Abstract: A controller for the conference session generates a speaker signal for speakers in a conference room. The controller correlates the speaker signal with network timing information and generates speaker timing information. The controller transmits the correlated speaker signal and timing information to a mobile device participating in the conference session. The mobile device generates an echo cancelled microphone signal from a microphone of the mobile device, and transmits the echo cancelled signal back to the controller. The controller also receives array microphone signals associated with an array of microphones at known positions in the room. The controller estimates a relative location of the mobile device within the conference room. The controller dynamically selects as audio output corresponding to the mobile device location either the echo cancelled microphone signal from the mobile device or an echo cancelled array microphone signal associated with the relative location of the mobile device.

Abstract: An electronic device, a method of cancelling an echo signal thereof, and a non-transitory computer readable recording medium are provided. The electronic device includes a speaker configured to output a sound corresponding to a first audio signal, a microphone configured to acquire a surrounding sound of the electronic device to generate a second audio signal; a double-talk detector (DTD) configured to determine whether a double-talk state, in which an echo signal and a noise signal of the first audio signal are included in the second audio signal, occurs; and an acoustic echo canceller (AEC) configured to generate an estimated echo signal corresponding to the echo signal of the first audio signal based on an adaptive filter (AF) in response to the DTD determining that the double-talk state does not occur and cancel the generated estimated echo signal from the second audio signal to detect an error signal.

Abstract: A multi-channel echo cancellation system that dynamically adapts to changes in acoustic conditions. The system does not require a sequence of “start-up” tones to determine the impulse responses. Rather, the adaptive filters approximate estimated transfer functions for each channel. A secondary adaptive filter adjusts cancellation to adapt to changes in the actual transfer functions over time after the adaptive filters have been trained, even if the reference signals are not unique relative to each other.

Abstract: A speech enhancement device which includes: a speech production section detection unit configured to detect a speech production section in which a speaker produces speech, from an input signal generated by a speech input unit; a timer unit configured to measure an elapsed time from a starting point of the speech production section; a gain determination unit configured to determine a gain, which represents a level of enhancement of the input signal, according to the elapsed time; and an enhancement unit configured to enhance the input signal or a spectrum signal of the input signal in the speech production section according to the gain, whereby the input signal is enhanced only at necessary portions thereof.

Abstract: Present disclosure provide a linear prediction-based noise signal processing method includes: acquiring a noise signal, and obtaining a linear prediction coefficient according to the noise signal; filtering the noise signal according to the linear prediction coefficient, to obtain a linear prediction residual signal; obtaining a spectral envelope of the linear prediction residual signal according to the linear prediction residual signal; and encoding the spectral envelope of the linear prediction residual signal. According to the noise processing method, the noise generation method, the encoder, the decoder, and the encoding and decoding system that are in the embodiments of the present disclosure, more spectral details of an original background noise signal can be recovered, so that comfort noise can be closer to original background noise in terms of subjective auditory perception of a user, and subjective perception quality of the user is improved.

Abstract: An example of an apparatus configured to be worn by a person who has an ear and an ear canal includes a first microphone adapted to be worn about the ear of the person, and a second microphone adapted to be worn at a different location than the first microphone. The apparatus includes a sound processor adapted to process signals from the first microphone to produce a processed sound signal, a receiver adapted to convert the processed sound signal into an audible signal to the wearer of the hearing assistance device, and a voice detector to detect the voice of the wearer. The voice detector includes an adaptive filter to receive signals from the first microphone and the second microphone.

Abstract: A conditional switched rate filter system having a two stage digital filter to remove noise from the input signal. The first stage of the filter effectively removes the random noise from the input signal higher in frequency (cut-off frequency) than outside the characteristic response of the closed loop system. The second stage of the filter is used to filter out noise that is at same frequency response as the closed loop system.

Abstract: An acoustic echo removing apparatus detects space information representing a location of a sound source using a plurality of microphone input signals that are received through a plurality of microphones, and generates an acoustic echo estimation signal from a far-end talker voice signal using an adaptive filter coefficient. The acoustic echo removing apparatus detects a double talk segment using the space information, and determines update of the adaptive filter coefficient according to whether the double talk segment exists.

Abstract: In one implementation, systems and processes are provide for local cross muting between two or more dispatch consoles in a dispatch center. Sound is received from a microphone of a first console. The first console generates an outgoing media signal from the received sound and a supplementary outgoing data. The supplementary outgoing data may be an identification number for the first console or a duplicate media signal from the first console. The first console sends the outgoing media signal to a forwarding device and sends the supplementary outgoing signal to a second console.

