Abstract: Computer-implemented methods, computer program products, and computer systems for testing a voice assistant device may include one or more processors configured for receiving test data from a database, wherein the test data may include a first set of coding parameters and a first user utterance having an expected device response. Further, the one or more processors may be configured for generating a first modified user utterance by applying the first set of coding parameters to the first user utterance, wherein the first modified user utterance is acoustically different than the first user utterance. The one or more processors may be configured for audibly presenting the first modified user utterance to a voice assistant device, receiving a first device response from the voice assistant device, and determining whether the first voice assistant response is substantially similar to the expected device response.

Abstract: Techniques for improving acoustic echo cancellation are described. Energy levels of audio data received from a microphone and representing near-end audio and reference audio data representing far-end audio are determined. If near-end audio is detected but far-end audio is not detected, a controller turns of or bypasses an acoustic echo cancellation system until far-end audio is again detected, thereby decreasing or eliminating distortion of the near-end audio by the acoustic echo cancellation system.

Abstract: A method of pre-processing audio for playback through an audio greeting card having at a speaker array of two or more speakers and an audio control circuit, by defining one or more device parameters associated with the greeting card, measuring an audio response from the greeting card when playing an original audio file, deriving correction gains to correct magnitude response characteristics of the audio response, deriving compressor threshold values to minimize level distortion of the audio response, deriving a virtualization transfer function from the one or more device parameters, and applying the correction gains, compressor threshold values, and the virtualization transfer function to generate an output audio file.

Abstract: A signal processing apparatus includes a generator, an output controller, and an echo canceller. The generator generates an output sound signal by combining an over-the-phone sound signal with a system sound signal different from the over-the-phone sound signal. The output controller outputs, to a loudspeaker, the output sound signal generated by the generator. The echo canceller cancels the output sound signal from an input sound signal input via a microphone located in a vicinity of the loudspeaker. The output controller suppresses a level of the system sound signal to be output from the loudspeaker so as not to be greater than a predetermined value within a range in which a volume value for the over-the-phone sound signal is settable.

Abstract: Audio systems and methods for suppressing residual echo are provided. First and second audio program content signals are received, and a residual signal from an echo canceler is received. A first spectral mismatch is determined based at least upon a cross power spectral density of the first program content signal and the residual signal. The first spectral mismatch is associated with an operational state of the audio system. The residual signal is filtered to reduce residual echo, based at least upon the first spectral mismatch, a spectral density of the first program content signal, and a spectral density of the residual signal.

Abstract: Audio systems and methods for suppressing residual echo are provided. First and second audio program content signals are received, and a residual signal from an echo canceler is received. A first spectral mismatch is determined based at least upon a cross power spectral density of the first program content signal and the residual signal. A second spectral mismatch is determined based at least upon a cross power spectral density of the second program content signal and the residual signal. The residual signal is filtered to reduce residual echo, based at least upon the first spectral mismatch, the second spectral mismatch, a spectral density of the first program content signal, a spectral density of the second program content signal, and a spectral density of the residual signal.

Abstract: Audio distortion compensation methods to improve accuracy and efficiency of audio content identification are described. The method is also applicable to speech recognition. Methods to detect the interference from speakers and sources, and distortion to audio from environment and devices are discussed. Additional methods to detect distortion to the content after performing search and correlation are illustrated. The causes of actual distortion at each client are measured and registered and learnt to generate rules for determining likely distortion and interference sources. The learnt rules are applied at the client, and likely distortions that are detected are compensated or heavily distorted sections are ignored at audio level or signature and feature level based on compute resources available. Further methods to subtract the likely distortions in the query at both audio level and after processing at signature and feature level are described.

Abstract: A device for reducing communications crosstalk including a plurality of channel inputs each configured to receive an incoming signal from a respective device. The device further includes at least one control input configured such that, when the control input is triggered, the control input activates a voltage divider that attenuates at least one of the incoming signals, thereby reducing crosstalk between a first channel carrying the at least one attenuated incoming signal and a second channel not carrying the at least one attenuated incoming signal. The at least one attenuated incoming signal and the remaining incoming signals are output to a respective receiver.

Abstract: An apparatus for eliminating impulse interference includes an impulse interference detecting unit, an impulse interference eliminating unit and a control unit. The impulse interference detecting unit detects whether impulse interference exists in an input signal according to a predetermined condition, and generates an impulse interference eliminating request when having detected that the input signal satisfies the predetermined condition. In response to the impulse interference eliminating request, the impulse interference eliminating unit performs an impulse interference eliminating process on the input signal to generate an output signal. According to an output signal quality index associated with the output signal, the control unit selectively adjusts the predetermined condition used for detecting whether the impulse interference exists.

