| Entries |
| Document | Title | Date |
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| 20080205661 | SYSTEM AND METHOD FOR CANCELLING ECHO - A system and a method for canceling an echo is disclosed. In accordance with the system and the method, a plurality of independently and variably delayed adaptive algorithm blocks are selectively applied to a delayed feedback signal to generate a plurality of echo components in parallel, thereby canceling the echo component from an input signal. | 08-28-2008 |
| 20080219463 | ACOUSTIC ECHO CANCELLATION SYSTEM - An embodiment of an acoustic echo cancellation system is disclosed. The system comprises an echo cancellation unit, a second filter and a subtraction unit. The echo cancellation unit comprises a first attenuator, a first filter and a first subtractor. The first attenuator has a first down-scaling factor for attenuating a first signal. The first filter generates a first echo signal estimate based on the attenuated first signal. The first subtractor generates a third signal by subtracting the first echo signal estimate from a second signal. The second filter generates a second echo signal estimate based on the first signal. The subtraction unit subtracts the second echo signal estimate from the third signal. | 09-11-2008 |
| 20080226090 | Oscillation/Echo Canceller System - An oscillation/echo canceller system ( | 09-18-2008 |
| 20080247557 | Information Processing Apparatus and Program - According to one embodiment, a signal processing apparatus includes a speaker configured to output the received input signal on which a delay detection signal which has a frequency component of an inaudible frequency on a received input signal is superposed to an acoustic space, an extracting section configured to extract the delay detection signal from the sending input signal outputted from microphone configured to collect sound in the acoustic space a calculating section configured to calculate a delay time between the received input signal and an acoustic echo component contained in the sending input signal, a delay section configured to delay the received input signal by a time corresponding to the delay time and generate a delayed received input signal, and an echo suppression processing section configured to suppress the acoustic echo component contained in the sending input signal by use of the delayed received input signal. | 10-09-2008 |
| 20080247558 | Robust and Efficient Frequency-Domain Decorrelation Method - An audio signal is processed by transforming the signal into a frequency domain representation having a plurality of frequency subbands. A decorrelated signal is derived from the frequency domain representation using a phase rotation. | 10-09-2008 |
| 20080247559 | ELECTRICITY ECHO CANCELLATION DEVICE AND METHOD - An electricity echo cancellation device applied at a terminal includes: an input buffer memory module, a network echo delay computation module and an adaptive filtering module. The adaptive filtering module includes an adaptive filter, a subtracter and a dual-ended voice detection module. An electricity echo cancellation method includes: calculating a network echo delay according to relevant information of an RTCP packet transmitted from the network; and dynamically adjusting a terminal input signal to be adaptively filtered according to the network echo delay. The present invention ensures the electricity echo cancellation effect at the final user end on the whole, and improves the effectiveness of electricity echo cancellation. Meanwhile method of the present invention can be realized with software, thus avoiding influences of hardware memory restricts on the echo cancellation effect. In addition, the present invention only needs a single-point deployment, and thus the cost is saved. | 10-09-2008 |
| 20080260172 | Echo Canceller and Speech Processing Apparatus - An echo canceller used for hands-free communication systems in which hands-free communication is performed by using a speaker and a microphone is disclosed. The echo canceller includes a step size control unit calculating a step size value in an adaptive filter and an adaptive filter unit estimating an echo component of a feedback path from an input signal to the feedback path by adaptively identifying an impulse response of the feedback path formed by an acoustical coupling and the like of the speaker and the microphone, and subtracting the echo component from an output signal from the feedback path, in which the step size control unit calculates a step size value by using an echo reduction amount defined based on the ratio between the output signal from the feedback path and a residual signal and outputs the value to the adaptive filter unit. | 10-23-2008 |
| 20080267420 | Methods and systems for reducing acoustic echoes in multichannel audio-communication systems - Various embodiments of the present invention are directed to adaptive real-time, acoustic echo cancellation methods and systems. One method embodiment of the present invention is directed to reducing acoustic echoes in microphone-digital signals transmitted from a first location to a second location. The first location includes a plurality of loudspeakers and microphones, each microphone produces one of the microphone-digital signals including sounds produced at the first location and acoustic echoes produced by the loudspeakers. The method includes determining approximate impulse responses, each of which corresponds to an echo path between the microphones and the loudspeakers. The method includes determining a plurality of approximate acoustic echoes, each approximate acoustic echo corresponds to convolving a digital signal played by one of the loudspeakers with a number of the approximate impulse responses. The acoustic echo in at least one of the microphone-digital signals is reduced based on the corresponding approximate acoustic echo. | 10-30-2008 |
| 20080292109 | Echo Detection - An echo detector includes means ( | 11-27-2008 |
| 20080298601 | Double Talk Detection Method Based On Spectral Acoustic Properties - A method of detecting double talk condition in hands free communication devices is disclosed. In general, the method in accordance with the teachings of this invention detects double talk conditions based on inherent frequency response differences between the transducers used and acoustical effect on the spectrum of the returned echo signal. An input signal from a far-end talker and an input signal from the output from an echo canceler are received by the detector. K spectral subbands are created for each input signal. From this K subbands q subbands are selected based on inherent frequency differences between the far-end transducer and a near-end transducer. The spectral echo residual power is estimated at each subband. The estimated spectral echo power and the output signal from the echo canceler for a selected subband are compared to a predetermined threshold. Based on this comparison, it is determined whether double talk conditions exist based on the comparison. | 12-04-2008 |
| 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 |
| 20080304675 | Acoustic Echo Canceller - An acoustic echo cancellation device ( | 12-11-2008 |
| 20080304676 | METHOD AND SYSTEM FOR CLEAR SIGNAL CAPTURE - A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control. | 12-11-2008 |
| 20080310643 | METHOD AND SYSTEM FOR CLEAR SIGNAL CAPTURE - A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control. | 12-18-2008 |
| 20080310644 | METHOD AND SYSTEM FOR CLEAR SIGNAL CAPTURE - A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control. | 12-18-2008 |
| 20090003615 | Audio System Providing For Filter Coefficient Copying | 01-01-2009 |
| 20090010444 | METHOD AND DEVICE FOR PERSONALIZED VOICE OPERATED CONTROL - An earpiece ( | 01-08-2009 |
| 20090010445 | ECHO SUPPRESSOR, ECHO SUPPRESSING METHOD, AND COMPUTER READABLE STORAGE MEDIUM - An apparatus is provided for suppressing an echo signal included in a measured signal corresponding to a measured sound. In the apparatus, the measured signal and a reference signal in a time domain are transformed into a frequency domain, and calculated for obtaining each value of a ratio and a correlation between the measured signal and the reference signal in the frequency domain. With executing a comparison of the values of the ratio and the correlation, a coefficient is derived, where a product of the coefficient and the measured sound in the frequency domain gives an estimated value of the echo signal. The echo in the measured signal is suppressed with subtracting the estimation of the echo signal from the measured signal, respectively in the frequency domain. | 01-08-2009 |
| 20090028354 | Echo Canceller Employing Dual-H Architecture Having Split Adaptive Gain Settings - An echo canceller circuit is set forth. The echo canceller circuit includes a digital filter having adaptive tap coefficients to simulate an echo response occurring during a call. The adaptive tap coefficients are updated during the call using a Means Squares process. A tap energy detector is also employed. The tap energy detector identifies and divides groups of taps having high energy from groups of taps having low energy. The high energy tap groups are smaller in number than the low energy tap groups. The high energy tap groups are adapted separately from the low energy tap groups using the Least Squares process. Still further, the high energy tap groups may be adapted using an adaptive gain constant a while the low energy tap groups are adapted using an adaptive gain constant a′, wherein a>a′. | 01-29-2009 |
| 20090028355 | Double-talk detector with accuracy and speed of detection improved and a method therefor - A double-talk detector finds an estimated power value of near end background noise based on a residual signal by a noise estimator; the average power of a transmitter input signal by a transmitter average power calculator; the average power of a receiver input signal by a receiver average power calculator; and an estimated echo path attenuation value through a predetermined echo path attenuation value estimating process based on the estimated power value of the near end background noise, the average power of the transmitter input signal and the average power of the receiver input signal by an attenuation value estimator. The double-talk detector detects a double-talk state based on the estimated echo path attenuation value, the average power of the transmitter input signal and the average power of the receiver input signal by a double-talk determiner to control update of the coefficient of an adaptive filter. | 01-29-2009 |
| 20090046866 | APPARATUS CAPABLE OF PERFORMING ACOUSTIC ECHO CANCELLATION AND A METHOD THEREOF - An apparatus capable of performing acoustic echo cancellation and a method thereof are provided. The apparatus comprises a mapping matrix, first and second speakers, first and second microphones, a reference generator, and a multi-channel acoustic echo canceller. The mapping matrix generates an output signal according to the first and second far end signals. The first and second speakers, coupled to the mapping matrix, play the output signal. The first and second microphones receive the first and second echo signals that are acoustically coupled from the first and second speakers to the first and second microphones, wherein the first and second echo signals are correlated to the output signal. The reference generator generates a reference signal linearly correlated to the output signal according to the first and second far end signals. The multi-channel acoustic echo canceller, coupled to the reference generator and the first and second microphones, filters the reference signal to generate the first and second filtered signals to be indicative of the estimated echo signals at the first and second microphones, subtracts the first filtered signal from the first echo signal to generate a first error signal, and subtracts the second filtered signal from the second echo signal to generate a second error signal, and then transmits the first and second error signals to a far end terminal. | 02-19-2009 |
| 20090052683 | Echo cancellation - Implementations related to echo cancellation are depicted and described herein. | 02-26-2009 |
| 20090052684 | AUDIO CONFERENCING APPARATUS - Microphones arranged in an array shape along a longitudinal direction are respectively formed in both the longitudinal side surfaces of a housing | 02-26-2009 |
| 20090067637 | Echo control retrofit - The present invention provides an echo control retrofit apparatus and method that advantageously reduces echo when communicating over packet-switched networks. An echo control retrofit apparatus, including an echo control circuit, is operably coupled between a headset or handset device and an audio source. The echo control circuit receives a sound signal from the audio source and a transmit signal from the headset or handset device and provides an adjusted sound signal to the audio source. Advantageously, a variety of existing headsets and handsets may be used in accordance with the present invention to provide reduced caller echo without the need to purchase new headsets or handsets. | 03-12-2009 |
| 20090086986 | EFFICIENT AUDIO SIGNAL PROCESSING IN THE SUB-BAND REGIME - A signal processing system enhances an audio signal. The audio signal is divided into audio sub-band signals. Some audio sub-band signals are excised. Other audio sub-band signals are processed to obtain enhanced audio sub-band signals. At least a portion of the excised audio sub-band signals are reconstructed. The reconstructed audio sub-band signals are synthesized with the enhanced audio sub-band signals to form an enhanced audio signal. | 04-02-2009 |
| 20090103743 | ECHO CANCELLER - An echo canceller includes a residual signal generation unit, a double talk detection unit, a nonlinear processor, a speech detection unit, and an input/output characteristic change unit. The residual signal generation unit generates a pseudo echo signal, and generates a residual signal by using the pseudo echo signal. The double talk detection unit detects the state of the transmission signal. The nonlinear processor attenuates the residual signal that has been inputted thereto to a signal level which is based on a predetermined input/output characteristic, and that outputs the attenuated residual signal. The speech detection unit detects whether or not speech is included in the reception signal. The input/output characteristic change unit changes the input/output characteristic of the nonlinear processor to a predetermined input/output characteristic when a single talk state has been detected at the double talk detection unit and speech has been detected at the speech detection unit. | 04-23-2009 |
| 20090110207 | Method and Apparatus for Speech Dereverberation Based On Probabilistic Models Of Source And Room Acoustics - Speech dereverberation is achieved by accepting an observed signal for initialization ( | 04-30-2009 |
| 20090123002 | System and method for providing step size control for subband affine projection filters for echo cancellation applications - A system and method for Acoustic Echo Cancellation. The system and method include a subband affine projection filter and a variable step size controller configured to cancel an estimated echo from a near-end signal. The system and method also include a divergence detector adapted to reset the subband affine projection filter in response to determining a divergence is occurring. Additionally, the system and method include a double talk detector adapted to transmit a signal to mask an output signal when double talk is detected. | 05-14-2009 |
| 20090147964 | APPARATUS AND METHOD FOR REMOVING AN ECHO SIGNAL IN A SIGNAL TRANSMISSION/RECEPTION APPARATUS OF A COMMUNICATION SYSTEM - A method and apparatus for removing an echo signal in a signal transmission/reception apparatus of a communication system are provided. A signal transmission/reception apparatus determines an echo channel impulse response using a reception signal, generates an echo signal removing coefficient using the echo channel impulse response, removes an echo signal from the reception signal using the echo signal removing coefficient, and transmits a signal in which the echo signal is removed. | 06-11-2009 |
| 20090154717 | Echo Suppressing Method and Apparatus | 06-18-2009 |
| 20090161884 | ETHERNET ISOLATOR FOR MICROPHONICS SECURITY AND METHOD THEREOF - A system and method for providing microphonic isolation on a transmission line. The transmission line has a first part and a second part. The first part of transmission line carries a data signal and a microphonic signal. The microphonic signal has frequencies that include those in a range of substantially 20 Hz to substantially 20 kHz. The system includes an isolation apparatus. The isolation apparatus has an input in electrical communication with a first part of the transmission line, an output in electrical communication with the second part of the transmission line, and a filter in electrical communication with the input and the output. The filter is arranged to substantially remove the microphonic signal received at the input from first part of transmission line and pass the data signal to the output. | 06-25-2009 |
| 20090169027 | ECHO SUPPRESSOR - It is an object of the present invention to provide an echo suppressor which is simple in construction in comparison with the conventional echo suppressor, and which can suppress echoes corresponding to the spatial transfer paths, and reduce the amount of calculations necessary to suppress echoes. The echo suppressor is operative under the condition that transfer functions corresponding to spatial transfer paths between two or more loudspeakers ( | 07-02-2009 |
| 20090185695 | METHOD AND SYSTEM FOR CLOCK DRIFT COMPENSATION - Different sampling rates between a playout unit and a capture unit are compensated for via a system, method and computer program product. The playout unit receives samples from a computational unit, and the capture unit sends samples to the computational unit. A playout FIFO buffer operates in a playout time domain, and a capture FIFO buffer operates in a capture time domain. The computational unit is synchronized to a common clock. A first relationship is calculated between the common clock and a playout fifo buffer read pointer, and a second relationship is calculated between the common clock and a capture FIFO buffer write pointer. For each sample in the playout time domain a corresponding sample in the samples from said computational unit is found and sent to the playout FIFO buffer. For each sample in the common clock time domain the corresponding sample in the capture time domain is found and sent to the computational unit. | 07-23-2009 |
