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: In a communication machine room wideband noise suppression system, a sensing unit detects a noise source produced by a fan during the operation thereof and generates a noise input signal and a feedback signal, which are sent to a signal amplifying unit for signal amplification. The amplified signals are then sent to a signal converting unit and converted into digital signals. A multi-channel hybrid controller receives the digital signals and makes corrections and conduct rapid convergence algorithm to derive a reverse digital signal, which is sent to the signal converting unit and converted into a reverse analog signal. The reverse analog signal is sent to the signal amplifying unit for power amplification to generate a control signal for driving a loudspeaker unit to produce interfering acoustic wave, so as to cancel out the noise source and achieve the purpose of eliminating wideband noise.

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: An apparatus comprises a first audio amplifier circuit configured to provide an analog audio signal and an analog switch circuit including a first input configured to receive the analog audio signal, a second input configured to receive a first digital data signal, and a first output configured to provide one of the digital data signal or the analog audio signal. The apparatus also includes a first feedback circuit coupled to the first audio amplifier circuit and the analog switch circuit output, the feedback circuit configured to bias the first audio amplifier circuit.

Abstract: A hearing aid (200) with multiple microphones comprises a first microphone (1) for converting sound into a first audio signal, a second microphone (20) for converting sound into a second audio signal, directional processing means for combining the first and said second audio signal according to a mixing ratio to form a spatial signal, estimating means for estimating a first acoustic feedback signal entering the first microphone and a second acoustic feedback signal entering the second microphone, processing means (4) for processing said spatial signal by applying a gain not exceeding a resulting maximum gain limit to form a hearing loss compensation signal, wherein the resulting maximum gain limit is derived from the first and second acoustic feedback signals and the mixing ratio, and an output transducer (3) for converting the hearing loss compensation signal into an acoustic output. The invention further provides a method and a computer program product.

Abstract: A method and apparatus for increasing phase margin in a feedback circuit of an active noise reduction headphone. The method includes providing an acoustic block comprising an acoustic driver comprising a voice coil mechanically coupled along an attachment line to an acoustic energy radiating diaphragm, the acoustic block further comprising a microphone positioned along a line parallel to an intended direction of vibration of the acoustic diaphragm and intersecting the attachment line, the acoustic block characterized by a magnitude frequency response compensating the magnitude frequency response by a compensation pattern that has a positive slope over at least one spectral range above 10 kHz.

Abstract: Feedback detection, adaptive notch filtering, and gain adjustment are combined for managing feedback. Feedback detection includes detection of short term and long term spectral peaks and assessing their magnitude, shape, rate of growth, and power concentration ratio. A plurality of notch filters are available and are allocated according to the spectral magnitude of the feedback detected. Wide band gain adjustment supplements the notch filters.

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: An apparatus and method of canceling a vocal component includes a first vocal canceling unit to delay each of the left and right channel input signals for a predetermined time and to feed-forward cross mix the delayed left and right channel signals with the left and right channel input signals, a sound stage widening unit to delay each of the left and right channel signals output from the first vocal canceling unit for a predetermined time and to feedback cross mix the signal of each delayed channel signals with the left and right channel signals, and a second vocal canceling unit to low-pass filter the left and right channel signals output from the sound stage widening unit and to mix the low-pass filtered left and right low frequency components and a high frequency component of the difference between the left and right channels to cancel the vocal component from an audio signal.

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: Method and apparatus for microphone matching for wearable directional hearing assistance devices are provided. An embodiment includes a method for matching at least a first microphone to a second microphone, using a user's voice from the user's mouth. The user's voice is processed as received by at least one microphone to determine a frequency profile associated with voice of the user. Intervals are detected where the user is speaking using the frequency profile. Variations in microphone reception between the first microphone and the second microphone are adaptively canceled during the intervals and when the first microphone and second microphone are in relatively constant spatial position with respect to the user's mouth.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: A speaker system includes an interface, a powered mixer, a first amplifier, a switch, a feedback killer and a second amplifier. The interface can receives an audio signal from a sound source. The powered mixer adjusts the audio signal by amplifying some of the frequencies of the audio signal. The first amplifier preliminarily amplifies the adjusted audio signal. The switch connects the first amplifier to the feedback killer if the amplitudes of any frequencies of the preliminarily amplified audio signal are outside a predetermined range and otherwise connects the first amplifier to the second amplifier. If the first amplifier is connected to the second amplifier, the second amplifier amplifies the gain of the preliminarily amplified audio signal. If the first amplifier is connected to the feedback killer, the feedback killer executes feedback killing on the preliminarily amplified audio signal before sending it to the second amplifier.

