СПОСОБЫ И УСТРОЙСТВА ДЛЯ ОБЕСПЕЧЕНИЯ КОМФОРТНОГО ШУМА В СИСТЕМАХ СВЯЗИ

Изобретение относится к системам связи и более конкретно к подавителю эхо-сигнала в двусторонней линии связи. Технический результат заключается в обеспечении комфортного шума, который близко и последовательно соответствует действительному шуму, исходя из спектрального содержания и громкости. Для этого в соответствии с приведенным в качестве примера вариантом осуществления модель фонового шума основана на наборе параметров модели шума, которые, в свою очередь, основаны на измерениях действительного фонового шума в системе подавления эхо-сигнала. Приведенные в качестве примера варианты осуществления включают в себя авторегрессионную модель, модель авторегрессионного скользящего среднего и модель частотной области. Приведенные в качестве примера модель авторегрессионного скользящего среднего первого порядка включает в себя единичный фиксированный нуль и единичный переменный полюс. Единичный нуль и единичный полюс являются достаточными для обеспечения соответственного отклонения спектра в получающемся ...

УПРАВЛЕНИЕ АКУСТИЧЕСКИМИ ЭХО-СИГНАЛАМИ НА ОСНОВЕ ВРЕМЕННОЙ ОБЛАСТИ

... 1. Реализуемый с помощью процессора способ для обработки сигнала, при этом способ содержит этапы, на которых: ! (a) определяют (например, 202, 204, 206), во временной области, то, какую величину акустического эхо-сигнала следует подавлять в сигнале; и ! (b) обрабатывают (например, 208) сигнал во временной области на основе определения того, какую величину акустического эхо-сигнала следует подавлять. ! 2. Способ по п. 1, в котором: ! - этап (a) содержит этап, на котором формируют значение усиления, чтобы применять к сигналу, согласно тому, какую величину акустического эхо-сигнала следует подавлять в сигнале; и ! - этап (b) содержит этап, на котором применяют значение усиления к сигналу. ! 3. Способ по п. 2, в котором этап (a) содержит этапы, на которых: ! (a1) формируют (например, 202, 204, 242, 244, 246) значение счетчика, указывающее то, в течение какого времени акустический эхо-сигнал обнаруживается в сигнале без обнаружения одновременного разговора; ! (a2) сравнивают (например, 248) ...

AKUSTISCHES ECHOAUSLÖSCHUNGS-REFERENZSIGNAL

Es werden Systeme und Verfahren zur akustischen Echoauslöschung bereitgestellt. Ein anschauliches Verfahren umfasst ein Empfangen eines Referenzsignals. Das Referenzsignal repräsentiert mindestens ein Geräusch bzw. Schallereignis, das im Inneren eines Gehäuses eines Lautsprechers erfasst wird. Der Lautsprecher ist ausgebildet, ein Signal des fernen Endes wiederzugeben. Das Verfahren umfasst ferner ein Empfangen eines akustischen Signals. Das akustische Signal repräsentiert mindestens ein Geräusch bzw. Schall, das bzw. der außerhalb des Gehäuses des Lautsprechers erfasst wird. Das akustische Signal enthält zumindest ein Signal des nahen Endes und ein Signal des fernen Endes. Das Verfahren umfasst eine Abschwächung des Signals des fernen Endes in dem akustischen Signal unter Verwendung des Referenzsignals. Das Referenzsignal kann durch ein Mikrofon mit geringer Empfindlichkeit erfasst werden, das im Inneren des Gehäuses des Lautsprechers angeordnet ist. Die Abschwächung des Signals des fernen ...

Verfahren und Vorrichtung zur Echounterdrückung bei einer Freisprecheinrichtung, insbesondere für ein Telefon

The invention relates to a method and device for suppressing echo in a hands free device such as a telephone. Control signals Vin, Vout with an amplitude and dynamic performance depending upon the mode of communication are produced by a control circuit (13) on the basis of a Uf remote subscriber signal, a ULRM output signal from the loudspeaker-room-microphone system and a UM output signal pertaining to an adaptive filter (14) acting as a reference. Amplification VE,VS of the adjustable amplifiers (11,16) arranged along the reception or transmission path is adapted to the mode of communication using the Vin and Vout control signals, resulting in reduced residual echo. The conversation is made to sound natural by progressively transforming the amplifications VE, VS from a quiescent value to a value which is attenuated by a factor alpha or beta with an exponential time characteristic when the mode of communication is modified.

Vollduplex-digitaler Lautfernsprecher.

DYNAMISCHE OPTIMIERUNG FÜR MIKROFONVERSTÄRKUNG IN EINER FREISPRECHEINRICHTUNG

SYMMETRISCH BASIERTER TEILBANDECHOKOMPENSATOR

Kommunikationsendgerät mit Bandbreitenerweiterung und Echokompensation

VERFAHREN UND GERÄT ZUR VERBESSERUNG VON ADAPTIVER TEILBANDFILTERUNG IN ECHOKOMPENSATIONSSYSTEMEN

VERFAHREN UND ANORDNUNG ZUR DURCHFÜHRUNG VON ECHOKOMPENSATION IN EINEM ÜBERTRAGUNGSNETZWERK UNTER VERWENDUNG EINES GEMISCHTEN ADAPTIVEN BALANCEFILTERS NACH DEM VERFAHREN DES KLEINSTEN QUADRATISCHEN FEHLERS

AUDIOSYSTEM MIT VORKEHRUNGEN ZUM FILTERKOEFFIZIENTEN KOPIEREN

Akustischer Teilband-Echokompensator

Anordnung zur aktiven unterdrückung von akustischem Echo und Geräusch

Microphone array with beamforming means and echo-cancelling means

Audio signals are received at an array of microphones 302. A characteristic of at least one of the audio signals received by the plurality of microphones is measured 508. A beamformer 504 applies beamformer coefficients to the received audio signals, thereby generating a beamformer output. Echo cancelling means 506 are applied to the beamformer output thereby suppressing, from the beamformer output, an echo resulting from audio signals output from the audio output means 210. An operating parameter of the echo cancelling means 506 is controlled based on the measured characteristic of the at least one of the audio signals received by the plurality of microphones.

Frequency spectrum control of received acoustic signal in a telephone set

Non-linear echo path detection

Audio signal processing

Detecting double-talk in an echo canceller

A double-talk detection device (400) detects the presence of near-end speech signals for an echo canceller including an adaptive filter to generate an echo estimate. A control device (300) generates the coefficients for the adaptive filter. The control device (300) includes a coefficient updating circuit (320). A buffer (314) holds the previous filter coefficients and a buffer (318) holds the current filter coefficients. The double-talk detection device (400) detects the presence of near-end speech signals based upon a detected variation of the coefficients. The detection device (400) is used to detect a double-talk condition during a single-talk condition and a single-talk condition during the double-talk condition.

Audio apparatus

VOICE CONFERENCE SYSTEM USING ECHO CANCELLERS

A Voice activity detector for an echo suppressor

A voice activity detector is described suitable for use in an echo suppressor. It comprises a whitening filter (19) for levelling the spectrum of the audio signal on the transmit path to provide a levelled signal and decision means (20) coupled to the whitening filter to measure energy in the levelled signal and thereby to detect voice on the transmit path. The whitening filter is a voice whitening filter which is adapted (8, 19) according to the voice parameters received by the voice decoder in the receive path. In a second aspect, a near-end voice entered into the transmit path is distinguished from a far-end voice resulting from an echo from the receive path and transmit and receive attenuators (26, 27) are arranged in a first disposition when voice on the transmit path is substantially due to echo from the receive path and in a second disposition when voice on the transmit path is at least partially due to near-end voice. ...

Howling suppression using echo cancellation

A method for reducing howling in a communication system (200) containing collocated mobile devices (540, 610, 640, 710) is presented. In a transmitter (610, 710), an audio signal is received at a microphone (512). 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 (540, 640). The encoded signal is decoded at the transmitter (610, 710), in addition to at the receiver (640, 740), 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 (512) 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 (102) to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

Echo cancellation using wavelet packet techniques

Device for capturing and outputting audio

Wideband speakerphone

Acoustic echo cancellation

Controlling operational characteristics of an acoustic echo canceller

Echo controlling method and apparatus for a video conference system

Dynamic gain controller

DEVICE AND PROCEDURE FOR THE MULTI-CANAL ACOUSTIC ECHO COMPENSATION WITH VARIABLE CHANNEL NUMBER

VERFAHREN ZUR ADAPTIVEN KOMPENSATION EINES ECHOS IN EINER KOMMUNIKATIONSEINRICHTUNG

An echo compensator (EK) for the autonomous speech system of a mobile telephone contains a digital signal processor (DSP) connected via analog/digital converters (A/D) to the emission and reception lines (SL, EL) in the low-frequency section of the mobile telephone. The acoustic and line-side electric echo is adaptively compensated. The relevant compensation value (c, c') is subtracted from the emission value (c) and the reception value (e). The acoustic and electric compensation value (c, c') is established in the corresponding adaptive filter (AF, AF'). The acoustic compensation value (c) is established by scalar multiplication of a summation vector obtained from previously scanned reception values (e) with a coefficient vector. The latter is determined from the previous coefficient vector by adding the product of the summation vector, divided by the energy of the reception signal, and the emission value(s) reduced by the acoustic compensation value (c). The compensation of the electric ...

AUDIO SYSTEM WITH PRECAUTIONS TO THE FILTER COEFFICIENT COPYING

SYSTEM AND PROCEDURE FOR EXTENDED STEREO AUDIO

AUDIOSIGNALDEKORRELATOR

PROCEDURE FOR THE SUPPRESSION OF ACOUSTIC REMAINDER ECHOES AFTER ECHO KILLING DURING A FREE SPEECH MECHANISM

PROCEDURE AND DEVICE FOR THE ACOUSTIC ECHO COMPENSATION WITH BELIEBIEGEN SCANNING RATES OF THE RENDITION

ECHO KILLING

REINFORCEMENT AND SPECTRAL CURVE ADJUSTMENT WITH THE PROCESSING OF AUDIO SIGNALS

ON INTOXICATION ENVIRONMENT OF BASING ECHO COMPENSATOR

PROCEDURE FOR THE ADAPTIVE COMPENSATION OF AN ECHO IN A COMMUNICATION DEVICE

ECHO PROCESSING MECHANISMS FOR SINGLE-CHANNEL OR MULTI-CANAL COMMUNICATION SYSTEMS

NONLINEAR ACOUSTIC ECHO COMPENSATOR

LANGUAGE SIGNAL PROCESSING FOR THE COMMON ADAPTIVE REDUCTION OF INTERFERENCES AND OF ACOUSTIC ECHOES

DELAY ESTIMATE EQUIPMENT

PROCEDURE FOR THE LANGUAGESTEERED DAEMPFUNGSREGELUNG IN TELEPHONE TRANSMISSION CIRCLES.

VOLLDUPLEXFERNSPRECHHOERER FUER RADIO AND UEBERLANDLINIEN.

COMMUNICATION TERMINAL WITH RANGE EXTENSION AND ECHO COMPENSATION

ACOUSTIC ECHO KILLING EQUIPMENT

MULTIPLE ENTRANCE WIRELESS TELEPHONE WITH ECHO KILLING

NOISE REDUCTION ARRANGEMENT

ECHO ELIMINATOR AND NON LINEAR PROCESSOR OF AN ECHO COMPENSATOR

SYMMETRICALLY BASING VOLUME ECHO COMPENSATOR

PROCEDURE AND DEVICE FOR THE ECHO COMPENSATION

Methods and apparatus for improved sub-band adaptive filtering in echo cancellation systems

VOICE CONFERENCE SYSTEM USING ECHO CANCELLERS

TRAINING METHOD FOR AN ECHO CANCELLER FOR USE IN A VOICE CONFERENCE SYSTEM

Spectral noise compensation for echo cancellation

Apparatus and method for detecting far end speech

Echo attenuating method and device

Method and apparatus for acoustic echo cancellation in a communication system providing TTY/TDD service

FULL DUPLEX ECHO CANCELLING CIRCUIT

ECHO CANCELING IN A SPEECH PROCESSING SYSTEM

Hands-free telephone

Adaptive filter and adapting method thereof

Echo canceller employing dual-h architecture having improved non-linear echo path detection

Residual echo suppression

ANTI-HOWLING STRUCTURE

A howling control structure for a full duplex communication system. The structure is implemented as part of an acoustic echo canceller having a conventional transversal adaptive filter. A second transversal adaptive filter, shorter than the conventional filter, that adapts even in the absence of speech in its reference signal, is provided. The short filter adapts quickly and provides enough echo cancellation to prevent howling from occurring, even if the echo path is changed significantly during silence periods.

METHOD AND APPARATUS FOR ADAPTIVE ECHO AND NOISE CONTROL

An method (500) and apparatus (100) for adaptive echo and noise control. A signal can be received at an input (140) to a communication or electronic device. Background noise in the signal can be determined. The order of noise suppression and echo cancellation can be adaptively determined (110) based on the background noise in the signal. Adaptively determining the order of noise suppression (230) and echo cancellation (220) can be performed by comparing the background noise to at least one threshold, performing echo cancellation prior to noise suppression on the signal if the background noise is below the at least one threshold, and performing noise suppression prior to echo cancellation on the signal if the background noise is above the at least one threshold.

METHOD OF ACOUSTIC ECHO CANCELLATION IN FULL-DUPLEX HANDS FREE AUDIO CONFERENCING WITH SPATIAL DIRECTIVITY

A method is set forth of controlling an acoustic echo canceller at the output of a beamformer in an audio conferencing device. Information is saved to, and retrieved from, memory that characterizes each of a finite number of look directions, or regions of focus, covering the entire spatial span of the conferencing device. Each time a change occurs from a first look direction to a second look direction, information relating to the workspace captured by the acoustic echo canceller is saved for the first look direction, and previously saved information for the second look direction is retrieved from memory. The acoustic echo cancellation then takes place for the second look direction with the retrieved information.

ACOUSTIC ECHO CANCELLATION

A multiple channel steered spatialised signal is generated from a signal input (12) modified according to respective spatialisation gain functions (g1, g2) to generate a plurality of audio channels (x1(t), x2(t)). An echo cancellation signal e(t) is applied to a return path (22) using a combined spatialisation and echo path estimate (ĥ1, ĥ2). The estimate is derived from the gain functions applied to the respective channels 14L, 14R. When the gain functions applied in the respective channels are changed, for instance to represent a different apparent position of the sound source (10), a new estimate (h1, h2) of the echo paths (ĥ1, ĥ2) is generated, based on a previous estimate of the echo path and on the new gain functions (g1, g2).

METHOD AND DEVICE FOR CANCELLING ACOUSTIC ECHO BY AUDIO WATERMARKING

Procédé d'annulation d'écho acoustique dans un premier signal comportant un signal d'écho d'un deuxième signal, comportant; insérer, de manière inaudible, dans le deuxième signal une séquence pseudo aléatoire dont l'autocorrélation circulaire comporte une impulsion unité et une composante continue, caractériser, dans le premier signal, au moyen de la séquence insérée, un canal acoustique suivi par le signal d'écho, estimer le signal d'écho dans le premier signal au moyen de la caractérisation du canal acoustique, et annuler le signal d'écho au moyen de l'estimation obtenue.

SYSTEMS AND METHODS FOR COMPUTER AND VOICE CONFERENCE AUDIO TRANSMISSION DURING CONFERENCE CALL VIA PSTN PHONE

A new approach is proposed that contemplates systems and methods to support canceling audio streams leaked from a speaker to a PSTN so that only the audio stream of a presenter or viewer speaking at a conference call is transmitted. Here, the audio streams being canceled includes the audio stream of an application being run or a material being presented online during a web conference. The voice audio stream of a presenter or viewer is then transmitted in addition to the computer audio stream, clean from any other audio stream of echoes or feedbacks captured by the hosting device during the conference call.

SPEECH SUPPRESSION BY PREDICTIVE FILTERING

ECHO CANCELLING DEVICE WITH FREQUENCY SUB-BAND FILTERING

An echo cancelling device for use between a line receiving an incoming signal and a line transmitting an outgoing signal, for cancelling out echo, comprising a plurality of processing channels connected in parallel relation and assigned to successive mutually adjacent sub-bands of the spectral band of the outgoing signal, each channel having: a first analysis band-pass filter receiving the echo-affected signal to be transmitted, whose output is connected to the additive input of a subtractor; a second analysis band-pass filter, identical to the first filter, receiving the incoming signal and feeding an adaptive filter delivering an estimated echo value in the sub-band to the subtractive input of the subtractor; and a synthesis filter, symmetrical with the analysis filters and whose output feeds the transmission line. Each processing channel receives an estimation of the aliasing component originating from another sub-band at least and that component is eliminated by adding it to the signal ...