| 20090190769 | SOUND QUALITY BY INTELLIGENTLY SELECTING BETWEEN SIGNALS FROM A PLURALITY OF MICROPHONES - Sound signal reception is improved by utilizing a plurality of microphones to capture sound signals which are then weighed to dynamically adjust signal quality. A first sound signal and a second sound signal are obtained from first and second microphones, respectively, where the first and second sound signals originate from one or more sound sources. A first signal characteristic (e.g., signal power, signal signal-to-noise ratio, etc.) is obtained for the first sound signal and a second signal characteristic is obtained for the second sound signal. The first and second sound signals are weighed or scaled based on their respective first and second signal characteristics. The weighed first and second sound signals are then combined to obtain an output sound signal. | 07-30-2009 |
| 20090214048 | HARMONIC DISTORTION RESIDUAL ECHO SUPPRESSION - Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal. | 08-27-2009 |
| 20090238373 | System and method for envelope-based acoustic echo cancellation - Systems and methods for envelope-based acoustic echo cancellation in a communication device are provided. In exemplary embodiments, a primary acoustic signal is received via a microphone of the communication device, and a far-end signal is received via a receiver. Frequency analysis is performed on the primary acoustic signal and the far-end acoustic signal to obtain frequency sub-bands. An echo gain mask based on magnitude envelopes of the primary and far-end acoustic signals for each frequency sub-band is generated. A noise gain mask based on at least the primary acoustic signal for each frequency sub-band may also be generated. A combination of the echo gain mask and noise gain mask may then be applied to the primary acoustic signal to generate a masked signal. The masked signal is then output. | 09-24-2009 |
| 20090245527 | LINEAR FULL DUPLEX SYSTEM AND METHOD FOR ACOUSTIC ECHO CANCELLATION - A linear full duplex system and method for acoustic echo cancellation is disclosed. In one embodiment, a method includes calculating a residual echo after subtraction of an echo estimate from a near end signal associated with a communication, refining a far end and a residual signal associated with the communication, updating, based on the far end signal, the echo estimate of an echo associated with the communication, subtracting the echo from the near end signal based on the echo estimate to cancel the echo associated with the communication, updating, based on the refined far end and refined residual signal, the adaptive filter module used for echo estimation, and detecting a steady state and, during the steady state, dynamically detecting internal substates and switching between the internal substates and detecting and managing double talk associated with the communication. | 10-01-2009 |
| 20090245528 | METHOD FOR REDUCING ECHO AND RELATED ECHO REDUCING DEVICE AND VOICE APPARATUS THEREOF - A method for reducing echo includes detecting whether an output sound volume is greater than a threshold value, and setting an input sensitivity from a first designated sensitivity value to a second designated sensitivity value when the output sound volume is detected to be greater than the threshold value. The method further includes detecting whether an interrupt signal is received, determining whether the interrupt signal is triggered by detecting that the output sound volume is greater than the threshold value when receiving the interrupt signal, detecting whether the input sensitivity is the second designated sensitivity value when determining that the interrupt signal is triggered by detecting that the output sound volume is greater than the threshold value, and setting the input sensitivity as the second designated sensitivity value when detecting that the input sensitivity is not the second designated sensitivity value. | 10-01-2009 |
| 20090252343 | INTEGRATED LATENCY DETECTION AND ECHO CANCELLATION - In an audio system having a microphone, a speaker coupled to a source of audio output, and an echo canceller coupled to the speaker and microphone, latency between the source of audio output and the speaker may be compensated in echo cancellation performed by the echo canceller. The echo canceller may use a reference signal derived from a signal from the source of audio output in echo cancellation. The latency may be compensated by measuring the latency between the signal from the source of audio output and the speaker, determining a delay amount from the latency, delaying the reference signal by the delay amount to produce a delayed reference signal, and using the delayed reference signal as the reference signal in the echo canceller. | 10-08-2009 |
| 20090262950 | MULTI-CHANNEL ACOUSTIC ECHO CANCELLATION SYSTEM AND METHOD - Techniques for multi-channel acoustic echo cancellation include adaptive filtering. An adaptive filter can use a lattice predictor of order M coupled to an adaptive LMS/Newton filter of length N, wherein M| 10-22-2009 | |
| 20090268920 | Cardioid beam with a desired null based acoustic devices, systems and methods - An acoustic device is provided with first and second one or more acoustic elements to generate a first signal that includes mostly undesired audio and substantially void of desired audio, and a second signal that includes desired as well undesired audio respectively. The first one or more acoustic elements are designed and arranged to generate a Cardioid beam with a null at an originating direction of the desired audio. The second one or more acoustic elements are designed and arranged to generate a complementary beam that includes the desired audio. A system is provided with an appropriate signal processing logic to recover the desired audio using the first and second signals. The signal processing logic may practice echo cancellation like techniques or blind signal separation techniques. | 10-29-2009 |
| 20090274315 | METHOD AND APPARATUS TO REDUCE NON-LINEAR DISTORTION - Techniques to reduce distortion in acoustic signals in mobile computing devices are described. For example, a mobile computing device may comprise a speaker operative to receive a first signal and output a second signal. The mobile computing device may further comprise a first microphone operative to receive the second signal and a second microphone operative to receive a third signal. An echo canceller may be coupled to the first microphone and the second microphone and may be operative to compare the second signal and the third signal and reduce distortion in the third signal based on the comparison. Other embodiments are described and claimed. | 11-05-2009 |
| 20090304197 | DISTRIBUTED AUDIO SIGNAL PROCESSING SYSTEM HAVING VIRTUAL CHANNELS - A distributed audio signal processing system having a plurality of linked audio signal processing units is disclosed. Each audio signal processing unit has physical channels for receiving and sending local audio signals and a high bandwidth interface for exchanging audio signals with other linked audio signal processing units. Each of the physical channels of each of the audio signal processing units is mapped to a corresponding global channel. Global channels can be combined to form virtual channels that can be processed as a signal channel. | 12-10-2009 |
| 20090304198 | AUDIO SIGNAL DECORRELATOR, MULTI CHANNEL AUDIO SIGNAL PROCESSOR, AUDIO SIGNAL PROCESSOR, METHOD FOR DERIVING AN OUTPUT AUDIO SIGNAL FROM AN INPUT AUDIO SIGNAL AND COMPUTER PROGRAM - An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal. | 12-10-2009 |
| 20090310794 | AUDIO CONFERENCE APPARATUS AND AUDIO CONFERENCE SYSTEM - To provide an audio conference apparatus and an audio conference system which can smoothly proceed with the audio conference by removing a recursion sound of the conference voice is achieved. An audio conference apparatus | 12-17-2009 |
| 20090316923 | MULTICHANNEL ACOUSTIC ECHO REDUCTION - A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal. | 12-24-2009 |
| 20090316924 | ACCOUSTIC ECHO CANCELLATION AND ADAPTIVE FILTERS - In one embodiment, a two-way telecommunication device may perform acoustic echo cancellation on incoming signals. An audio decoding module may produce an audio render signal. An audio capture interface may receive an audio capture signal. A short length adaptive filter may determine a time delay between the audio render signal and the audio capture signal by adaptively predicting a sub-band of the audio capture signal using a corresponding sub-band of the audio render signal. | 12-24-2009 |
| 20100046768 | METHOD AND SYSTEM FOR ELIMINATION OF ACOUSTIC FEEDBACK - A method and system is provided for eliminating acoustical feedback in a system. The method determines a parameter for at least one notch filter, adjusting the notch filter based on the parameter, processing the digital signals through the notch filter, testing at the effect of the notch filter in the system, and removing the notch filter if the notch filter is not effective. Also disclosed is a method and system of selecting candidate frequencies which might be feedback, as opposed to other wanted sound frequencies. The selection method sampling the digital signals, converting the time domain digital signal samples by a fast Fourier transform algorithm into the frequency domain, using a ballistics approach to find prominences in the frequency spectrum, and testing the sizes of the prominences. | 02-25-2010 |
| 20100054489 | DEREVERBERATION SYSTEM AND DEREVERBERATION METHOD - Provided is a dereverberation system or the like which copes with an arbitrary condition flexibly and is capable of recognizing a sound or a sound source signal. According to the dereverberation system, an inverse filter (h) is set by using a pseudo-inverse matrix (R | 03-04-2010 |
| 20100067712 | Echo Cancelling Device, Signal Processing Device and Method Thereof, and Program - An echo cancelling device includes a filter-coefficient polarity inverter configured to invert the polarity of a filter coefficient at intervals of a predetermined frame length, an adaptive filter configured to estimate a signal to be inputted from an unknown system by multiplying a signal to be outputted from a speaker by the filter coefficient, and generate the resultant estimated signal, a subtracter configured to calculate an error signal from an input signal from a microphone and the estimated signal, and an error-signal polarity inverter configured to invert the polarity of the error signal in synchronization with the inversion of the polarity of the filter coefficient at intervals of the predetermined frame length. The adaptive filter updates the filter coefficient on the basis of the error signal by using binary fixed-point arithmetic, and therein, performing a truncation operation. | 03-18-2010 |
| 20100074452 | ACOUSTIC ECHO CONTROL - Processing requirements for echo cancellation in voice communications are significant and are even more so as the bandwidth of the communication increases. Whilst voice communication occupies a relatively narrow band of frequencies the processing requirements and so forth for wideband communication render echo cancellation difficult to achieve in a cost effective manner. The invention provides for provides echo cancellation within wideband communications by dividing the communications into sub-bands and applying echo cancellation to some sub-bands whilst processing other sub-bands according to the status of the communications. Additional sub-bands are transmitted at either full-duplex or half-duplex. | 03-25-2010 |
| 20100080398 | METHOD AND SYSTEM FOR HEARING DEVICE FITTING - The method for manufacturing an adjusted hearing device ( | 04-01-2010 |
| 20100092003 | INTEGRATING ACOUSTIC ECHO CANCELLATION AS A SUBSYSTEM INTO AN EXISTING VIDEOCONFERENCE AND TELEPRESENCE SYSTEM - The present invention is embodied in a computer-readable program in a computer-readable medium for upgrading a video conference system, the computer-readable program comprising acoustic echo canceling control software having an application programming interface. The acoustic echo canceling control software is implemented on a computer system that operates the video conference system and macros are configured to couple the acoustic echo canceling control software to hardware components of the video conference system and to interface with the application programming interface. The macros are user configurable for providing real time adjustments of echo canceling runtime parameters of the hardware components during a video conference session. | 04-15-2010 |
| 20100150363 | Method for testing echo cancellers - The performance of an echo canceller is assessed using a) a test signal launched from originating test equipment and b) a simulated echo of the test signal launched from terminating test equipment. The launch of the simulated echo signal is timed in such a way that it arrives at the tandem echo canceller(s) at a point in time relative to the arrival of the test signal thereat that the tandem echo canceller(s) is (are) not able to cancel the simulated echo signal. The latter thus arrives uncanceled at the target echo canceller. The launch of the simulated echo signal is further timed in such a way that it arrives at the target echo canceller at a point in time relative to the arrival of the test signal thereat that the target echo canceller IS able to cancel the simulated echo signal. As a result, any residual echo received at the originating test equipment is a measure of the performance of the target echo canceller exclusive of the performance of the tandem echo canceller(s). | 06-17-2010 |
| 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 |
| 20100166199 | ACOUSTIC ECHO REDUCTION CIRCUIT FOR A "HANDS-FREE" DEVICE USABLE WITH A CELL PHONE - The device comprises: a circuit for picking up acoustic signals with the microphone ( | 07-01-2010 |
| 20100166200 | Feedback Elimination Method and Apparatus - A method and apparatus for detecting a singing frequency in a signal processing system using two neural-networks is disclosed. The first one (a hit neural network) monitors the maximum spectral peak FFT bin as it changes with time. The second one (change neural network) monitors the monotonic increasing behavior. The inputs to the neural-networks are the maximum spectral magnitude bin and its rate of change in time. The output is an indication whether howling is likely to occur and the corresponding singing frequency. Once the singing frequency is identified, it can be suppressed using any one of many available techniques such as notch filters. Several improvements of the base method or apparatus are also disclosed, where additional neural networks are used to detect more than one singing frequency. | 07-01-2010 |
| 20100183163 | SOUND SIGNAL PROCESSOR AND DELAY TIME SETTING METHOD - An echo canceller formed of an adaptive filter is designed such that even under a condition where a system transmission delay is undefined, an appropriate delay time can be set in a delay circuit that absorbs a system delay, and that an effective echo cancellation effect can always be achieved. A time difference of a transmission path until a reproduction audio signal input to the delay circuit is input as a processing target signal of an adaptive filter system through a space between a speaker and a microphone is determined, and the delay time corresponding to this time difference is set in the delay circuit. At this time, the speaker and the microphone are placed so that the distance therebetween is small, and the delay time of the delay circuit is set to 0. Thus, the determined time difference indicates a system transmission delay in the above transmission path. That is, an accurate delay time corresponding to the system transmission delay can be set in the delay circuit. | 07-22-2010 |
| 20100189274 | DEVICE FOR AND A METHOD OF PROCESSING AUDIO SIGNALS - A method is provided which is suitable to cope with non-linear echo paths during acoustic echo cancellation in speakerphones. Non-linear paths occur particularly in hands-free operation of, e.g., a mobile phone, due to driving the amplifier and loudspeaker in the non-linear range. The idea is to combine the commonly known one microphone approach of linear acoustic echo cancellation using an adaptive filter and a post-processor together with a multiple microphone approach using beam forming which separately removes the non-linear part of the echo. | 07-29-2010 |
| 20100189275 | PASSENGER COMPARTMENT COMMUNICATION SYSTEM - A communication system for a passenger compartment includes at least two microphone arrays arranged within first and second regions, respectively, in the passenger compartment, and at least two loudspeakers and a signal processor connected to the microphone arrays and to the loudspeaker. Each microphone array has at least two microphones and provides an audio signal. Each loudspeaker is located within a different one of the first and the second regions. The signal processor processes the audio signal from the microphone array within the first region and provides the processed audio signal to the loudspeaker located within the second region. | 07-29-2010 |
| 20100208908 | ECHO SUPRESSING METHOD AND APPARATUS - The coefficient generating section receives a first signal which is the output signal of the microphone of a signal generated by subtracting the output signal of a linear echo canceller from the output signal of the microphone and a second signal which is the output signal of the linear echo canceller. The coefficient generating section detects the minimum value of the variation with time of the ratio of the amplitude of the first signal to that of the second signal and outputs the value of constant times the detected minimum value as a crosstalk coefficient indicating the degree of crosstalk of the echo. The converting section corrects the first signal according to the crosstalk coefficient and the second signal to generate a near-end signal which is the resultant signal of when the echo is removed from the first signal and outputs the near-end signal to an output terminal. | 08-19-2010 |
| 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 |
| 20100226503 | ECHO CANCELLER - An echo canceller capable of suppressing an echo more effectively than the case of performing filter processing by a signal of a fixed-point format while suppressing an increase in cost is provided. An echo canceller | 09-09-2010 |
| 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 |
| 20100260344 | Method and Arrangement for Echo Cancellation of Voice Signals - A method and arrangement for cancelling echoes in a communication terminal ( | 10-14-2010 |
