Abstract: A howling suppression device that can reduce quality deterioration of processed sound includes: a delay unit delaying the input signal to output the delayed input signal as a reference signal; a signal separation unit including an adaptive filter extracting a periodic signal component from the reference signal by adaptively updating a filter coefficient; a howling detection unit detecting an occurrence of howling using at least a signal of the periodic signal component output from the adaptive filter; and a howling suppression unit. The howling suppression unit includes: a suppression filter obtaining the updated filter coefficient from the adaptive filter with timing when the howling detection unit detects the occurrence of the howling, to extract the periodic signal component from the reference signal based on the filter coefficient; and a subtractor subtracting the periodic signal component from the input signal so as to output a signal obtained by the subtraction.

Abstract: 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.

Abstract: An acoustic-feedback detection apparatus includes: a first level detecting section configured to detect a signal level of sound signals obtained from a position in a sound-signal system in which a microphone and speaker are connected; a first extracting section configured to extract, from the sound signals of which the signal level is detected, signals in a band having a bandwidth predetermined for each of at least one predetermined center frequency; a second level detecting section configured to detect a signal level of the signals in each band, the signals being extracted by the first extracting section; and a determining section configured to determine whether or not acoustic feedback is occurring, on the basis of a threshold determined according to the signal level detected by the first level detecting section and a waveform of each signal level detected by the second level detecting section.

Abstract: 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.

Abstract: 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.

Abstract: A method and device for transforming ambient audio are provided. Example embodiments may include monitoring ambient audio proximate to a sound processing device located in an environment. The device may receive a selection from a user interface. The selection may comprise one of a number of available first selections, each available first selection identifying one of multiple transformation modes. The device may access memory to obtain transformation audio and process the transformation audio, based on the ambient audio and the selection. The device may also use the transformation audio to provide modified output audio for propagation into the environment.

Abstract: A method of changing at least two parameter settings of a device includes detecting an abnormal change in an external feedback path and an input signal generated by an abnormal pressure wave, and activating a pressure wave detection switch and an abnormal feedback path detection switch for changing the at least one parameter setting in the device. A device includes a digital signal processor configured to implement a detection algorithm to detect an abnormal change in an external feedback path, an adaptive internal feedback cancellation system for continually monitoring and responsively adapting to changes occurring in an external feedback path, a pattern recognition algorithm for detecting input signals generated when an abnormal pressure wave is generated, at least two parameter settings for adjusting the characteristics of the device, and an abnormal feedback path detection switch for switching the device to a next available parameter setting.

Abstract: An audio processing system is provided for controlling the acoustics of a loudspeaker-room system. The loudspeaker-room system having a listening room and loudspeakers located in said listening room, and transfer functions with linear and non-linear components. The audio processing system comprises a compensator with a transfer function for obtaining at least two compensated signals from the input signals. The transfer functions of the compensator may include linear and non-linear components and are inverse to the transfer functions of the loudspeaker-room system to the extent that a desired overall transfer function is established.

Abstract: A handsfree call apparatus includes: a speaker held in the vicinity of a user's ear by a holding unit with a speaker unit thereof not being attached to a baffle board so as to mix sounds emitted from the front and back of a diaphragm thereof; a microphone, which is provided to a region where sounds emitted from the front and back of the diaphragm of the speaker are mixed and canceled out, configured to collect the call voice of the user; an output unit configured to output a call voice signal of the user from the microphone to transmit this to a call partner; and an input unit configured to receive a call voice signal from the call partner to supply this to the speaker.

Abstract: Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In particular, one embodiment of the present invention relates to a method for determining an approximate impulse-response vector for canceling an acoustic echo resulting from an audio signal transmitted from the first location to the second location. This method includes forming a trust region within a search space based on computing a recursive specification vector defining the trust region. The method also includes computing a recursive shadow-impulse-response vector that lies substantially within the trust region, and computing the approximate impulse-response vector based on the recursive shadow-impulse-response vector and the recursive specification vector.