Abstract: The present invention relates to a voice detector being responsive to an input signal being divided into sub-signals representing a frequency sub-band, comprising: means to calculate, for each sub-band, an SNR value snr[n] based on a corresponding sub-signal for each sub-band and a background signal for each sub-band. The voice detector further comprises: means to calculate a power SNR value for each sub-band, wherein at least one of said power SNR values is calculated based on a non-linear function, means to form a single value snr_sum based on the calculated power SNR values, and means to compare said single value snr_sum and a given threshold value vad_thr to make a voice activity decision vad_prim presented on an output port. The invention also relates to a voice activity detector, a node and a method for selectively suppressing sub-bands in a voice detector.

Abstract: Disclosed is a voltage control device including a voltage detector detecting an output voltage output from an output port of a mobile communication device; and a voltage control circuit bypassing the output voltage to an earpiece in which a speaker and a microphone are integrally formed when the output voltage detected by the voltage detector is between a first voltage and a second voltage higher than the first voltage.

Abstract: Sound enhancement techniques through dereverberation are described. In one or more implementations, a method is described of enhancing sound data through removal of reverberation from the sound data by one or more computing devices. The method includes obtaining a model that describes primary sound data that is to be utilized as a prior that assumes no prior knowledge about specifics of the sound data from which the reverberation is to be removed. A reverberation kernel is computed having parameters that, when applied to the model that describes the primary sound data, corresponds to the sound data from which the reverberation is to be removed. The reverberation is removed from the sound data using the reverberation kernel.

Abstract: A method of echo cancellation in an acoustic system, comprising a first user device and at least one further user device, the method comprising, at the first user device: detecting an audio signal received from the at least one further user device over a communications network; supplying the audio signal when received to an audio signal processing module of the first user device, wherein the audio signal processing module processes the audio signal and outputs a processed audio signal to audio output means of the first user device; and controlling an echo cancellation process for cancelling echo from an audio signal received via audio input means of the first user device based on a loopback signal provided by the processed audio signal and selectively based on the received audio signal when detected.

Abstract: A method for cancelling/reducing acoustic echoes in speech/audio signal enhancement processing comprises using a received reference signal to excite an adaptive filter wherein the output of the adaptive filter forms a replica signal of acoustic echo; the replica signal of acoustic echo is reduced adaptively by multiplying a gain to get a gained replica signal of acoustic echo wherein the gain is smaller in speech area and/or double-talk area than non-speech area; the gained replica signal of acoustic echo is subtracted from a microphone input signal to suppress the acoustic echo in the microphone input signal.

Abstract: A method including: receiving at least a first input audio channel and a second input audio channel; and using an inter-channel prediction model to form at least an inter-channel direction of reception parameter.

Abstract: An audio signal processing device includes: a frequency-domain conversion unit that generates a plurality of pieces of frequency-domain information from a plurality of audio input signals acquired at different positions; a relative value calculation unit that calculates, for each piece of frequency-domain information, a relative value between a time-frequency component included in one frequency-domain information and a time-frequency component included in another frequency-domain information; a mask generation unit that compares the relative value with an emphasized range set based on a relative value threshold stored in advance to generate a time-frequency mask that decreases a value of the frequency-domain information corresponding to the relative value which is outside the emphasized range; a mask multiplication unit that multiplies the time-frequency mask by the frequency-domain information to generate emphasized frequency-domain information; and a time-domain conversion unit that converts the emphasized

Abstract: A method (800) for determining an estimate (215, 261) of an echo path property of an electronic device (200, 250, 300, 600). The electronic device is configured to render a total audio signal using a loudspeaker (102), and the electronic device is configured to record an echo of the rendered audio signal using a microphone (103), thereby yielding a recorded audio signal (112). The method comprises inserting (801), in an inaudible manner, an auxiliary audio signal (212) into the total audio signal to be rendered; wherein the auxiliary audio signal (212) comprises a tonal audio signal at a first frequency; isolating (803) the echo of the auxiliary audio signal (212) from the recorded audio signal (112); and determining (804) the estimate (215, 261) of the echo path property based on the inserted auxiliary audio signal (212) and based on the isolated echo of the auxiliary audio signal (212).

Abstract: A multi-channel echo cancellation system that dynamically adapts to changes in acoustic conditions. The system does not require a sequence of “start-up” tones to determine the impulse responses. Rather, the adaptive filters approximate estimated transfer functions for each channel. A secondary adaptive filter adjusts cancellation to adapt to changes in the actual transfer functions over time after the adaptive filters have been trained, even if the reference signals are not unique relative to each other.