| 20100278351 | Methods and systems for reducing acoustic echoes in multichannel communication systems by reducing the dimensionality of the space of impulse resopnses - Various embodiments of the present invention are directed to adaptive methods for reducing acoustic echoes in multichannel audio communication systems. Acoustic echo cancellation methods determine approximate impulse responses characterizing each echo path between loudspeakers and microphones within a room and improve performance based on previously determined impulse responses. In particular, the methods adapt to changes in the room by inferring approximate impulse responses that lie within a model of an impulse response space. Over time the method improves performance by evolving the model into a more accurate space from which to select subsequent approximate impulse responses. | 11-04-2010 |
| 20100296663 | System and Method for the Application on an LMS Method to Updating an Echo Canceller in an ADSL Modem - An echo cancellation device relies on the known characteristics of the sync frame to monitor, update in an off-line fashion and determine the accuracy of an echo canceller in, for example, a modem, such as an ADSL modem. Specifically, time domain samples are read from the transmit (Tx) and receive (Rx) paths of the modem. These samples are stored in memory. When the sync frame has received a predetermined number of the same Tx samples and Rx samples, the samples are stored. Running averages, over the sync frames, of the TX and RX samples are maintained. These averages are subtracted from a sync frame of samples, to allow LMS updating of the echo canceller taps, free of extraneous signals. Updating, i.e., tracking of changes in the echo channel, is done for the echo canceller in an off-line fashion. The coefficients for the in-line version are updated, while the off-line version is updated over several sync frames. Periodically, the performance of the off-line version is compared with the in-line version. The coefficients of the in-line version are replaced by those of the off-line version only if it is determined the off-line version, which is tracking echo channel changes, has better performance. After replacement of the in-line coefficients, the off-line tracking is continued in the off-line version. | 11-25-2010 |
| 20100316228 | METHODS AND SYSTEMS FOR BLIND DEREVERBERATION - Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal. | 12-16-2010 |
| 20100329472 | REVERBERATION SUPPRESSING APPARATUS AND REVERBERATION SUPPRESSING METHOD - A reverberation suppressing apparatus separating sound source signals based on input signals output from microphones collecting the plurality of sound source signals, includes a sound signal output unit generating sound signals and outputting the generated sound signals, a sound acquiring unit acquiring the input signals from microphones, a first evaluation function calculation unit calculating a separation matrix, the input signals, and the sound source signals, and calculating a first evaluation function, a reverberation component suppressing unit calculating an optimal separation matrix, and suppressing a reverberation component by separating the sound source signals other than the generated sound signals, and a separation matrix updating unit dividing a step-size function, approximating each segment to a linear function, calculating step sizes based on the approximated linear functions, and repeatedly updating the separation matrix so that the degree of separation of the sound source signals exceeds the predetermined value. | 12-30-2010 |
| 20110002473 | DEREVERBERATION APPARATUS, DEREVERBERATION METHOD, DEREVERBERATION PROGRAM, AND RECORDING MEDIUM - A sound source model storage section stores a sound source model that represents an audio signal emitted from a sound source in the form of a probability density function. An observation signal, which is obtained by collecting the audio signal, is converted into a plurality of frequency-specific observation signals each corresponding to one of a plurality of frequency bands. Then, a dereverberation filter corresponding to each frequency band is estimated by using the frequency-specific observation signal for the frequency band on the basis of the sound source model and a reverberation model that represents a relationship for each frequency band among the audio signal, the observation signal and the dereverberation filter. A frequency-specific target signal corresponding to each frequency band is determined by applying the dereverberation filter for the frequency band to the frequency-specific observation signal for the frequency band, and the resulting frequency-specific target signals are integrated. | 01-06-2011 |
| 20110013781 | SYSTEM AND PROCESS FOR REGRESSION-BASED RESIDUAL ACOUSTIC ECHO SUPPRESSION - A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well. | 01-20-2011 |
| 20110019831 | Echo Suppression Method and Apparatus Thereof - In an echo suppression apparatus, an adaptive filter estimates an echo path of a near end, and generates a pseudo echo signal of a reception signal received from a far end. A subtractor subtracts the pseudo echo signal from a near-end signal including an echo signal of the reception signal, a near-end sound and a background noise, thereby generating an echo-canceled signal. A background noise estimation unit estimates a spectrum of the background noise. A non-linear processing unit performs spectrum subtraction of the estimated spectrum of the background noise from the spectrum of the echo-canceled signal, and controls a gain of the spectrum of the echo-canceled signal in response to the result of the spectrum subtraction, thereby obtaining a spectrum of a transmission signal transmitted to the far end. A threshold calculation unit calculates a threshold value used to determine presence or absence of the residual echo in the echo-canceled signal. The non-linear processing unit compares the result of the spectrum subtraction with the threshold value, then controls the spectrum of the echo-canceled signal with a high gain in case that the result of the spectrum subtraction is higher than the threshold value, and with a low gain in case that the result of the spectrum subtraction is not higher than the threshold value. | 01-27-2011 |
| 20110019832 | SOUND PROCESSOR, SOUND PROCESSING METHOD AND RECORDING MEDIUM STORING SOUND PROCESSING PROGRAM - A sound processor includes a conversion unit converts a reference sound signal corresponding to a base of sound to be output and an observation sound signal based on each of sound signals output by a plurality of sound receiving units into frequency components, an echo suppression unit estimates echo derived from sound based on a converted reference sound signal and suppressing the estimated echo in a converted observation sound signal, a noise suppression unit estimates noise based on an arrival direction of sound and suppressing the estimated noise in the converted observation sound signal and an integrating process unit suppresses, with respect to each frequency component, echo and noise in the converted sound signal based on a observation sound signal obtained after echo suppression and a observation sound signal obtained after noise suppression. | 01-27-2011 |
| 20110019833 | APPARATUS AND METHOD FOR COMPUTING FILTER COEFFICIENTS FOR ECHO SUPPRESSION - A preferred embodiment of an apparatus for computing filter coefficients for an adaptive filter for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal includes an extractor for extracting a stationary component signal or a non-stationary component signal from the loudspeaker signal or from a signal derived from the loudspeaker signal, and a computer for computing the filter coefficients for the adaptive filter on the basis of the extracted stationary component signal or the extracted non-stationary component signal. | 01-27-2011 |
| 20110044461 | APPARATUS AND METHOD FOR COMPUTING CONTROL INFORMATION FOR AN ECHO SUPPRESSION FILTER AND APPARATUS AND METHOD FOR COMPUTING A DELAY VALUE - An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value. | 02-24-2011 |
| 20110044462 | SIGNAL ENHANCEMENT DEVICE, METHOD THEREOF, PROGRAM, AND RECORDING MEDIUM - The initial values of parameter estimates are set, including reverberation parameter estimates, which includes a regression coefficient used in a linear convolutional operation for calculating an estimated value of reverberation included in an observed signal, source parameter estimates, which includes estimated values of a linear prediction coefficient and a prediction residual power that identify the power spectrum of a source signal, and noise parameter estimates, which include noise power spectrum estimates. Then, the maximum likelihood estimation is used to alternately repeat processing for updating at least one of the reverberation parameter estimates and the noise parameter estimates and processing for updating the source parameter estimates until a predetermined termination condition is satisfied. | 02-24-2011 |
| 20110110526 | ACOUSTIC ECHO CANCELLER AND ACOUSTIC ECHO CANCELLATION METHOD - An adaptive filter generates a pseudo echo sound signal based on a sound emission sound signal. An adder subtracts the pseudo echo sound signal from a low band component of a collected sound signal, thereby generating a sound signal with a first-adjusted low band component. An echo spectrum estimation section estimates and calculates a frequency spectrum of a reverberation echo this time from a spectrum of the pseudo echo sound signal this time, a frequency spectrum of the preceding reverberation echo, and an update coefficient based on an audio environment. An adder subtracts the frequency spectrum of the reverberation echo and the frequency spectrum of stationary noise from a spectrum of the sound signal with the first-adjusted low band component. | 05-12-2011 |
| 20110116644 | SIMULATED BACKGROUND NOISE ENABLED ECHO CANCELLER - An apparatus may include a noise estimation unit configured to determine a noise spectrum associated with a noise signal, and generate a filter based on the determined noise spectrum. The apparatus may also include a noise synthesis unit configured to generate a colored noise using the filter generated by the noise estimation unit. The apparatus may be incorporated in an echo canceller. | 05-19-2011 |
| 20110129096 | METHOD AND SYSTEM FOR REDUCING ACOUSTICAL REVERBERATIONS IN AN AT LEAST PARTIALLY ENCLOSED SPACE - A method of increasing the intelligibility of an audio broadcast in an at least partially enclosed space from at least one amplified audio source. An input microphone receives an incident audio wavefront at a first position in the at least partially enclosed space. An active noise control system is employed to generate a cancelling audio wavefront having a magnitude substantially equal to the magnitude of incident audio wavefront and a phase substantially opposite to the phase of the incident audio wavefront. The cancelling audio wavefront is broadcast at a second position in the at least partially enclosed space adjacent to a reflective surface of the at least partially enclosed space so as to attenuate the incident audio wavefront substantially at or near the reflective surface in order to reduce reverberations of the incident audio wavefront. In this manner, reverberations which could reduce the intelligibility of the audio broadcast to an audience is reduced. | 06-02-2011 |
| 20110135105 | ECHO CANCELLER - An adaptive filter unit | 06-09-2011 |
| 20110158418 | DEREVERBERATION AND NOISE REDUCTION METHOD FOR MICROPHONE ARRAY AND APPARATUS USING THE SAME - A dereverberation and noise reduction method adapted for a microphone array and an apparatus using the same are proposed. The microphone array receives a plurality of audio signals from an audio source. The dereverberation and noise reduction method includes the following steps. The received audio signals are processed by a beamforming processing, and a first audio signal is generated. Besides, the received audio signals are processed by a suppression processing, and a suppression audio vector is generated. Further, suppression audio vector is processed by an adaptive filtering processing, and a second audio signal is generated. In addition, the second audio signal is subtracted from the first audio signal to acquire an audio output signal, where parameters of the adaptive filtering processing are adjusted according to a feedback of the audio output signal. | 06-30-2011 |
| 20110176687 | APPARATUS AND ASSOCIATED METHODOLOGY FOR SUPPRESSING AN ACOUSTIC ECHO - A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low | 07-21-2011 |
| 20110176688 | SIGNAL PROCESSING METHOD, SIGNAL PROCESSING DEVICE, AND SIGNAL PROCESSING PROGRAM - Provided is a signal processing method which receives a plurality of reception signals and subtracts pseudo-echoes generated by a plurality of adaptive filters which input the reception signals, from a plurality of echoes generated by the reception signals, thereby reducing the echoes. The method delays two or more of the reception signals so as to generate delayed reception signals and inputs the reception signals and the delayed reception signals to the adaptive filters so as to generate pseudo-echoes. | 07-21-2011 |
| 20110206212 | Microphone System and Method of Operating the Same - A microphone system is provided, wherein the microphone system comprises a microphone array comprising a plurality of microphone units each adapted to generate a primary signal indicative of an acoustic wave received from the respective microphone unit, a first echo cancellation unit, an integrator unit, and a combination unit, wherein the microphone system is adapted to generate a first dipole response and a monopole response from the primary signals, wherein the integrator unit is adapted to generate a first integrated dipole response by integrating the first dipole response, wherein the first echo cancellation unit is adapted to generate a first echo cancelled integrated dipole response from the first integrated dipole response, and wherein the combination unit is adapted to combine the monopole response and the first echo cancelled integrated dipole response. | 08-25-2011 |
| 20110211706 | SOUND EMISSION AND COLLECTION DEVICE AND SOUND EMISSION AND COLLECTION METHOD - Provided is a sound emission and collection device capable of estimating the azimuth of a sound source (such as a main utterer) precisely without any processing load. The sound emission and collection device ( | 09-01-2011 |
| 20110255702 | SIGNAL DEREVERBERATION USING ENVIRONMENT INFORMATION - An audio processing system includes an audio processing device and an external device, the audio processing device having a signal processing unit adapted for running an algorithm for processing an input signal representing an acoustic signal from the environment of the user and providing a processed output signal. The external device includes in an information signal a measure, characteristic of the reverberation of the room or location where the audio processing device is located, and the audio processing device is adapted to extract the measure of the reverberation from the information signal and use the measure as an input to an algorithm that includes a directional algorithm for providing a directional characteristic of the input signal. | 10-20-2011 |
| 20110255703 | DETERMINING AN ACOUSTIC COUPLING BETWEEN A FAR-END TALKER SIGNAL AND A COMBINED SIGNAL - The present invention provides an improved determination of an acoustic coupling between a loudspeaker ( | 10-20-2011 |
| 20110268287 | LOUDSPEAKER SYSTEM AND SOUND EMISSION AND COLLECTION METHOD - A loudspeaker system with a high degree of versatility, which, even when a microphone or a person speaking moves, can adequately response to the movement is provided. A plurality of linkage terminals | 11-03-2011 |
| 20110293104 | AUDIO COMMUNICATION DEVICE AND METHOD USING FIXED ECHO CANCELLATION FILTER COEFFICIENTS - Methods, apparatuses, systems, and software are disclosed for providing a phone or other audio communication device with a fixed-path acoustic echo cancellation feature that compensates for a fixed-path acoustic coupling caused by transmission of sound from one or more speakers to one or more microphones through, for instance, the phone body itself. | 12-01-2011 |
| 20110311066 | Echo Cancellers and Echo Cancelling Methods - Methods and systems for updating the adaptive filter of an echo canceller. A method of updating the adaptive filter of an echo canceller in which an estimated echo is resolved from a received signal and then subtracted from an incoming echo-contaminated signal so as to produce a filtered output signal, includes: obtaining a corrected impulse response of an echo reconstruction filter (ERF); calculating specified decision measures usable to decide whether to prospectively apply the corrected ERF impulse response or a current ERF impulse response; determining whether application of the corrected ERF impulse response would result in improved echo cancellation; and updating the ERF to apply the corrected impulse response, when it is determined that the updating would result in a lower residual echo. | 12-22-2011 |
| 20120027218 | Multi-Microphone Robust Noise Suppression - A robust noise reduction system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to cochlea domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain. | 02-02-2012 |
| 20120045069 | Dynamic Audibility Enhancement - A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that, each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component, at that respective frequency; d) estimating a sat of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-specific lower bound gain and respective frequency-specific upper bound gain; and e) applying the frequency-specific gains to the primary audio input signal. | 02-23-2012 |
| 20120045070 | ACTIVE NOISE CONTROL DEVICE - The active noise control device includes: a signal obtaining section that obtains an electric signal relating to the predetermined sound; a control section that adjusts an amplitude and a phase of the electric signal obtained by the signal obtaining section; a vibrating section having a diaphragm and a vibrator, the vibrator vibrating in accordance with an output from the control section. Because a sound radiated from the diaphragm toward the first region is substantially in opposite phase to that toward the second region, the control section controls the vibrator so that the diaphragm generates a sound that attenuates the predetermined sound in the first region, and causes the predetermined sound to have a desired frequency characteristic in the second region. | 02-23-2012 |