Abstract: A sound emission and collection device includes a speaker, a filter processing a sound emission signal, microphones, echo cancellers cancelling regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones, a first integration section integrating adaptive filter coefficients taken out from the plurality of echo cancellers, a reverberation time estimation section estimating the reverberation time for each frequency band in the space in which the speaker and the plurality of microphones are present on the basis of the integrated adaptive filter coefficient, and an arithmetic operation section specifying a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time, calculating a filter coefficient for suppressing power of the specified frequency band, and setting the filter coefficient to the filter.

Abstract: A system according to an aspect of the present invention switches between the activation/deactivation of the speaker and microphone in speakerphone mode to help ensure the quality of audio communications. The system includes a processor, a microphone, a speaker, and a data relay transceiver. The microphone, speaker, and data relay transceiver are each in communication with the processor. The system further includes a memory in communication with the processor which stores instructions that, when executed by the processor, cause the processor to determine a threshold based on an average amplitude of a signal provided to the speaker by the data relay transceiver; sample the amplitude of the signal; and disable the microphone and enable the speaker if the sampled is greater than the threshold.

Abstract: Systems and methods described herein modify audio content on an electronic device. Embodiments can be configured to detect a mode of the electronic device to determine whether the device is in a telephone mode; receive a speech signal from a speech source while the device is in the telephone mode; and process the speech signal to improve the perceived quality of the speech at a recipient when the electronic device is in a telephone mode; wherein processing the speech signal to improve the perceived quality of the speech comprises, decreasing the signal level of audio content outside of a determined frequency band relative to the signal level of the audio content within the determined frequency band; and wherein the determined frequency band is a frequency band associated a vocal range of the anticipated speech content. The electronic device further includes a graphical user interface which allows a user to adjust any or all audio parameters including very high frequency attack or release times.

Abstract: 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.

Abstract: Before a party is admitted into a conference call, the party's communication connection to the conference bridge is tested for quality. If the quality is acceptable, the party is admitted into the conference. If the quality is not acceptable, an attempt is made to improve the quality. If the attempt fails, the party is given the choice of either hanging up and calling back or being admitted into the conference in a restricted mode that limits the adverse effects of the poor connection on the conference.

Abstract: Disclosed is an apparatus and method for cancelling a wideband acoustic echo that controls a data flow of data that is transmitted and received using a central processing unit, and calculates data using a calculation processing unit configured to be distinct from the central processing unit.

Abstract: An audio conferencing system having a base station, a loud speaker and one or more microphones generally operates to receive a F.E. audio signal, play the signal, and to receive a N.E. audio signal at the microphone that includes the F.E. audio signal modified by an echo path and a N.E. audio signal. The base station includes an adaptive filter whose length is controlled to be variable depending upon activity detected at each of one or more filter elements comprising the adaptive filter.

Abstract: Systems and methods for adaptive power control are provided. In exemplary embodiments, a primary signal is received. A noise power level of the primary signal is then estimated. The noise power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the noise power level to the power threshold.

Abstract: A system and method in accordance with a non-limiting aspect minimizes echo return loss in a communications system. The dial tone is broken on an analog telephone line connected to a hybrid circuit having selectable impedance configurations and that interfaces a digital communications circuit and the analog telephone line. A white noise signal as a test call is inserted through the hybrid circuit onto the analog telephone line corresponding to a selected impedance configuration. The returned white noise signal is used for measuring the echo return loss. The test call is dropped. This process is repeated at different impedance configurations to obtain echo return loss measurements at each of the impedance configurations. The optimum impedance setting configuration is determined based on the results and applied to the hybrid circuit and typically set during a running telephone configuration.

Abstract: A signal processing device includes an adaptive filter (5), a noise estimation circuit (10), and a double talk detection circuit (81) and operates so that the double talk detection circuit (81) detects a double talk by using the estimated noise obtained by the noise estimation circuit (10). The signal processing device further includes noise estimation means and detects a double talk by using an estimated noise, a microphone signal, and pseudo-echo. An echo removal method and device detect a double talk by using a reliability coefficient expressed as continuous values between 0 and 1. By using continuous values instead of two values 0 and 1, it is possible to reduce the affect of a detection error.