REAL TIME PROCESSING AND MANAGEMENT METHOD FOR CANCELING OUT THE ECHO BETWEEN A LOUDSPEAKER AND A MICROPHONE OF A COMPUTER TERMINAL

L'invention concerne un procédé temps réel de traitement et de gestion pour l'annulation d'écho entre haut-parleur (HP) et microphone (M) d'un terminal informatique. Il consiste à établir le signal de haut-parleur ou de micro comme signal de référence et à synchroniser (A) l'autre signal par rapport à ce dernier ainsi que (B) les tâches d'acquisition et de restitution audio. Le retard existant (rej) entre l'autre signal et le signal de référence est mesuré (C) et la valeur de retard existant (rej) est validée (D) pour commander le retard appliqué au signal de référence à partir du retard courant (rej). Application aux terminaux informatiques équipés de systèmes d'exploitation multi-tâches.

IMPROVED SYSTEM AND METHOD FOR IMPLEMENTATION OF AN ECHO CANCELLER

A system and method for cancelling an echo signal. An input waveform is provided to an acoustic processor, and a determination is made whether the input waveform includes information representative of an echo signal. If the input waveform includes information representative of an echo signal, an output waveform is formed by attenuating a residual waveform with the acoustic processor. The residual waveform is attenuated by an attenuation factor that gradually changes from an initial attenuation value to a final attenuation value during the attenuation step. A system and method for adjusting an acoustic signal from a muted state to an unmuted state by varying an attenuation factor applied to an acoustic signal by an acoustic processor. The acoustic signal is provided to an acoustic processor. An output waveform is formed from the acoustic processor by adjusting the attenuation factor from a muted state to a first attenuation value associated with the non-muted state. After the attenuation ...

INTEGRATED VOICE PROCESSING SYSTEM FOR PACKET NETWORKS

A next-generation voice processing system (NGVPS) is provided. Voice- processing blocks within prior art system have been opened up revealing common functions and interblock dependencies. By opening up and consolidating portions of these blocks, the NGVPS enhances the functionality of some functions by using processing and signals that were previously only available to a single block. By taking into account the interaction of these various sub- systems and elements, the NGVPS provides the best overall voice performance. This holistic approach provides new implementations for optimizing voice processing from an end-to-end systems approach.

METHOD AND APPARATUS FOR ACOUSTIC ECHO CANCELLATION IN A COMMUNICATION SYSTEM PROVIDING TTY/TDD SERVICE

A method and apparatus are disclosed for canceling acoustic echo generated during transmission of text information in a communication system supporting a TTY/TDD service. When an mobile-to-land call is established between a mobile subscriber and a PSTN subscriber, a decoder notifies an encoder of detection of TTY/TDD text in a BS vocoder. The encoder then transmits a packet having silence information, which prevents acoustic echo reproduction of the text in the TTY/TDD device.

ACOUSTIC ECHO CANCELLER

A method and apparatus for cancelling acoustic echoes that enhances the hands- free operation of audio/video conferencing equipment, wireless and cellular telephones, internet and intranet telephones, etc. is disclosed. The method and apparatus use a constrained and orthogonalized, frequency domain, block, least mean square adaptive filter (5) to generate an estimate of an acoustic echo signal. The estimate of the acoustic echo signal is subtracted from a near-end microphone signal to provide an echo reduced communication signal. The echo reduced communication signal is then either transmitted or processed further. The further processing can include non-linear processing using an adaptive speech filter (18). The method and apparatus include a novel method for updating the coefficients of an adaptive filter (5).

TELEPHONE APPARATUS, METHOD OF CONTROLLING TELEPHONE APPARATUS, AND PROGRAM

To provide a telephone apparatus that can prevent generation of a howling noise more reliably, a method of controlling the telephone apparatus, and a program. A control unit (113) of a telephone device (100) is configured to perform a first processing of: estimating a line echo amount based on a volume of a DTMF signal generated by a DTMF signal generation unit (108) and a volume of the DTMF signal input from a line (200); calculating a howling noise threshold value based on the maximum value of the acoustic echo generated at a handset (104) that is measured in advance and set values of a sidetone volume (105), a transmission volume (106), and a reception volume (111); and calculating a difference between the line echo amount and the howling noise threshold value as an adjustment requisite amount and lowering the set value of the sidetone volume (105) by the adjustment requisite amount when the line echo amount is larger than the howling noise threshold value.

AUTOMATICALLY TUNING AN AUDIO COMPRESSOR TO PREVENT DISTORTION

A system and method automates the tuning of one or more multiband compressors or multiband limiters to minimize loudspeaker distortion. The system and method render one or more test loudspeaker signals that vary in frequency and in amplitude out of a loudspeaker and records the responses of the loudspeaker at a microphone. The system and method measure a distortion of the loudspeaker, relative to a frequency and an amplitude, with respect to the one or more test loudspeaker signals and a microphone signal and calculate tunable parameters of the multiband compressor that includes cutoff frequencies in response to the measured distortion.

CLOUD-BASED ACOUSTIC ECHO CANCELLER

A cloud based echo canceller is set forth for recreating an estimate of a lost packet or data at a server without requiring redundant data over the network or freezing operation of the echo canceller. In an exemplary embodiment, the echo cancelling function is not located in a single device, but is shared between the end-point and a cloud service, where the function of the end-point is to provide a time synchronized copy of the signal from the end-point loudspeaker and the signal received by the end-point microphone. Consequently, the high CPU intensive operations can be offloaded to a server such as a cloud server. In addition, several users can share the echo canceller, thereby reducing the cost of the overall function. According to an additional aspect, a further synchronization block is provided, in the form of a packet estimator, to compensate for packet or data loss in the send direction.

METHOD OF CAPTURING CONSTANT ECHO PATH INFORMATION IN A FULL DUPLEX SPEAKERPHONE USING DEFAULT COEFFICIENTS

A method of determining when to save default coefficients in an echo canceller so as to ensure that the capture of coefficients that correspond to the best possible echo cancellation in a current condition. Coefficients are saved at varying times depending on the amount of echo removed by the echo canceller. More particularly, the present method involves constantly monitoring the error signal to the echo canceller and comparing it with the error signal that would be obtained if default coefficients were to be used instead of the current coefficients. This ensures that the default coefficients are upgraded each time the current set of coefficients is better than the saved default coefficients.

Feedback Level Estimator Between Loudspeaker and Microphone

MULTI-CHANNEL ECHO CHANCELLATION WITH ADAPTIVE FILTERS HAVING SELECTABLE COEFFICIENT VECTORS

Multi-Channel Echo Cancelling Method and a Device Thereof

TRAINING METHOD FOR AN ECHO CANCELLER FOR USE IN A VOICE CONFERENCE SYSTEM

SUBBAND ECHO CANCELLATION METHOD USING PROJECTION ALGORITHM

A received signal is output to an echo path and, at the same time, it is divided into a plurality of subbands to generate subband received signals, which are applied to estimated echo paths in the respective subbands to produce echo replicas. The echo having propagated over the echo path is divided into a plurality of subbands to generate subband echoes, from which the corresponding echo replicas are subtracted to produce misalignment signals. Based on the subband received signal in each subband and the misalignment signal corresponding thereto, a coefficient to be provided to each estimated echo path is adjusted by a projection or ES projection algorithm.

OPTIMIZATION OF ADAPTIVE FILTER TAP SETTINGS FOR SUBBAND ACOUSTIC ECHO CANCELERS IN TELECONFERENCING

A subband acoustic echo canceller for a telecommunications teleconferencing room is equiped with subband adaptive filters and filter taps associated with said subbands. A data store is provided containing a weight list for controlling allocation of the filter taps among the subbands. An optimum tap profile as determined by the weights is a composite of room acoustic impulse response and weighting adjustments based on one or more measures of perceived human acoustic sensitivity experienced by far-end users. An active periodic adjustment of the weights responsive to detection of sibilant energy in the incoming path further improves echo cancellation. Active measurement and updating of the room acoustic impulse response provides further refinement.

ECHO CANCELLING SYSTEM SUITABLE FOR VOICE CONFERENCE

An echo cancelling system has two echo cancellers for cancelling a channel echo and a room echo, respectively. In order to obviate echo estimation errors, the echo canceller at the channel side performs estimation only when an input signal from a channel is absent and a microphone input signal is present. The echo canceller at the room or acoustic side executes estimation only when a microphone input signal i s absent and an input signal from the channel is present. A necessary amount of suppression is applied to either the receipt side or the transmission side until the echo canceller estimates an echo, so that a howling margin is guaranteed.

METHOD AND APPARATUS FOR MULTI-CHANNEL ACOUSTIC ECHO CANCELLATION

A variation in the cross-correlation between current received signals of different channels is extracted which corresponds to the cross-correlation between previous received signals, and the extracted variation is used as an adjustment vector to iteratively adjust the estimation of the impulse response of each echo path. Furthermore, by additionally providing a function of actively varying the cross-correlation between the received signals to such an extent as not to produce a jarring noise, it is possible to reconstruct acoustic signals by individual loudspeakers and utilize received signal added with the cross-correlation variation to obtain the adjustment vector for an estimated echo path vector.

ACOUSTIC ECHO CANCELER

Acoustic echos are canceled by employing a first echo canceler having a comparatively short first impulse response synthesis capability which is connected between a transmit path and receive path for generating a first error signal and for canceling echo signals in the transmit path, and at least a second echo canceler having a comparatively long second impulse response synthesis capability connected in parallel with the first echo canceler between the transmit and the receive path. The second echo canceler is supplied with the first error signal from the first echo canceler and is adaptively operating simultaneously with but independent of the first echo canceler to further cancel echos in the transmit path. More specifically, the first echo canceler is intended to capture the direct path acoustic echo and any early arriving echos that are stable which are not time varying. Since the direct path acoustic echo and the stable early arriving echos are not time varying, the adaptive adaptation ...

ECHO SUPPRESSOR AND NON-LINEAR PROCESSOR OF ECHO CANCELLER

The invention relates to an acoustic echo suppressor and a non-linear processor for an echo canceller in a 4-wire data transmission network. An echo canceller is typically controlled by comparing the power levels (PWR IN, PS IN) of the near-end and far-end signals (R IN, S IN). The transfer function of an acoustic echo path is usually very non- uniform. According to the invention, the spectrum of the far-end signal (R IN) is treated, before the signal power level is calculated, with a weighted filter (21), which models the effect of the transfer function of acoustic echo. As a result, the double talk dynamics is improved , since high-energy vowels of far-end speech, which normally cause low-energy vowels of near-end speech to be clipped in an echo suppressor, will be attenuated.

ADAPTIVE FILTER AND ADAPTING METHOD THEREOF

An object of this invention is to provide an adaptive filter capable of reducing a convergence time and residual error even when there exists an interval having a small input signal power for a long time, and an adapting method thereof. The adaptive filter comprises a signal power monitoring circuit for receiving an input signal and outputs of delay elements and outputting a first stop signal for stopping tap position control and a second stop signal for stopping coefficient updating and a coefficient updating control circuit for receiving the second stop signal and an error signal outputted by a subtracter and outputting the error signal or zero to coefficient generating circuits as an updating signal. The signal power monitoring circuit calculates a sum of powers of all or part of signals of the delay elements, so that if the power sum is smaller than a first threshold, it turns on the first stop signal to stop the tap position control and if the sum is smaller than a second threshold ...

IMPROVED METHOD OF OPERATING A FULL DUPLEX SPEAKERPHONE ECHOCANCELLER

A method of improving convergence of an echo canceller in a full duplex speakerphone, wherein the echo canceller includes LEC (Line Echo Canceller) and AEC (Acoustic Echo Canceller) portions, comprising the steps of capturing LEC coefficients during operation, storing the coefficients, and utilizing the stored coefficients as default values during start-up of a subsequent call. The method of the present invention reduces the overall convergence time of the echo canceller by alleviating the requirement to wait for a suitable reference signal in order to converge the LEC.

DISTRIBUTED AUDIO SIGNAL PROCESSING IN A NETWORK EXPERIENCING TRANSMISSION DELAY

Apparatus and method for the prevention of echo signals in terminals connect ed to a network carrying telephony traffic. The echo prevention occurs prior to the introduction of signal delay by a network. It provides a method of controlling echo contained in audio signals transmitted between the nodes of a network, wherein the audio signals experience delay in transit from node to node and wherein the echo controlling method is distributed across the nodes, the method comprising: a ) feeding a signal from a local source of audio information to the node; b) canceling echo present in the local source signal to produce an echo canceled signal; c) suppressing any residual echo present in the echo canceled signal to produce an echo suppressed signal; and d) delivering the echo suppressed signal to the network.

DOUBLE TALK AND ECHO PATH CHANGE DETECTION IN A TELEPHONY SYSTEM

A double talk and echo path change detector (100) is provided for use in an echo canceller (10) that makes a determination about whether a residual signal is dominated by echo or by a "near end" signal. In order to make this determination, a first measure of linear dependency is computed between the residual signal and an echo estimate, and a second measure of linear dependency is computed between the residual signal and a desired signal. The two results are compared with each other, and if they are of about the same order, no further action is needed. However, if the comparison determines that the dependence between the residual signal and the desired signal is much stronger than the dependence between the residual signal and the echo estimate, the present detector (100) assumes that double talk has been detected and a signal denoting that result is output. On the other hand, if the dependence between the residual signal and the desired signal is much weaker than the dependence between ...

METHOD AND APPARATUS FOR CANCELLING ECHO ORIGINATING FROM A MOBILE TERMINAL

A novel method and apparatus are disclosed for cancelling echoes originating from a digital mobile handset using an echo canceller (10) located in the network. The method and apparatus thus solve a practical echo problem that many mobile telephone operators have long been faced with. The method includes an algorithm that differs significantly from a conventional network echo canceller algorithm, because of the following basic considerations that influenced a solution to the echo problem: (I) the algorithm should not worsen speech quality for digital mobiles that do not generate echoes (30); and (2) the ec ho path is non-linear due to two speech coder/decoder pairs in the echo path and a low echo level (38).

Adaptive control of electrical signal amplification involves using Wiener filter, deriving normalized gain factor from estimated useful signal, estimated noise signal

The method involves feeding the signal (d(n)) to a Wiener filter whose normalized gain factor is computed by deriving an estimated useful signal from the smoothed power of the electrical signal and then deriving the estimated noise signal power and the normalized gain factor from a stated equation from these parameters. An Independent claim is also included for an application of the method to a hands-free device.

Method, device and equipment for controlling echo suppressor, and storage medium

Signal processing method, apparatus and program

In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.

Echo canceler

An echo canceler 10 generates an echo elimination signal by filtering through adaptive filters 101 and 102 reference signals input from sound sources causing echoes. It includes a sound source number detecting unit 103 for detecting the number of the sound sources causing echoes from the reference signals, and a control unit 105 for making the number of taps of the adaptive filters 101 and 102 variable in accordance with the number of the sound sources detected by the sound source number detecting unit 103.

Apparatus And Method For Cancelling Echo In Joint Time Domain And Frequency Domain

Disclosed in the present invention is a method for cancelling echo in joint time domain and frequency domain. The method includes: receiving an input receiver signal and an input transmitter signal; implementing echo cancellation on the received transmitter signal, based on the received receiver signal, by using a first echo canceller which is either a time domain echo canceller or a frequency domain echo canceller, to obtain a first echo-cancelled signal; implementing echo cancellation again on the first echo-cancelled signal, based on the received receiver signal, by using a second echo canceller which is the other one of the time domain echo canceller and the frequency domain echo canceller, to obtain a second echo-cancelled signal; wherein, the filter parameters of the second filter of the second echo canceller is updated based on the second echo-cancelled signal, and the first and second echo canceller respectively include the corresponding first and second filters. By using said method in the present invention, fast response to echo reflecting environment can be achieved with little residual echo, thus the effect of echo cancellation is entirely improved.

System and method for echo reduction in audio and video telecommunications over a network

A method and a system use an intermediate server to process the communication between two parties, so as to eliminate echoes between them. The server performs echo cancellation in a network-based voice communication system handling a large number of conversations. In one implementation, the server allocates two echo cancellation modules to each conversation, with each echo cancellation module including (a) a communication interface for communicating with a client program associated with the echo cancellation module; (b) a first buffer for storing audio data received from the client program for transmission to another echo cancellation module; (c) a second buffer for storing audio data received from the other echo cancellation module for transmitting to the associated client program; and (d) a set of filters using the audio data in both the first buffer and the second buffer to cancel echoes in the audio data in the second buffer.

Device and method for controlling damping of residual echo

The present invention relates to a device, such as a communication device, comprising an adaptive foreground filter configured to calculate a first echo estimation signal based on a first input signal, and an adaptive background filter being more rapidly adapting than the foreground filter and configured to calculate a second echo estimation signal based on said first input signal. Embodiments of the device further comprise damping control means for controlling damping of an echo-cancelled output signal. The device in various embodiments includes that the damping control means is configured to calculate a maximum echo estimation signal using both the first and the second echo estimation signals, and control the damping of the echo-cancelled output signal based on said maximum echo estimation signal and/or a signal derived from said maximum echo estimation signal.

Speech quality enhancement in telecommunication system

Technologies are generally described for an echo cancelling device of a telecommunication system. In some examples, an echo canceling device may include a noise reduction unit configured to reduce a background noise around a near-end talker from a near-end signal provided by a microphone, a double talk detector configured to detect a double talk event based on the noise-reduced near-end signal and a far-end signal, and a filtering unit configured to receive the far-end signal and the near-end signal provided by the microphone.