Abstract: An electronic audio device with a digital audio output channel in which an amplifier output voltage is gradually ramped up and down to avoid causing a popping sound when the device is turned on and off. This is accomplished without employing any additional hardware, by incrementally changing a digital input word applied to a digital audio source, such as a DSP, so as to gradually change the amplifier output voltage between a minimum, such as zero volts, and a DC working voltage. On powering up, the amplifier is only turned on after the digital word is applied, but while it still results in a minimum amplifier output, and on powering down the amplifier is turned off after it's output has been ramped down, but before removing the digital input word. Sources and output channels can also be switched over by powering down, and then powering up, following the same method.

Abstract: An electronic device comprises a microphone, a transceiver circuit, a loudspeaker, a sense element and a processing circuit. The microphone is configured to receive a first audio signal. The transceiver circuit is configured to communicate the first audio signal to a remote device and to receive a second audio signal from the remote device. The loudspeaker is configured to provide an audible signal based on the second audio signal. The sense element is configured to sense the audible signal provided by the loudspeaker. The sense element may be positioned at a distance from the loudspeaker different than a distance between the microphone and the loudspeaker. The processing circuit is configured to process at least one of the first audio signal and the second audio signal based on a sensed signal from the sense element.

Abstract: Improving the acoustical communication between interlocutors in at least two positions in a room includes generating electrical signals representative of acoustical signals present at the respective interlocutor positions, amplifying each of the electrical signals and converting the amplified electrical signals into acoustical signals. A time delay is applied to the electrical signals such that the acoustical signal arriving first at one of the interlocutor positions originates from the direction of the other interlocutor position.

Abstract: A system and method for analyzing a signal to monitor the dynamics of its magnitude and frequency characteristics over time. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention may comprise a feedback control block operable to determine a candidate frequency having potential feedback such that the feedback control block is further operable to perform an iterative analysis of the magnitude of the audio signal at the candidate frequency to determine the growth characteristics of the signal. The electronic circuit may further include a test filter block operable to deploy a test filter at a candidate frequency and a permanent filter block operable to deploy a permanent filter at the candidate frequency if the feedback control block determines that the growth characteristics of the signal at the candidate frequency comprises feedback characteristics after the test filter has been deployed.

Abstract: 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.

Abstract: A method and a hearing device for cancelling or preventing feedback are disclosed. The hearing device comprises a microphone (1), a transfer function (2) and a receiver (3), wherein the transfer function (2) defines relation between an input signal (12) of the hearing device and an output signal (13, 13?) of the hearing device. The method according to the present invention comprises the steps of estimating an external transfer function (21) of an external feedback path (11) defined by sound traveling from the receiver (3) to the microphone (1), estimating the input signal (12?) having no feedback components of the external feedback path (11) using an auxiliary signal (15), which does not comprise feedback components of the external feedback path (11), and using the estimated input signal (14) for estimating the external transfer function (21) of the external feedback path (11).

Abstract: A system and method for analyzing a signal to identify one or more candidate frequencies that may have a potential feedback problem. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention, may comprise a feedback control block operable to: 1) determine a magnitude of a first dominant spectral component in a wide-band span of frequencies of a signal, 2) determine a magnitude of a second dominant spectral component in a mid-band span of frequencies of the signal, and 3) determine a magnitude of a third dominant spectral component in a narrow-band span of frequencies of the signal. The feedback control block may then further analyze the magnitude and frequency of the first, second, and third dominant spectral components to determine the presence of feedback characteristics in the signal.

Abstract: A microphone apparatus is obtained capable of preventing howling effectively, while having no time difference between a timing of voice utterance to a microphone and a timing of sound emission from a speaker and having simple and low-cost physical and electrical configurations. There are provided microphone units or microphones disposed at locations with the same acoustic conditions, and switching units disposed corresponding to the microphone units or microphones, respectively, for switching sequentially an audio signal transformed in each of the microphone units or microphones to generate an output signal. The switching units may switch sequentially the audio signal transformed in the microphone unit or microphone at an shorter time interval than a time period from a time point the microphone unit or microphone is turned on to a time point an output thereof grows to reach a maximum level by howling.