Abstract: An audio processing device for reducing the effect on a first signal of echo from a second signal, the device comprising: an echo reduction processor for processing the first signal to reduce echo in it, the echo reduction unit having: a first mode of operation for reducing echo of a first function from the first signal; and a second mode of operation for reducing echo of a second function from the first signal, the second function being more complex than the first function and the echo reduction processor being such as to consume more power in the second mode of operation than in the first mode of operation; and an echo reduction controller for controlling the echo reduction processor to operate in a selected one of the first mode of operation and the second mode of operation.

Abstract: 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.

Abstract: The present document discloses a method and system for improving echo cancellation in the presence of non-linear distortions. A circuit attenuates a transmit signal in one or more spectral bands based on a receive signal. The distorted echo comprised within the transmit signal results from distortions incurred by the receive signal when being rendered by an electronic device. The circuit compares energy values of the receive signal in multiple spectral bands with multiple corresponding spectral band dependent distortion thresholds and determines that for a first spectral band from multiple spectral bands. A first energy value from multiple energy values exceeding a first distortion threshold from multiple distortion thresholds attenuates the transmit signal in one or more spectral bands at higher frequencies than the first spectral band with corresponding spectral band dependent attenuation factors.

Abstract: The present invention relates to call voice signal processing, and particularly, to apparatus and method for echo control in parameter domain. The apparatus includes: an echo detection module configured to extract parameter domain characteristic parameters of a far-end output signal and a near-end input signal, respectively, through parameter domain decoding, and detect whether a near-end input signal frame is an echo frame according to the parameter domain characteristic parameters; and a transition module configured to re-quantize the fixed codebook gain, the adaptive codebook gain and an LSF prediction error in the near-end input signal frame when the echo detection module detects the near-end input signal frame as a non-echo frame, and replace the original fixed codebook gain, the original adaptive codebook gain and the original LSF in the near-end input signal frame.

Abstract: A packet voice transceiver adapted to reside at a first end of a communication network and to send an ingress communication signal comprising voice packets to, and receive an egress communication signal comprising voice packets from, a second packet voice transceiver residing at a second end of the communication network. The packet voice transceiver includes a far-end echo canceller that reduces echo that is present in the egress communication signal. The far-end communicates with other functional components of the transceiver system and cancels echo or refrains from canceling echo based on the activity of the other functional components.

Abstract: Disclosed is an apparatus and method for cancelling a wideband acoustic echo that controls a data flow of data that is transmitted and received using a central processing unit, and calculates data using a calculation processing unit configured to be distinct from the central processing unit.

Abstract: Architecture that employs a signal (e.g., audible or inaudible sounds) to detect if endpoints of a communications session are sufficiently close to each other to induce echo, and then control (e.g., muting) is applied to one or more of the endpoints to prevent echo. The signals can be played and detected from the endpoints or a central conferencing component such as a multiple control unit (MCU). The MCU can provide support for legacy endpoints as well. When echo is detected, the offending endpoint(s) can be controlled to mute one or more onboard devices such as a speaker or microphone. The device(s) can be muted from a remote component or for a local component or locally by the endpoint user. A notification can be sent that notifies the endpoint user that the mute operation has been applied or should be applied to one or more of the local devices.

Abstract: System and method for processing communication signals in a communication system having a detector for detecting a parameter of a communication signal. A communication signal is provided to a non-linear processor (NLP) adapted to examine the signal and to decide whether or not to enter an active state based upon a parameter of the signal. If the NLP enters an active state, the NLP performs non-linear processing on the signal. The NLP communicates to the detector whether the NLP is active or inactive. If the NLP is active, a processing step of the detector is disabled.

Abstract: A filter coefficient setting device for setting a filter coefficient of an echo prevention device including a first FIR filter, and a second FIR filter, comprises: a filter coefficient initial setting portion configured to set a predetermined filter coefficient for the first and second FIR filters when the echo prevention device is started.

Abstract: An acoustic echo cancellation system buffers remote side signals to one or more storage areas. A delay controller filters and delays the remote side signals to correlate the remote side signals to local side signals. An acoustic echo cancellation core pulls the remote side signals from the buffer through a delay controller. A delay estimator generates an estimate of a delay between the buffered remote side signals and the local side signals. The delay estimator renders a confidence level that indicates a degree of likelihood for each potential match between delayed remote side signals and the local side signals. The acoustic echo cancellation core substantially cancels a portion of the echo or reverberation within the local side signals based on the estimate of the delay and the confidence level.

Abstract: Techniques to characterize and reduce acoustic echo are described. For example, a mobile computing device may comprise an audio characterization module to send a first signal to a first component and receive a second signal from a second component. The first signal may be a deterministic audio training sequence and the second signal may be a recovered audio training sequence. The audio characterization module compares the first signal and the second signal to determine an audio echo path. Other embodiments are described and claimed.