Abstract: Technology for dynamic characterization of a communication device is disclosed. One method comprises sending an acoustic characterization excitation signal to be output by an acoustic transducer in the communication device. A received acoustic coupling signal that is received by a microphone of the communication device in response to the output of the acoustic transducer is measured. An acoustic signal conditioning algorithm is applied to at least one of the microphone or the acoustic transducer to reduce an amplitude of the acoustic signal or the received acoustic coupling signal to a level below a selected threshold.

Abstract: A feedback cancellation assembly for an electroacoustic communication apparatus may include a signal transmission path for generation and emission of an outgoing sound signal to an external environment through an electrodynamic loudspeaker and a signal reception path comprising a microphone for generation of a microphone input signal corresponding to sound received from the external environment. The signal reception path may generate a digital microphone signal. The outgoing sound signal may be acoustically coupled to the microphone. An electronic feedback cancellation path may be coupled between a tapping node and a summing node to produce a feedback cancellation signal to the summing node.

Abstract: Systems and methods for controlling echo in audio communications between a near-end system and a far-end system are described. The system and method may intelligently assign a plurality of microphone beams to a limited number of echo cancellers for processing. The microphone beams may be classified based on generated statistics to determine beams of interest (e.g., beams with a high ratio of local-voice to echo). Based on this ranking/classification of microphone beams, beams of greater interest may be assigned to echo cancellers while less important beams may temporally remain unprocessed until these beams become of higher importance/interest. Accordingly, a limited number of echo cancellers may be used to intelligently process a larger number of microphone beams based on interest in the beams and properties of echo cancellation performed for each beam.

Abstract: There is provided an audio signal processing device including a feature amount extraction unit configured to extract, from an audio signal obtained by a sound collection unit, a feature amount that indicates a pulsed operation sound generated from a sound generation unit provided in a same housing as the sound collection unit, and a noise determination unit configured to determine whether or not the operation sound is included in the audio signal based on the feature amount.

Abstract: The present invention relates to a desktop speakerphone constructed to make the desktop speakerphone less space-consuming while having improved audio qualities. The desktop speakerphone preferably has two microphone clusters 6, 7 mounted at the upper side of the housing 2 closer towards respective longitudinal ends 8 of the latter, so that each microphone cluster 6, 7 can receive voice sound Av from one or more of the users. Each microphone cluster 6, 7 preferably comprises three pressure microphones 10, 11, 12. Furthermore, within each microphone cluster 6, 7, the second and third sound inlets 14, 15 are preferably arranged symmetrically on opposite sides of the respective median plane 18. Within each microphone cluster 6, 7, the relative arrangement of the three sound inlets 13, 14, 15 defines a respective microphone axis 9, 19 for each of the microphone pairs 10, 11, 10, 12.

Abstract: In a sound processing device, an index value calculation unit calculates a first index value that follows change of a sound signal at a first following degree and a second index value that follows the change of the sound signal at a second following degree which is lower than the first following degree. An adjustment value calculation unit calculates an adjustment value effective to adjust a reverberation component of the sound signal based on difference between the first index value and the second index value. A reverberation adjustment unit applies the adjustment value to the sound signal.

Abstract: In an audio system having acoustic echo, incoming audio signals are decorrelated by applying both phase modulation and amplitude modulation. The resulting decorrelated signals are rendered by loudspeakers, and co-located microphones generate outgoing audio signals that include echo signals caused by the existence of acoustic echo paths from the loudspeakers to the microphones. Acoustic echo cancellation (AEC) is applied to the outgoing audio signals to reduce the amount of echo in those signals. Decorrelating the incoming audios signals using amplitude modulation in addition to phase modulation can increase the effectiveness of the AEC processing. In a frequency-domain implementation, the amount of amplitude modulation can be different for different frequency bands (e.g., greater amplitude modulation for lower frequencies corresponding to human speech and/or no amplitude modulation for higher frequencies).

Abstract: A harmonic echo power estimator estimates power of echo generated by harmonic loudspeaker nonlinearities in a user equipment having an echo path between a loudspeaker input and a microphone output. The estimator includes a frequency band mapper that maps each frequency band in a set of loudspeaker output signal frequency bands into a corresponding array of loudspeaker input signal frequency bands, where each frequency band in the set is mapped into several frequency bands in the corresponding array. A power estimator determines a power estimate of each input signal in each array of frequency bands. A power estimate combiner combines determined power estimates in each array of frequency bands into a corresponding estimate of loudspeaker input power generating harmonic loudspeaker nonlinearities. A power estimate transformer transforms the estimates of loudspeaker input power across the echo path into power estimates of the echo generated by the harmonic loudspeaker nonlinearities.