Nonlinear reference signal processing for echo suppression

An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

Method and apparatus for reducing noise pumping due to noise suppression and echo control interaction

An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Methods and apparatuses for echo cancelation with beamforming microphone arrays

Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of microphones oriented to cover a corresponding plurality of direction vectors and to develop a corresponding plurality of microphone signals. A processor is operably coupled to the plurality of microphones. The processor is configured to perform a beamforming operation to combine the plurality of microphone signals to a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals. The processor is also configured perform an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals and select one of the plurality of combined echo-canceled signals for transmission.

Echo Cancelling-Codec

Echo-cancellation is utilized in terminal devices such as speakerphones to compensate for acoustic echoes and interaction of the audio signal with the surrounding environment. An echo-cancelling codec incorporates encoding, decoding and acoustic echo-cancellation in a single device, enabling processing to be utilized that reduces processing and memory resources. The configuration enables processing information to also be shared between encoding, decoding and acoustic echo-cancellation functions to optimize operational characteristics. The acoustic echo cancelling codec interfaces between the amplitude signal domain, speaker and microphone, and an encoded data domain, a data interface, reducing component requirements required to provide echo-cancellation and coding functions.

Methods and apparatuses for multi-channel acoustic echo cancelation

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.

ECHO CANCELLER

The proposed echo canceller comprises:—an adaptive filter for receiving a signal affected by an echo, and for supplying a filtered signal that is an estimate of the echo;—a subtractor for subtracting this estimate from the received signal and supplying a residual signal;—means for detecting () a ring back tone in said residual signal;—means for blocking () the received signal and replacing it by a locally generated ring back tone if a ring back tone is detected in the received signal;—a timer to determine a time period;—and means (-) for, during said time period, replacing the residual signal by some synthetic comfort noise when there is no ring back tone. 1) An echo canceller comprising:an adaptive filter for receiving a signal affected by an echo, and for supplying a filtered signal that is an estimate of the echo;a subtractor for subtracting this estimate from the received signal and supplying a residual signal; means for comparing the difference between the energy of the received signal and the energy of a transmitted signal with a first threshold, and then concluding that a ring back tone is detected when this difference is greater than this first threshold,', 'and means for comparing the energy of the transmitted signal to a second threshold, and comparing the cumulated time during which this energy is greater than the second threshold with a fourth threshold, then assuming that the ringing phase is elapsed when this cumulated time is greater than this fourth threshold;, 'means for detecting a ring back tone in said residual signal, comprisingmeans for blocking the received signal and replacing it by a locally generated ring back tone when a ring back tone has been detected in the received signal, and until the ringing phase is assumed to be elapsed.2) An echo canceller according to claim 1 , further comprising:a timer to determine a time period;and means for, during said time period, replacing the residual signal by some synthetic comfort noise when there is ...

VOICE SWITCHING FOR VOICE COMMUNICATION ON COMPUTERS

A voice communication end device performs quality checks to determine whether acoustic echo cancellation would be ineffective, such as due to noise or clock drift or discontinuities between incoming and outgoing voice channels. In the case where echo cancellation would prove ineffective, the device falls back on a tri-state voice switching operation that includes a bi-direction state in which both channels are on in full duplex operation, which provides a smoother transition switching between active channels. The tri-state voice switching supports both voluntary transitions where the active user voluntarily stops to yield the active channel, and forced transitions where the active user is forcedly interrupted by the other user speaking more loudly. 1. A computer-readable media storing instructions thereon for executing a method of preventing acoustic echo in a two-way voice communication end device , the method comprising:upon starting a communication session with another communication end device, operating in a full duplex voice communication with acoustic echo cancellation mode;determining whether the voice communication with the other communication end device has sufficient quality for effective acoustic echo cancellation; andin the event that the voice communication is determined to lack sufficient quality, operating in a tri-state voice switching mode; operating in an outgoing state when voice activity is detected on an outgoing channel;', 'operating in incoming state when voice activity is detected on an incoming channel;', 'operating in a bi-directional state when voice activity ceases for over a threshold period of time while in the incoming or outgoing states;', 'from the incoming state, forcing a transition to the outgoing state when there is voice activity on both the incoming and outgoing channels, but a first energy level associated with the volume of the voice activity on outgoing channel exceeds a second energy level of the voice activity on the ...

Optimizing audio processing functions by dynamically compensating for variable distances between speaker(s) and microphone(s) in a mobile device

Mobile communication devices, having multiple speakers and/or microphones to perform a number of audio functions, for use with mobile devices, are provided. The microphones may be housed within the communication device housing. To compensate for the unwanted signal feedback between the speakers and microphones, acoustic echo cancellation may be implemented to determine the proper distance and relative location between the speakers and microphones. Acoustic echo cancellation removes the echo from voice communications to improve the quality of the sound. The removal of the unwanted signals captured by the microphones may be accomplished by characterizing the audio signal paths from the speakers to the microphones (speaker-to-microphone path distance profile), including the distance and relative location between the speakers and microphones. The optimal distance and relative location between the speakers and microphones is provided to the user to optimize performance.

Echo Canceler Circuit and Method

An echo canceler circuit () and method attenuates at least post-echo canceler uplink data () to produce attenuated uplink data () in response to uplink echo return loss based attenuation data (). The echo canceler circuit () includes an echo return loss based attenuation data generator () and at least an uplink data attenuator (). The echo return loss based attenuation data generator () produces the uplink echo return loss based attenuation data () in response to echo return loss data (). The echo return loss data () is based on at least one of: attenuated downlink data (), pre-echo canceler uplink data (), and/or amplifier gain data (). The uplink data attenuator () attenuates the post-echo canceler uplink data () to produce attenuated uplink data () based on the uplink echo return loss based attenuation data (). 1. An echo canceler circuit comprising:an uplink data attenuator operative to receive at least post-echo canceler uplink data and uplink echo return loss based attenuation data and in response to attenuate the post-echo canceler uplink data to produce attenuated uplink data; andan echo return loss (ERL) based attenuation data generator operatively coupled to the uplink data attenuator and operative to produce the uplink echo return loss based attenuation data in response to echo return loss data, wherein the echo return loss data is calculated based on a ratio of the attenuated downlink data to the pre-echo canceler uplink data.2. (canceled)3. The echo canceler circuit of wherein the uplink data attenuator is operative to attenuate the post-echo canceler uplink data over a period of time to produce the attenuated uplink data.4. An echo canceler circuit comprising:an uplink data attenuator operative to receive post-echo canceler uplink data and uplink echo return loss based attenuation data and in response to attenuate the post-echo canceler uplink data to produce attenuated uplink data;a downlink data attenuator operative to receive downlink data and ...

ECHO CANCELLATION USING CLOSED-FORM SOLUTIONS

A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal. 1. An echo cancellation system , comprising:a filter that is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal;filter parameter determination logic that is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics; anda combiner that is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal.2. The echo cancellation system of claim 1 , wherein the filter processes a frequency-domain representation of the far-end audio signal.3. The echo cancellation system of claim 1 , wherein the filter comprises a hybrid frequency-domain filter that generates the estimated echo signal by passing each of a plurality of frequency components of a frequency-domain representation of the far-end audio signal through a respective one of a plurality of time direction filters.4. The echo cancellation system of ...

Method and System for Stereo Echo Cancellation for VOIP Communication Systems

An exemplary embodiment of the present invention is directed toward a method and system for cancelling line echo in the presence of a known secondary audio signal. Filter adaptation is enabled in the presence of a known secondary audio source such as the sound of a computer game, a music signal or other secondary audio sources that would otherwise prevent echo cancellation due to an apparent double talk condition. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

Methods, apparatus and articles of manufacture to cancel echo for communication paths having long bulk delays

Example methods, apparatus and articles of manufacture to cancel echo for a communication path having long bulk delays are disclosed. A disclosed example method includes determining a first location of a first coefficient having a largest magnitude of a first plurality of magnitudes associated with a first plurality of coefficients of a first phase; determining a second location of a second coefficient having a largest magnitude of a second plurality of magnitudes associated with a second plurality of coefficients of a second phase different than the first phase; comparing a difference between the first and second locations to a threshold; and, when the difference is less than the threshold, selecting a first offset based on a greater of the magnitude of the first coefficient and the magnitude of the second coefficient; and cancelling an echo contained in a signal using the first offset.

Reduced-delay subband signal processing system and method

A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.

Method and apparatus for a frequency domain echo suppression filter

Residual frequency components of a reference signal are suppressed from an error signal. A magnitude of the frequency domain representation of the reference signal is divided by a magnitude of the frequency domain representation of LMS-filtered representation of the error signal to obtain a frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal. The frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal is multiplied by the frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal, to obtain a frequency domain signal having reduced residual frequency components of the reference signal.

COMPUTATION SAVING ECHO CANCELLER FOR A WIDE BAND AUDIO SIGNAL

A canceller splits a signal, transmitted from a near end terminal (ET) to a far ET, into a sub-sampled signal corresponding to a higher frequency sub-band of the signal transmitted to the far ET, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal transmitted to the far ET; and splits a signal received from a far ET into a sub-sampled signal corresponding to a higher frequency sub-band of the signal received from the far ET, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal received from the far ET. The canceller includes a first adaptive filter for filtering the sub-sampled signal corresponding to the lower frequency sub-band, a second adaptive filter for filtering the sub-sampled signal corresponding to the higher frequency sub-band, and controls the adaptation of the first and second adaptive filters so that these two adaptations are never simultaneous. 1. A wide band echo canceller comprising:means for splitting a signal transmitted from a near end terminal to a far end terminal, into a sub-sampled signal corresponding to a higher frequency sub-band of the signal transmitted to the far end terminal, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal transmitted to the far end terminal,means for splitting a signal received from a far end terminal into a sub-sampled signal corresponding to a higher frequency sub-band of the signal received from the far end terminal, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal received from the far end terminal,a first adaptive filter for filtering the sub-sampled signal corresponding to the lower frequency sub-band of the signal transmitted to the far end terminal, and restituting a first filtered signal,a second adaptive filter for filtering the sub-sampled signal corresponding to the higher frequency sub-band of the signal transmitted to the far end terminal, and restituting a second filtered ...

CALL QUALITY IMPROVEMENT SYSTEM, APPARATUS AND METHOD

Provided is a call quality improvement method configured to operate a call quality improvement system and a call quality improvement apparatus by executing an artificial intelligence (AI) algorithm and/or a machine learning algorithm in a 5G environment connected for the Internet of Things. According to one embodiment of the present disclosure, the call quality improvement method may include receiving a voice signal from a far-end speaker, receiving a sound signal including a voice signal from a near-end speaker, receiving an image of a face of the near-end speaker, including lips, and extracting the voice signal of the near-end speaker from the received sound signal. 1. A call quality improvement system using lip-reading , the call quality improvement system comprising:a microphone configured to collect a sound signal including a voice signal of a near-end speaker;a speaker configured to output a voice signal from a far-end speaker;a camera configured to photograph a face of the near-end speaker, including lips; anda sound processor configured to extract the voice signal of the near-end speaker from the sound signal collected from the microphone,wherein the sound processor comprises an echo reduction module including an adaptive filter configured to filter out an echo component from the sound signal collected through the microphone based on a signal inputted to the speaker, and a filter controller configured to control the adaptive filter, andthe filter controller changes parameters of the adaptive filter based on lip movement information of the near-end speaker.2. The call quality improvement system according to claim 1 , wherein the sound processor further comprises:a noise reduction module configured to reduce a noise signal in the sound signal from the echo reduction module; anda voice reconstructor configured to reconstruct the voice signal of the near-end speaker damaged during a noise reduction process through the noise reduction module, based on the lip ...

VARIABLE STEP SIZE ECHO CANCELLATION WITH ACCOUNTING FOR INSTANTANEOUS INTERFERENCE

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. 1. A system capable of converging in the presence of constant double talk using a variable step size (VSS) adaptive filter , said system comprising:a memory area for storing long-term statistics describing a reference signal, a near-end noise signal, and an error signal; and monitor a present value corresponding to the error signal during a conversation;', 'determine a first function based on the monitored present value for the error signal;', 'determine a second function based on the stored long-term statistics; and', 'calculate, based on applying a first weight to the determined first function and a second weight to the determined second function, an adaptation coefficient for the VSS adaptive filter, the first weight and the second weight representing a level of confidence in the first function and the second function, respectively; and, 'a processor programmed toreduce the near-end noise signal during the conversation by applying the calculated adaptation coefficient to the VSS adaptive filter.2. (canceled)3. The system of claim 1 , wherein the processor is further programmed to update claim 1 , in real-time claim 1 , a ...

SOUND EMISSION AND COLLECTION DEVICE, AND SOUND EMISSION AND COLLECTION METHOD

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. 110-. (canceled)11. A sound emission and collection device comprising:a speaker;at least one microphone;a reverberation time estimation section configured to estimate a reverberation time for each frequency band in a space where the speaker and the at least one microphone are present; andan arithmetic operation section configured to specify a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time and to suppress power of the specified frequency band.12. The sound emission and collection device according to claim 11 , further comprising:at least one echo canceller configured to cancel a regression sound signal of sound emitted by the speaker from a sound collection signal output by the at least one microphone, wherein:the reverberation time estimation section estimates a reverberation time for each frequency band in a space where the speaker and the at least one microphone are present based on an adaptive filter coefficient obtained from the at least one echo canceller, andthe ...

HOWLING SUPPRESSION APPARATUS, AND METHOD AND PROGRAM FOR THE SAME

A howling suppression apparatus includes: an integration processing part that obtains the maximum value among L values corresponding to n-th frames of L i-th signals, for i=1, 2, . . . , L, L being any integer equal to or greater than 2, the L i-th signals being frequency-domain signals obtained from sound signals collected by multiple microphones; and a howling suppression processing part that performs howling suppression processing on at least any of the L i-th signals using the maximum value. 1. A howling suppression apparatus comprising:an integration processing part that obtains a maximum value among L values corresponding to n-th frames of L i-th signals, for i=1, 2, . . . , L, L being any integer equal to or greater than 2, the L i-th signals being frequency-domain signals obtained from sound signals collected by a plurality of microphones; anda howling suppression processing part that performs howling suppression processing on at least any of the L i-th signals using the maximum value.2. The howling suppression apparatus according to claim 1 , wherein the howling suppression processing part performs the howling suppression processing by utilizing a fact that howling components are out of phase among the sound signals collected by the plurality of microphones.3. The howling suppression apparatus according to or claim 1 , wherein claim 1 , at least either (i) if the maximum value is greater than a value indicating predetermined power or (ii) if a value indicating a variation in the maximum value is greater than a value indicating a predetermined variation claim 1 , the howling suppression processing part performs the howling suppression processing by multiplying at least any of the L i-th signals by a smaller one of a first gain obtained based on the maximum value and a second gain obtained based on the value indicating the variation in the maximum value.4. The howling suppression apparatus according to or claim 1 , comprising a smoothing processing part that ...

ACOUSTIC ECHO SUPPRESSION DEVICE AND ACOUSTIC ECHO SUPPRESSION METHOD

A microphone picks-up voice of a driver. A first echo suppression unit outputs a voice signal after first echo suppression based on a voice signal of the driver and a voice signal after echo suppression in the past (first reference signal) stored in a buffer memory. A second echo suppression unit outputs a voice signal after second echo suppression based on a voice signal of the driver and a voice signal after the echo suppression in the past (second reference signal) stored in a buffer memory. An output signal selector selects one of the voice signals after the first echo suppression or the voice signal after the second echo suppression according to a detection result of the presence or absence of a system variation by a system variation detector, and causes a speaker to output the selected voice signal. 1. An acoustic echo suppression device that suppresses acoustic echo in a room where a sound pick-up unit is installed , the device comprising:a first filter processor which is connected to the sound pick-up unit and outputs a first sound signal obtained by updating an echo component included in a picked-up sound signal acquired by the sound pick-up unit at a first rate;a second filter processor which is connected to the sound pick-up unit and outputs a second sound signal obtained by updating the echo component included in the picked-up sound signal at a second rate faster than that of the first filter processor, against a sudden variation in a sound field environment in the room;a detector which detects presence or absence of the variation in the sound field environment in the room; andan output selector which selects one of the first sound signal and the second sound signal according to a detection result of the presence or absence of the variation in the sound field environment in the room and causes a voice output unit to output the selected sound signal.2. The acoustic echo suppression device of claim 1 ,wherein an update rate of a coefficient of a filter ...