Abstract: A noise suppression system reduces low-frequency noise in a speech signal using linear predictive coefficients in an adaptive filter. A digital filter may update or adapt a limited set of linear predictive coefficients on a sample-by-sample basis. The linear predictive coefficients may be used to provide an error signal based on a difference between the speech signal and a delayed speech signal. The error signal represents an enhanced speech signal having attenuated and normalized low-frequency noise components.

Abstract: In an example embodiment, an example method is provided for echo mitigation in a conference call. In this method, a test audio signal is transmitted to a conference endpoint and as a result, an echo associated with the transmittal of the test audio signal is received. One or more parameters of the echo are then identified and an echo mitigation process is selected from multiple echo mitigation processes based on the identified parameters. The selected echo mitigation process is then applied.

Abstract: Sound quality is enhanced in a sound system including handsets and headsets. Handset sound enhancing algorithms are implemented in a handset. The handset automatically determines which, if any, of a plurality of headset sound enhancing algorithms are active in a headset in communication with the handset. The handset determines how to use the handset sound enhancing algorithms in a sound processing channel based on which of the headset sound enhancing algorithms are active in the headset.

Abstract: The invention relates to methods for controlling adaptive filters, where a first step involves a first input signal, which has a narrowband frequency range and occurs at least intermittently during an adaptation phase in the filter, at the input of the filter prompting the adaptation phase to be continued until an echo signal which is conditional upon the first input signal and in the same frequency range is compensated for completely, a second step involves the adaptation phase being suspended when complete compensation for the echo signal has been achieved, and a third step involves the adaptation phase being continued if the first input signal is followed by a second input signal which has a narrowband frequency range and which differs from the frequency range of the first input signal at least by a frequency value, with the above steps being repeated starting at the first step, or with a fourth step involving a new adaptation phase being performed if the first input signal is followed by a third input sig

Abstract: A voice communication end device provides echo reduction when operating in a full duplex mode using acoustic echo cancellation, which includes periodic audio queue adjustments to account for drift. The end device performs a quality assessment by calculating consistency statistics for the queue adjustments to determine whether acoustic echo cancellation would be ineffective, and if so falls back to a half duplex mode using voice switching.

Abstract: In one embodiment, the presence of double talk (DT) is detected in a telecommunications network having a near-end user and a far-end user. The energies of both (1) a signal received from the far-end user by the near-end user and (2) a signal to be communicated from the near-end user to the far-end user are computed. An echo return loss (ERL) estimate is calculated based on the energy calculations, and a preliminary decision is made as to whether DT is present based on the ERL estimate and the energy calculations. If DT is detected, then a counter is set to a hangover value. If DT is not detected, then the counter is reduced. This process is repeated, and, for each iteration, a final decision as to whether DT is present is made based on the counter value.

Abstract: Sound quality is enhanced in a sound system including handsets and headsets. Handset sound enhancing algorithms are implemented in a handset. The handset automatically determines which, if any, of a plurality of headset sound enhancing algorithms are active in a headset in communication with the handset. The handset determines how to use the handset sound enhancing algorithms in a sound processing channel based on which of the headset sound enhancing algorithms are active in the headset.

Abstract: In a data processing system, a method includes receiving samples of a send path input signal (Sin) and samples of a receive path input signal (Rin); in response to receiving each sample of the send path input signal, providing a double-talk indicator to indicate whether or not double-talk has been detected; using the double-talk indicator to determine a double-talk density within a predetermined number of samples of the send path input signal or the receive path input signal; and using the double-talk density to provide a heavy double-talk indicator which indicates whether or not heavy double-talk has been detected. In response to the heavy double-talk indicator indicating that heavy double-talk has been detected, a heavy double-talk mode may be entered.

Abstract: Disclosed herein are conferencing systems that include a conferencing device and a head-processing system, both including selectively enabled speakers, the conferencing device including an acoustic echo canceler that remains operable in the system regardless of whether its internal speaker is selected to be enabled. Detailed information on various example embodiments of the inventions are provided in the Detailed Description below, and the inventions are defined by the appended claims.

Abstract: Sound quality is enhanced in a sound system including handsets and headsets. Handset sound enhancing algorithms are implemented in a handset. The handset automatically determines which, if any, of a plurality of headset sound enhancing algorithms are active in a headset in communication with the handset. The handset determines how to use the handset sound enhancing algorithms in a sound processing channel based on which of the headset sound enhancing algorithms are active in the headset.