Abstract: According to an embodiment, a signal processing device includes a background calculator, a signal generator, an extractor, a similarity calculator, and a mixer. The background calculator is configured to calculate a first background signal in which a speech signal is removed, based on the acoustic signals. The signal generator is configured to generate a reference signal from at least one of the acoustic signals. The extractor is configured to extract a second background signal by removing a speech signal from the reference signal. The similarity calculator is configured to calculate a similarity between feature data of the background signals. The mixer is configured to calculate a weighted sum of the background signals in such a way that a greater weight is given to the first background signal as the similarity is higher and a greater weight is given to the second background signal as the similarity is lower.

Abstract: The invention discloses a method and a device for reducing voice reverberation based on double microphones. The method comprises the steps of calculating a transfer function h(t) from a secondary microphone to a primary microphone according to an input signal x2(t) of the primary microphone and an input signal x1(t) of the secondary microphone; judging the strength of reverberation according to h(t) and calculating a regulatory factor ? of a gain function by taking a tail section hr(t) of the h(t); obtaining a late reverberation estimation signal {circumflex over (r)}(t) of x2(t) with the convolution of x1(t) and hr(t); calculating the gain function according to the frequency spectrum of x2(t), ? and frequency spectrum of {circumflex over (r)}(t); obtaining the reverberation removed frequency spectrum of x2(t) by multiplying the frequency spectrum of x2(t) by the gain function; and obtaining a late reverberation removed time-domain signal of x2(t) by frequency-time conversion.

Abstract: According to one embodiment, a communication device includes a clock to measure time, communication circuitry and processing circuitry. The communication circuitry transmits a request signal to the other communication device in a certain cycle, and receives a response signal transmitted in response to the request signal, from the other communication device. The processing circuitry changes a transmission timing of the request signal; compares first round-trip delay time with second round-trip delay time; and if a result of comparison shows that the second round-trip delay time is shorter than the first round-trip delay time, adjusts the time measured by the clock using a transmission time point of the second request signal and the reception time point of the second request signal in the other communication device, and the transmission time point of the second response signal from the other communication device and a reception time point of the second response signal.

Abstract: Examples of the disclosure provide variable step size (VSS) adaptive echo cancellation in the presence of near-end noise such as dense double talk without using an explicit double talk detector and/or without using a dual-filter. During a conversation, the present value for an error signal is monitored. Based on the monitored present value for the error signal, a first function is determined. A second function is determined based on long-term statistics describing a reference signal, a near-end noise signal, and the error signal. An adaptation coefficient is calculated for the VSS adaptive filter based on the determined first function and the determined second function. The calculated adaptation coefficient is used in the VSS adaptive filter for echo cancellation against interference due to the near-end noise signal during the conversation.

Abstract: The present technology provides adaptive noise and echo reduction of an acoustic signal which can overcome or substantially alleviate problems associated with mistaken adaptation of speech and noise models to acoustic echo. The present technology carries out a multi-faceted analysis to identify echo within the near-end acoustic signal to derive an echo model. Echo classification information regarding the derived echo model is then utilized to build near-end speech and noise models. These echo, speech, and noise models are then used to generate one or more signal modifications applied to the acoustic signal to preserve the desired near-end speech signal and reduce the echo and near-end noise signals. By building near-end speech and noise models utilizing echo classification information, the present technology can prevent adaptation of the speech and noise model to the acoustic echo.

Abstract: A controller for an audio device is provided. The controller receives a first collected sound signal and a second collected sound signal respectively provided by two microphones, and includes an echo cancellation module and a beamforming module. The echo cancellation module performs echo cancellation on the first collected sound signal to accordingly provide an intermediate signal. The beamforming module performs beamforming by utilizing the echo-cancelled intermediate signal and the non-echo-cancelled second collected sound signal.

Abstract: A bidirectional communication system includes—a first compensating unit to which a first output unit and a first input unit are assigned that is configured to compensate for undesired feedback from the first output unit in the first input unit with generation of first compensation data and at least one second compensating unit that is configured like the first compensating unit and is physically separate from the first compensating unit. The second compensating unit has a second output unit but no input unit is assigned to it. The first compensating unit and the second compensating unit are connected by interfaces via which the first compensation data generated by the first compensating unit can be sent to the second compensating unit so that second compensation data can be generated in the second compensating unit based on the first compensation data.