Abstract: An anti-feedback device includes an anti-feedback filter provided in a closed loop. The anti-feedback device down-samples a signal of specific band selected from an output signal of an adaptive target signal transfer system and a signal of the same band selected from an input signal of the transfer system, and a filtering coefficient of the adaptive filter is updated by use of the down-sampled signals. The filter controller controls a filtering characteristic of the anti-feedback filter so that a peak gain of a frequency of an amplitude characteristic within a specific band of a closed loop determined from the filtering coefficient of the adaptive filter is suppressed. Moreover, the filter controller estimates a gain of the closed loop outside the specific band from the amplitude characteristic in the specific band and controls the amount of suppression of the anti-feedback filter outside the band in accordance with a result of estimation.

Abstract: 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.

Abstract: Disclosed herein are conferencing products implementing an acoustic echo cancellation system that utilizes converging coefficients and a detector of turn-off and/or turn-on events of push-to-talk microphones, and, further, that mitigates against divergence and/or drift of coefficients and other variables of an echo canceller. A push-to-talk detector may be used that includes a high-pass filter, a transient detector, or an adjustable high-pass filter. An echo canceller may be disabled as to a push-to-talk microphone that has been turned off.

Abstract: An automatic gain control unit controls the gain applied to an input signal produced by a microphone subject to ambient noise. The automatic gain control circuit continually monitors the signal level of said input signal. A first gain control circuit decreases the gain applied to the input signal in increments of a first size when the input signal exceeds a first predetermined level. A second gain control circuit increases the gain applied to said input signal in increments of a smaller size when the input signal falls below a second predetermined level and in response to the presence of a speech present signal. The second predetermined level is less than said first predetermined level. In one embodiment, the first gain control circuit controls the analog gain of a codec and the second gain control circuit controls the digital gain of the codec.

Abstract: 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.

Abstract: A control method to automatically adjust the volume of a sound transmitter based on the measurement and the computation of the acoustic statistics of the room where the sound is emitted. The system comprises at least one sensor, a calculator determining statistics of the signal collected by the at least one sensor and a controller using these statistics provided by the calculator to adjust the volume of the transmitter in the room.

Abstract: A system for reducing an echo and/or a residual echo in a microphone output signal includes an echo compensation filter configured to receive audio input signals and generate an estimated echo signal. A speech activity detector is configured to detect speech of a local speaker, and a combining circuit is configured to generate an echo compensated signal by subtracting the estimated echo signal from the microphone output signal. A residual echo reduction circuit is configured to suppress the residual echo in the echo compensated signal based on the detected speech activity, and output an echo suppressed output signal. The echo reduction system may reduce local background noise and residual echoes in a microphone output signal so that local speech may be transmitted free from undesirable signals.

Abstract: A system and method to detect and remediate unacceptable levels of speech intelligibility evaluates received test audio transmitted across and received in a space or region of interest. Intelligibility is improved by altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: A sound field controlling apparatus for a public-address system comprises a microphone that picks up a sound of a speaker, a loudspeaker that sound a sound signal based on the sound picked up by the microphone, a sound source position detector that detects a position of a sound source, and a signal processor that controls a level, delay time and equalizing property of the sound signal output to the loudspeaker in accordance with the sound source position detected by the sound source position detector.

Abstract: A system and method to detect and measure remediated speech intelligibility by evaluating received test audio transmitted across and received in a space or region of interest. Remediation of the test audio may include altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: An equalization system enhances the quality of communications between a remote party and a local party. The equalization system includes an equalization filter that equalizes an acoustic signal received from the remote party. The equalized acoustic signal is transmitted to a speaker based on the equalized acoustic signal. A device converts sound into electrical signals. The electrical signals are transmitted to an echo compensation filter that compensates for reflected sound. Filter characteristics of the equalization filter are based on filter characteristics of the echo compensation filter.

Abstract: A system reduces echoes in an audio system by de-correlating audio signals. Adaptive filters provide compensation based on the de-correlated signals. A controller controls the de-correlation based upon the adaptation state of the adaptive filters.