| 20120063608 | SYSTEM FOR EXTRACTION OF REVERBERANT CONTENT OF AN AUDIO SIGNAL - A reverberant characteristic of an acoustic space is superimposed on an audio signal that is received by an apparatus. The apparatus decomposes the audio signal into an estimated original dry signal component and an estimated reverberant characteristic of the acoustic space. Estimation of the original dry signal component and the reverberant characteristic of the acoustic space is based on determination of an estimated impulse response of the acoustic space from the received audio signal. Once the audio signal is decomposed, the estimated original dry signal component and the estimated reverberant characteristic of the acoustic space may be independently modified by the apparatus. The modified or unmodified estimated original dry signal component and estimated reverberant characteristic of the acoustic space may be combined by the apparatus to produce one or more adjusted frequency spectra. | 03-15-2012 |
| 20120063609 | ACOUSTIC MULTI-CHANNEL CANCELLATION - A multi-channel acoustic echo canceller arrangement comprises a microphone ( | 03-15-2012 |
| 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 |
| 20120114129 | MIXER WITH ADAPTIVE POST-FILTERING - A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherency levels of the signals. | 05-10-2012 |
| 20120128168 | METHOD AND APPARATUS FOR NOISE AND ECHO CANCELLATION FOR TWO MICROPHONE SYSTEM SUBJECT TO CROSS-TALK - A method and apparatus for joint noise and echo cancellation of a two microphone system subject to cross-talk. The method includes estimating the reference output by removing the cross-talk and the estimated echo from the reference channel, when an echo is detected in the reference echo signal, adapting filters H | 05-24-2012 |
| 20120140939 | METHOD AND DEVICE FOR CANCELLING ACOUSTIC ECHO - An acoustic echo cancellation device for generating a pseudo echo signal by filtering an input remote speaker signal based on a plurality of adaptive filters and controlling the adaptive filters to filter the same based on a filter coefficient. The acoustic echo cancellation device generates an error signal by subtracting the pseudo echo signal from a nearby speaker signal, determines a convergence state of the filter coefficient based on the error signal, and sets at least one filter coefficient with a previously used value to stop the operation for calculating new values for the corresponding filter coefficients when the filter coefficient is determined to be converged. | 06-07-2012 |
| 20120148059 | Controlling Audio Signals - Method, user terminal and computer program product for controlling audio signals at the user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in (i) a first mode in which secondary audio signals output from the user device are likely to disturb the primary audio signal received at the audio input means, or (ii) a second mode in which secondary audio signals output from the user device are not likely to disturb the primary audio signal received at the audio input means. In dependence upon determining that the user device is operating in the first mode, the secondary audio signals are selectively suppressed from being output from the user device during the communication session, such that when the user device is operating in the first mode said secondary audio signals do not disturb the primary audio signal received at the audio input means for transmission to the remote node in the communication session. | 06-14-2012 |
| 20120155665 | Echo Canceller With Adaptive Non-Linearity - An echo canceller ( | 06-21-2012 |
| 20120163612 | Communication System - An echo component of a first signal received at an audio input device is removed. A second signal is output from an audio output device. The echo component in the first signal is the result of the second signal traversing an echo path. The characteristics of the first and second signals are compared, and if the first signal only comprises the echo, an estimate of the echo path is determined by comparing the first and second signals. The echo path estimate is applied to the first signal to determine an equalised first signal, which is is compared with the second signal to determine an estimate of the echo component. The echo component from the first signal is removed in dependence on the estimate of the echo component. | 06-28-2012 |
| 20120195438 | Echo suppression for wireless handsets and headsets - The enhancements provided herein are designed to allow an acoustic echo suppressor algorithm to operate with significant echo path delay between an audio block R | 08-02-2012 |
| 20120207315 | AUDIO PROCESSING APPARATUS AND METHOD OF CONTROLLING THE AUDIO PROCESSING APPARATUS - An audio processing apparatus includes first and second audio pickup units. The second audio pickup unit includes an audio resistor provided to cover a sound receiving portion to suppress external wind introduction while passing an external audio. A first filter attenuates a signal having a frequency lower than a first cutoff frequency of the output signal of a first A/D converter. A second filter attenuates a signal having a frequency higher than a second cutoff frequency of the output signal of a second A/D converter. A third filter is provided between the first audio pickup unit and the first A/D converter to attenuate a signal having a frequency lower than a third cutoff frequency for suppressing the wind noise. | 08-16-2012 |
| 20120224708 | INFORMATION PROCESSING APPARATUS, AUXILIARY DEVICE THEREFOR, INFORMATION PROCESSING SYSTEM, CONTROL METHOD THEREFOR, AND CONTROL PROGRAM - Disclosed is a noise suppression technology for suppressing various types of noise including unknown noise without storing a large number of noise information in advance. | 09-06-2012 |
| 20120237047 | NONLINEAR REFERENCE SIGNAL PROCESSING FOR ECHO SUPPRESSION - An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity. | 09-20-2012 |
| 20120243698 | Dynamic Beamformer Processing for Acoustic Echo Cancellation in Systems with High Acoustic Coupling - Near-end equipment for a communication channel with far-end equipment. The near-end equipment includes at least one loudspeaker, at least two microphones, a beamformer, and an echo canceller. The communication channel may be in one of a number of communication states including Near-End Only state, Far-End Only state, and Double-Talk state. In one embodiment, when the echo canceller determines that the communication channel is in either the Far-End Only state or the Double-Talk state, the beamformer is configured to generate a nearfield beampattern signal that directs a null towards a loudspeaker. When the echo canceller detects the Near-End Only state, the beamformer is configured to generate a farfield beampattern signal that optimizes reception of acoustic signals from the near-end audio source. Using different beamformer processing for different communication states allows echo cancellation processing to be more successful at reducing echo in the signal transmitted to the far-end equipment. | 09-27-2012 |
| 20120250871 | Nonlinear Echo Suppression - Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band. | 10-04-2012 |
| 20120294452 | METHOD AND APPARATUS FOR REDUCING NOISE PUMPING DUE TO NOISE SUPPRESSION AND ECHO CONTROL INTERACTION - An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component. | 11-22-2012 |
| 20120294453 | METHOD AND APPARATUS FOR REDUCING NOISE PUMPING DUE TO NOISE SUPPRESSION AND ECHO CONTROL INTERACTION - An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component. | 11-22-2012 |
| 20120294454 | MOBILE TERMINAL AND EARPHONE IDENTIFYING METHOD - A mobile phone apparatus | 11-22-2012 |
| 20130010976 | Efficient Audio Signal Processing in the Sub-Band Regime - A signal processing system enhances an audio signal. The audio signal is divided into audio sub-band signals. Some audio sub-band signals are excised. Other audio sub-band signals are processed to obtain enhanced audio sub-band signals. At least a portion of the excised audio sub-band signals are reconstructed. The reconstructed audio sub-band signals are synthesized with the enhanced audio sub-band signals to form an enhanced audio signal. | 01-10-2013 |
| 20130028431 | MULTIFUNCTIONAL ELECTRONIC ACCESSORY - An electronic accessory for an electronic device includes an audio processing circuit. The audio processing circuit includes a first switch circuit, an echo cancellation circuit and an amplification circuit. The echo cancellation circuit removes echoes from a first voice signal from a user in a call and a second voice signal from a caller, outputs the first voice signal to the electronic device via the first switch circuit, and outputs the second voice signal to the amplification circuit. The amplification circuit amplifies the second voice signal. The amplification circuit further directly receives and amplifies a third voice signal from multimedia content stored in the electronic device via the first switch circuit. | 01-31-2013 |
| 20130028432 | REVERBERATION SUPPRESSION DEVICE, REVERBERATION SUPPRESSION METHOD, AND COMPUTER-READABLE RECORDING MEDIUM STORING REVERBERATION SUPPRESSION PROGRAM - A reverberation suppression device includes, a first storage unit configured to store, in advance, information representing a first impulse response obtained from a signal output from a microphone when a sound source positioned according to directivity of either a speaker or the microphone, which are mounted on a mobile terminal, outputs an impulse; a second storage unit configured to store information representing a second impulse response obtained from a signal output from the microphone when the speaker mounted on the mobile terminal outputs an impulse in a room where reverberation sound is to be suppressed; a response correction unit configured to obtain a corrected impulse response, which reflects the room's environment, by correcting the second impulse response, which is represented by the information stored in the second storage unit, using the information representing the first impulse response; and a sound correction unit configured to correct a sound signal | 01-31-2013 |
| 20130039503 | BEAMFORMING APPARATUS AND METHOD BASED ON LONG-TERM PROPERTIES OF SOURCES OF UNDESIRED NOISE AFFECTING VOICE QUALITY - A beamforming technique for a microphone array is described to attenuate a source of undesired noise that is deemed the most limiting to audio quality in an acoustic environment. Possible sources of undesired noise include echo, background noise (stationary) and other interference signals (non-stationary). The beamforming technique is updated based on long-term evaluations. Once an evaluation occurs and a decision is made, the beamformer adapts with a maximum responsiveness and without intentional delay, and therefore not affecting the beamformer's tracking ability. When fixed beamforming is utilized, one of several fixed beamformers having different attenuation targets are selected to implement noise attenuation. When adaptive beamforming is utilized, the beamformer adapts whenever the selected target is deemed dominant. | 02-14-2013 |
| 20130044890 | INFORMATION PROCESSING DEVICE, INFORMATION PROCESSING METHOD AND PROGRAM - n information processing device includes: an estimating section which estimates an amplitude frequency function from a first signal output to a speaker and a second signal input from a microphone; a generating section which generates an estimated echo signal from the first signal and the amplitude frequency function; and a suppressing section which suppresses the estimated echo signal from the second signal, wherein the estimating section changes a coefficient of the amplitude frequency function on the basis of the correlation between the estimated amplitude frequency function and a short-time average amplitude frequency function. | 02-21-2013 |
| 20130070934 | ENCASEMENT FOR ABATING ENVIRONMENTAL NOISE, HAND-FREE COMMUNICATION AND NON-INVASIVE MONITORING AND RECORDING - An encasement such as an electronic pillow including a pillow unit encasing at least one error microphone and at least one loudspeaker in electrical connection with a controller unit, the pillow unit also including a power source, and a reference sensing unit including at least one reference microphone in electrical connection with the controller unit, the controller unit including an algorithm for controlling interactions between the error microphone, loudspeaker, and reference microphone. A method of abating unwanted noise, by detecting an unwanted noise with a reference microphone, analyzing the unwanted noise, producing an anti-noise corresponding to the unwanted noise in a pillow, and abating the unwanted noise. | 03-21-2013 |
| 20130077798 | REVERBERATION SUPPRESSION DEVICE, REVERBERATION SUPPRESSION METHOD, AND COMPUTER-READABLE STORAGE MEDIUM STORING A REVERBERATION SUPPRESSION PROGRAM - A reverberation suppression device includes an analyzer configured to analyze change over time in the power of an input signal obtained from a microphone in response to sound input, and thereby compute the decrease per unit time in the power of the input signal in a reverb segment following the end of a segment in which the sound is produced; and a suppression controller configured to control a suppression gain which indicates the rate at which the input signal is attenuated, on the basis of analysis results from the analyzer. | 03-28-2013 |
| 20130083936 | Processing Audio Signals - Audio signals are processed for use in a communication event. A data store may be queried to obtain an indication of an echo direction, which relates to a direction from which audio signals output from the audio output are likely to be received at a microphone array (plurality of microphones) of a device. Beamformer coefficients of an adaptive beamformer of the device are determined in dependence upon the received indication of the echo direction. Audio signals are received at the microphone array. The adaptive beamformer applies the determined beamformer coefficients to the received audio signals, thereby generating a beamformer output for use in the communication event. The beamformer coefficients are determined such that echo suppression is applied to audio signals received at the microphone array from the indicated echo direction. | 04-04-2013 |
| 20130083937 | Speaker Device, Sound Source Simulation System, and Echo Cancellation System - A speaker device includes an enclosure provided with a speaker unit and a bass reflex port, wherein the bass reflex port has an outer opening opened to outside of the enclosure and an inner opening opened to inside of the enclosure, and has a tubular body whose hollow cross-sectional area is gradually reduced along an axial direction of the bass reflex port from its ends to its center, and the tubular body is formed so that a length of the hollow cross-section in one direction does not change along the axial direction, and wherein the inner opening of the bass reflex port is located opposite to the speaker unit, with the bass reflex port sandwiched therebetween. | 04-04-2013 |
| 20130121497 | System and Method for Acoustic Echo Cancellation Using Spectral Decomposition - A method and apparatus for canceling an echo in audio communication is disclosed. The method comprises receiving an audio signal from a network and subsequently detecting a mixture audio signal comprising a target audio signal and an echo audio signal, the echo signal corresponding to the received audio signal. The method then comprises estimating the target audio signal by determining magnitude spectrograms for the mixture and received audio signals respectively, estimating a magnitude spectrogram of the target audio signal dependent on those of the mixture and received audio signal, and generating an output audio signal that estimates the target audio signal, the output audio signal being dependent on the estimated magnitude spectrogram. | 05-16-2013 |
| 20130121498 | NOISE REDUCTION USING MICROPHONE ARRAY ORIENTATION INFORMATION - A handheld device includes: an orientation sensor; an audio processor connected to the orientation sensor and adapted to receive orientation information from the orientation sensor; and a plurality of microphones through which audio content is captured, wherein the audio processor modifies the noise reduction algorithm applied to the audio content captured based, at least in part, on the orientation information. | 05-16-2013 |
| 20130129100 | PROCESSING AUDIO SIGNALS - In an embodiment, a method of processing audio signals at a device includes receiving audio signals at a plurality of microphones of the device; processing at least one of the audio signals received by the plurality of microphones to generate a first characteristic; a beamformer applying beamformer coefficients to the received audio signals, thereby generating a beamformer output; processing the beamformer output to generate a second characteristic. An echo canceller is applied to the beamformer output, thereby suppressing, from the beamformer output, an echo resulting from audio signals output from an audio output. An operating parameter of the echo canceller is determined, using a relationship between the first and second characteristics. | 05-23-2013 |
| 20130129101 | MULTICHANNEL ACOUSTIC ECHO REDUCTION - A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal. | 05-23-2013 |
| 20130142348 | Method and System for Bone Conduction Sound Propagation - A wearable ambient sound reduction system, the system may include: a microphone, adapted to detect an ambient sound signal; a processor adapted to generate an output signal in response to the ambient sound signal; wherein the output signal, when conveyed to a bone of the user, reduces an affect that an ambient sound signal has upon the user; wherein the microphone is coupled to the processor; and a bone conduction speaker, coupled to the processor, adapted to convey the output signal to a bone of a user. | 06-06-2013 |
| 20130156209 | OPTIMIZING AUDIO PROCESSING FUNCTIONS BY DYNAMICALLY COMPENSATING FOR VARIABLE DISTANCES BETWEEN SPEAKER(S) AND MICROPHONE(S) IN A MOBILE DEVICE - Mobile communication devices, having multiple speakers and/or microphones to perform a number of audio functions, for use with mobile devices, are provided. The microphones may be housed within the communication device housing. To compensate for the unwanted signal feedback between the speakers and microphones, acoustic echo cancellation may be implemented to determine the proper distance and relative location between the speakers and microphones. Acoustic echo cancellation removes the echo from voice communications to improve the quality of the sound. The removal of the unwanted signals captured by the microphones may be accomplished by characterizing the audio signal paths from the speakers to the microphones (speaker-to-microphone path distance profile), including the distance and relative location between the speakers and microphones. The optimal distance and relative location between the speakers and microphones is provided to the user to optimize performance. | 06-20-2013 |