COMMUNICATION DEVICE WITH ECHO SUPPRESSION

The application relates to a communication device, e.g. a speakerphone, comprising a microphone signal path, MSP, and a loudspeaker signal path, SSP, the microphone signal path comprising a microphone unit, an MSP-filter, and a transmitter unit operationally connected to each other and configured to transmit a processed signal originating from an input sound picked up by the microphone, the loudspeaker signal path comprising a receiver unit, an SSP-filter, and a loudspeaker unit operationally connected to each other and configured to provide an acoustic sound signal originating from a signal received by the receiver unit. The communication device comprises a control unit for dynamically controlling the filtering characteristics of the MSP and SSP-filters based on one or more control input signals. This has the advantage of providing a simple and flexible scheme for decreasing echo in a communication device, while ensuring an acceptable sound quality in the transmitted signal. 1. A communication device comprising a microphone signal path , termed MSP , and a loudspeaker signal path , termed SSP , the microphone signal path comprising a microphone unit , an MSP-signal processing unit comprising an MSP-filter , and a transmitter unit operationally connected to each other and configured to transmit a processed signal originating from an input sound picked up by the microphone , the loudspeaker signal path comprising a receiver unit , an SSP-signal processing unit comprising an SSP-filter , and a loudspeaker unit operationally connected to each other and configured to provide an acoustic sound signal originating from a signal received by the receiver unit , wherein the filtering characteristics of the MSP-filter and/or the SSP-filter are configurable , and wherein the communication device comprises a control unit for dynamically controlling the configurable filtering characteristics of the MSP and SSP-filters based on one or more control input signals dependent of a ...

Double talk detection method, double talk detection apparatus and echo cancellation system

A double talk detection method, a double talk detection apparatus and an echo cancellation system are provided. The double talk detection method comprises: determining, according to an energy ratio between a far-end digital voice signal and a near-end digital voice signal, and a frequency coherence value between the near-end digital voice signal and the far-end digital voice signal, whether a near-end speaker's digital voice signal is present in the near-end digital voice signal. The double talk detection method avoids missing detection and false detection, improves the accuracy of double talk detection, cancels the echo in the near-end voice signal thoroughly when applied in the field of echo cancellation, and improves the communication experience of both talk parties.

ACOUSTIC ECHO MITIGATION APPARATUS AND METHOD, AUDIO PROCESSING APPARATUS AND VOICE COMMUNICATION TERMINAL

The present application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.

Full duplex wireless communications on devices with limited echo cancellation capabilities

Disclosed in some examples are methods, systems, and machine readable mediums which allow for wireless devices with limited echo cancellation capabilities to participate in full-duplex communications. In some examples, by carefully controlling transmission powers and the modulation and coding schemes (MCS) used in the transmissions, both devices can engage in full-duplex communication.

CLOUD-BASED ACOUSTIC ECHO CANCELLER

A cloud based echo canceller is set forth for recreating an estimate of a lost packet or data at a server without requiring redundant data over the network or freezing operation of the echo canceller. In an exemplary embodiment, the echo cancelling function is not located in a single device, but is shared between the end-point and a cloud service, where the function of the end-point is to provide a time synchronized copy of the signal from the end-point loudspeaker and the signal received by the end-point microphone. Consequently, the high CPU intensive operations can be offloaded to a server such as a cloud server. In addition, several users can share the echo canceller, thereby reducing the cost of the overall function. According to an additional aspect, a further synchronization block is provided, in the form of a packet estimator, to compensate for packet or data loss in the send direction. 1. A method of compensating for lost packets in a distributed echo canceler , where successive input packets are stored in memory of the distributed echo canceler , comprising:a) detecting a lost signal packet Ro′(n);b) freezing operation of the distributed echo canceler;c) invoking a packet loss compensation (PLC) algorithm for one of either recreating an estimated output packet from previous output packets or halting transmission of output packets;d) estimating the lost signal packet Ro′(n) by performing a correlation of a previously received signal packet with an input signal packet Rin, 'A) upon receipt of a next packet repeating steps b) and c) until effect of the lost signal packet Ro′(n) is flushed out of an echo canceler memory and resuming operation of the distributed echo canceler; or', 'i) if said correlation is poor then'} B) using a relative shift offset of the lost signal packet Ro′(n) to the input signal packet Rin to read an estimated buffer value (Ro″) out of the successive input packets and replacing the signal packet Ro′(n) by the estimated buffer value (Ro ...

Robust Short-Time Fourier Transform Acoustic Echo Cancellation During Audio Playback

Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each niteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nframe of the model signal from an nframe of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC. 1. A playback device comprising:an audio input interface;an audio stage comprising an audio processor and an audio amplifier;one or more speakers;one or more microphones;at least one processor; and receive, via the audio input interface, one or more audio signals;', 'play back at least one audio signal of the one or more audio signals via the one or more speakers and the audio stage;', 'while playing back the at least one audio signal, capture, via the one or more microphones, audio within an acoustic environment, wherein at least a portion of the captured audio represents sound produced by the one or more speakers in playing back the at least one audio signal via the one or more speakers;', 'receive at least one playback signal from the audio stage representing the at least one audio signal being played back by the one or more speakers and the audio stage;', 'transform into a short time Fourier transform (STFT) domain the captured audio within the acoustic environment to generate a measured signal representing actual acoustic echo;', 'transform into the STFT domain the ...

ADAPTIVE FILTER UNIT FOR BEING USED AS AN ECHO CANCELLER

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(f, A(t, . . . , t)) in the frequency domain; to calculate a transformed second audio signal Y(f, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal. 1. An adaptive filter unit , in particular for being used as an echo canceller , comprisinga first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t);a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t); receive the first and second electric audio signal;', {'sub': n', '1', 'M(fn)', 'n', '1', 'N', 'n, 'calculate and provide audio estimation data X(f, A(t, . . . , t)) in the frequency domain by calculating a FFT transform of the first audio signal A(t), for frequencies f=f, . . . , f, wherein N is a number of FFT bins, and with a number of sampling points M(f) of the first audio signal A(t);'}, {'sub': 'n', 'calculate a transformed second audio signal Y(f, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain;'}, 'calculate a filtered audio signal by ...

Pre-distortion system for cancellation of nonlinear distortion in mobile devices

A pre-distortion system for improved mobile device communications via cancellation of nonlinear distortion is disclosed. The pre-distortion system may transmit an acoustic signal from a network to a device, wherein the acoustic signal includes a linear signal and a nonlinear cancellation signal that cancels at least a portion of nonlinear distortions created once a loudspeaker in the device emits the linear signal. Thus, when a loudspeaker of a mobile device is operating and nonlinear distortions are generated by the loudspeaker or adjacent components of the mobile device in close proximity to the loudspeaker, the pre-distortion system may create one or more nonlinear cancellation signals in the network. The nonlinear cancellation signal may be combined with the linear signal sent to the loudspeaker to cancel the nonlinear distortion signal created by the loudspeaker emitting acoustic sounds from the linear signal. Thus, the nonlinear cancellation signal becomes a pre-distortion signal.

METHOD AND DEVICE OF SUSTAINABLY UPDATING COEFFICIENT VECTOR OF FINITE IMPULSE RESPONSE FILTER

A method and a device of sustainably updating a coefficient vector of a finite impulse response FIRfilter. The method includes obtaining () a time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing a preset signal; updating () the coefficient vector of the FIR filter according to the time-varying regularization factor. 172.-. (canceled)73. A sustainable adaptive updating method of a coefficient vector of a Finite Impulse Response (FIR) filter , comprising:obtaining a time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing a preset signal;updating the coefficient vector of the FIR filter according to the time-varying regularization factor.74. The sustainable adaptive updating method according to claim 73 , wherein the preset signal comprises one of combined pairs of following:a far-end reference speech signal inputted in an Acoustic Echo Canceller (AEC) and a near-end speech signal received by a microphone;a noise reference signal and a system input signal in an adaptive noise cancellation system;an interference reference signal and a system input signal in an adaptive interference cancellation system; andan excitation input signal and an unknown system output signal to be identified in adaptive system identification.75. The sustainable adaptive updating method according to claim 73 , wherein the preset signal comprises a far-end reference speech signal inputted in an Acoustic Echo Canceller (AEC) and a near-end speech signal received by a microphone;obtaining the time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing the preset signal, comprises:obtaining a power of a signal received ...

ECHO CANCELLER FOR VOIP NETWORKS

An echo canceller in an IP network includes an adaptive filter that models the echo path between a receiving output port of the echo canceller and a sending input port. The adaptive filter filters a receiving input signal to generate an estimate of an echo signal. The estimate of the echo signal is subtracted from a sending input signal to cancel the echo in the sending input signal and to generate a sending output signal. A packet loss detection circuit detects when packet loss occurs in the echo path. Responsive to detection of packet loss in the echo path, the echo canceller applies packet loss concealment to either the sending output signal or the receiving input signal.

ECHO CANCELLER FOR VOIP NETWORKS

An echo canceller for an IP network includes an adaptive filter that models the echo path and generates an estimate of the echo signal from a receiving input signal. The echo canceller subtracts the estimate of the echo signal from a sending input signal to generate a sending output signal with reduced echo. Variation in the echo delay is detected. A delay circuit compensates for the changes in the echo delay to provide proper time-alignment between the estimate of the echo signal and the sending input signal so that the echo signal will be more effectively cancelled. 1. A method of echo cancellation to handle variation of an echo delay , said method comprising:generating, from a receiving input signal received on a first input port of an echo canceller, a first estimate of an echo signal using an adaptive filter that models an echo path between a first output port and a second input port of the echo canceller;computing a first estimate of the echo delay by correlating the first estimate of the echo signal with a sending input signal received on said second input port;time-aligning the first estimate of the echo signal with the sending input signal based on the first estimate of the echo delay; andsubtracting the time-aligned first estimate of the echo signal from the sending input signal to generate a first sending output signal with reduced echo for output over a second output port of the echo canceller.2. The method of wherein computing the first estimate of the echo delay by correlating the estimate of the echo signal with the sending input signal comprises:correlating the first estimate of the echo signal with a sending input signal received on said second input port to generate a correlation signal; andcomputing the first estimate of the echo delay by locating a peak in the correlation signal.3. The method of further comprising:detecting variation in the echo delay;wherein computing the first estimate of the echo delay comprises computing the estimate of the ...

SOUND EMISSION AND COLLECTION DEVICE, AND SOUND EMISSION AND COLLECTION METHOD

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. 1. A sound emission and collection device comprising:a speaker;a filter configured to process a sound emission signal serving as a sound signal to be supplied to the speaker;a plurality of microphones;a plurality of echo cancellers provided so as to respectively correspond to the plurality of microphones and configured to cancel regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones;a first integration section configured to integrate adaptive filter coefficients taken out from the plurality of echo cancellers;a reverberation time estimation section configured to estimate 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; andan arithmetic operation section configured to specify a frequency band having a long reverberation time from the sound emission signal on the basis of the estimated reverberation time, to calculate a filter coefficient for ...

MULTI-CHANNEL ACOUSTIC ECHO CANCELLATION

A playback device is configured to receive, via a network interface, a source stream of audio including first and second channel streams of audio, and to produce, via respective first and second speaker drivers, a first channel audio output and a second channel audio output. The playback device is also configured to receive, via one or more microphones, a captured stream of audio including first and second portions corresponding to the respective first and second channel audio outputs. The playback device is also configured to combine at least the first channel stream of audio and the second channel stream of audio into a compound audio signal and perform acoustic echo cancellation on the compound audio signal and thereby produce an acoustic echo cancellation output, then to apply the acoustic echo cancellation output to the captured stream of audio and thereby increase a signal-to noise ratio of the captured stream of audio. 1. A playback device comprising:a first speaker driver;at least a second speaker driver;at least one processor;a network interface;a non-transitory computer-readable medium; and receiving, via the network interface, a source stream of audio comprising source audio content to be played back by the playback device, wherein the source audio content comprises a first channel stream of audio and a second channel stream of audio;', 'producing a first channel audio output by playing back, via the first speaker driver, the first channel stream of audio;', 'producing a second channel audio output by playing back, via the second speaker driver, the second channel stream of audio;', 'receiving, via one or more microphones, a captured stream of audio comprising (i) a first portion corresponding to the first channel audio output and (ii) a second portion corresponding to the second channel audio output, wherein the captured stream of audio has a first signal-to-noise ratio;', 'combining at least the first channel stream of audio and the second channel ...

Robust Short-Time Fourier Transform Acoustic Echo Cancellation During Audio Playback

Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each niteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nframe of the model signal from an nframe of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC. 1. A system comprising:an audio stage comprising an audio processor and an audio amplifier;one or more speakers;one or more microphones;one or more processors;data storage storing instructions executable by the one or more processors that cause the system to perform functions comprising:while audio content is playing back via the one or more speakers, capturing, via the one or more microphones, audio within an acoustic environment, wherein the captured audio comprises audio signals representing sound produced by the one or more speakers in playing back the audio content;receiving a playback signal from the audio stage representing the audio content being played back by the one or more speakers;transforming into a short time Fourier transform (STFT) domain the captured audio within the acoustic environment to generate a measured signal representing actual acoustic echo;transforming into the STFT domain the received playback signal from the audio stage to generate a reference signal;{'sup': 'th', 'claim-text': [{'sup': th', 'th, 'claim-text': [{'sup': th', 'th', 'th, ' ...

EFFICIENT REUTILIZATION OF ACOUSTIC ECHO CANCELER CHANNELS

Audio systems and methods are provided to reduce echo content in an audio signal. The systems and methods receive an audio signal and sound stage rendering parameter(s), and select a set of filter coefficients to filter the audio signal to provide an estimated echo signal. The estimated echo signal is subtracted from a microphone signal to provide an output signal with reduced echo content. The set of filter coefficients are selected based upon the sound stage rendering parameter(s) from among a plurality of stored sets of filter coefficients. 1. A method of reducing echo content of an audio signal , comprising:receiving an audio program content signal;receiving a sound stage rendering parameter;selecting a set of echo filter coefficients, from among a plurality of stored sets of echo filter coefficients, based upon the sound stage rendering parameter;filtering the audio program content signal, using the selected set of filter coefficients, to generate an estimated echo signal;receiving a microphone signal, configured to include a signal component representative of an echo of the audio program content signal; andsubtracting the estimated echo signal from the microphone signal to generate an output audio signal.2. The method of further comprising loading the selected set of echo filter coefficients to an audio filter and activating the audio filter to perform the filtering.3. The method of further comprising rendering the audio program content signal into an acoustic signal claim 1 , based upon the selected sound stage rendering parameter.4. The method of further comprising loading the selected set of echo filter coefficients to an adaptive filter claim 1 , adapting the adaptive filter coefficients claim 1 , and copying the adaptive filter coefficients to an active audio filter that performs the filtering.5. The method of further comprising loading the selected set of echo filter coefficients to an adaptive filter claim 1 , the adaptive filter performing the ...

MULTI-CHANNEL RESIDUAL ECHO SUPPRESSION

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. 1. A method of suppressing residual echo , comprising:receiving a residual signal from an echo cancelation subsystem;receiving a first program content signal;determining a first spectral mismatch based at least upon a cross power spectral density of the first program content signal and the residual signal;receiving a second program content signal;determining a second spectral mismatch based at least upon a cross power spectral density of the second program content signal and the residual signal; andcontrolling a filter to filter the residual signal based 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.2. The method of wherein controlling the filter includes calculating filter coefficients and providing the filter coefficients to the filter.3. The method of wherein controlling the filter to filter the residual signal is based upon a previously determined first spectral mismatch and a previously determined second spectral mismatch during a period of time when a double-talk condition is detected.4. The method of further ...

MITIGATING IMPACT OF DOUBLE TALK FOR RESIDUAL ECHO SUPPRESSORS

Audio systems and methods of suppressing residual echo are provided that determine spectral mismatches by comparing a spectral density of a residual signal from an acoustic echo canceler to a spectral density of a program content signal. At least one spectral mismatch is stored in memory. The systems and methods select a spectral mismatch to use for calculating a filter coefficient, from among one or more of the stored or actively determined spectral mismatches, and filter the residual signal based upon the calculated filter coefficient. 1. A method of suppressing residual echo , comprising:determining a first spectral mismatch of an acoustic echo canceler based upon a program content signal and a residual signal;storing the first spectral mismatch in a memory;determining a second spectral mismatch of the acoustic echo canceler based upon the program content signal and the residual signal at a different time than the first spectral mismatch;selecting one of the first spectral mismatch or the second spectral mismatch;calculating a filter coefficient based upon the selected spectral mismatch; andfiltering the residual signal based upon the calculated filter coefficient.2. The method of wherein selecting one of the first spectral mismatch or the second spectral mismatch is based at least in part upon detecting a double-talk condition.3. The method of wherein selecting one of the first spectral mismatch or the second spectral mismatch is based at least in part upon a sound stage configuration selected for rendering the program content signal into an acoustic signal.4. The method of further comprising storing additional spectral mismatches in the memory to provide a plurality of stored spectral mismatches.5. The method of further comprising detecting a double talk condition and wherein selecting one of the first spectral mismatch or the second spectral mismatch includes selecting one of the plurality of stored spectral mismatches based on an amount of time for the double ...

Acoustic Echo Cancellation for Audio System with Bring Your Own Devices (BYOD)

A controller for the conference session receives at least one audio signal to generate a speaker signal. 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 removes acoustic echo from the array microphone signals, and estimates a relative location of the mobile device. The controller dynamically selects as audio output corresponding to the mobile device location either (a) the array microphone signal, or (b) the echo cancelled microphone signal from the mobile device. 1. A method comprising:receiving at least one audio signal at a controller associated with a conference session;receiving, at the controller, a plurality of array microphone signals associated with an array of microphones at corresponding known positions in a room of the conference session;generating a speaker signal from the at least one audio signal and the plurality of array microphone signals;correlating, at the controller, the speaker signal with network timing information to generate speaker timing information;transmitting the speaker signal with the speaker timing information via a network to a mobile device that is participating in the conference session to enable the mobile device to generate an echo cancelled microphone signal from a microphone of the mobile device;receiving, at the controller, the echo cancelled remote microphone signal;removing an acoustic echo from the plurality of array microphone signals to generate a plurality of echo cancelled array microphone signals;estimating a ...