Abstract: The array microphone device has a microphone array composed of a plurality of microphone units, having a signal input section for inputting a signal from the microphone units to be corrected as a signal to be corrected; a reference signal input section; a gain variable section for making the levels of the signal to be corrected and the reference signal equal; and a gain control section. The gain control section includes a high-speed gain update section for changing the gain with a first amount of change per unit time upon time elapsed since the array microphone device is starting-up being below a predetermined period of time; and a low-speed gain update section for changing the gain with a second amount of change per unit time upon the elapsed time being above the predetermined period of time, the second amount of change being smaller than the first amount of change.

Abstract: A signal processing method operating to generate a target blocked signal in which a target signal is suppressed by processing a mixed signal with a first set of adaptive filters, the method including estimating a noise contained in the mixed signal to determine an estimated noise, detecting a target signal contained in the mixed signal using a value that corrected similarity with a reference signal and the mixed signal by the estimated noise to determine a value corresponding to a reliability in the detection, and adaptively controlling a degree of coefficient update for first and second sets of adaptive filters according to the value corresponding to the reliability.

Abstract: The present disclosure relate to a spectrum sensing method and a spectrum sensing device. The spectrum sensing method includes: (a) sampling a received signal: (b) calculating a first operation value relating to a strength of the received signal based on the sampled signal; (c) searching for a first frequency band having the largest correlation with an input signal based on the sampled signal; (d) calculating a first residual component signal by removing a signal component in the first frequency band from the received signal; (e) calculating a second operation value relating to a strength of the first residual component signal; and (f) determining whether the frequency band occupied in the received signal is present based on a relative value of the first operation value and the second operation value.

Abstract: Systems, methods, and other embodiments associated with echo cancellation are described. According to one embodiment, an apparatus includes a cable tester that determines whether a fault in a cable exists by using echo cancellation values.

Abstract: An apparatus includes an echo canceller having an audio signal input and an audio signal output and dynamic pre-conditioning logic. The dynamic pre-conditioning logic is operatively coupled to the echo canceller audio signal output as a feedback signal and has a dynamic pre-conditioning logic output operatively coupled to the echo canceller audio signal input. The dynamic pre-conditioning logic is also operative to receive an audio signal input from at least one microphone. The dynamic pre-conditioning logic is operative to analyze the feedback signal to obtain at least one characteristic, and pre-condition the audio signal input, based on the at least one characteristic of the feedback signal, and provide a pre-conditioned audio signal at the echo canceller audio signal input. The echo canceller audio signal output is then provided to a noise suppressor for the send path of a full duplex communication channel.

Abstract: An amplifier circuit and a method of amplification using automatic gain control (AGC) are disclosed. A method for reducing distortions incurred by an audio signal when being rendered by an electronic device is described. The method comprises receiving an input signal; determining signal strength; determining a frequency-dependent AGC filter; wherein the frequency-dependent AGC filter is adapted to selectively attenuate the input signal within a number N of predetermined frequency ranges, according to corresponding N degrees of attenuation; wherein N predetermined frequency ranges depend upon a rendering characteristic of the electronic device; and wherein the N-degrees of attenuation depend upon the signal strength; and attenuating the input signal using the frequency-dependent AGC filter to obtain an output signal for rendering by the electronic device.

Abstract: Embodiments include methods and apparatuses for echo cancelation involving multiple audio channels and the production, sensing, or a combination thereof of multiple audio channels in conferencing systems. A conferencing apparatus with a plurality of speakers configured to generate outgoing acoustic waves responsive to a multi-channel audio signal. One or more microphones are configured to sense incoming acoustic waves from the plurality of speakers and from locally produced acoustic waves from a participant of a conference to generate one or more incoming audio signals. A processor is operably coupled to the plurality of speakers and the one or more microphones. The processor is configured to perform acoustic echo cancelation on the one or more incoming audio signals relative to at least two different channels of the multi-channel audio signal.

Abstract: Methods and devices are disclosed for performing echo cancellation with a communication device for use by a hearing-impaired user. The communication device comprises communication elements, a memory device storing a plurality of sets of training parameters for an echo canceller, and a processor operably coupled with the communication elements and the memory device. The communication elements are configured to receive audio signals from a far end communication device, and to receive text captions corresponding to the audio signals from a relay service for display by the communication device during a call. Each set of training parameters corresponding to a different audio mode for the communication device. The processor is configured to execute an echo canceller by loading a first set of training parameters when a first audio mode is selected, and a second set of training parameters when a second audio mode is selected.

