835 | Speaker Localization - The present invention relates to a method for localizing a sound source, in particular, a human speaker, comprising detecting sound generated by the sound source by means of a microphone array comprising more than two microphones and obtaining microphone signals, one for each of the microphones, selecting from the microphone signals a pair of microphone signals for a predetermined frequency range based on the distance of the microphones to each other and estimating the angle of the incidence of the sound on the microphone array based on the selected pair of microphone signals. | 01-27-2011 |
20110026732 | System for Detecting and Reducing Noise via a Microphone Array - A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment. | 02-03-2011 |
20120009878 | Vehicle Communication System - A vehicle communication system detects the presence of a passenger wearable communication device. The system receives audio signals from multiple sources inside or outside of a vehicle. The system processes the signals before routing the signals to multiple destinations. The destinations may include wearable personal communication devices, front and/or rear speakers, and/or a remote mobile device. | 01-12-2012 |
20120106749 | Microphone Non-Uniformity Compensation System - A microphone compensation system responds to changes in the characteristics of individual microphones in an array of microphones. The microphone compensation system provides a communication system with consistent performance despite microphone aging, widely varying environmental conditions, and other factors that alter the characteristics of the microphones. Furthermore, lengthy, complex, and costly measurement and analysis phases for determining initial settings for filters in the communication system are eliminated. | 05-03-2012 |
20120278041 | Measurement and Tuning of Hands Free Telephone Systems - An arrangement is described for measuring performance characteristics of a hands free telephone system. There is a measurement system which is coupleable over a telephone audio interface directly to the hands free telephone system for measuring the performance characteristics. | 11-01-2012 |
20120294118 | Acoustic Localization of a Speaker - A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals. | 11-22-2012 |
20130024196 | SYSTEMS AND METHODS FOR USING A MOBILE DEVICE TO DELIVER SPEECH WITH SPEAKER IDENTIFICATION - Systems, methods, and apparatus for using at least one mobile device to receive a representation of at least one audio signal. In some embodiments, the at least one audio signal comprises speech of at least one of a plurality of first participants of a meeting, the plurality of first participants participating in the meeting from a first location, and the at least one audio signal may be audibly rendered to at least one second participant of the meeting at a second location different from the first location. In some embodiments, the at least one mobile device may further receive an indication of an identity of a leading speaker of the speech in the at least one audio signal, the leading speaker being identified from among the plurality of first participants, and may render the identity of the leading speaker to the at least one second participant. | 01-24-2013 |
20130136271 | Method for Determining a Noise Reference Signal for Noise Compensation and/or Noise Reduction - The invention provides a method for determining a noise reference signal for noise compensation and/or noise reduction. A first audio signal on a first signal path and a second audio signal on a second signal path are received. The first audio signal is filtered using a first adaptive filter to obtain a first filtered audio signal. The second audio signal is filtered using a second adaptive filter to obtain a second filtered audio signal. The first and the second filtered audio signal are combined to obtain the noise reference signal. The first and the second adaptive filter are adapted such as to minimize a wanted signal component in the noise reference signal. | 05-30-2013 |
20130179163 | IN-CAR COMMUNICATION SYSTEM FOR MULTIPLE ACOUSTIC ZONES - An In-Car Communication (ICC) system supports the communication paths within a car by receiving the speech signals of a speaking passenger and playing it back for one or more listening passengers. Signal processing tasks are split into a microphone related part and into a loudspeaker related part. A sound processing system suitable for use in a vehicle having multiple acoustic zones includes a plurality of microphone In-Car Communication (Mic-ICC) instances coupled and a plurality of loudspeaker In-Car Communication (Ls-ICC) instances. The system further includes a dynamic audio routing matrix with a controller and coupled to the Mic-ICC instances, a mixer coupled to the plurality of Mic-ICC instances and a distributor coupled to the Ls-ICC instances. | 07-11-2013 |
20130251159 | System for Detecting and Reducing Noise via a Microphone Array - A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment. | 09-26-2013 |
20130325458 | DYNAMIC MICROPHONE SIGNAL MIXER - A system and method of signal combining that supports different speakers in a noisy environment is provided. Particularly for deviations in the noise characteristics among the channels, various embodiments ensure a smooth transition of the background noise at speaker changes. A modified noise reduction (NR) may achieve equivalent background noise characteristics for all channels by applying a dynamic, channel specific, and frequency dependent maximum attenuation. The reference characteristics for adjusting the background noise may be specified by the dominant speaker channel. In various embodiments, an automatic gain control (AGC) with a dynamic target level may ensure similar speech signal levels in all channels. | 12-05-2013 |
20140105338 | LOW-DELAY FILTERING - A method of frequency-domain filtering is provided that includes a plurality of filters, the plurality of filters including at least one constrained filter(s) | 04-17-2014 |
20140153740 | BEAMFORMING PRE-PROCESSING FOR SPEAKER LOCALIZATION - Methods and apparatus to beamform a first plurality of microphone signals using at least one beamforming weight to obtain a first beamformed signal, beamform a second plurality of microphone signals using the at least one beamforming weight to obtain a second beamformed signal, and adjust the at least one beam forming weight so that the power density of at least one perturbation component present in the first or the second plurality of microphone signals is reduced. | 06-05-2014 |
20140247953 | SPEAKER LOCALIZATION - Methods and apparatus for determining phase shift information between the first and second microphone signals for a sound signal, and determining an angle of incidence of the sound in relation to the first and second positions of the first and second microphones from the phase shift information of a band-limited test signal received by the first and second microphones for a frequency range of interest. | 09-04-2014 |
20140307883 | METHOD FOR DETERMINING A SET OF FILTER COEFFICIENTS FOR AN ACOUSTIC ECHO COMPENSATOR - Methods and apparatus for beamforming and performing echo compensation for the beamformed signal with an echo canceller including calculating a set of filter coefficients as an estimate for a new steering direction without a complete adaptation of the echo canceller. | 10-16-2014 |
20140337016 | Speech Signal Enhancement Using Visual Information - 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. | 11-13-2014 |
20150046157 | User Dedicated Automatic Speech Recognition - A multi-mode voice controlled user interface is described. The user interface is adapted to conduct a speech dialog with one or more possible speakers and includes a broad listening mode which accepts speech inputs from the possible speakers without spatial filtering, and a selective listening mode which limits speech inputs to a specific speaker using spatial filtering. The user interface switches listening modes in response to one or more switching cues. | 02-12-2015 |