MUTED COMPONENT DETECTION

One embodiment provides a method, comprising: transmitting, from a communication component, a signal down a communication channel; determining, using a processor, whether an echo associated with the signal is detected by the communication component; and providing, responsive to determining that the echo is not detected, a notification to a user that a mute control is enabled at another communication component along the communication channel. Other aspects are described and claimed. 1. A method , comprising:transmitting, from a communication component, a signal down a communication channel, wherein the signal is shaped to comprise a predetermined waveform;determining, using a processor, whether the signal is detected again by the communication component; andproviding, responsive to determining that the signal is not detected, a notification to a user that a mute control is enabled at another communication component along the communication channel, wherein the notification comprises an identification of the another communication component at which the mute control is enabled.2. The method of claim 1 , wherein the communication component is a component selected from the group consisting of a headset microphone claim 1 , a VoIP client claim 1 , and a VoIP server.3. (canceled)4. The method of claim 1 , wherein the predetermined waveform is able to circumvent at least one noise cancellation mechanism present in the communication channel.5. The method of claim 1 , wherein the notification comprises identification of the another communication component at which the mute control is enabled.6. The method of claim 5 , further comprising dynamically deactivating the mute control on the identified another communication component.7. The method of claim 6 , further comprising aligning the mute control with a global operating system.8. The method of claim 1 , wherein the providing the notification comprises providing an audible notification through a headset.9. The method of claim ...

SINGLE-CHANNEL, BINAURAL AND MULTI-CHANNEL DEREVERBERATION

A method is presented for estimating and suppressing reverberation from a digital reverberant signal. A method for changing a first reverberation estimation according to another reverberation estimation is further provided. A method for controlling the reverberation suppression rate is also presented. 118.-. (canceled)19. A method , in a multimedia signal processing system , that allows a user to adjust suppression of reverberation or noise from a digital signal that represents a sound signal , comprising:analyzing the digital signal by determining a plurality of time-frequency frames of the digital signal;deriving a first estimation of reverberation or noise from one of the frames in a first time instant;deriving a first suppression gain from the first estimation;selecting, by the user, a first exponent from a predetermined set of exponents;deriving a modified first suppression gain based on the first suppression gain raised to a power related to the first exponent;wherein the modified first suppression gain is applied to the one frame in the first time instant;deriving a second estimation of reverberation or noise from one of the frames in a second time instant;deriving a second suppression gain from the second estimation;selecting, by the user, a second exponent from a predetermined set of exponents;deriving a modified second suppression gain based on the second suppression gain raised to a power related to the second exponent;wherein the modified second suppression gain is applied to the one frame in the second time instant;outputting a signal that involves processing the frame in the first time instant utilizing the first modified suppression gain;and outputting a signal that involves processing the frame in the second time instant utilizing the second modified suppression gain;wherein the first exponent and the second exponent are different from one another, andwherein the second time instant is subsequent to the first time instant.20. The method of claim 19 , ...

SYSTEM AND METHOD FOR ADDRESSING ACOUSTIC SIGNAL REVERBERATION

A method, computer program product, and computer system for addressing acoustic signal reverberation is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and a reverberation audio signal. Embodiments may further include processing at least one of the first audio signal and the reverberation audio signal. Embodiments may also include limiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at least one of the first audio signal and the reverberation audio signal. 1. A computer-implemented method comprising:receiving, at one or more microphones, a first audio signal;receiving, at the one or more microphones, a reverberation audio signal;processing at least one of the first audio signal and the reverberation audio signal; andlimiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at least one of the first audio signal and the reverberation audio signal.2. The computer-implemented method of claim 1 , further comprising:receiving the one or more outputs from the model based reverberation equalizer at a postfilter.3. The computer-implemented method of claim 1 , further comprising:receiving a beamformer output at a postfilter.4. The computer-implemented method of claim 1 , further comprising:adjusting the model based reverberation equalizer to obtain a particular direct-to-noise ratio.5. The computer-implemented method of claim 4 , further comprising:measuring the direct-to-noise ratio using, at least in part, at least one temporal criteria.6. The computer-implemented method of claim 4 , further comprising:using the model based reverberation equalizer for the particular direct-to- noise ratio as a constraint ...

ARRAY MICROPHONE AND SOUND COLLECTION METHOD

A sound collection method and a microphone array estimate at least one sound source direction and form a plurality of sound collection beams in the estimated plurality of sound source direction, using sound collection signals of a plurality of microphones. The number of sound source directions estimated is smaller than the number of sound collection beams formed. 1. A microphone array comprising:a plurality of microphones;an estimator that estimates at least one sound source direction;a beam former that forms, using sound collection signals from the plurality of microphones, a plurality of sound collection beams larger in number than the number of the estimated at least one sound source direction but no more than a predetermined maximum number;a memory storing information indicating a beam direction of each of the plurality of sound collection beams;a determiner that determines whether or not a number of the plurality of sound collection beams reaches the predetermined maximum number; andan updater that updates at least one of the stored beam directions to the estimated at least one sound source direction upon the number of the plurality of sound collection beams being determined to reach the predetermined maximum number.2. The microphone array according to claim 1 , further comprising a mixing processor that mixes an audio signal corresponding to one sound collection beam claim 1 , among the plurality of sound collection beams claim 1 , by a gain according to volume of the one sound collection beam.3. The microphone array according to claim 1 , wherein the updater updates the direction of an earliest updated sound collection beam among the plurality of sound collection beams.4. The microphone array according to claim 1 , wherein the plurality of microphones are configured as a ceiling tile.5. The microphone array according to claim 1 , wherein the updater updates the direction of a sound collection beam with the direction thereof closest to the estimated at least ...

METHOD AND APPARATUS FOR ACOUSTIC ECHO CANCELLATION IN VOIP TERMINAL

A method of acoustic echo cancellation in the VoIP terminal using processing of the far-end signal with the digital adaptive filter in order to obtain the echo estimate that is subtracted from the microphone signal in which the far-end signal, before is is converted to the analog from and passed to the loudspeaker (), is marked by embedding an encoded digital signature obtained from the signature generator () and then detection of the digital signature is performed in the signal collected by the microphone () and converted to digital form, depending on the result of the digital signature detection, adaptation of the digital adaptive filter () is resumed or stopped. A circuit for acoustic echo cancellation in VoIP terminal contains the digital adaptive filter with the control block situated between the far-end speech signal path and the near-end speech signal path, and the double-talk detector () that comprises the signature generator () connected by the signature encoder (). 141479. A method of acoustic echo cancellation in a VoIP terminal in which a far-end speech signal is processed with a digital adaptive filter in order to obtain a echo estimate that is subtracted from a microphone signal and the result is used for adaptation of the digital adaptive filter , said adaptation is stopped while a double-talk is present characterized in that the far-end speech signal , before it is converted to an analog form and passed to a loudspeaker () , is marked by adding an encoded digital signature obtained from a signature generator () and then detection of the digital signature is performed in a signal collected by a microphone () and converted to a digital form and depending on detection of presence or absence of the digital signature in the signal , adaptation of a digital adaptive filter () is resumed or stopped , respectively.26547. A method as in characterized in that the digital signature is a sequence of bytes chosen so that the digital signature is suppressed by a ...

METHOD FOR IMPROVING ECHO CANCELLATION EFFECT AND SYSTEM THEREOF

A method for improving an echo cancellation effect and a system thereof are disclosed. The method comprises includes: performing a non-linear compensation for a non-linear response portion of an excitation signal using an NLC algorithm to obtain a compensated excitation signal; outputting the compensated excitation signal to an echo cancellation system; and performing echo cancellation for the compensated excitation signal by the echo cancellation system. According to the present disclosure, using the NLC algorithm, non-linear compensation is performed for the non-linear portion of the excitation signal, non-linear outputs generated due to non-linear characteristics of the system are pre-compensated when being input to the echo cancellation system, such that the echo signal output by the echo cancellation system is minimized and the echo cancellation effect is improved. 1. A method for improving echo cancellation effect , which comprising the following steps of:Step S1, performing a non-linear compensation for a non-linear response portion of an excitation signal using an NLC algorithm to obtain a compensated excitation signal;Step S2, outputting the compensated excitation signal to the an echo cancellation system; andStep S3, performing echo cancellation for the compensated excitation signal by the echo cancellation system.2. The method according to claim 1 , wherein the compensated excitation signal comprises a linear response portion and a background noise portion of the excitation signal.3. The method according to claim 2 , wherein the non-linear response portion of the excitation signal is converted to at least one of a linear response portion or a smaller non-linear response portion upon the non-linear compensation.4. An echo cancellation system claim 2 , comprising:a non-linear compensation module; andan echo cancellation device;wherein the non-linear compensation module is configured to perform a non-linear compensation for an excitation signal using an NLC ...

ECHO CANCELLATION FOR ULTRASOUND

A method includes accessing signal data descriptive of a transmission sequence and pre-determined values associated with the transmission sequence. The signal data and the pre-determined values may be stored in a memory. The method includes transmitting a signal from a speaker of an electronic device according to the transmission sequence. The method includes generating a frame based on one or more signals received at a microphone of the electronic device. The one or more signals include an echo signal associated with the transmitted signal. The method includes processing the frame using the pre-determined values to produce an output frame in which a contribution associated with the echo signal is reduced. 1. A method comprising:accessing signal data descriptive of a transmission sequence and pre-determined values associated with the transmission sequence, wherein the signal data and the pre-determined values are stored in a memory;transmitting a signal from a speaker of an electronic device according to the transmission sequence;generating a frame based on one or more signals received at a microphone of the electronic device, the one or more signals including an echo signal associated with the transmitted signal; andprocessing the frame using the pre-determined values to produce an output frame in which a contribution associated with the echo signal is reduced.2. The method of claim 1 , wherein the pre-determined values correspond to a fast Fourier transform (FFT) of the transmission sequence.3. The method of claim 2 , wherein processing the frame further comprises:performing an FFT on the frame to produce a first processed frame; andproviding the first processed frame to echo cancellation logic, wherein the echo cancellation logic is configured to process the first processed frame based on the pre-determined values.4. The method of claim 3 , wherein processing the first processed frame based on the pre-determined values comprises:multiplying the first processed ...

Dynamic Speaker Management with Echo Cancellation

A system for echo cancellation includes a dynamic speaker management (DSM) module, a current/voltage sensing amplifier, a sound pressure level (SPL) model module and an echo canceller. The example DSM module is receptive to a far-end signal and is operative to develop a modified far-end signal and a plurality of parameter outputs. The example current/voltage sensing amplifier is coupled to the modified far-end signal and develops an amplifier output, a voltage (V) parameter output, and a current (I) parameter output. The example sound pressure level (SPL) model module is coupled to the plurality of parameter outputs of the of the DSM module and is operative to develop a predicted SPL. The example echo canceller module is responsive to the predicted SPL and to a near-end signal and operative to develop an echo-canceled output signal. 1. A system for echo cancellation comprising:a dynamic speaker management (DSM) module receptive to a far-end signal and operative to develop a modified far-end signal and a plurality of parameter outputs;a current/voltage sensing amplifier coupled to the modified far-end signal and having an amplifier output, a voltage (V) parameter output, and a current (I) parameter output;a sound pressure level (SPL) model module coupled to the plurality of parameter outputs of the of the DSM module and operative to develop a predicted SPL; andan echo canceller module responsive to the predicted SPL and to a near-end signal and operative to develop an echo-canceled output signal.2. A system for echo cancellation as recited in further comprising a speaker coupled to the amplifier output of the current/voltage sensing amplifier and a microphone providing the near-end signal.3. A system for echo cancellation as recited in wherein the DSM module includes a DSM control block and a speaker modeler.4. A system for echo cancellation as recited in wherein the speaker modeler uses the V and I parameter outputs of the current/voltage sensing amplifier to ...

Detecting Double Talk in Acoustic Echo Cancellation Using Zero-Crossing Rate

A method for acoustic echo cancellation in a communication device is provided that includes receiving a first near-end audio signal in the communication device, wherein the first near-end audio signal comprises acoustic echo of a far-end audio signal reproduced by the communication device, and performing echo cancellation on the first near-end audio signal to generate a second near-end audio signal with at least some of the acoustic echo removed, wherein the echo cancellation is performed responsive to presence or absence of double-talk (DT), and wherein a zero-crossing rate (ZCR) is used to detect the presence or absence of DT.

Acoustic echo cancellation system

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

IN-CAR COMMUNICATION HOWLING PREVENTION

Howling or oscillation in an in-vehicle communications system is prevented when the speaker gain or microphone gain is detected as giving rise to a loop gain equal to or greater than one for a particular frequency or band of frequencies or when the rate of gain of a frequency or band of frequencies increases at a rate that indicating howling will occur. Howling is prevented and not just stopped or suppressed, by adjusting the gain of all frequencies prior to the howling actually starting. 1. A method of preventing an in-vehicle audio communication system from howling , the in-vehicle audio communications system comprising an audio feedback loop , the loop comprising an in-vehicle microphone that detects in-vehicle sounds , an audio amplifier that receives signals from the microphone , and , a loudspeaker coupled to the audio amplifier and which transduces audio frequency signals from the amplifier into sound that is output into the vehicle , the signals from the loudspeaker being acoustically coupled to the microphone through the vehicle's interior , the amplifier having a changeable gain factor by which the audio signals from the microphone are amplified or suppressed , then output to the loudspeaker , the method comprising:determining a frequency-domain model of acoustic characteristics of the vehicle's interior, for a plurality of different audio frequency bands in order to detect changes in acoustic coupling between the loudspeaker and microphone in substantially real time;detecting audio frequency acoustic signals in the vehicle by the microphone;amplifying the detected audio frequency signals;monitoring magnitudes of audio signals detected by the microphone, in predetermined frequency bands of audio signals, and which were output from the loudspeaker;determining a loop gain for each of the plurality of different frequency bands;decreasing the loop gain of the audio amplifier for all audio frequencies, responsive to a determination that the loop gain for an ...

SYSTEMS AND METHODS TO DISRUPT PHASE CANCELLATION EFFECTS WHEN USING HEADSET DEVICES

In applications where assisted listening headphones are worn inside of a theater, phase cancellation effects cause the headset wearer to perceive the audio as reduced in volume and distorted. These undesirable phase cancellation effects may be disrupted through preprocessing or real time processing of the headset audio track by summing acoustical noise with the original headset audio track and providing this altered audio track to the headset. The acoustical noise is modulated such that it is imperceptible to the headset wearer while at the same time disrupting undesirable phase cancellation effects, which would otherwise occur if the headset audio track was provided unaltered. Thus, the preprocessing of the headset audio preserves the integrity of the intended headset audio, as perceived by the headset wearer, in headsets worn in a theater environment. 1. A headset audio preprocessing method comprising:storing an audio track in memory of a mobile computing device;receiving in a microphone of the mobile computing device, contemporaneously played back audio;synchronizing playback of the stored audio track with the contemporaneously played back audio;acquiring acoustic noise; and,summing the acquired acoustic noise with the synchronized playback of the stored audio track in order reduce phase cancelation effects otherwise present in the synchronized playback of the stored audio track.2. The method of claim 1 , wherein the acoustic noise is pink noise.3. The method of claim 1 , wherein the acoustic noise is dithered noise.4. The method of claim 1 , wherein the contemporaneously played back audio is included as part of a motion picture played in a movie theater.5. The method of claim 1 , wherein the mobile computing device is a mobile phone.6. A computer program product for headset audio preprocessing claim 1 , the computer program product comprising:a non-transitory computer readable storage medium comprising a memory device having computer readable program code ...

FULL DUPLEX VOICE COMMUNICATION SYSTEM AND METHOD

A full duplex voice communication method constituted of: estimating an acoustic echo within a near-end signal; cancelling the estimated acoustic echo; detecting whether, or not, a change has occurred in a near-end acoustic echo path, the received near-end signal represents speech and the received far-end signal represents silence, wherein, responsive to the results thereof, the method is further constituted of: alternately attenuating frequency components of the echo cancelled near-end signal by a first frequency domain attenuation value and by a second greater frequency domain attenuation value; alternately attenuating a first function of the frequency component attenuated echo cancelled near-end signal by a first switchable attenuation value and by a second greater switchable attenuation value; and alternately attenuating a second function of the received far-end signal by a third switchable attenuation value and by a fourth greater switchable attenuation value. 1. A full duplex voice communication system comprising:a near-end input port arranged to receive a near-end signal;a far-end input port in communication with a far-end communication device and arranged to receive a far-end signal from said far-end communication device;an acoustic echo estimation functionality arranged, responsive to said received far-end signal, to estimate an acoustic echo within said received near-end signal;an acoustic echo cancellation functionality arranged to cancel said estimated acoustic echo from said received near-end signal;a frequency domain processing functionality;a first switchable attenuation functionality;a second switchable attenuation functionality;an echo path change detection functionality arranged to detect a change in a near-end acoustic echo path;a near-end speech detection functionality arranged to detect whether said received near-end signal represents speech;a far-end silence detection functionality arranged to detect whether said received far-end signal ...