| 20130156210 | TRAINING AN ECHO CANCELLER IN SEVERE NOISE - An apparatus includes a non-adaptive filter, an adaptive filter, and a controller. The non-adaptive filter may have non-adaptive filter coefficients and be configured to develop a non-adaptive error signal as a function of the non-adaptive filter coefficients. The adaptive filter may have adaptive filter coefficients and be configured to develop an adaptive error signal as a function of the adaptive filter coefficients. The controller may be configured to monitor a quality of the non-adaptive and adaptive error signals and perform one or more of a full coefficient update, a partial coefficient update and a fractional coefficient update of the non-adaptive filter coefficients based on a comparison of the quality of the adaptive error signal to a determined current best-attained performance measurement. | 06-20-2013 |
| 20130156211 | WIDEBAND ACOUSTIC ECHO CANCELLATION APPARATUS WITH ADAPTIVE TAIL LENGTH IN EMBEDDED SYSTEM, AND WIDEBAND ACOUSTIC ECHO CANCELLATION METHOD - A wideband acoustic echo cancellation apparatus with an adaptive tail length in an embedded system, and a wideband acoustic echo cancellation method are provided, and the wideband acoustic echo cancellation apparatus may include a delay length calculating unit to calculate a delay length of an echo path, using a near-end signal and a far-end signal, an adaptive filter implementing unit to implement an adaptive filter based on the calculated delay length, using selected coefficients, and an error calculating unit to search for three intervals having a largest impulse response value from all intervals of a tail of the adaptive filter, and to calculate an error during an interval in which the selected coefficients are used. | 06-20-2013 |
| 20130177162 | NOVEL PRE-PROCESS (AMPLITUDE DISTORTION) AND POST-PROCESS (PHASE SYNCHRONIZATION) FOR LINEAR AEC SYSTEM - An acoustic processing apparatus is provided. The apparatus includes a pre-processing component, a filter and a first signal processing component. The pre-processing component compensates a non-linearity of a reference signal to generate an input signal. The filter coupled to the pre-processing component, the filter executes filtering on the input signal to generate an output signal. The first signal processing component, coupled to the pre-processing component, the reference signal obtains a gain from the first signal processing component to generate a first signal, and the first signal processing component passes the gain to the pre-processing component. | 07-11-2013 |
| 20130188797 | Over-Sampled Single-Sided Subband Echo Cancellation - A method of echo cancellation, particularly for use in high definition applications, splits an input signal and reference signal into M single-sided sub-band. The subbanded signals are downsampled at a downsampling rate N, where N≦M, adaptively filtered, and recombined to produce an output signal. The sub-bands are preferably oversampled such that N| 07-25-2013 | |
| 20130188798 | REVERBERATION REDUCTION DEVICE AND REVERBERATION REDUCTION METHOD - A reverberation reduction device includes, a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, calculating reverberation characteristics in response to an impulse response of a path of a sound from an audio output unit to an audio input unit by determining the impulse response from a first audio signal and a second audio signal that represents a sound that the audio input unit has picked up from the first audio signal reproduced by the audio output unit, and estimating a distance from the audio input unit to a sound source in accordance with at least one of a volume and a frequency characteristic of a third audio signal that represents a sound that the audio input unit has picked up from a sound from the sound source; correcting the reverberation characteristics so that the reverberation characteristics. | 07-25-2013 |
| 20130188799 | AUDIO PROCESSING DEVICE AND AUDIO PROCESSING METHOD - An audio processing device includes a reverb property estimating unit that estimates a reverb property at each frequency on the basis of a first audio signal and a second audio signal representing sounds of the first audio signal output by an audio output unit and collected by an audio input unit, a gain calculating unit that determines an attenuating ratio for a component of the first audio signal at each frequency such that the larger the reverb property at the frequency is, the larger the attenuating ratio for the component at the frequency becomes, and a correcting unit that attenuates the first audio signal at the each frequency in accordance with the attenuating ratio determined for each frequency. | 07-25-2013 |
| 20130208905 | Temporal Interpolation Of Adjacent Spectra - Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal. Computational complexity of signal processing systems can, therefore, be reduced. | 08-15-2013 |
| 20130216056 | NON-LINEAR ECHO CANCELLATION - A two-stage structure for performing non-linear echo cancellation is described in which a first echo canceller is used to attenuate linear echo components of a microphone signal and a second echo canceller is used to attenuate non-linear echo components of the output signal generated by the first echo canceller. One or both of the echo cancellers may be implemented using closed-form solutions, including a closed form solution for a hybrid method in the frequency domain. | 08-22-2013 |
| 20130216057 | ECHO CANCELLATION USING CLOSED-FORM SOLUTIONS - A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal. | 08-22-2013 |
| 20130230183 | ECHO CANCELLATION - The invention is directed to echo cancellation for a microphone system. An exemplary microphone system comprises a first transistor, wherein a gate terminal of the first transistor is connected to a ground terminal via a microphone electret element, the microphone electret element being associated with a capacitance and a voltage, the microphone electret element reverse biasing the first transistor; and a second transistor in parallel with the first transistor, wherein a gate terminal of the second transistor is connected to the ground terminal via a capacitor, the capacitance of the capacitor being selected to suppress at least a portion of a common mode signal, and wherein the gate terminal of the second transistor is not connected to the microphone electret element. The common mode signal comprises the echo, which may be the output of a speaker system that is received as input to the microphone system. | 09-05-2013 |
| 20130230184 | ECHO SUPPRESSION COMPRISING MODELING OF LATE REVERBERATION COMPONENTS - An apparatus for computing filter coefficients for an adaptive filter is disclosed. The adaptive filter is used for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal. The apparatus has: an echo decay modeling means for modeling a decay behavior of an acoustic environment and for providing a corresponding echo decay parameter; and computing means for computing the filter coefficients of the adaptive filter on the basis of the echo decay parameter. A corresponding method has: providing echo decay parameters determined by means of an echo decay modeling means; and computing the filter coefficients of the adaptive filter on the basis of the echo decay parameters. | 09-05-2013 |
| 20130251167 | CONTROL OF A LOUDSPEAKER OUTPUT - A loudspeaker drive circuit has a microphone which forms part of an acoustic echo cancellation system. An input signal is processed before application to a loudspeaker driver, and the processing is controlled in dependence on the echo cancellation system performance, such as to control the extent to which the loudspeaker is driven into a non-linear operating region. In this way, the linearity can be controlled so as to provide an excursion limit, without needing a model of the loudspeaker or additional dedicated sensors. | 09-26-2013 |
| 20130251168 | AMBIENT INFORMATION NOTIFICATION APPARATUS - An ambient information notification apparatus includes: an interior sound acquisition device acquiring an interior sound in a compartment of the vehicle; an ambient information presentation sound generator generating an ambient information presentation sound; and an ambient information presentation sound output device outputting the ambient information presentation sound. The ambient information presentation sound satisfies that a sound pressure level of the ambient information presentation sound is higher than the interior sound in a predetermined frequency band, and is lower than or equal to the interior sound in other frequency band, and that the ambient information presentation sound is provided by stereophony, in which a sound image localization direction approximately directs to a virtual sound source. | 09-26-2013 |
| 20130251169 | ECHO CANCELER AND ECHO DETECTOR - An echo canceler includes a signal-to-echo ratio calculating unit | 09-26-2013 |
| 20130259247 | AUDIO CONTROL SYSTEM - Technologies are generally described for a system for controlling an audio signal based on a proximity of, e.g., a user's hand to at least one component of an audio control system. In some examples, an audio control system may include a filter configured to provide an echo signal and a control decision unit configured to provide a control signal based on the echo signal. | 10-03-2013 |
| 20130266149 | COMMUNICATION SYSTEM AND METHOD HAVING ECHO-CANCELLING MECHANISM - A communication system having echo-cancelling mechanism is provided. The communication system communicates with a remote communication device. The communication system comprises a computer system end, a digital playback device, a sound-receiving module, a digital-to-analog (D/A) converter and a digital processing module. The computer system end receives a digital signal from the remote communication device. The digital playback device plays the digital signal to generate a sound signal. The sound-receiving module generates an analog audio signal comprising the sound signal. The D/A converter receives a digital sound signal of the digital signal and converts the digital sound signal to an analog sound signal. The digital processing module performs an echo-cancelling process on the analog audio signal according to the analog sound signal. | 10-10-2013 |
| 20130272533 | HOWLING SUPPRESSION DEVICE, HEARING AID, HOWLING SUPPRESSION METHOD, AND INTEGRATED CIRCUIT - A howling suppression device includes a subtractor which subtracts a pseudo feedback signal from an input signal; an adaptive filter which produces a pseudo feedback signal for a next input signal; and a coefficient update control unit which controls an update rate of a filter coefficient of the adaptive filter and includes: a level calculation unit which calculates a signal level of the input signal; a signal-rising-edge detection unit which detects a rising-edge point; a reverberation section detection unit which detects a reverberation section; and an update rate control unit which sets the update rate to a first rate in the reverberation section and to a second rate in other sections. The adaptive filter updates the filter coefficient at the update rate set by the update rate control unit. | 10-17-2013 |
| 20130294611 | SOURCE SEPARATION BY INDEPENDENT COMPONENT ANALYSIS IN CONJUCTION WITH OPTIMIZATION OF ACOUSTIC ECHO CANCELLATION - Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals and perform acoustic echo cancellation. Independent component analysis may be used to perform the source separation in conjunction with acoustic echo cancellation on the time-frequency domain mixed signals to generate at least one estimated source signal corresponding to at least one of the original source signals. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims. | 11-07-2013 |
| 20130294612 | AUTOMATIC MICROPHONE MUTING OF UNDESIRED NOISES BY MICROPHONE ARRAYS - Methods and systems for cancelation of table noise in a speaker system used for video or audio conferencing are disclosed. Table noise is cancelled by using a vertical microphone array to distinguish the tilt angle of sound received by a microphone. If the sound is close to horizontal, the audio is muted. If the sound is above a given angle from horizontal, it is not muted, as this indicates a person speaking. This eliminates paper rustling, keyboard clicks and the like. | 11-07-2013 |
| 20130301840 | METHODS FOR PROCESSING AUDIO SIGNALS AND CIRCUIT ARRANGEMENTS THEREFOR - A method for processing audio signals is provided comprising outputting an audio signal; receiving the output audio signal via a first receiving path as a first received audio signal; receiving the output audio signal via a second receiving path as a second received audio signal; determining an echo suppression gain based on the first received audio signal and the second received audio signal; and filtering echo suppression of the audio signal based on the first received audio signal and the echo suppression gain. | 11-14-2013 |
| 20130301841 | AUDIO PROCESSING DEVICE, AUDIO PROCESSING METHOD AND PROGRAM - An audio processing device includes: a directivity adjustment unit adjusting directivity and sharpness thereof in audio picked up by plural microphones picking up audio; and a howling suppression adjustment unit adjusting intensity of suppressing howling of audio picked up by the plural microphones, wherein the directivity adjustment unit adjusts the directivity and sharpness thereof in preference to the howling suppression of audio performed by the howling suppression adjustment unit. | 11-14-2013 |
| 20130315407 | METHOD AND DEVICE FOR IN EAR CANAL ECHO SUPPRESSION - An earpiece ( | 11-28-2013 |
| 20130315408 | ADAPTIVE EQUALIZER, ACOUSTIC ECHO CANCELLER DEVICE, AND ACTIVE NOISE CONTROL DEVICE - A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer. | 11-28-2013 |
| 20130322638 | METHOD FOR ACOUSTIC ECHO CANCELLATION AND SYSTEM THEREOF - The spirit of the present invention is to vary the step size parameter in accordance with the error signal and the output acoustic signal, wherein the filter is easy to implement, nonparametric VSS-NLMS algorithm which employs the mean-square error and the estimated system noise power to control the step-size update. The new nonparametric VSS-NLMS algorithm has been shown to perform with fast convergence rate, good tracking, and low mis-adjustment. In comparison with existing VSS-NLMS algorithms, the proposed algorithm has demonstrated consistently superior performance both in convergence and for final error level relative to published algorithms in application on both simulated data and real speech data. | 12-05-2013 |
| 20130322639 | SYSTEMS AND METHODS FOR ACOUSTIC ECHO CANCELLATION WITH WIRELESS MICROPHONES AND SPEAKERS - Systems and methods for acoustic echo cancellation with wireless microphone and speaker systems are described herein. In one embodiment, an acoustic echo canceller is provided that can, among other things, cancel an acoustic echo component of an audio signal that is produced when an audio signal transmitted to a remote speaker on a lossy wireless link is picked up by a remote microphone and transmitted back to a base station on a lossy wireless link. | 12-05-2013 |
| 20130322640 | POST-PROCESSING INCLUDING MEDIAN FILTERING OF NOISE SUPPRESSION GAINS - A method of post-processing raw banded gains for applying to an audio signal, an apparatus to generate banded post-processed gains, and a tangible computer-readable storage medium comprising instructions that when executed carry out the method. The raw banded gains are determined by input processing one or more input audio signals. The method includes applying post-processing to the raw banded gains to generate banded post-processed gains, generating a particular post-processed gain for a particular frequency band, including median filtering using raw gain values for frequency bands adjacent to the particular frequency band. One or more properties of the post-processing depend on classification of the one or more input audio signals. | 12-05-2013 |
| 20130336494 | AGC Circuit for an Echo Cancelling Circuit - An amplifier circuit and a method of amplification using automatic gain control (AGC) are disclosed. A method for reducing distortions incurred by an audio signal when being rendered by an electronic device is described. The method comprises receiving an input signal; determining signal strength; determining a frequency-dependent AGC filter; wherein the frequency-dependent AGC filter is adapted to selectively attenuate the input signal within a number N of predetermined frequency ranges, according to corresponding N degrees of attenuation; wherein N predetermined frequency ranges depend upon a rendering characteristic of the electronic device; and wherein the N-degrees of attenuation depend upon the signal strength; and attenuating the input signal using the frequency-dependent AGC filter to obtain an output signal for rendering by the electronic device. | 12-19-2013 |
| 20140003611 | SYSTEMS AND METHODS FOR SURROUND SOUND ECHO REDUCTION | 01-02-2014 |
| 20140003612 | INFORMATION TERMINAL AND COMPUTER PROGRAM PRODUCT | 01-02-2014 |
| 20140010382 | AUDIO SIGNAL PROCESSING SYSTEM AND ECHO SIGNAL REMOVING METHOD THEREOF - An audio signal processing system and an echo signal removing method thereof are provided. The audio signal processing system includes a speaker that is configured to output an audio signal; a microphone that is configured to receive the audio signal output by the speaker including an echo signal generated by the audio signal; an echo signal delay unit that is configured to delay the echo signal for a bulk delay time, and output the echo signal that is delayed; and an echo signal removing unit that is configured to remove the echo signal that is delayed and output by the echo signal delay unit from the audio signal received by the microphone, wherein the echo signal delay unit includes a bulk delay measuring unit that is configured to measure a bulk delay by analyzing impulse response characteristics of an echo path. | 01-09-2014 |
| 20140016794 | ECHO CANCELLATION SYSTEM AND METHOD WITH MULTIPLE MICROPHONES AND MULTIPLE SPEAKERS - An audio processing system comprising two or more microphones and an echo cancellation system configured to apply a fast converging adaptive filtering algorithm to low frequency bands of a first microphone signal to generate first synthesized echo signal components and an adaptive filtering algorithm to high frequency bands of the first microphone signal to generate second synthesized echo signal components and to apply the first synthesized echo signal components and the second synthesized echo signal components to the first microphone signal to cancel an echo signal of the first microphone signal. An echo estimate and suppression system is configured to receive the first synthesized echo signal components and the second synthesized echo signal components and to apply them to estimate powers of echo signals in one or more additional microphones. | 01-16-2014 |