Abstract: The present invention relates to a method and apparatus of a digital filter for noise suppression of a signal representing an acoustic recording. The method comprises determining a desired frequency response (H(?)) of the digital filter; and generating a noise suppression filter based on the desired frequency response. The desired frequency response is determined in a manner so that the desired frequency response does not exceed a maximum level, wherein the maximum level is determined in response to the signal to be filtered.

Abstract: Various embodiments are directed to cooperation among communications devices having microphones to employ their microphones in unison to provide voice detection with noise reduction for voice communications. A first communications device comprises a processor circuit; a first microphone; an interface operative to communicatively couple the processor circuit to a network; and a storage communicatively coupled to the processor circuit and arranged to store a sequence of instructions operative on the processor circuit to store a first detected data that represents sounds detected by the first microphone; receive a second detected data via the network that represents sounds detected by a second microphone of a second communications device; subtractively sum the first and second data to create a processed data; and transmit the processed data to a third communications device. Other embodiments are described and claimed herein.

Abstract: Method, user device and computer program product for suppressing echo. An audio signal is outputted from a speaker. A microphone receives an audio signal which includes echo resulting from said outputted audio signal. An echo path of the echo is modelled using a plurality of models. A first of the models is a FIR based model and a second of the models is different to the first model. The first model is used to determine a first model estimate of the echo power of a first component of the echo. The second model is used to determine a second model estimate of the echo power of a second component of the echo. The first and second model estimates of the echo power are combined. The combined estimate of the echo power is used to apply echo suppression to the received audio signal, thereby suppressing the echo.

Abstract: An apparatus and a method thereof, processes a voice signal of a mobile terminal in a mobile communication system. The apparatus to process a received-voice signal received through a wireless channel in a mobile terminal includes a digital signal processing unit to generate an encoded packet and frame type information defining a characteristic of the encoded packet by performing voice encoding on an audible signal input from a microphone. The apparatus also includes a received-voice controlling unit to determine a noise level in consideration of the frame type information and a level of the audible signal, and to control at least one of a tone and a volume of a received voice by the determined noise level.

Abstract: An audio processing apparatus includes a first microphone, a second microphone, and a masking unit configured to mask movement of air from outside of the apparatus to the second microphone. A filter coefficient is estimated and learned so as to minimize the difference between the output signal of the first microphone and the output signal of the second microphone, thereby suppressing a reverberation component generated in the closed space between the masking unit and the second microphone out of the output signal of the second microphone.

Abstract: An audio amplifier may comprise a signal limiting circuit and a power amplifier. The signal limiting circuit may be configured to limit an audio signal received at an input and to provide it as a limited audio signal at an output. The power amplifier may have a supply connection which may be coupled to a power supply unit in order to supply power to the power amplifier. The power amplifier may be configured to amplify a signal, which may be based on the limited audio signal, and to provide it as a level-limited audio signal at an output which may be coupled to a load, so that the load may be operated at limited power. The signal limiting circuit may be configured to produce an audio signal which may be limited depending on the load.

Abstract: A method for processing multichannel acoustic signals which is characterized by calculating the feature quantity of each channel from the input signals of a plurality of channels, calculating similarity between the channels in the feature quantity of each channel, selecting channels having high similarity, and separating signals using the input signals of the selected channels.

Abstract: An acoustic echo cancellation (AEC) system includes a remote device, for capturing a remote captured sound, a server coupled to the remote device, and a local device coupled to the server. The server transmits the remote captured sound from the remote device to the local device. The local device receives, stores and plays the remote captured sound as a local playback sound. An echo is generated from reflection of the local playback sound. The local device captures the echo and a local sound into a local captured sound, and transmits both the remote captured sound and the local captured sound to the server. The server performs AEC on the local captured sound by using the remote captured sound from the local device and transmits the AEC processed local captured sound to the remote device.

Abstract: A wideband acoustic echo cancellation apparatus with an adaptive tail length in an embedded system, and a wideband acoustic echo cancellation method are provided, and the wideband acoustic echo cancellation apparatus may include a delay length calculating unit to calculate a delay length of an echo path, using a near-end signal and a far-end signal, an adaptive filter implementing unit to implement an adaptive filter based on the calculated delay length, using selected coefficients, and an error calculating unit to search for three intervals having a largest impulse response value from all intervals of a tail of the adaptive filter, and to calculate an error during an interval in which the selected coefficients are used.