Automatic volume control of a voice signal provided to a captioning communication service

Apparatuses and methods are disclosed for automatic volume control of an audio stream reproduced by a captioning communication service for use by a call assistant in generating a text transcription of a communication session between a hearing-impaired user and a far-end user. The automatic volume control automatically adjusts a volume of the audio stream reproduced by the captioning communication service responsive to a volume control command identifying which of the far-end voice signal and the near-end voice signal is active at a given time. The system further includes an echo modifier configured to add distortion to an echo portion of the far-end voice signal when generating the audio stream.

NONLINEAR ACOUSTIC ECHO CANCELLATION BASED ON TRANSDUCER IMPEDANCE

An acoustic echo cancellation (AEC) system within an audio playback system of an electronic device, such as a mobile phone, may calculate an estimation of an acoustic echo based on parameters describing the transducer reproducing the audio playback signals. Those parameters may include, for example, a resistance and/or inductance of the transducer and a current through and/or a voltage across the transducer. The acoustic echo cancellation system may predict, for example, a coil velocity of the transducer based on the transducer impedance. Then, an echo may be estimated using the predicted coil velocity. That estimated echo may be output to the transducer to cancel echo in the playback signal. Additionally, that estimated echo may be used to predict nonlinearities in the transducer output and an appropriate signal generated to cancel nonlinear behavior. 1. An apparatus , comprising:a current input node for receiving a current signal that is indicative of a measured current into an audio speaker;a voltage input node for receiving a voltage signal that is indicative of a voltage value measured across the audio speaker; and calculating an impedance of the audio speaker based, at least in part, on the current signal received at the current input node and the voltage signal received at the voltage input node by using an adaptive filter; and', 'generating an acoustic echo cancellation signal based, at least in part, on the calculated impedance of the audio speaker., 'a processing circuit coupled to the current input node and to the voltage input node and configured to perform the steps of2. The apparatus of claim 1 , wherein the processing circuit is configured to generate the acoustic echo cancellation signal to cancel a nonlinear response of the audio speaker.3. The apparatus of claim 1 , wherein the processing circuit is configured to generate the acoustic echo cancellation signal by performing the step of calculating a back electromagnetic force (bemf) of the audio ...

ECHO SUPPRESSOR USING PAST ECHO PATH CHARACTERISTICS FOR UPDATING

In an echo suppressor, a frequency bin component detector compares a far-end signal amplitude spectrum with a threshold value for each frequency bin to determine whether or not each frequency bin includes a frequency component. A frequency bin echo path characteristic estimator uses the far-end signal amplitude spectrum in the frequency bins determined to have a frequency component by the frequency bin component detector and the near-end input signal amplitude spectrum of corresponding frequency bins to estimate the echo path characteristics of the frequency bins. An estimated echo signal calculation-and-echo suppressor section calculates estimated echo signals based on the echo path characteristic in each frequency bin and the far-end signal amplitude spectrum to suppress the estimated echo signals from the near-end input signal amplitude spectrum. 1. An echo suppressor apparatus comprising:a far-end signal amplitude spectrum calculator converting an incoming far-end signal into a frequency domain and calculating a far-end signal amplitude spectrum of the far-end signal;a near-end input signal amplitude spectrum calculator converting a near-end input signal into a frequency domain and calculating a near-end input signal amplitude spectrum of the near-end input signal;a frequency bin component detector using the far-end signal amplitude spectrum for a frequency bin to determine whether or not there is a frequency component in the frequency bin;a frequency bin echo path characteristic estimator using a far-end signal amplitude spectrum of the frequency bin determined to have a frequency component by said frequency bin component detector and a near-end input signal amplitude spectrum of a corresponding frequency bin of the near-end input signal to estimate an echo path characteristic of the frequency bin;an estimated-echo signal calculator calculating an estimated echo signal from the echo path characteristic for the frequency bin estimated by said frequency bin echo ...

Subband Domain Echo Masking for Improved Duplexity of Spectral Domain Echo Suppressors

A method and system for improving a perceived duplexity of handsfree telephone applications is disclosed. An echo suppression circuit for a device comprising loudspeaker and microphone is described. A circuit attenuates a subband of a transmit signal, wherein the transmit signal is captured by the microphone and wherein the transmit signal comprises an echo of a far-end signal rendered by the loudspeaker and a near-end signal. The attenuation circuit further determines a subband far-end indicator of a voice activity in the far-end signal; determines a subband near-end indicator of a voice activity of the near-end signal; determines a subband masking weight; determines a subband attenuation for the transmit signal in the subband; and attenuates the subband of the transmit signal using the determined subband attenuation.

A METHOD AND APPARATUS FOR SUPPRESSION OF UNWANTED AUDIO SIGNALS

A device for removal of unwanted components in an audio signal, the device comprising a processor, coupled to memory, configured to receive reference and processed inputs into memory where the processed input is a result of a reduction process of unwanted components of the audio signal, estimate envelope values for processed and reference inputs at a plurality of time and frequency instances, for each time and frequency instance: compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input, apply a nonlinear process to said first gain to produce a second gain, compute an output gain as the ratio between second gain and first gain and, apply the output gain to processed input, thereby producing a filtered output with unwanted components suppressed. 1. A device for removal of unwanted components in an audio signal , the device comprising:a processor operatively coupled to a memory, said processor configured to:receive a reference input and a processed input into the memory, where said processed input is a result of at least a reduction process of unwanted components of the audio signal;estimate envelope values for the processed input and for the reference input at a plurality of time and frequency instances;for each said time and frequency instance:compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input;apply a nonlinear process to said first gain to produce a second gain;compute an output gain as the ratio between said second gain and said first gain; and,apply said output gain to the processed input; and,thereby producing a filtered output with unwanted components suppressed.2. The device of claim 1 , wherein said processed input and said reference input are in a time domain representation and said processor is configured to convert said processed input and said reference input ...

Managing Telephony and Entertainment Audio in a Vehicle Audio Platform

A signal input module receives at least one of an entertainment audio signal and a telephony audio signal from vehicle sound circuitry. A level control module executes gain level control logic to balance the entertainment audio signal and a telephony audio signal according to a ratio. A gain control signal is applied to at least one of the entertainment audio signal and the telephony audio signal. A routing module mixes the entertainment audio signal and the telephony audio signal and routes the mixed signal to an output channel associated with a speaker. 1. An apparatus comprising:a signal input module configured to receive an entertainment audio signal and a telephony audio signal from vehicle sound circuitry;a level control module configured to execute gain level control logic to balance the entertainment audio signal and the telephony audio signal relative to each other according to a predetermined ratio of gain levels, and to apply a gain control signal to at least one of the entertainment audio signal and the telephony audio signal; anda routing module configured to mix the entertainment audio signal and the telephony audio signal and route the mixed signal to an output channel associated with a speaker.2. The apparatus of claim 1 , wherein the predetermined ratio of gain levels is set to satisfy a Speech Transmission Index (STI) parameter claim 1 , so as to make the presentation of the entertainment audio signal and the telephony audio signal acceptable.3. The apparatus of any of the foregoing claims claim 1 , wherein the level control module is configured to balance the entertainment audio signal and the telephony audio signal in such a manner as that both the telephony and entertainment audio are audible and intelligible.4. The apparatus of any of the foregoing claims claim 1 , adapted to be used in an automobile passenger compartment including multiple car seats and speakers claim 1 , wherein the level control module is configured to balance the ...

Correlation based filter adaptation

Example apparatus and methods concern performing stereo acoustic echo cancellation using a correlation based filter adaptation control approach and without using stereo de-correlation. An embodiment includes a stereo adaptive filter that produces an echo removed microphone signal from received audio signals. The embodiment includes a mono adaptive filter that produces an echo removed microphone signal from the received audio signals. A correlation detector determines a level of correlation between the received signals and provides a signal to an adaptive filter controller. The adaptive filter controller controls how the stereo adaptive filter and the mono adaptive filter adapt audio echo cancellation as a function of the correlation between the received signals. A signal selector may select for output the signal from either the stereo adaptive filter or the mono adaptive filter based, for example, on the power level of the signals.

EMERGENCY REPORT APPARATUS

In an emergency report apparatus, diagnostic data is modulated based on a preselected diagnostic modulation method by using a diagnostic carrier wave having a carrier wave frequency that is set within a range of frequencies detectable by a vehicle-mounted voice input instrument and is outside a voice band. A diagnostic electric signal is outputted to a voice output instrument. Demodulated data is generated by demodulating an input voice electric signal being a voice electric signal representing a voice detected by the voice input instrument based on a preselected diagnostic demodulation method. It is determines whether the demodulated data includes data that matches the diagnostic data. 1. An emergency report apparatus that is mounted in a vehicle to establish wireless data communication in an event of an emergency in the vehicle and enable an occupant of the vehicle to converse with an emergency report center , the vehicle including a voice input instrument and a voice output instrument ,the emergency report apparatus comprising: modulates preselected diagnostic data based on a preselected diagnostic modulation method by using a diagnostic carrier wave having a preselected carrier wave frequency that is within a range of frequencies detectable by the voice input instrument and is outside a voice band, and', 'outputs a diagnostic electric signal to the voice output instrument, the diagnostic electric signal being obtained by modulating the diagnostic data;, 'a modulation section that'}a demodulator that generates demodulated data by demodulating an input voice electric signal being a voice electric signal representing a voice detected by the voice input instrument based on a preselected diagnostic demodulation method; anda diagnostic data determination section that determines whether the demodulated data generated by the demodulator includes data that matches the diagnostic data.2. The emergency report apparatus according to claim 1 , further comprising:a plurality ...

SPECTRAL BLENDING WITH INTERIOR MICROPHONE

A headphone can include plurality of exterior microphones, that generates corresponding exterior microphone signals, an accelerometer that generates an accelerometer signal; and an interior microphone, not directly exposed to the environment, that generates an interior microphone signal. A processor of the headphone can be configured to generate an audio signal containing voice of a user, based on a) the accelerometer signal, b) the interior microphone signal, and c) the plurality of exterior microphone signals. 1. A headphone , comprising:a plurality of exterior microphones, directly exposed to an environment of the headphone, that generates corresponding exterior microphone signals;an accelerometer that generates an accelerometer signal;an interior microphone, not directly exposed to the environment, that generates an interior microphone signal; anda processor, configured togenerate an audio signal containing voice of a user, based on a) the accelerometer signal, b) the interior microphone signal, and c) the plurality of exterior microphone signals.2. The headphone of claim 1 , wherein the audio signal has a) a first frequency band generated based on the accelerometer signal claim 1 , b) a second frequency band generated based on the interior microphone signal claim 1 , and c) a third frequency band generated based on the exterior microphone signals claim 1 , the second frequency band having a higher frequency than the first frequency band claim 1 , and the third frequency band having a higher frequency than the second frequency band.3. The headphone of claim 1 , wherein generating the audio signal includes beamforming at least two of the exterior microphone signals to form an exterior beamformed signal used to generate the audio signal.4. The headphone of claim 1 , wherein the interior microphone is echo cancelled.5. The headphone of claim 1 , wherein the headphone is at least partially worn in the user's ear canal.6. The headphone of claim 1 , wherein the ...

Detection of Acoustic Echo Cancellation

An echo cancellation detector for controlling an acoustic echo canceller that is configured to cancel an echo of a far-end signal in a near-end signal in a telephony system, the echo cancellation detector comprising a comparison generator configured to compare the far-end signal with the near-end signal, a decision unit configured to make a determination about a first acoustic echo canceller based on that comparison and a controller configured to control an operation of a second acoustic echo canceller in dependence on the determination. 1. An echo cancellation detector for controlling an acoustic echo canceller that is configured to cancel an echo of a far-end signal in a near-end signal in a telephony system , the echo cancellation detector comprising:a comparison generator configured to compare the far-end signal with the near-end signal;a decision unit configured to make a determination about a first acoustic echo canceller based on a result of comparison by the comparison generator; anda controller configured to control an operation of a second acoustic echo canceller in dependence on the determination by the decision unit.2. An echo cancellation detector as claimed in claim 1 , wherein the decision unit is further configured to make a determination as to whether the first acoustic echo canceller is present in said telephony system or not.3. An echo cancellation detector as claimed in claim 2 , wherein the controller is further configured to:control the second acoustic echo canceller to be in a state in which it is not operating in response to a determination that the first acoustic echo canceller is present; andcontrol the second acoustic echo canceller to be in a state in which it is operating in response to a determination that the first acoustic echo canceller is not present.4. An echo cancellation detector as claimed in claim 2 , wherein the controller comprises:a monitoring unit configured to monitor whether the first acoustic echo canceller is ...

Sound Signal Processing Device and Sound Signal Processing Method

A sound signal processing device includes: a microphone terminal to which a sound signal derived from sound received by a microphone is input; a loudspeaker terminal from which a sound signal directed to a loudspeaker is output; a first input terminal to which a sound signal from another proximal-end device is input; a first output terminal from which a sound signal directed to the other device is output; a distal-end input terminal to which a distal-end sound signal is input via a network; a distal-end output terminal from which a sound signal directed to the network is output; and at least one processor configured to execute stored instructions to establish at least one signal path from at least one of the microphone terminal, the first input terminal, or the distal-end input terminal, to at least one of the loudspeaker terminal, the first output terminal, or the distal-end output terminal. 1. A sound signal processing device comprising:a microphone terminal to which a sound signal derived from sound received by a microphone is input;a loudspeaker terminal from which a sound signal directed to a loudspeaker is output;a first input terminal to which a sound signal from another device at a proximal-end is input;a first output terminal from which a sound signal directed to the other device is output;a distal-end input terminal to which a distal-end sound signal is input via a network;a distal-end output terminal from which a sound signal directed to the network is output; andat least one processor configured to execute stored instructions to establish at least one signal path from at least one of the microphone terminal, the first input terminal, or the distal-end input terminal, to at least one of the loudspeaker terminal, the first output terminal, or the distal-end output terminal.2. The sound signal processing device according to claim 1 ,wherein the at least one processor is further configured to detect whether the subject device is connected to the network, ...

Acoustic echo path change detection apparatus and method

An acoustic echo path change detection apparatus constituted of: a time domain path change detection functionality arranged to: detect a change in a near-end acoustic echo path responsive to a time domain analysis of a near-end signal and a signal output by an acoustic echo canceller; and output an indication of the detected change, a frequency domain path change detection functionality arranged to: detect a change in the near-end acoustic echo path responsive to a frequency domain analysis of a far-end signal and the signal output by the acoustic echo canceller; and output an indication of the detected change, and a combination path change detection functionality arranged to: determine a first function of the output indication of the time domain path change detection functionality and the output indication of the frequency domain path change detection functionality; and output the determined first function.

ECHO SUPPRESSION

Method, user device and computer program product for suppressing echo. An audio signal is output from a speaker. A microphone receives an audio signal, wherein the received audio signal includes an echo resulting from the outputted audio signal. A Finite Impulse Response filter estimate ĥ(n) is dynamically adapted in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(n) related to the echo in the received audio signal. The filter estimate ĥ(n) is used in an estimate of the echo power in the received audio signal, and the estimated echo power is used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal. 1. A method of suppressing echo , the method comprising:outputting an audio signal;receiving an audio signal, wherein the received audio signal includes an echo resulting from said outputted audio signal;dynamically adapting a Finite Impulse Response filter estimate ĥ(n) in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(n) of the echo in the received audio signal;using the filter estimate ĥ(n) in an estimate of the echo power in the received audio signal; andusing the estimated echo power to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal.2. The method of wherein said echo suppression is applied to the received audio signal without a prior step of applying echo cancellation to the received audio signal.3. The method of wherein the outputted audio signal is used in the estimate of the echo power in the received audio signal.4. The method of further comprising applying echo cancellation to the received audio signal claim 1 , wherein said echo suppression is applied after the echo cancellation in the processing of the received audio signal.5. The method of wherein said echo suppression which is applied to the received audio signal is ...

ACOUSTIC ECHO CANCELLATION WITH ROOM CHANGE DETECTION

Acoustic echo cancelling includes receiving a source signal and a sink signal; providing a first error signal representative of an echo-free residual signal based on a first set of coefficients based on the source signal and the sink signal, the first error signal forming an output signal of the controller; providing a second error signal based on a second set of coefficients based on the source signal and the sink signal; detecting a room change if the evaluated first second error signal is greater than a sum or product of the evaluated second first error signal and a first threshold; copying one of sets of reference coefficients stored in a memory to the second acoustic echo canceller; and copying the first set of coefficients from the first acoustic echo canceller as a set of reference coefficients into at least one of the second acoustic echo canceller and the memory. 1. An acoustic echo cancelling controller configured to receive a source signal representative of sound broadcast at a first position in a room and a sink signal representative of sound picked up at a second position in the room , the sound picked up at the second position being transferred from the first position according to a transfer characteristic , the controller comprising:a first acoustic echo canceller configured to receive the source signal and the sink signal, and to model the transfer function in an adaptive manner based on a first set of coefficients, the first acoustic echo canceller being further configured to provide a first error signal representative of an echo-free residual signal, the first error signal forming an output signal of the controller;a second acoustic echo canceller configured to receive the source signal and the sink signal, and to model the transfer function in a non-adaptive manner based on a second set of coefficients, the second acoustic echo canceller being further configured to provide a second error signal;a memory operatively coupled with the first acoustic ...