| 20140044273 | DIRECT SOUND EXTRACTION DEVICE AND REVERBERANT SOUND EXTRACTION DEVICE - A direct sound extraction device includes: a spectrum transform unit that transforms an input signal, which includes a reverberant sound in a direct sound and on which a Fourier transform process has been performed, to a first amplitude spectrum signal Lfa; a low-pass filter unit ( | 02-13-2014 |
| 20140056435 | NOISE ESTIMATION FOR USE WITH NOISE REDUCTION AND ECHO CANCELLATION IN PERSONAL COMMUNICATION - A method comprises processing M subband communication signals and N target-cancelled signals in each subband with a set of beamformer coefficients to obtain an inverse target-cancelled covariance matrix of order N in each band; using a target absence signal to obtain an initial estimate of the noise power in a beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with a noise correction factor to obtain a refined estimate of the power of the beamformer output noise signal component in each subband; processing the refined estimate with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value to obtain a postfilter output signal in each subband; and processing the postfilter output subband signals to obtain an enhanced beamformed output signal. | 02-27-2014 |
| 20140064506 | ELECTRONIC DEVICE AND METHOD FOR BLOCKING ECHO GENERATION BY ELIMINATING SOUND OUTPUT FROM SPEAKER - A method for operating an electronic device including a first microphone and a second microphone to block the echo generation the electronic device is operated in a loudspeaker mode. The method includes: determining whether a sensor located within a predetermined range from the second microphone has sensed a peripheral object; blocking an operation of the second microphone when the sensor has sensed a peripheral object; and blocking echo generation by eliminating a sound that is output from a speaker and input to the first microphone. | 03-06-2014 |
| 20140079232 | AUDIO PROCESSING DEVICE, AUDIO PROCESSING METHOD, AND RECORDING MEDIUM RECORDING AUDIO PROCESSING PROGRAM - The present invention provides an audio processing device that appropriately suppresses echo generated in a stereophonic audio output. The audio processing device includes: means for generating a first artificial linear echo signal and a second artificial linear echo signal that are estimated to be generated by first audio and second audio travelling to audio input means; means for suppressing a linear echo signal mixed to an input audio signal based on the first artificial linear echo signal and the second artificial linear echo signal: means for estimating a non-linear echo signal based on the first artificial linear echo signal and the second artificial linear echo signal; and means for suppressing the non-linear echo signal. | 03-20-2014 |
| 20140098967 | OUTPUT PHASE MODULATION ENTRAINMENT CONTAINMENT FOR DIGITAL FILTERS - Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents. | 04-10-2014 |
| 20140105410 | ECHO CANCELLATION METHOD AND DEVICE - The present invention discloses an echo cancellation method. The method includes: dividing an audio signal into a high-band audio signal and a low-band audio signal; performing adaptive filtering on the low-band audio signal, and performing synthesis filtering on a signal obtained after the low-band audio signal undergoes the adaptive filtering and on the high-band audio signal to generate a preliminary echo cancellation signal; performing envelope predication echo suppression on a high-band signal in the preliminary echo cancellation signal, and calculating and outputting a residual echo suppression coefficient; performing echo suppression on a low-band signal in the preliminary echo cancellation signal, and outputting a processing result; and multiplying the output result by the residual echo suppression coefficient, and outputting a signal of which echoes are canceled. | 04-17-2014 |
| 20140105411 | METHODS AND SYSTEMS FOR KARAOKE ON A MOBILE DEVICE - Systems and methods for providing karaoke recording and playback on mobile devices are provided. The mobile device may play music audio and associated video, and receive via one or more microphones a mix of a user voice, the music, and background noise. The mix is stored both in its original form and as processed to enhance voice and sound through noise suppression and other processing. Stored audio may be uploaded through a communications network to a cloud based computing environment for listening on other mobile devices. Selectable playing control and recording options may be provided. Audio cues may be determined during signal processing of the original acoustic sound and be stored on the mobile device. During playback of recorded audio and, optionally, associated video, the original acoustic sound, recorded cues, and user selectable optional processing may be used to remix during playback, while retaining the original recording. | 04-17-2014 |
| 20140112487 | USING AN AUXILIARY DEVICE SENSOR TO FACILITATE DISAMBIGUATION OF DETECTED ACOUSTIC ENVIRONMENT CHANGES - Audio input may be received at one or more microphones of a mobile device. Based on the audio input, a change in an acoustic environment of the device, such as a change in a direction of arrival of the audio input or a degradation in a quality of acoustic echo cancellation being performed upon the audio input, may be detected. A determination may be made that the detected change coincides with a non-acoustic physical event detected using an auxiliary sensor at the mobile device. The event may for example be device motion, a new proximate object, a change in a proximity of an object, a new heat source or a change in a heat level from a known heat source. Based on the determining, a signal processor, possibly comprising an audio beamformer or echo canceller, may be recalibrated, e.g. the audio beamformer or echo canceller may be caused to reconverge. | 04-24-2014 |
| 20140112488 | METHOD AND APPARATUS FOR CANCELING ACOUSTIC ECHO, AND METHOD FOR UPDATING ADAPTIVE FILTER COEFFICIENT BY THE APPARATUS - An apparatus for canceling an acoustic echo signal caused by a far-end talker signal is provided. The apparatus for canceling an acoustic echo signal includes: a variance estimating unit configured to estimate a variance of a first audio signal of a near-end talker signal and a first noise signal of the near-end talker signal; a step size determining unit configured to determine a step size by using the variance of the first audio signal and the variance of the first noise signal; an adaptive filter coefficient updating unit configured to update an adaptive filter coefficient of an adaptive filter by using the step size; and an acoustic echo canceling unit configured to estimate an acoustic echo signal by using the adaptive filter coefficient, and cancel the acoustic echo signal from a microphone input signal by using the estimated acoustic echo signal. | 04-24-2014 |
| 20140119552 | LOUDSPEAKER LOCALIZATION WITH A MICROPHONE ARRAY - Embodiments of the present disclosure provide systems and methods for source localization using a microphone array. Such an embodiment, among others, utilizes circuitry to process audio signals and microphone signals to remove an echo (produced by a nearby loudspeaker) from the microphone signals using a plurality of adaptive echo cancellation filters. Location information is then determined for the loudspeaker based on the plurality of adaptive echo cancellation filters used to remove the echo. | 05-01-2014 |
| 20140133666 | SIGNAL PROCESSING SYSTEM AND SIGNAL PROCESSING METHOD - A signal processing system includes microphone units connected in series and a host device connected to one of the microphone units. Each of the microphone units has a microphone, a temporary storage memory, and a processing section for processing the sound picked up by the microphone. The host device has a non-volatile memory in which a sound signal processing program for the microphone units is stored. The host device transmits the sound signal processing program read from the non-volatile memory to each of the microphone units. Each of the microphone units temporarily stores the sound signal processing program in the temporary storage memory. The processing section performs a process corresponding to the sound signal processing program temporarily stored in the temporary storage memory and transmits the processed sound to the host device. | 05-15-2014 |
| 20140146975 | ACOUSTIC ECHO CANCELLATION SYSTEM - An acoustic echo cancellation (AEC) system includes a remote device, for capturing a remote captured sound, a server coupled to the remote device, and a local device coupled to the server. The server transmits the remote captured sound from the remote device to the local device. The local device receives, stores and plays the remote captured sound as a local playback sound. An echo is generated from reflection of the local playback sound. The local device captures the echo and a local sound into a local captured sound, and transmits both the remote captured sound and the local captured sound to the server. The server performs AEC on the local captured sound by using the remote captured sound from the local device and transmits the AEC processed local captured sound to the remote device. | 05-29-2014 |
| 20140177856 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD - According to one embodiment, a first processing module adds, to a first queue, output sound data output from a first task, with a time stamp attached thereto. A second processing module adds, to a second queue, input sound data received from a microphone, with a time stamp attached thereto. A controller fetches first output sound data as reference data from the first queue, the first output sound data having a time stamp whose time difference from a time stamp of first input sound data in the second queue falls within a predetermined range. An echo canceller performs echo cancelling processing to cancel an echo component in the first input sound data based on the reference data. | 06-26-2014 |
| 20140177857 | METHOD OF PROCESSING A SIGNAL IN A HEARING INSTRUMENT, AND HEARING INSTRUMENT - A method of processing a signal in a hearing instrument includes the steps of calculating a coherence between two microphone signals or microphone combination signals having different directional characteristics, determining an attenuation from the coherence, and applying the attenuation to the signal. | 06-26-2014 |
| 20140185818 | SOUND PROCESSING DEVICE, SOUND PROCESSING METHOD, AND PROGRAM - A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal. | 07-03-2014 |
| 20140198923 | REAL TIME AUDIO ECHO AND BACKGROUND NOISE REDUCTION FOR A MOBILE DEVICE - An audio enhancement system includes a display unit configured to exhibit a waveform corresponding to a microphone signal that is subject to an audio interference. The audio enhancement system also includes an interference reduction unit coupled to the microphone signal and configured to provide a reduction in the audio interference, wherein a reduced audio interference is indicated by the waveform in real time. A microphone signal enhancement method is also provided. | 07-17-2014 |
| 20140205105 | AUDIO SYSTEM SIGNAL PROCESSING CONTROL USING MICROPHONE MOVEMENT INFORMATION - An audio conferencing system has a base station, a speaker and one or more mobile microphones, and it operates to receive audio information from both a far-end audio source and a near-end audio source and to process the audio information for transmission to a far-end audio conferencing system. At least one of the mobile microphones has a motion detection device and operates to send detected motion information to the base station. The base station operates to control the processing of audio information it receives from the microphone according to the detected microphone motion information. | 07-24-2014 |
| 20140211955 | MICROPHONE HISS MITIGATION - A system and method for mitigating microphone hiss may obtain a frequency spectrum characteristic for a microphone. A microphone that has limited dynamic range may create microphone hiss in an output signal. The microphone hiss may prevent a reproduction of a sound field, represented in an output signal of the microphone, from being perceived as a natural environment. The microphone frequency spectrum may be obtained using static measurements or calculated dynamically. A virtual noise floor may be calculated responsive to the microphone frequency spectrum and a desired noise floor. Gain coefficients may be calculated responsive to the output signal of the microphone. The gain coefficients may be calculated to mitigate undesirable signal content including background noise and echoes. The calculated gain coefficients may be modified responsive to the virtual noise floor. The modified gain coefficients may allow a reproduction of the sound field to be perceived as a natural environment. | 07-31-2014 |
| 20140233746 | EARPHONE MICROPHONE - An acoustic space including a sound output path, first and second sound input paths is formed in a main body casing of an earphone microphone. Output sound from a speaker propagates in the sound output path. Sound input to a first microphone propagates in the first sound input path communicating with outside. Sound input to a second microphone propagates in the second sound input path. The sound output path branches into one path communicating with the outside of the main body casing and the other path communicating with the second sound input path. The earphone microphone amplifies a sound signal output from at least one of the first and second microphones so as to input sound from a sound source outside the main body casing, and suppresses input of the output sound. | 08-21-2014 |
| 20140254813 | AUDIO PROCESSOR CIRCUITS FOR ACOUSTIC ECHO CANCELLATION AND METHOD THEREFOR - In one form, an audio processor circuit includes a first digital signal processing circuit, a second digital signal processing circuit, and an interleaver. The first digital signal processing circuit has an input for receiving a far-end audio signal, and an output. The second digital signal processing circuit has an input for receiving a digital near-end audio signal, and an output. The interleaver has a first input coupled to the output of the first digital signal processing circuit, a second input coupled to the output of the second digital signal processing circuit, and an output for alternatively providing signals received from the first and second inputs to the output. | 09-11-2014 |
| 20140270216 | SINGLE-CHANNEL, BINAURAL AND MULTI-CHANNEL DEREVERBERATION - A method is presented for estimating and suppressing reverberation from a digital reverberant signal. A method for changing a first reverberation estimation according to another reverberation estimation is further provided. A method for controlling the reverberation suppression rate is also presented. | 09-18-2014 |
| 20140270217 | Apparatus with Adaptive Microphone Configuration Based on Surface Proximity, Surface Type and Motion - An apparatus includes a group of microphones and a surface compensator that is operatively coupled to switch logic and to a signal conditioner that provides a control channel to voice recognition logic. The surface compensator may detect surfaces in proximity to the apparatus as well as the surface's acoustic reflectivity or acoustic absorptivity and may accordingly configure the group of microphones including selecting appropriate signal conditioning and beamforming based on the surface acoustic reflectivity or acoustic absorptivity and the orientation of the apparatus. Voice recognition performance is thus improved when microphones are impeded or occluded by proximate surfaces. A group of sensors of the apparatus is used by the surface compensator to detect surfaces and surface type, and to determine apparatus orientation and motion. | 09-18-2014 |
| 20140286497 | MULTI-MICROPHONE SOURCE TRACKING AND NOISE SUPPRESSION - Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described. | 09-25-2014 |
| 20140286498 | ECHO CANCELLING DEVICE - An echo cancelling device splits a low-band signal through LPFs having characteristics which do not cause aliasing during downsampling of downsamplers, and splits a high-band signal through HPFs having characteristics which do not cause aliasing during downsampling of downsamplers. The echo canceling device generates a mid-band signal by subtracting the low-band signal and the high-band signal from a pre-split signal by adder-subtractors, and cancels an echo on a band-by-band basis. | 09-25-2014 |
| 20140307882 | ACOUSTIC ECHO CANCELLATION WITH INTERNAL UPMIXING - Methods, systems, and apparatuses are described for performing acoustic echo cancellation with internal upmixing that allow for a more effective handling of acoustic echo cancellation of audio components that are provided via different channels. In an embodiment in which audio is played back using two loudspeakers, audio components that are panned equally among the loudspeakers form a “phantom center image.” Acoustic echo cancellation is performed by initially upmixing the different channels to internally create modified versions of these channels and a virtual channel representative of the phantom center image. Each of these channels is passed through a respective adaptive filter that is configured to estimate an acoustic echo produced by each respective channel. These estimates are then subtracted from the signal received from one or more microphones (or from a signal obtained by combining multiple microphone signals) to suppress or eliminate the acoustic echo. | 10-16-2014 |
| 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 |
| 20140328490 | MULTI-CHANNEL ECHO CANCELLATION AND NOISE SUPPRESSION - A method for multi-channel echo cancellation and noise suppression is described. One of multiple echo estimates is selected for non-linear echo cancellation. Echo notch masking is performed on a noise-suppressed signal based on an echo direction of arrival (DOA) to produce an echo-suppressed signal. Non-linear echo cancellation is performed on the echo-suppressed signal based, at least in part, on the selected echo estimate. | 11-06-2014 |
| 20140341383 | NOISE REDUCTION VIA TUNED ACOUSTIC ECHO CANCELLATION - An embodiment provides a method, including: accessing a tuned corrective mask stored in a memory device; forming, using a processor, a tuned acoustic echo cancellation mask utilizing the tuned-corrective mask; applying, using a processor, the tuned acoustic echo cancellation mask to a digital audio signal; and outputting an echo-cancelled audio signal. Other aspects are described and claimed. | 11-20-2014 |