Abstract: A system automatically determines an equalizing filter characteristic for a communication system within a vehicle. The communication system includes a loudspeaker and a microphone or microphone array. The system transmits a predetermined test signal through the loudspeaker and receive the test signal through the microphone or microphone array. Based on the predetermined test signal and the received test signal, a transfer function is developed. The equalizing filter characteristic is then developed from the transfer function.

Abstract: Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In one embodiment of the present invention, a first digital signal encoding sounds produced at the first location is output from the first location, and a second digital signal encoding an acoustic echo and sounds produced at the second location is output from the second location. The method computes a control state that depends on the signals transmitted between the first and the second locations and computes an approximate acoustic echo based on the control state.

Abstract: A method and apparatus for automatically adjusting volume of an audio signal on a mobile device, comprising playing the audio signal at an initial volume, sampling the audio signal, estimating a transfer function based on an echo path characteristic between the played audio signal and sampled audio signal, selecting a volume policy based on the estimated transfer function, and adjusting the volume of the audio signal in accordance with the selected volume policy.

Abstract: A signal processing circuit includes a delaying unit that is configured to carry out delay processing on the basis of periodicity information synchronized with the periodicity of a periodic noise included in an input signal and a filter unit that is configured to receive the input signal and has a notch characteristic at a frequency f. The frequency f satisfies f=N/T, where N represents an integer equal to or greater than one and T represents a delaying time applied by the delaying unit.

Abstract: An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

Abstract: An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

Abstract: A hearing assistance system having adjustable bulk delay for cancellation of a time varying acoustic feedback path. The hearing assistance system including an FIR filter, coefficient update module, and delay rules module for programmable adaptive filtering. The hearing assistance system adjustable for continuous bulk delay adjustments. The hearing assistance system providing a number of coefficient update routines, including, but not limited to an LMS coefficient update process and a normalized LMS coefficient update process.

Abstract: A procedure and device for linearizing the characteristic curve of a vibration signal transducer such as a microphone that includes collecting signals, transmitting the signals, extracting information from the signals, dephasing such information by 180 degrees compared to the initial signals and taking the algebraic sum of the initial signals and dephased information.

Abstract: The invention concerns an audio system comprising a microphone, audio signal processing means, an output transducer and mans for detecting a possible feedback tone and the corresponding frequency of the feedback tone in the audio system between the output transducer and the microphone. According to the invention means for counteracting feedback are provided. Further, mans are provided for changing the phase of the audio signal at a given frequency.

Abstract: In a method and equipment for operating a voice-supported system, such as a communications and/or intercom/two-way intercom device in a motor vehicle, using at least one microphone and at least one loudspeaker to reproduce a signal generated by the microphone, as well as a bandpass filter configured between the microphone and the loudspeaker, a power of the signal as a function of a frequency is determined, and the bandpass filter is adjusted as a function of at least one local maximum of the power of the signal as a function of the frequency.

Abstract: A howling detection device detects a dominance ratio, which indicates a risk of howling to occur when a mixed signal obtained by mixing a plurality of sound signals collected by a plurality of microphones is outputted by a speaker. The howling detection device detects levels of the plurality of sound signals, compares, in a same time domain, the mixed signal with a signal regarding a sound to be outputted by the speaker as a noise reference signal, detects a time period, as a word ending section, during which the mixed signal is inputted after the noise reference signal falls, and calculates a dominance ratio by extracting only a level of the plurality of sound signals corresponding to the word ending section and determining a ratio of each of the extracted levels of each of the sound signals to a sum of the extracted levels of the plurality of sound signals.

Abstract: An adaptive howling canceller has a plurality of adaptive filters. A delay adds a time delay of an acoustic feedback path to an electric signal fed from an amplifier of a sound-reinforcement system. Each adaptive filter filters the output signal of the delay with a filter coefficient, which is periodically updated at an update interval. The update interval of each adaptive filter is set to decrease successively from a first one to a last one of the adaptive filters. Adders are arranged in correspondence to the adaptive filters in series between a microphone and the amplifier. Each adder subtracts the output signal of the corresponding adaptive filter from an output signal fed from a preceding adder to thereby provide an output signal to a succeeding adder. The output signal from each adder is inputted into the corresponding adaptive filter.