Abstract: An audio feedback suppression device is provided that outputs a sound while changing a phase so as not to generate audio feedback without performing a relatively complicated process. A phase control section 2 generates a phase adjustment value ? gradually varying with the lapse of time and gives the value ? to an adjustment signal generation section 121 of a phase shifter 1. Phase adjustors 111 and 112 of the phase shifter 1 output two first stage adjusted signals having phases which are shifted from each other by ?/2 in accordance with a shifter input signal. An adjusting signal generation section 121 generates a cosine value cos(?) and a sine value sin(?) of a phase adjustment value 0 and gives the values to respective multipliers 123A and 123B. The multiplier 123A multiplies one of the first stage phase adjusted signals to the cosine value cos(?) and the multiplier 123B multiplies the other first stage phase adjusted signal to the sine value sin(?).

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 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: 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 method for reducing howling in a communication system containing collocated mobile devices is presented. In a transmitter, an audio signal is received at a microphone. Acoustic feedback is removed from the audio signal and the resulting signal is encoded and transmitted either using direct or trunked mode operation to a receiver. The encoded signal is decoded at the transmitter, in addition to at the receiver, and fed back to an echo canceller with sufficient delay to account for substantially the entirety of a loop delay from encoding of the audio signal to reception of the acoustic feedback at the microphone to enable removal of the acoustic feedback. An estimate of the acoustic feedback is used to initially remove the acoustic feedback, the error being fed back to the processor to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

Abstract: An audio system includes a signal processor for processing an audio signal, and a feedback suppressor circuit configured for modelling a feedback signal path of the audio system by provision of a feedback compensation signal based on sets of feedback model parameters for the feedback signal path that are stored in a repository.

Abstract: This invention relates to a method for virtual bass synthesis. The low frequency signal is attained by applying a low pass filter to the original. In order to reduce the operations, process of down sampling the low frequency signal, moving the low frequency signal to a series of harmonics whose frequencies are integral times as large as the frequency of low frequency signals, and then up sampling them are provided. By means of psycho-acoustic theory, the weights of harmonics are attained and applied to the harmonics. Finally the weighted harmonics are combined to produce the bass signal. As the result, the virtual bass effect which is almost the same as the low frequency of the original audio signal can be accomplished. Because the harmonic signals are high frequency ones, the virtual effect can be made in the panel speakers or ordinary low-end speakers.

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: 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: 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: 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 semiconductor die with an integrated circuit providing a signal conditioner (106) for a capacitive transducer (105), comprising: a gain stage (101) configured to receive an analogue transducer signal; an analogue-to-digital converter (102) coupled to receive a signal outputted from the gain stage (101) and to provide a digital signal. A feedback signal is provided via a digital-to-analogue converter (104) and a digital signal processor (103) that receives the digital signal; and the gain stage (101) is configured with a first input (107) and second input (108) coupled to receive the analogue transducer signal and the feedback signal, respectively.

Abstract: An advanced video teleconferencing (AVTC) system uniquely combines a number of features to promote a realistic “same room” experience for meeting participants. These features include an autodirector to select audio and video sources and to compose shots, a collaboration interface for each participant to communicate nonverbal information, directional LEDs to privately alert participants, audio reflected from the main display, and a collaborative table to share a video of objects or papers on a table. When implemented with sufficient bandwidth for take advantage of these features and to keep latency time low, this AVTC system results in a highly realistic and productive teleconferencing experience.

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: This invention concerns a method, and a device, for feedback cancellation. This invention also concerns a computer program product comprising computer program code means to make a computer execute a procedure for feedback cancellation. The method comprises providing an adaptive feedback cancellation filter which adapts under the control of a control module, and filtering at least one input of the control module to suppress correlated signals from the input prior to the control module operating upon the input. The device comprises an adaptive feedback cancellation filter, a control module and at least one filter. The control module controls adaptation of the adaptive feedback cancellation filter. The filter suppresses correlated signals from an input to the control module prior to the control module operating upon the input.

Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.

Abstract: 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.

Abstract: A first set of signals from an array of one or more microphones, and a second signal from a reference microphone are used to calibrate a set of filter parameters such that the filter parameters minimize a difference between the second signal and a beamformer output signal that is based on the first set of signals. Once calibrated, the filter parameters are used to form a beamformer output signal that is filtered using a non-linear adaptive filter that is adapted based on portions of a signal that do not contain speech, as determined by a speech detection sensor.