| 20140363008 | USE OF VIBRATION SENSOR IN ACOUSTIC ECHO CANCELLATION - Methods and systems are provided for acoustic echo cancellation in electronic devices. The echo cancellation may comprise applying, as a first step, echo cancellation filtering to an acoustic input obtained via an acoustic input element (e.g., microphone), and applying, as a second step, echo suppression to the acoustic input, wherein the echo suppression comprises suppressing residual echo in the acoustic input. The echo cancellation filtering may comprise identifying and/or filtering out echo components, both linear and nonlinear, in the acoustic input, with the echo components corresponding to an echo signal caused by an acoustic output outputted via the acoustic output element (e.g., speaker). A sensor signal, generated by a vibration sensor that detects vibrations in the electronic device including vibrations caused by the outputting of the acoustic output, may be used as reference signal in the echo cancellation filtering and/or the echo suppression. | 12-11-2014 |
| 20150016622 | DEREVERBERATION PARAMETER ESTIMATION DEVICE AND METHOD, DEREVERBERATION/ECHO-CANCELLATION PARAMETERESTIMATIONDEVICE,DEREVERBERATIONDEVICE,DEREVERBERATION/ECHO-CANC- ELLATION DEVICE, AND DEREVERBERATION DEVICE ONLINE CONFERENCING SYSTEM - In a conventional dereverberation system, when there is a fluctuating reverberation component, it has been difficult to determine, with high accuracy, a linear dereverberation filter for removing a non-fluctuating reverberation component. An algorithm integrating a dereverberation system using a linear filter and a dereverberation system using a non-linear filter includes the function of measuring the amount of fluctuation in transfer function in a latter-stage non-linear filter over time, and controls the strength of the non-linear filter over time based on the function. In this configuration, a strong non-linear process is implemented only when the fluctuation in transfer function is large, whereby distortion in speech components can be minimized. Further, by feeding the amount of fluctuation in transfer function over time by the function back to the estimation of a previous-stage linear filter, and thus reducing the weight for a time band in which the transfer function of a linear filter estimated value fluctuates, the influence causing a decrease in estimation accuracy of the transfer function can be reduced. | 01-15-2015 |
| 20150023514 | Method and Apparatus for Acoustic Echo Control - Embodiments of method and apparatus for acoustic echo control are described. According to the method, an echo energy-based doubletalk detection is performed to determine whether there is a doubletalk in a microphone signal with reference to a loudspeaker signal. A spectral similarity between spectra of the microphone signal and the loudspeaker signal is calculated. It is determined that there is no doubletalk in the microphone signal if the spectral similarity is higher than a threshold level. Adaption of an adaptive filter for applying acoustic echo cancellation or acoustic echo suppression on the microphone signal is enabled if it is determined that there is no doubletalk in the microphone signal through the echo energy-based doubletalk detection, or there is no doubletalk through the spectral similarity-based doubletalk detection. | 01-22-2015 |
| 20150030171 | ACOUSTIC SIGNAL PROCESSING DEVICE AND ACOUSTIC SIGNAL PROCESSING METHOD - Provided is an acoustic signal processing device for producing an output sound meeting listener's preferences by adjusting attack sound, reverberation, and noise component. The device includes: an FFT section for transforming an input audio signal from a time-domain to a frequency-domain to calculate a frequency spectrum signal and for generating a first amplitude spectrum signal and a phase spectrum signal; an attack component controller ( | 01-29-2015 |
| 20150043740 | METHOD USING ARRAY MICROPHONE TO CANCEL ECHO - A method using an may microphone to cancel echo applies to a sound receiving system and comprises steps: an array microphone receiving a sound source and outputting a plurality of analog acoustic signals formed from the sound source; an A/D converter converting the analog acoustic signals into a plurality of digital acoustic signals; a digital signal processor respectively using an adaptive beamforming process and a blocking matrix filtering process to convert the digital acoustic signals into a primary acoustic signal and at least one noise signal; and the digital signal processor using a multiple-input cancelling process to subtract the noise signal from the primary acoustic signal to obtain an acoustic signal where the echo has been cancelled. Thereby, the present invention can eliminate the systematic errors of the array microphone of the sound receiving system and improves the robustness of the acoustic signal. | 02-12-2015 |
| 20150043741 | WIRED AND WIRELESS EARSET USING EAR-INSERTION-TYPE MICROPHONE - The present invention relates to an earset, comprising: a first earphone portion, which includes a first speaker for outputting sound signals or voice signals that are provided from an external device, and which can be inserted into a first external auditory canal of a user; a second earphone portion, which includes a first microphone for receiving inputted user voice signals that are provided through the external auditory canal of the user, and which can be inserted into a second auditory canal of the user; and a main body connected to each of the first and the second earphone portion. When the main body is wirelessly connected to the external device, the main body comprises; a signal transceiving portion for transceiving the signals with the external device; and a control portion for outputting via the first speaker the voice signals received from the external device through the signal transceiving portion. | 02-12-2015 |
| 20150043742 | HEARING DEVICE WITH INPUT TRANSDUCER AND WIRELESS RECEIVER - The present disclosure regards a hearing device comprising a power source, electric circuitry, a loudspeaker, at least one microphone for sound from an acoustic environment, and at least one wireless receiver for wirelessly received sound signals. The microphone is configured to generate an environment sound signal. The wireless receiver is configured generate a source sound signal. The electric circuitry is configured to estimate at least one parameter of an impulse response from the location of the origin of the wirelessly received signal to the location of a user of the hearing device in dependence on the source sound signal and the environment sound signal. The electric circuitry is further configured to process the environment sound signal in dependence on the estimated at least one impulse-response parameter, thereby generating an output sound signal. The output sound signal is processed into sound by the loudspeaker. | 02-12-2015 |
| 20150063579 | Acoustic Echo Cancellation for Microphone Array with Dynamically Changing Beam Forming - Acoustic echo cancellation is improved by receiving a speaker signal that is used to produce audio in a room, and receiving audio signals that capture audio from an array of microphones in the room, including an acoustic echo from the speakers. To cancel the acoustic echo, one adaptive filter is associated with a corresponding subspace in the room. Each of the audio signals is assigned to at least one of the adaptive filters, and a set of coefficients is iteratively determined for each of the adaptive filters. The coefficients for an adaptive filter are determined by selecting each of the audio signals assigned to that adaptive filter and adapting the filter to remove an acoustic echo from each of the selected audio signals. At each iteration, a different audio signal is selected from the audio signals assigned to the adaptive filter in order to determine the set of coefficients. | 03-05-2015 |
| 20150063580 | CONTROLLER FOR AUDIO DEVICE AND ASSOCIATED OPERATION METHOD - A controller for an audio device is provided. The controller receives a first collected sound signal and a second collected sound signal respectively provided by two microphones, and includes an echo cancellation module and a beamforming module. The echo cancellation module performs echo cancellation on the first collected sound signal to accordingly provide an intermediate signal. The beamforming module performs beamforming by utilizing the echo-cancelled intermediate signal and the non-echo-cancelled second collected sound signal. | 03-05-2015 |
| 20150078564 | ECHO CANCELLATION ALGORITHM FOR LONG DELAYED ECHO - This document discloses one or more systems, apparatuses, methods, etc. for implementing an echo cancellation algorithm for long delayed echo that is created during a wire or wireless voice communications. In an implementation, a WiDi feature in a device during the wire or wireless voice communications may add an additional echo delay in addition to channel multipath delay when an audio sound signal travels from a WiDi component to a microphone of the device. In this implementation, a separate delay estimator is configured to estimate total delay. The estimated total delay is fed back to an adoptive filter component for long delay echo cancellation. | 03-19-2015 |
| 20150078565 | Dual-Diaphragm Microphone Module to Reduce Vibration Noise - A microphone module disposed in an electronic device for reducing echo noise. The microphone module includes a casing, a first diaphragm disposed in the casing, a second diaphragm disposed in the casing and a substrate disposed between the first diaphragm and the second diaphragm and joined to the casing to define a first space and a second space which are isolated and separated from each other. The first diaphragm is disposed in the first space, the second diaphragm is disposed in the second space, and the substrate is electrically connected with the first diaphragm and the second diaphragm. | 03-19-2015 |
| 20150078566 | Acoustic Echo Cancellation - A method and system for acoustic echo cancellation stores received far-end data in a first buffer. When the far-end data in the first buffer exceeds a predefined length, the stored far-end data is used to calculate echo estimate data. The echo estimate data is stored in a second buffer. Whenever microphone data is received the error data is calculated independent of echo estimate data availability. In particular, subsequent to sufficient echo estimate data being stored in the second buffer and responsive to the reception of the microphone data, the error data is calculated by subtracting, from the microphone data, corresponding echo estimate data stored in the second buffer. | 03-19-2015 |
| 20150078567 | Varying Adaptive Filter Step Size in Acoustic Echo Cancellation - A method and system for acoustic echo cancellation varies a step size of an adaptive filter in an acoustic echo canceller. Far-end data is received and echo estimate data is calculated using the received far-end data. Microphone data is received and error data is calculated using the received microphone data and the echo estimate data. A first average of the microphone data and a second average of the error data are computed over a predefined number of samples. An echo leakage is estimated using the first average and the second average wherein the echo leakage indicates an extent to which the far-end data is present in the error data, and the step size of the adaptive filter is varied based on the echo leakage and a maximum allowed step size. | 03-19-2015 |
| 20150092950 | Matching Reverberation in Teleconferencing Environments - A system and method of matching reverberation in teleconferencing environments. When the two ends of a conversation are in environments with differing reverberations, the method filters the reverberation so that when both signals are output at the near end (e.g., the audio signal from the far end and the sidetone from the near end), the reverberations match. In this manner, the user does not perceive an annoying difference in reverberations, and the user experience is improved. | 04-02-2015 |
| 20150098578 | CANCELLATION OF INTERFERING AUDIO ON A MOBILE DEVICE - Generally, this disclosure provides devices, systems and methods for cancelling an interfering audio signal. The system may include a mobile device including a microphone configured to capture an acoustic audio signal, the acoustic audio signal a combination of the interfering audio signal and a desired audio signal, the desired audio signal generated by a user of the mobile device. The system may also include a wireless communication module incorporated in the mobile device, to receive a reference signal through a side-channel, the reference signal associated with the interfering audio signal. The system may further include an acoustic echo cancellation module incorporated in the mobile device, the acoustic echo cancellation module to cancel the interfering audio signal from the captured acoustic audio signal, the cancellation based on the reference signal. | 04-09-2015 |
| 20150110282 | ACOUSTIC ECHO CONTROL FOR AUTOMATED SPEAKER TRACKING SYSTEMS - A processing system can include a tracking microphone array; audio tracker circuitry connected to the tracking microphone array to track an audio source based on an audio input from the array; communication microphones; and a processor. The processor can include audio circuitry to receive an audio input from the communication microphones and process the audio input to apply one or more of acoustic echo cancellation (AEC) and acoustic echo suppression (AES) processing to the audio input. The processor can further include calculating circuitry to calculate a ratio of signal power after and before the AEC and/or the AES processing, and control circuitry to generate an acoustic echo presence indication based on the ratio calculated by the calculating circuitry. The processor can transmit, via transmitting circuitry, the acoustic echo presence indication to an audio tracking device via a data communication channel between the processor and the audio tracker. | 04-23-2015 |
| 20150117656 | SAMPLING POINT ADJUSTMENT APPARATUS AND METHOD AND PROGRAM - There is provided a sampling point adjustment apparatus including: a frequency conversion unit that converts a first signal and a second signal into a first narrow band signal and a second narrow band signal through frequency conversion; a central position determination unit that determines a central position of a window of the frequency conversion for the second signal based on an estimated value of a sampling interval offset between the first narrow band signal and the second narrow band signal; and a phase control unit that controls a phase of the second narrow band signal based on the estimated value. | 04-30-2015 |
| 20150124986 | METHOD AND DEVICE FOR ELININATING ECHOES - Provided are a method and device for eliminating echo. The method comprises: an echo path characteristic parameter of an echo signal is estimated; a source signal of the echo signal is taken as a reference signal, and an echo estimation signal is generated according to the echo path characteristic parameter; and the echo estimation signal is subtracted from a speech signal to be processed. The disclosure solves the problem in the related art that self-adaptation cannot be achieved when a returned near-end audio is suppressed, thus facilitating the enhancement of the speech quality of a conference and the improvement of user experience. | 05-07-2015 |
| 20150124987 | ENHANCEMENT OF REVERBERANT SPEECH BY BINARY MASK ESTIMATION - The invention is directed to a single channel mask estimation method capable of improving reverberant speech identification for CI users. The method is based on the energy of the reverberant signal and the residual signal computed from linear prediction (LP) analysis. The mask is estimated by comparing the energy ratio of the two signals at different frequency bins with an adaptive threshold. As the threshold is updated for each frame of speech based on the energy ratios of the reverberant and LP residual signals computed from previous frames, it is amenable for real-time implementation. It can thus be used as a specialized (for reverberant environments) sound coding strategy used for cochlear implant applications. | 05-07-2015 |
| 20150131807 | Systems And Methods For Split Echo Cancellation - Systems and methods for performing videoconferencing using split endpoints are disclosed herein. These endpoints can include a Main Unit and a Satellite Unit that are associated with each other, and where the tasks of video and audio encoding and decoding are separated between the two. In some embodiments the Main Unit performs video and audio encoding, and the Satellite Unit performs video and audio decoding. In some embodiments the Satellite Unit obtains media data through its associated Main Unit, whereas in other embodiments the Satellite Unit obtains media data directly from the Server. | 05-14-2015 |
| 20150341501 | NONLINEAR ECHO SUPPRESSION - Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency hand. | 11-26-2015 |
| 20150341718 | EAR MICROPHONE - Disclosed is an ear microphone which includes: a microphone for converting a voice signal provided from the external auditory meatus of a user to an electric signal; a speaker for converting the electric signal provided from an external device to the voice signal; and a soundproof member having a microphone receiving groove for fixing and supporting the microphone to the inside of the housing and a speaker receiving groove for fixing and supporting the speaker to the inside of the housing. | 11-26-2015 |
| 20150341722 | METHODS AND DEVICES FOR REVERBERATION SUPPRESSION - In one embodiment, a process for suppressing reverberation begins with a device of a user obtaining a reverberant speech signal from a voice of the user. The device determines a first estimated reverberation component of the reverberant speech signal. The device generates a first de-reverberated output signal with a first reverberation suppression based on the reverberant speech signal and the first estimated reverberation component. Then, the device generates a second improved reverberation component using the first de-reverberated output signal. The device generates a second de-reverberated output signal with a second reverberation suppression based on the reverberant speech signal and the second improved reverberation component. | 11-26-2015 |
| 20150350777 | METHOD AND APPARATUS FOR IMPROVED RESIDUAL ECHO SUPPRESSION AND FLEXIBLE TRADEOFFS IN NEAR-END DISTORTION AND ECHO REDUCTION - A system and method is described for estimating residual echo and performing residual echo suppression on a signal. In particular, the system and method described herein: 1) provides a more accurate coherence-based approach where the sub-band nature of coherence-based residual echo estimation is leveraged using sub-band-based coherence measures and upper-bound envelopes to guide residual echo estimates; 2) provides joint estimation of the residual echo and a background stationary noise-floor; 3) provides more accurate residual echo estimation by combining the above two measures with information on near-end activity to selectively adjust the residual-echo estimation in sub-bands; and 4) provides improved control of near-end distortion by leveraging the noise-floor estimate and the internally calculated signal-to-echo ratios as additional guides/limits to help balance, flexibly, and as needed, near-end distortion with residual echo suppression. The resultant residual echo suppressor design provides significant improvement in both residual echo suppression and background noise preservation. | 12-03-2015 |