Audio Signal Processing

An estimated system gain spectrum of an acoustic system is generated, and updated in real-time to respond to changes in the acoustic system. Peak gains in the estimated system gain spectrum are tracked as the estimated system gain spectrum is updated. Based on the tracking, at least one frequency at which the estimated system gain spectrum is currently exhibiting a peak gain is identified. Based on the identification of the at least one frequency, an audio equalizer is controlled to apply, to a first speech containing signal to be played out via an audio output device of the audio device and/or to a second speech containing signal received via an audio input device of the audio device, an equalization filter to reduce the level of that signal at the identified frequency. The equalization filter is applied continuously throughout intervals of both speech activity and speech inactivity in that signal.

RECEIVE-PATH SIGNAL GAIN OPERATIONS

Operations related to performing gain operations with respect to a receive-path signal of a first device may be performed. The operations may include obtaining the receive-path signal, which includes an echo speech signal and a receive speech signal originating at a second device. In addition, the operations may include identifying a portion of the receive-path signal that includes, at a particular time, a first frequency component that corresponds to the echo speech signal and a second frequency component that corresponds to the receive speech signal in which the first frequency component is different from the second frequency component. Moreover, the operations may include attenuating the first frequency component of the portion while avoiding attenuating the second frequency component of the portion based on the first frequency component corresponding to the echo speech signal and the second frequency component corresponding to the receive speech signal. 1. A method comprising:obtaining a transmit speech signal generated by a first device during a communication session between the first device and a second device;obtaining a receive-path signal that includes an echo speech signal of the transmit speech signal and a receive speech signal originating at the second device during the communication session;identifying a portion of the receive-path signal that includes, at a particular time, a first frequency component that corresponds to the echo speech signal and a second frequency component that corresponds to the receive speech signal in which the first frequency component is different from the second frequency component; andattenuating the first frequency component of the portion while avoiding attenuating the second frequency component of the portion based on the first frequency component corresponding to the echo speech signal and the second frequency component corresponding to the receive speech signal.2. The method of claim 1 , further comprising amplifying ...

Audio Device, Sound Processing Method, Sound Processing Program, Sound Output Method, And Sound Output Program

There is provided an audio device including a control section configured to cause an audio signal to be output, the audio signal including a sound signal obtained through playback of content and a sound signal received from a communication partner device, and a sound processing section configured to generate an elimination signal obtained by eliminating a given sound signal from a microphone detection signal, which is the audio signal that is propagated and detected by a microphone. The control section causes the communication partner device to transmit the elimination signal.

Deep neural network-based method and apparatus for combining noise and echo removal

Disclosed is a deep neural network-based method and apparatus for combining noise and echo removal. The deep neural network-based method for combining noise and echo removal according to one embodiment of the present invention may comprise the steps of extracting a feature vector from an audio signal that includes noise and echo; and acquiring a final audio signal from which both noise and echo have been removed, by using a combined nose and echo removal gain estimated by means of the feature vector and deep neural network DNN.

DEVICE FOR ASSISTING TWO-WAY CONVERSATION AND METHOD FOR ASSISTING TWO-WAY CONVERSATION

A two-way conversation assisting device includes a first microphone that enters a first voice, a first loudspeaker that outputs the first voice, a second microphone that enters a second voice, a second loudspeaker that outputs the second voice, and a first echo and crosstalk canceller. The first echo and crosstalk canceller estimates and calculates, using an input signal into the second loudspeaker, a first interference signal indicative of degrees of a first echo caused when the second voice output from the second loudspeaker enters into the first microphone and first crosstalk caused when the second voice enters into the first microphone, and removes the calculated first interference signal from an output signal of the first microphone. 1. A two-way conversation assisting device for amplifying and assisting a two-way conversation ,the two-way conversation assisting device comprising:a first microphone that enters a first voice;a first loudspeaker that outputs the first voice;a second microphone that enters a second voice;a second loudspeaker that outputs the second voice; anda first echo and crosstalk canceller that estimates and calculates, using an input signal into the second loudspeaker, a first interference signal indicative of degrees of a first echo caused when the second voice output from the second loudspeaker enters into the first microphone and first crosstalk caused when the second voice enters into the first microphone, and that removes the calculated first interference signal from an output signal of the first microphone.2. The two-way conversation assisting device according to claim 1 , further comprisinga first acoustic feedback canceller that estimates and calculates a first acoustic feedback signal indicative of a degree of first acoustic feedback caused when the second voice output from the second loudspeaker enters into the second microphone, and that removes the calculated first acoustic feedback signal from an output signal of the second ...

Acoustic Echo Suppression

A controller for an echo suppressor configured to suppress a residual echo of a far-end signal included in a primary error signal, the controller adapted for operation with a primary adaptive filter configured to form a primary echo estimate of the far-end signal included in a microphone signal and an echo canceller configured to cancel that primary echo estimate from the microphone signal so as to form the primary error signal, the controller comprising: a secondary adaptive filter configured to form a secondary echo estimate of the far-end signal comprised in the microphone signal; and control logic operable in at least two modes selected in dependence on a convergence state of the primary adaptive filter, the control logic being configured to control activation of the echo suppressor in dependence one or more transient or steady state decision parameters. 1. A controller for an echo suppressor configured to suppress a residual echo of a far-end signal included in a primary error signal , the controller adapted for operation with an adaptive echo estimation filter configured to form a primary echo estimate of the far-end signal included in a microphone signal and an echo canceller configured to cancel that primary echo estimate from the microphone signal so as to form the primary error signal , the controller comprising:decision logic operable in at least two modes selected in dependence on a convergence state of the adaptive echo estimation filter, the decision logic comprising steady-state decision logic and transient state decision logic and being configured to:in its first mode selected when the adaptive echo estimation filter is in a non-converged state, use the transient-state decision logic to control activation of the echo suppressor according to a transient-state decision path in dependence on a state of the microphone signal; andin its second mode selected when the adaptive echo estimation filter is in a converged state, use the steady-state decision ...

Audio Group Identification For Conferencing

Systems and methods are disclosed for audio group identification for conferencing. For example, methods may include joining a conference call using a network interface; accessing an audio signal that has been captured using a microphone; detecting a control signal in the audio signal; and, responsive to detection of the control signal, invoking modification of an audio path of the conference call.

Automatic Tuning of a Gain Controller

A gain control system for applying gain to a far-end signal, the system comprising: a signal identifier configured to detect an echo of the far-end signal in a microphone signal; and a path estimator configured to estimate a characteristic of an echo path of the detected echo, wherein: the signal identifier is further configured to detect a near-end signal from the microphone signal; and in response to detecting the near-end signal, the gain control system is configured to adjust the gain applied to the far-end signal in dependence on the estimated characteristic of the echo path.

ECHO CANCELLATION METHOD AND TERMINAL, COMPUTER STORAGE MEDIUM

Described is an echo cancellation method including buffering the at least one frame of a first voice signal, the at least one frame of the first voice signal including an echo signal, when the at least one frame of signal is preprocessed, separately obtaining, in a first timing period, the at least one frame of the first voice signal from a buffer and at least one frame of a reference signal matching the at least one frame of first voice signal from a reference signal queue, so that an estimated latency value, between each frame of first voice signal and a corresponding reference signal, remains within a preset range, performing echo cancellation processing on the echo signal in the at least one frame of first voice signal using the at least one frame of reference signal, to obtain a second voice signal, and outputting the second voice signal. 1. An echo cancellation method , comprising:collecting at least one frame of a first voice signal;buffering the at least one frame of the first voice signal, the at least one frame of the first voice signal including an echo signal;when the at least one frame of the first voice signal is preprocessed, separately obtaining, in a first timing period, the at least one frame of the first voice signal from a buffer and at least one frame of a reference signal matching the at least one frame of the first voice signal from a reference signal queue, so that an estimated latency value, between each frame of the first voice signal and a corresponding reference signal, remains within a preset range;performing, by processing circuitry of an information processing apparatus, echo cancellation processing on the echo signal in the at least one frame of the first voice signal using the at least one frame of the reference signal, to generate at least one frame of a second voice signal; andoutputting the at least one frame of the second voice signal.2. The method according to claim 1 , wherein before the at least one frame of the first voice ...

TERMINAL, AND OPERATION METHOD FOR TERMINAL

An operation method of a terminal may include: executing a voice call service between the terminal and at least one terminal; determining at least one short-range terminal existing within a preset range from among the at least one terminal, based on location information of the terminal; detecting a howling frequency band in which howling occurs between the terminal and the at least one short-range terminal from among a plurality of frequency bands in which the voice call service is performed, based on information of the plurality of frequency bands; and removing the howling by adjusting a gain of the howling frequency band. 1. An operation method of a terminal , the operation method comprising:executing a voice call service between the terminal and at least one terminal;determining at least one short-range terminal existing within a preset range from among the at least one terminal, based on location information of the terminal;detecting a howling frequency band in which howling occurs between the terminal and the at least one short-range terminal from among a plurality of frequency bands in which the voice call service is performed, based on information of the plurality of frequency bands; andremoving the howling by adjusting a gain of the howling frequency band.2. The operation method of claim 1 , wherein the information of the plurality of frequency bands comprises at least one from among a peak value and an energy value of each of the plurality of frequency bands.3. The operation method of claim 2 , wherein the detecting of the howling frequency band in which the howling occurs between the terminal and the at least one short-range terminal from among the plurality of frequency bands in which the voice call service is performed claim 2 , based on the information of the plurality of frequency bands comprises:obtaining an energy change rate of each of the plurality of frequency bands based on input voice information of the at least one short-range terminal and ...

Comfort Noise Generation

A system for generating comfort noise for a stream of frames carrying an audio signal includes frame characterizing logic configured to generate a set of filter parameters characterising the frequency content of a frame; an analysis filter adapted using the filter parameters and configured to filter the frame so as to generate residual samples; an analysis controller configured to cause the residual samples to be stored in a store responsive to receiving an indication that the frame does not comprise speech; and a synthesis controller operable to select stored residual samples from the store and cause a synthesis filter, inverse to the analysis filter and adapted using filter parameters generated by the frame characterizing logic for one or more frames not comprising speech, to filter the selected residual samples so as to generate a frame of comfort noise.

Pure Delay Estimation

A system for estimating delay between a far-end signal and an echo of the far-end signal in a microphone signal, the system comprising: a buffer configured to store a time-series of far-end samples representing the far-end signal; a first delay estimator configured to form a first estimate of the delay in respect of a speech frame representing speech in the microphone signal; a second delay estimator configured to form a second estimate of the delay for the speech frame by operating a first set of one or more filters on far-end samples selected from the buffer in dependence on an operating delay; a second set of one or more filters for operation on far-end samples; and a controller configured to, in response to a determination that the first estimate of the delay for the speech frame differs from the operating delay for a previous frame by at least a predefined length of time, cause the one or more filters of the second set to operate on far-end samples selected from the buffer according to the first estimate of the delay and, if in respect of one or more speech frames a measure of convergence of the second set of filters exceeds a measure of convergence of the first set of filters by at least a first predefined threshold, update the operating delay using the first estimate of the delay. 1. A system for estimating delay between a far-end signal and an echo of the far-end signal in a microphone signal , the system comprising:a buffer configured to store a time-series of far-end samples representing the far-end signal;a first delay estimator configured to form a first estimate of the delay in respect of a speech frame representing speech in the microphone signal;a second delay estimator configured to form a second estimate of the delay for the speech frame by operating a first set of one or more filters on far-end samples selected from the buffer in dependence on an operating delay, wherein the second estimate of the delay is used by the system as an output estimate ...

Mobile Phone and Method for Processing Call Signal Thereof

The present invention discloses a method for processing a call signal of a mobile phone, the method includes detecting a current call state of the mobile phone; acquiring an environmental noise signal when the mobile phone enters an earphone call state; and performing a noise reduction processing to a call signal according to the environmental noise signal for acquiring a call signal with the noise reduction processing. The present invention further discloses a mobile phone. The present invention is capable of ensuring that a user is not affected by environmental noises for guaranteeing a better quality of a call when the user makes the call with an earphone. 1. A method for processing a call signal of a mobile phone , comprising:a. detecting a current call state of the mobile phone for judging whether the mobile phone enters an earphone call state;b. utilizing a main microphone for acquiring a first environmental noise signal when the mobile phone enters the earphone call state, wherein the step b further comprises: utilizing a sub-microphone for acquiring a second environmental noise signal when the mobile phone enters the earphone call state; andc. performing a noise reduction processing to a call signal acquired by an earphone microphone according to the first environmental noise signal for acquiring a call signal with the noise reduction processing, {'br': None, 'i': S=L', '−M, 'sub': 1', '1, '.'}, 'wherein the noise reduction processing is performed according to the following equation in the step c{'sub': 1', '1, 'where S is the call signal with the noise reduction, Lis the call signal acquired by the earphone microphone, and Mis the first environmental noise signal acquired by the main microphone.'}2. The method of claim 1 , wherein the step c further comprises:performing the noise reduction processing to the call signal acquired by the earphone microphone according to the first environmental noise signal and the second environmental noise signal for ...

Acoustic echo cancelling system and method

An audio system includes a loudspeaker, a first microphone, an echo canceller, and a second microphone within the loudspeaker enclosure coupled to the loudspeaker. The first microphone provides an environmental acoustic signal to the echo canceller. The second microphone can be a high acoustic overload microphone and be placed in a back cavity of the speaker enclosure. A speaker signal is used to drive the loudspeaker, which may produce non-linear distortions in the acoustic output. The second microphone senses a signal that includes both the linear and non-linear distortions. This sensed signal is used to remove both the linear and the non-linear distortions from the environmental acoustic signal picked up from the first microphone and processed by the echo canceller.

METHOD FOR PERFORMING FUNCTION AND ELECTRONIC DEVICE SUPPORTING THE SAME

An electronic device and a method for performing a function in the electronic device are provided. The method includes recognizing that the electronic device is at a first state, the first state including that the electronic device is in an overturned state or that a predetermined amount of a display is covered; detecting a specific event in the first state; outputting a color notification associated with the detected specific event through at least a portion of the display, the portion of the display wrapping around a side edge of the electronic device; receiving a specific user input corresponding to the color notification, while outputting the color notification; and performing a function associated with the received specific user input. 1. A method for performing a function in an electronic device including a display , the method comprising:recognizing that the electronic device is at a first state, the first state including that the electronic device is in an overturned state or that a predetermined amount of the display is covered;detecting a specific event in the first state;outputting a color notification associated with the detected specific event through at least a portion of the display, the portion of the display wrapping around a side edge of the electronic device;receiving a specific user input corresponding to the color notification, while outputting the color notification; andperforming a function associated with the received specific user input.2. The method of claim 1 , wherein recognizing the first state comprises recognizing that the electronic device is in the overturned state using a sensor module.3. The method of claim 2 , wherein the sensor module includes at least one of a proximity sensor claim 2 , an acceleration sensor claim 2 , and a geomagnetic sensor.4. The method of claim 1 , wherein recognizing the first state comprises determining that the predetermined amount of the display is covered using a sensor module.5. The method of claim 1 ...

Sound Modification for Close Proximity Shared Communications Path Devices

Systems and methods are provided for attenuating a communications channel on one or more speakerphones which are located in close physical proximity to one another and which are being used on the same call. 1. A system for attenuating a communication channel , the system comprising:a customer premises equipment (“CPE”);said CPE configured to electrically communicate with a plurality of speakerphones;wherein said CPE is configured to identify at least two of said plurality of speakerphones as neighbors;said CPE configured to communicate with a remote terminal via a communication channel;said CPE being configured to apply audio attenuation to the at least two of said plurality of speakerphones when the at least two of said plurality of speakerphones are simultaneously connected to the communication channel.2. The system according to wherein said audio attenuation is fixed.3. The system according to wherein said audio attenuation is variable.4. The system according to wherein said audio attenuation is the same for each of said at least two of said plurality of speakerphones.5. The system according to wherein said audio attenuation is different for each of said at least two of said plurality of speakerphones.6. The system according to wherein said customer premises equipment is a private branch exchange (PBX) and said PBX attenuates said communication channel.7. The system according to wherein at least one of said at least two speakerphones is configured to attenuate said communication channel.8. The system according to wherein said attenuation includes muting said communication channel.9. The system according to wherein each of said speakerphones includes a microphone and said audio attenuation includes attenuating said microphone of at least one of said at least two of said plurality of speakerphones.10. The system according to wherein each of said speakerphones includes a speaker and said audio attenuation includes attenuating said speaker of at least one of said at ...