Abstract: A sound pickup apparatus selecting one of a plurality of microphones and performing echo cancellation processing for a plurality of microphones with an echo canceller to output a sound. The sound pickup apparatus sets a “learning mode” when the power supply is turned on, outputs a calibration sound from an echo cancellation calibration sound generator via a speaker, detects an echo at that time with a microphone and obtains an echo cancellation use parameter canceling the echo.

Abstract: A state deciding unit 2 makes a decision of double talk when both received signal and transmission signal are in a speech burst, and when the amount of reduction in an echo removal level per prescribed time period exceeds a predetermined threshold, the echo removal level being represented as a level difference between the transmission signal and a residual signal passing through the echo elimination by an adaptive filter 1, and when the amount of changes of the signal level of the received signal is less than a prescribed reference value.

Abstract: Coefficients of an adaptive filter representative of an acoustic channel between an emitted acoustic signal and a microphone signal are determined and smoothed in time. An echo is then estimated by filtering the emitted acoustic signal with the smoothed coefficients. Properties of the estimated echo and of the microphone signal are estimated. The echo cancellation filter is controlled as a function of a comparison between the properties of the estimated echo and those of the microphone signal so as to take into account the potential presence of a signal other than an echo signal in the microphone signal.

Abstract: Signal detectors are described herein. By way of example, a system for detecting signals can include a microphone signal detector, a loudspeaker signal detector, a signal discriminator and a decision component. When the microphone signal detector detects the presence of a microphone signal, the loudspeaker signal detector detects the presence of a loudspeaker signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component can confirm the presence of doubletalk. When the microphone signal detector detects the presence of a microphone signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component confirms the presence of near-end signal.

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: An echo canceller which can respond to a sudden change in echo characteristics in real time and does not require an alteration outside the echo canceller, includes a smoothed sending-speech signal calculation means for calculating a smoothed sending-speech signal from the sending-speech signal, the smoothed sending-speech signal being obtained by smoothing the sending-speech signal; a smoothed receiving-speech signal calculation means for calculating a smoothed receiving-speech signal from the receiving-speech signal, the smoothed receiving-speech signal being obtained by smoothing the receiving-speech signal; a delay time information generation means for obtaining delay time information reflecting delay characteristics of an echo path in accordance with a correlation between the smoothed sending-speech signal and the smoothed receiving-speech signal; and an update information generation means for obtaining update information indicating execution or suspension of updating of the tap coefficients of the adapti

Abstract: An echo canceller is capable of reducing echo feeling in a conversation even if a tone signal is input. The echo canceller of this invention judges whether a far-end input signal is a calling control tone signal or not, and if that is the case, resets a coefficient of an adaptive filter. An echo canceller of another aspect judges whether a far-end input signal is a predetermined kind of a tone signal or not, and stops the update of the coefficient of the adaptive filter if the result is affirmative and an amount of echo cancellation is not larger than a threshold value. Further, when the tone signal ends, the echo canceller compares amounts of echo cancellation before and after the end of the tone signal, in accordance with a result of the comparison, restart the update of the coefficient of the adaptive filter after resetting the coefficient of the adaptive filter, or restarts the update of the coefficient of the adaptive filter without resetting the coefficient of the adaptive filter.

Abstract: A method for removing an echo component in a first signal received at an audio input device, said method comprising the steps of outputting a second signal from an audio output device, receiving at the audio input device the first signal wherein the echo component in the first signal is the result of the second signal traversing an echo path having a frequency response which defines the relationship between the echo component and the second signal; determining an echo path model indicating an estimate of the frequency response of the echo path; removing an estimate of the echo component from the first signal using the echo path model; detecting if the first signal is overloaded; and wherein if it is detected that the first signal is overloaded the step of determining an echo path model comprises determining an overload echo path model such that the estimate of the frequency response indicated by the overload echo path model is only allowed to increase over time regardless of whether the frequency response of

Abstract: Hybrid echo canceller controllers are described herein. By way of example, a system for controlling an echo canceller can include a signal indicator and an echo canceller controller. The signal indicator can be configured to indicate periods of near-end signal and to indicate periods of echo only with echo-path change in the corrupted signal based at least in part on cross-correlation between two signals associated with the echo canceller. The echo canceller controller can be configured to control the echo canceller according to indications from the signal indicator.

Abstract: A method for conversation like rendering of a stored audio information stream determines a first location in the stored audio information stream. The first location represents a point in time when the sound detection flag became active. The method next moves from the first location to a second location in the stored audio information stream. The second location is selected based upon a criterion to make playback of the stored audio information stream appear like actual conversation. Finally, the stored audio information stream is rendered starting with audio information stored at the second location.