Abstract: A system providing method and apparatus to detect occurrence of an entrainment artifact and address it. The system analyzing a feedback canceller filter for certain characteristics that are associated with an entrained filter. When an entrained filter is detected, the feedback canceller filter is reset to a good filter that ideally represents the current approximate external acoustic feedback path without the characteristics of the entraining signal.

Abstract: An audio amplifier is operated in an audio system with optimized source impedance to minimize distortion in a loudspeaker that is paired with the audio amplifier. The audio amplifier provides an amplified audio signal to drive the loudspeaker. A variable output impedance of the audio amplifier is controlled using a feedback control loop to allow negative output impedance at low frequencies that changes to positive output impedance at higher frequencies. The change from negative output impedance to positive output impedance occurs at a determined threshold frequency or determined transitional frequency band.

Abstract: A howling detector is provided which can discriminate between howling and a signal having a strong narrow-band component, thereby detecting howling with higher accuracy. The howling analyzer includes a frequency analyzing section for analyzing a frequency of a time signal, a level calculating section for calculating a level of a signal output from the frequency analyzing section, a howling detecting section for deciding whether howling occurs or not by analyzing the level having been calculated by the level calculating section, a periodic signal detecting section for deciding whether or not time progression of the level having been calculated by the level calculating section has periodicity, and a howling deciding section for finally deciding whether howling occurs or not based on decision results of the howling detecting section and the periodic signal detecting section.

Abstract: Duplex audio is provided for a mobile communication device and an accessory. In some embodiments, the accessory can selectably operate in a duplex audio mode, concurrently sending audio to and receiving audio from the mobile communication device, or in another audio mode. In duplex audio mode, the accessory can enable its internal audio processing operations (e.g., echo cancellation) while the mobile communication device disables its corresponding internal operations or vice versa. The mobile communication device can control when the accessory transitions into and/or out of duplex audio mode.

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: This invention describes a method for generating noise references for adaptive interference cancellation filters for applications in generalized sidelobe canceling systems. More specifically the present invention relates to a multi-microphone beamforming system similar to a generalized sidelobe canceller (GSC) structure, but the difference with the GSC is that the present invention creates noise references to the adaptive interference canceller (AIC) filters using steerable beams that block out the desired signal when the beam is steered away from the desired signal source location.

Abstract: There is provided an echo canceller which can appropriately cancel an echo even when a low-frequency component is included in the signal to be passed. The echo canceller includes an echo replica forming means for forming an echo replica signal from a far-end input signal by using an adaptive filter including a filter section and a coefficient update section, and an echo cancellation means for removing an echo component in a near-end input signal by subtracting the echo replica signal from the near-end input signal. The echo canceller further includes an offset removal means for removing an offset component produced under an effect of low frequencies from the filter coefficient of the adaptive filter.

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: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

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: 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: In a room with strong low-frequency modes the control of excessively long decays is problematic or impossible with conventional passive means. In this patent application a systematic methodology is presented for active modal equalization able to correct the modal decay behaviour of a loudspeaker-room system. Two methods of modal equalization are proposed. The first method modifies the primary sound such that modal decays are controlled. The second method uses separate primary and secondary radiators and controls modal decays with sound fed into at least one secondary radiator. Case studies of the first method of implementation are presented.

Abstract: A method and apparatus are provided for automatic alignment of a notch filter in a receiver. The method comprises the steps of determining a frequency of an interfering signal, monitoring an energy in the interfering signal, tuning the notch filter based on an initial tune value, detecting an energy content in the radio signal after the tuning step, incrementally tuning the notch filter away from the initial tune value while monitoring the energy in the interfering signal, repeating the step of detecting and the step of incrementally tuning until the energy in the interfering signal is minimized, and storing a new tune value as the initial tune value. The notch filter is configured to filter the radio signal. The new tune value indicates a minimized energy in the interfering signal.

Abstract: An active noise cancellation earphone (1) has an acoustic path including a cavity (36) and a pipe (20) leading to the auditory canal (40) which are arranged to form an oscillator in use which has the effect of recovering the open loop system phase characteristics at a selected frequency or frequency range. The earphone (1) also has two parts (5,18) which can be adjusted relative to each other to allow the earphone (1) to be comfortably and correctly positioned in use.

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.