| 20150350778 | PERCEPTUAL ECHO GATE APPROACH AND DESIGN FOR IMPROVED ECHO CONTROL TO SUPPORT HIGHER AUDIO AND CONVERSATIONAL QUALITY - To address issues with present echo gate control, a method and apparatus for more intelligently operating an echo gate is described herein. In particular, the decision of whether to mute an uplink signal, or not, is formulated herein as primarily a perceptual decision based on an appropriate analysis of the perceptual interaction of the current residual echo and the current near-end signal(s). By doing so, the application of muting through an echo gate may be minimized and/or more appropriately engaged. This will lead to fewer dropouts and muting of speech onsets and offsets 1) during periods such as double-talk or 2) during periods of downlink playback in the presence of low near-end signal levels, two cases of particular importance. | 12-03-2015 |
| 20150350781 | AUDIO PROCESSING DEVICE AND AUDIO PROCESSING METHOD - An audio processing device includes a reverb property estimating unit that estimates a reverb property at each frequency on the basis of a first audio signal and a second audio signal representing sounds of the first audio signal output by an audio output unit and collected by an audio input unit, a gain calculating unit that determines an attenuating ratio for a component of the first audio signal at each frequency such that the larger the reverb property at the frequency is, the larger the attenuating ratio for the component at the frequency becomes, and a correcting unit that attenuates the first audio signal at the each frequency in accordance with the attenuating ratio determined for each frequency. | 12-03-2015 |
| 20150371654 | ECHO CONTROL THROUGH HIDDEN AUDIO SIGNALS | 12-24-2015 |
| 20150371656 | Acoustic Echo Preprocessing for Speech Enhancement - A method for cancelling/reducing acoustic echoes in speech/audio signal enhancement processing comprises selecting a long-term filter based on an echo tail length detection or an echo reverberation time detection of an microphone input signal; a reference signal is pre-processed with the selected long-term filter; the pre-processed reference signal is used to excite an adaptive filter wherein the output of the adaptive filter forms a replica signal of acoustic echo and/or acoustic echo tail; the replica signal of acoustic echo and/or acoustic echo tail is subtracted from a microphone input signal to suppress the acoustic echo and/or acoustic echo tail in the microphone input signal. The echo tail length or the echo reverberation time is detected by analyzing and comparing the microphone input signal and a received signal which is sent to a speaker. A strong long-term filter is selected if the detected echo tail length or the detected echo reverberation time is long; a weak long-term filter is selected if the detected echo tail length or the detected echo reverberation time is not long. | 12-24-2015 |
| 20150371657 | Energy Adjustment of Acoustic Echo Replica Signal for Speech Enhancement - A method for cancelling/reducing acoustic echoes in speech/audio signal enhancement processing comprises using a received reference signal to excite an adaptive filter wherein the output of the adaptive filter forms a replica signal of acoustic echo; the replica signal of acoustic echo is reduced adaptively by multiplying a gain to get a gained replica signal of acoustic echo wherein the gain is smaller in speech area and/or double-talk area than non-speech area; the gained replica signal of acoustic echo is subtracted from a microphone input signal to suppress the acoustic echo in the microphone input signal. | 12-24-2015 |
| 20150371658 | Control of Acoustic Echo Canceller Adaptive Filter for Speech Enhancement - A method for cancelling/reducing acoustic echoes in speech/audio signal enhancement processing comprises using a received reference signal to excite an adaptive filter wherein the output of the adaptive filter forms a replica signal of acoustic echo; an adaptation step size is controlled for updating the coefficients of the adaptive filter; the adaptation step size is initialized by using an open-loop approach and optimized by using a closed-loop approach; one of the most important parameters with the open-loop approach is an energy ratio between an energy of a returned echo signal in an input microphone signal and an energy of the received reference signal; one of the most important parameters with the closed-loop approach is a normalized correlation or a square of the normalized correlation between the input microphone signal and the replica signal of acoustic echo; the replica signal of acoustic echo is subtracted from the microphone input signal to suppress the acoustic echo in the microphone input signal. | 12-24-2015 |
| 20160012813 | SYSTEM AND METHOD FOR ACOUSTIC MANAGEMENT | 01-14-2016 |
| 20160027421 | AUDIO SIGNAL ANALYSIS - An apparatus comprises a dereverberation module for generating a dereverberated audio signal based on an original audio signal containing reverberation, and an audio-analysis module for generating audio analysis data based on audio analysis of the original audio signal and audio analysis of the dereverberated audio signal. | 01-28-2016 |
| 20160029120 | ROBUST ACOUSTIC ECHO CANCELLATION FOR LOOSELY PAIRED DEVICES BASED ON SEMI-BLIND MULTICHANNEL DEMIXING - A method for echo cancellation in multichannel audio signals includes receiving a plurality of time-domain signals, including multichannel audio signals and at least one reference signal, transforming the time-domain signals to K under-sampled complex-valued subband signals using an analysis filter bank, and performing, for each of the K under-sampled complex-value subband signals, linear echo cancellation of the reference signal from each channel using an acoustic echo canceller. A probability of acoustic echo dominance is produced using a single-double talk estimator, and a semi-blind multichannel source separation is performed based on the probability and independent component analysis (“ICA”) to decompose the audio signals into a near-end source signal and a residual echoes using subband semi-blind source separation. The residual echo components are removed from the near-end source signal using a spectral filter bank, and the subband audio signals are reconstructed to a multichannel time-domain audio signal using a subband synthesis filter. | 01-28-2016 |
| 20160035366 | ECHO SUPPRESSION DEVICE AND ECHO SUPPRESSION METHOD - An echo suppression device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: generating a corrected sound signal by suppressing an echo signal representing an echo generated by collecting, by a sound input unit, a sound arising from a reproduction sound signal reproduced by a sound output unit; obtaining a gain to attenuate the corrected sound signal according to a degree of distortion of the echo signal with which intensity of the echo signal non-linearly changes with respect to an intensity change of the reproduction sound signal; and suppressing the corrected sound signal according to the gain. | 02-04-2016 |
| 20160050489 | SYSTEMS AND METHODS FOR NONLINEAR ECHO CANCELLATION - In accordance with embodiments of the present disclosure, an audio processing circuit for use in an audio device may perform non-linear acoustic echo cancellation by predicting a displacement associated with an audio speaker, wherein such prediction takes into account a nonlinear response of the audio speaker with a mathematical model that calculates the predicted displacement of the audio speaker as a function of a current signal associated with the audio speaker using a time-varying difference equation, wherein coefficients of the difference equation are based on a set of physical parameters of the audio speaker. From the predicted displacement, the processing circuit may calculate a predicted acoustic output of the audio speaker, which may be used to generate a reference signal to an acoustic echo canceller. | 02-18-2016 |
| 20160050491 | Reversed Echo Canceller - A method implemented during a communication event conducted between a user device and at least one further user device over a communications network, the method comprising: receiving an audio signal that is to be output from a speaker of said user device, said audio signal comprising at least one microphone signal transmitted from the at least one further user device; detecting that said audio signal comprises at least one audio component of a microphone signal transmitted from one or more of one of said at least one further user device that is co-located with said user device; and based on this detection, removing said at least one audio component from said audio signal to produce an output audio signal for output from said speaker. | 02-18-2016 |
| 20160050504 | Utilizing a Smartphone During a Public Address System Session - A technique utilizes a smartphone during a public address system session. The technique involves operating a public address system to provide audio from a local human presenter to a local human audience, the local human presenter and the local human audience residing at a common physical public address location. The technique further involves establishing, while operating the public address system to provide audio from the local human presenter to the local human audience, an electronic connection between the smartphone and the public address system. The technique further involves providing, after establishing the electronic connection between the smartphone and the public address system, audio from a user of the smartphone to the local human audience through the smartphone, the electronic connection, and the public address system. | 02-18-2016 |
| 20160055842 | SMART PILLOWS AND PROCESSES FOR PROVIDING ACTIVE NOISE CANCELLATION AND BIOFEEDBACK - A smart pillow unit and processes for providing active noise cancellation and biofeedback. | 02-25-2016 |
| 20160055862 | SYSTEM AND METHOD FOR ADDRESSING ACOUSTIC SIGNAL REVERBERATION - A system and method for addressing acoustic signal reverberation is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and a reverberation audio signal. Embodiments may further include processing at least one of the first audio signal and the reverberation audio signal. Embodiments may also include limiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at signal least one of the first audio signal and the reverberation audio signal. | 02-25-2016 |
| 20160064011 | REVERBERATION SUPPRESSION APPARATUS USED FOR AUDITORY DEVICE - A reverberation suppression apparatus includes: an instantaneous value calculation unit that calculates an instantaneous value/instantaneous values in an envelope of values correlating with the absolute value or the square of an input signal; a reverberation estimation unit that calculates an exponential moving average of the instantaneous value(s) as an estimated reverberation component; a gain derivation unit that derives a gain corresponding to the input signal according to the estimated reverberation component and the instantaneous value(s) when the each instantaneous value is larger than the estimated reverberation component, and derives a lower limit of the gain as the gain corresponding to the input signal when the each instantaneous value is smaller than the estimated reverberation component; a smoothing unit that performs a smoothing process on the gain; and a gain processing unit that applies the gain to amplitude adjustment of the input signal thereafter. | 03-03-2016 |
| 20160073206 | Emission of a Commencement Sound and a Conclusion Sound - Various embodiments associated with a commencement sound and a conclusion sound are described. The commencement sound can be a firing sound, such as a sound of a bullet exiting a rifle. The conclusion sound can be an impact sound, such as a sound of the bullet impacting a concrete wall. These sounds can replicate what it sounds like to have an experience around someone without actually subjecting that person to the experience. | 03-10-2016 |
| 20160142548 | CONFERENCING APPARATUS WITH AN AUTOMATICALLY ADAPTING BEAMFORMING MICROPHONE ARRAY - This disclosure describes a conferencing apparatus with an automatically adapting beamforming microphone array for teleconferencing applications. The conferencing apparatus includes a processor coupled to memory, storage, and a housing. The apparatus further includes a communication element, a plurality of microphones arranged in the housing and configured as a beamforming microphone array, and an orientation sensor. Additionally, the apparatus includes executing a series of steps including: (a) adapting a signal-processing characteristic of the beamforming microphone array responsive to the orientation signal, (b) beamforming to combine the plurality of microphone signals to a plurality of combined signals, (c) performing an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals, and (d) selecting one or more of the plurality of combined echo-canceled signals for transmission. | 05-19-2016 |
| 20160150315 | SYSTEM AND METHOD FOR ECHO CANCELLATION - Systems and methods are provided for improving acoustic echo cancellation in a cabin of a vehicle. A source of far end speech is detected using a far end speech control module. A main beam directed at a speaking occupant and based on the far end speech is formed using a beam forming module. An echo cancelation filter is formed based on the far end speech using an acoustic echo cancellation module. An audible communication from the speaking occupant is received with at least one microphone in a microphone array to generate a microphone signal. The microphone signal is filtered using a spatial filter based on the main beam and the echo cancellation filter to generate a cabin output signal which is broadcasted to the source of far end speech. | 05-26-2016 |
| 20160150337 | Reference Microphone For Non-Linear and Time Variant Echo Cancellation - A far end signal from a far end user and broadcasting the far end signal at a loudspeaker is received. A signal for use in an echo transfer function is determined, and the signal selected from between: the far end signal or an echo reference signal received from a near field microphone disposed in close proximity to the speaker. The near field microphone sensing the far end signal that is broadcast from the speaker while sensing the near-end speech and ambient noise at insignificant energy levels compared to the speaker signal. An echo transfer function is determined based at least in part upon the selected signal, and the echo transfer function represents characteristics of an acoustic path between the loudspeaker and a far field microphone that is disposed at a greater distance from the speaker than the near field microphone. An estimated echo is determined based at least in part upon the echo transfer function. The estimated echo is subtracted from a signal received from the far field microphone, the subtracting effective to cancel an echo present in the signal received from the far end microphone. | 05-26-2016 |
| 20160189700 | METHOD AND APPARATUS FOR REDUCING DISTORTION ECHO - Method and apparatus for reducing distortion echo are provided. K-path amplification and pre-distortion process are performed to the downlink reference signal to obtain K-path pre-distorted signals. Afterwards, filtering is performed using the self-adaptive filters which correspond to the downlink reference signal x(t) and the K-path pre-distorted signals to obtain the filtering signals. Error signals are obtained by calculating differences between the target signal and each of the filtering signals. The minimum-value fusion process is performed to the error signals to obtain the residual signal which is then output as the final self-adaptive echo cancellation. In embodiments of the present disclosure, the residual signal is relatively small as the minimum-value fusion process is performed to the error signals. That is to say, echo loss is relatively great. Therefore, the method may provide echo loss with high amplitude under a situation that a speaker has relatively serious distortion. | 06-30-2016 |
| 20160189727 | METHOD AND APPARATUS FOR REDUCING ECHO - The embodiments of the present disclosure provide a method and an apparatus for reducing an echo, which apply a first stage adaptive filter to a downlink reference signal to obtain a first stage filtered signal and a first error signal; after performing a K-path gain process and a K-path pre-distortion process to the downlink reference signal, apply the at least one second stage adaptive filter to a K-path pre-distorted signal to obtain a second stage filtered signal and a second error signal; and perform a minimum value fusion process to the first error signal and the second error signal so as to obtain a residue signal, which is considered as a final output of adaptive echo cancellation. Accordingly, the present disclosure realizes steadily providing a relatively high echo loss under a circumstance that a speaker suffers a relatively great distortion. | 06-30-2016 |
| 20160189731 | PROCESS AND ASSOCIATED SYSTEM FOR SEPARATING A SPECIFIED AUDIO COMPONENT AFFECTED BY REVERBERATION AND AN AUDIO BACKGROUND COMPONENT FROM AN AUDIO MIXTURE SIGNAL - Processes are described herein for transforming an audio mixture for which a specific component is affected by reverberation, into a specific dry component (i.e. unaffected by the reverberation) and a background component. In the process described herein, the long-term effects of reverberation are explicitly taken into account by modelling the spectrogram of the specific component as the result of a matrix convolution along time between the spectrogram of the specific dry component and a reverberation matrix. Parameters of the model are estimated iteratively by minimizing a cost-function measuring the divergence between the spectrogram of the mixture signal and the model of the spectrogram of the mixture signal. | 06-30-2016 |
| 20160196834 | WIRED AND WIRELESS EARSET USING EAR-INSERTION-TYPE MICROPHONE | 07-07-2016 |
| 20160205262 | ACOUSTIC ECHO CANCELLATION FOR AUDIO SYSTEM WITH BRING YOUR OWN DEVICES (BYOD) | 07-14-2016 |
| 20190147900 | Sound Collection Apparatus for Far-Field Voice | 05-16-2019 |
| 20190149915 | METHOD AND DEVICE FOR IN EAR CANAL ECHO SUPPRESSION | 05-16-2019 |