Abstract: Hearing aids with microphone and telephone coil are to be made simpler and more convenient. For this purpose it is provided to use an adaptive filter to compensate acoustic and electromagnetic feedback. In order to allow for the propagation delay differences, a delay element is connected downstream of the telephone coil. The microphone and telephone coil signals can be individually weighted with the factors a and b so that mixed mode is also possible.

Abstract: An acoustic system that eliminates the howling that occurs when the sound outputted by the speaker feeds back to the input device. The acoustic system comprises a digital signal processor (DSP) that divides the input audio signal into different frequency bands, and reduces the audio levels for the frequency bands where howling is most likely to occur. In one embodiment, the acoustic system comprises a sound source section that generates a test tone that substantially covers the entire human audible range such that the DSP can set the filter levels according to the feedback of the test tone. In another embodiment, the sound source section stores one waveform at a given pitch and generates waveforms of other pitches based on the stored waveform. In yet another embodiment, the pitches of the generated waveforms are dispersed into four frequency bands to create a test tone that resembles a chord or a musical tone.

Abstract: A method of processing a sound signal in an audio amplification device using frequency transposition, the method including the steps of: (a) receiving (11) an input sound signal, (b) determining (19) gains for amplifying the input sound signal at a plurality of input frequencies, (c) transposing (20) one or more of the input frequencies of the amplified sound signal to generate one or more output signals at transposed frequencies, (d) determining (31-33) the presence of an undesired feedback signal component resulting from the amplification and frequency transposition of the input sound signal at the input frequencies, and (e) correcting (34) the output signal at each of the transposed frequencies to compensate for the presence of the undesired feedback signal component.

Abstract: Suppressing one or more frequency ranges of a signal prevents the occurrence of feedback in a voice data communications application. A system recognizes a frequency range in a signal where feedback occurs, or anticipates a frequency range where feedback is anticipated. The signal includes a signal the input system generates or that the output system renders. The system suppresses the signal in the frequency range by disregarding one or more sampling bits representing the frequency range, or by applying one or more filters to attenuate or eliminate the signal in the frequency range. The system may monitor the signal to identify feedback resulting in different or additional frequency ranges and suppress the signal in the different or additional frequency ranges to prevent feedback from occurring.

Abstract: Howling, which occurs when amplifying a target sound collected by a first microphone through an amplification section and outputting the amplified sound as an intensified sound from a loudspeaker, is suppressed using a first and second acoustic signal. A first power spectrum is produced according to the first acoustic signal output from the first microphone collecting a sound. A second power spectrum is produced according to the second acoustic signal of a sound including at least the intensified sound and not including the target sound. Then, the first acoustic signal is filtered based on the first power spectrum, and the second power spectrum to output only an acoustic signal of the target sound to the amplification section.

Abstract: 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.

Abstract: A processor operates on samples of a digital output signal to determine samples of a digital correction signal. The output signal samples are directed to an output channel for transmission from a speaker. The digital correction signal samples are supplied to a first digital-to-analog converter for conversion into an analog correction signal. The subtraction circuit generates a difference between a first analog signal provided by a microphone and the analog correction signal. The analog correction signal is an estimate of a contribution to the first analog signal due to a direct path transmission between the speaker and the microphone. The processor also receives a digital input signal derived from the difference signal, and, performs acoustic echo cancellation on the digital input signal to obtain a resultant signal. The acoustic echo cancellation is configured to remove contributions to the digital input signal due to reflected path transmissions.

Abstract: A system for canceling acoustic feedback includes an input for receiving a digital audio signal and a processor configured to detect acoustic feedback signal in the digital audio signal and to determine the frequency of the feedback signal The system also includes a plurality of bandpass filters for attenuating the feedback signal. The processor is further configured to: select a bandpass filter from among the plurality of bandpass filters. The selected bandpass filter comprises a response characteristic that attenuates parts of the signal at the frequency of acoustic feedback signal.