Conferencing Apparatus that combines a Beamforming Microphone Array with an Acoustic Echo Canceller

The present disclosure discloses a conferencing apparatus for a conference between a local end and a far end that combines a beamforming microphone array with an acoustic echo canceller. The apparatus includes a beamforming microphone array that further comprises a plurality of microphones wherein each microphone is configured to sense acoustic waves and the plurality of microphones are oriented to develop a corresponding plurality of microphone signals. The apparatus further includes a processor, memory, and storage where the processor is configured to execute program instructions. The processor performs a beamforming operation to create a plurality of combined signals. In addition, the processor performs an acoustic echo cancellation operation to generate a plurality of combined echo cancelled signals. Further, the processor performs a direction of arrival determination; and, selects, in response to the direction of arrival determination, one of the combined echo cancelled signals for transmission to the far end. 1. A conferencing apparatus for a conference between a local end and a far end that combines a beamforming microphone array with an acoustic echo canceller , comprising:a beamforming microphone array that further comprises a plurality of microphones wherein each microphone is configured to sense acoustic waves and said plurality of microphones are oriented to develop a corresponding plurality of microphone signals; perform a beamforming operation with a beamformer to combine the plurality of microphone signals from said beamforming microphone array into a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals, each of the plurality of combined signals corresponding to a different fixed beam;', 'perform an acoustic echo cancellation operation with an acoustic echo canceller on the plurality of combined signals to generate a plurality of combined echo cancelled signals;', 'perform a ...

ACOUSTIC ECHO CANCELLATION BASED SUB BAND DOMAIN ACTIVE SPEAKER DETECTION FOR AUDIO AND VIDEO CONFERENCING APPLICATIONS

Systems, methods, and devices are disclosed for detecting an active speaker in a two-way conference. Real time audio in one or more sub band domains are analyzed according to an echo cancellor model. Based on the analyzed real time audio, one or more audio metrics are determined from output from an acoustic echo cancellation linear filter. The one or more audio metrics are weighted based on a priority, and a speaker status is determined based on the weighted one or more audio metrics being analyzed according to an active speaker detection model. For an active speaker status, one or more residual echo or noise is removed from the real time audio based on the one or more audio metrics. 1. A method for detecting an active speaker in a two-way conference comprising:analyzing real time audio in one or more sub band domains according to an echo cancellor model;determining, based on the analyzed real time audio, one or more audio metrics determined from output from an acoustic echo cancellation linear filter;weighting the one or more audio metrics based on a priority;determining a speaker status based on the weighted one or more audio metrics being analyzed according to an active speaker detection model, wherein the determination is based at least in part on a hysteresis model that stabilizes the speaker status over a period of time; andfor an active speaker status, removing one or more of residual echo or noise from the real time audio based on the one or more audio metrics.2. The method of claim 1 , wherein the one or more of residual echo or noise from the real time audio is removed nonlinearly based on the one or more audio metrics.3. The method of claim 1 , wherein the one or more audio metrics determines the speed of the hysteresis model.4. The method of claim 1 , wherein one or more weights used in weighting the one or more audio metrics is generated dynamically from a machine learned model.5. The method of claim 1 , wherein the one or more audio metrics include ...

SOUND PROCESSING METHOD, REMOTE CONVERSATION METHOD, SOUND PROCESSING DEVICE, REMOTE CONVERSATION DEVICE, HEADSET, AND REMOTE CONVERSATION SYSTEM

A sound processing method performs near-end side sound collection processing to collect a sound on a near-end side and generate a sound collection signal, near-end side filter processing to adjust the sound collection signal using transfer characteristics of a far-end side, and near-end side sound emission processing to emit the sound collection signal that has been adjusted, from a speaker on the near-end side. 1. A sound processing method comprising:performing near-end side sound collection processing to collect a sound on a near-end side and generate a sound collection signal;performing near-end side filter processing to adjust the sound collection signal using transfer characteristics of a far-end side; andperforming near-end side sound emission processing to emit the adjusted sound collection signal, from a speaker on the near-end side.2. The sound processing method according to claim 1 , further comprising performing transfer characteristics obtaining processing to obtain the transfer characteristics of the far-end side.3. The sound processing method according to claim 2 ,wherein the transfer characteristics obtaining processing obtains the transfer characteristics sequentially, andwherein the near-end side filter processing adjusts the sound collection signal using the transfer characteristics obtained sequentially.4. A remote conversation method comprising the sound processing method according to claim 1 , wherein the near-end side sound emission processing emits an audio signal of the far-end side and the adjusted sound collection signal claim 1 , from the speaker on the near-end side.5. The remote conversation method according to claim 4 , wherein the near-end side filter processing reduces the sound collection signal based on the audio signal of the far-end side claim 4 , which is diffracted from the speaker on the near-end side to a microphone on the near-end side and outputs the reduced sound collection signal and a filter coefficient.6. A sound ...

CONFERENCING APPARATUS WITH AN AUTOMATICALLY ADAPTING BEAMFORMING MICROPHONE ARRAY

This disclosure describes a conferencing apparatus with an automatically adapting beamforming microphone array for teleconferencing applications. The conferencing apparatus includes a processor coupled to memory, storage, and a housing. The apparatus further includes a communication element, a plurality of microphones arranged in the housing and configured as a beamforming microphone array, and an orientation sensor. Additionally, the apparatus includes executing a series of steps including: (a) adapting a signal-processing characteristic of the beamforming microphone array responsive to the orientation signal, (b) beamforming to combine the plurality of microphone signals to a plurality of combined signals, (c) performing an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals, and (d) selecting one or more of the plurality of combined echo-canceled signals for transmission. 1. A conferencing apparatus with an automatically adapting beamforming microphone array for teleconferencing applications , comprising:a processor coupled to memory, and storage included in a housing;a communication element coupled to said processor for communicating with other devices and or communication networks;a plurality of microphones arranged in said housing and configured as a beamforming microphone array, said beamforming microphone array oriented to cover a plurality of direction vectors and develop a plurality of microphone signals;an orientation sensor that generates an orientation signal indicative of an orientation of with a predetermined spatial arrangement relative to said beamforming microphone array, said orientation sensor couples to said processor; adapting a signal-processing characteristic of said beamforming microphone array responsive to said orientation signal;', 'beamforming to combine said plurality of microphone signals to a plurality of combined signals;', 'performing an acoustic echo ...

AUDIO DEVICE, SOUND PROCESSING METHOD, SOUND PROCESSING PROGRAM, SOUND OUTPUT METHOD, AND SOUND OUTPUT PROGRAM

There is provided an audio device including a control section configured to cause an audio signal to be output, the audio signal including a sound signal obtained through playback of content and a sound signal received from a communication partner device, and a sound processing section configured to generate an elimination signal obtained by eliminating a given sound signal from a microphone detection signal, which is the audio signal that is propagated and detected by a microphone. The control section causes the communication partner device to transmit the elimination signal. 118.-. (canceled)19. An audio processing apparatus that outputs a first audio signal including audio content and a second audio signal received from a communication apparatus , comprising:sound processing circuitry that receives a microphone detection signal comprising the first audio signal and the second audio signal and generates an elimination signal that is used to remove a predetermined audio signal from the microphone detection signal to produce an echo-eliminated signal,a transmitter that transmits the echo-eliminated signal to the communication apparatus upon generation of the echo-eliminated signal by the sound processing circuitry, andan output for producing audible sound comprising the first audio signal and the second audio signal.20. The audio processing apparatus of claim 19 , wherein the sound processing circuitry generates the elimination signal based on an amount of delay that occurs between the output generating the audible sound and detection of the microphone detection signal.21. The audio processing apparatus of claim 19 , wherein transmission of the echo-eliminated signal to the communication apparatus stops upon the sound processing circuitry ending generation of the elimination signal.22. The audio processing apparatus of claim 19 , wherein generation of the elimination signal is interrupted upon receipt of a suspend request.23. The audio processing apparatus of claim ...

RETROFIT DIGITAL NETWORK SPEAKER SYSTEM

Described herein is a retrofit digital speaker system comprising two or more retrofitted speaker enclosures, each of the two or more retrofitted speaker enclosures (enclosures) comprising: at least one speaker; and an analog-and-digital interface adapted to receive digitally encoded audio signals, electrical power, and digital command signals, and wherein the digitally encoded audio signals, electrical power, and digital command signals are transmitted over existing two wire analog audio cables, and further wherein the two or more retrofitted speaker enclosures are wired in a daisy chain fashion, via the existing two wire analog audio cables. 1. A retrofit digital speaker system comprising two or more retrofitted speaker enclosures , each of the two or more retrofitted speaker enclosures (enclosures) comprising:at least one speaker; andan analog-and-digital interface adapted to receive digitally encoded audio signals, electrical power, and digital command signals, and wherein the digitally encoded audio signals, electrical power, and digital command signals are transmitted over existing two wire analog audio cables, and further wherein the two or more retrofitted speaker enclosures are wired in a daisy chain fashion, via the existing two wire analog audio cables.2. The retrofit digital speaker system according to claim 1 , further comprising:a relay in each of the two or more retrofitted speaker enclosures adapted to remain normally open on power-up such that the digitally encoded audio signals, electrical power, and digital command signals received by a first enclosure and relay are not transmitted to a second enclosure unless and until specifically commanded to do so.3. The retrofit digital speaker system according to claim 1 , further comprising:a digital signal processor (DSP) adapted to receive and process the digitally encoded audio signals and digital command signals; anda coder-decoder device adapted to receive and decode the digitally encoded audio signals ...

SPEAKERPHONE SYSTEM OR SPEAKERPHONE ACCESSORY WITH ON-CABLE MICROPHONE

A portable speakerphone having a housing, a receiving transducer, an electrical cable, a transmitting transducer, and a processor. The receiving transducer is affixed to the housing and is configured to receive a first electrical signal from a mobile device. The electrical cable is coupled to and extends from the housing. The transmitting transducer is affixed to the electrical cable, remote from the housing. Also, the transmitting transducer is configured to transmit a second electrical signal, and the second electrical signal is based in part on the first electrical signal. The processor is configured to suppress acoustic echo by modifying the second electrical signal. The processor is also configured to output the modified second electrical signal to the mobile device. A related method is also disclosed. 1. A portable speakerphone comprising:a housing;a receiving transducer affixed to the housing and configured to receive a first electrical signal from a mobile device;an electrical cable coupled to and extending from the housing;a transmitting transducer affixed to the electrical cable, remote from the housing, and configured to transmit a second electrical signal, the second electrical signal being based in part on the first electrical signal and an analog voice signal; anda processor configured to suppress acoustic echo in the second electrical signal by modifying the second electrical signal, and the processor further configured to output the modified second electrical signal to the mobile device.2. The speakerphone of claim 1 , in which the receiving transducer is a loudspeaker claim 1 , and in which the transmitting transducer is a microphone.3. The speakerphone of claim 1 , in which the electrical cable is detachably attached to the housing.4. The speakerphone of claim 1 , in which the electrical cable has a connector at a distal end of the electrical cable claim 1 , opposite a proximal end of the electrical cable coupled to the housing.5. The speakerphone ...

SYSTEMS AND METHODS OF ECHO REDUCTION

Echo reduction. At least one example embodiment is a method including producing, by a loudspeaker, acoustic waves based on a far-microphone signal; receiving, at a local microphone, an echo based on the acoustic waves, and receiving acoustic waves generated locally, the receiving creates a local-microphone signal; producing an estimated-echo signal based on the far-microphone signal and a current step-size parameter; summing the local-microphone signal and the estimated echo signal to produce a resultant signal having reduced echo in relation to the local-microphone signal; and controlling the current step-size parameter. The controlling current step size may include: calculating a convergence value based on a cross-correlation of the local-microphone signal and the resultant signal; and updating the current step-size parameter based on the convergence value. 1. A method of echo cancellation , the method comprising:producing, by a loudspeaker, acoustic waves based on a far-microphone signal;receiving, at a local microphone, an echo based on the acoustic waves, and receiving acoustic waves generated locally, the receiving creates a local-microphone signal;producing an estimated-echo signal based on the far-microphone signal and a step-size parameter;summing the local-microphone signal and the estimated-echo signal to produce a resultant signal having reduced echo in relation to the local-microphone signal; and calculating a convergence value based on a cross-correlation of the local-microphone signal and the resultant signal; and', 'updating the step-size parameter based on the convergence value., 'controlling the step-size parameter by2. The method of wherein calculating the convergence value further comprises:converting the local-microphone signal into the frequency domain to create a local-microphone spectrum;converting the resultant signal into the frequency domain to create a resultant spectrum;performing a cross-correlation of the local-microphone spectrum and ...

EFFICIENT REUTILIZATION OF ACOUSTIC ECHO CANCELER CHANNELS

Audio systems and methods are provided to reduce echo content in an audio signal. The systems and methods receive an audio signal and sound stage rendering parameter(s), and select a set of filter coefficients to filter the audio signal to provide an estimated echo signal. The estimated echo signal is subtracted from a microphone signal to provide an output signal with reduced echo content. The set of filter coefficients are selected based upon the sound stage rendering parameter(s) from among a plurality of stored sets of filter coefficients. 1. A method of reducing echo content of an audio signal , comprising:receiving an audio program content signal;receiving a plurality of microphone signals;identifying an audio system configuration;selecting a set of filter coefficients from among a plurality of stored sets of filter coefficients based upon the audio system configuration, the selected set of filter coefficients representing an estimated echo response of the audio system including an array processing configuration of the plurality of microphone signals;array processing the plurality of microphone signals in accord with the array processing configuration to provide an arrayed microphone signal;filtering the audio program content signal, using the selected set of filter coefficients, to generate an estimated echo signal; andsubtracting the estimated echo signal from the arrayed microphone signal to generate an output audio signal.2. The method of further comprising loading the selected set of filter coefficients to an audio filter and activating the audio filter to perform the filtering.3. The method of further comprising rendering the audio program content signal into an acoustic signal claim 1 , based upon the identified audio system configuration.4. The method of further comprising loading the selected set of filter coefficients to an adaptive filter claim 1 , adapting the adaptive filter coefficients claim 1 , and copying the adaptive filter coefficients to an ...

COMMUNICATION DEVICE AND ECHO CANCELLATION METHOD

A communication device is disclosed. The communication device includes a transceiver circuit, an echo canceler, and a processor. The transceiver circuit is configured to transmit a test signal to a channel. The echo canceler is configured to obtain a plurality of echo power of a reflected signal corresponding to the test signal. The processor is configured to obtain a plurality of positions on the channel according to a parameter value. The parameter value is N, a number of the plurality of positions is N, and the plurality of positions corresponds to the top N largest of the plurality of echo power. The echo canceler is further configured to eliminate part of the plurality of echo power corresponding to the plurality of positions according to the plurality of positions. 1. A communication device , comprising:a transceiver circuit, configured to transmit a test signal to a channel;an echo canceler, configured to obtain a plurality of echo power of a reflected signal corresponding to the test signal; anda processor, configured to obtain a plurality of positions on the channel according to a parameter value, wherein the parameter value is N, a number of the plurality of positions is N, and the plurality of positions corresponds to the top N largest of the plurality of echo power;wherein the echo canceler is further configured to eliminate part of the plurality of echo power corresponding to the plurality of positions according to the plurality of positions.2. The communication device of claim 1 , wherein the processor is configured to obtain a first position of the plurality of positions during a first time period claim 1 , wherein the first position corresponds to a first largest echo power claim 1 , and the first largest echo power is the largest one of the plurality of echo power during the first time period claim 1 , wherein the echo canceler is further configured to eliminate the first largest echo power according to the first position.3. The communication device ...

ARRAY MICROPHONE AND SOUND COLLECTION METHOD

A sound collection method includes estimating at least one sound source direction and forming a plurality of sound collection beams in the estimated plurality of sound source direction, using sound collection signals of a plurality of microphones. The number of sound source directions estimated is smaller than the number of sound collection beams formed. 1. An array microphone comprising:a plurality of microphones;an estimator that estimates at least one sound source direction; anda beam former that forms a plurality of sound collection beams in the estimated at least one sound source directions, using sound collection signals of the plurality of microphones,wherein a number of the at least one sound source direction estimated by the estimator is smaller than a number of sound collection beams formed by the beam former.2. The array microphone according to claim 1 , further comprising a mixing processor that mixes an audio signal according to a sound collection beam claim 1 , among the plurality of sound collection beams claim 1 , by a gain according to volume of the sound collection beam.3. The array microphone according to claim 1 , wherein the plurality of microphones are arranged in a plane.4. The array microphone according to claim 1 , wherein the plurality of microphones are configured as a ceiling tile.5. The array microphone according to claim 4 , wherein the ceiling tile is configured to be replaceable.6. The array microphone according to claim 1 , further comprising an echo canceller that removes an echo component from an audio signal according to a sound collection beam claim 1 , among the plurality of sound collection beams.7. The array microphone according to claim 1 , further comprising:an echo canceller that removes an echo component from the plurality of sound collection signals of the plurality of microphones,wherein the estimator estimates each sound source direction, using the sound collection signals in which the echo component has been removed by ...

Method And System For Sampling Rate Mismatch Correction Of Transmitting And Receiving Terminals

Disclosed in the invention is a method and system for sampling rate mismatch correction of transmitting and receiving terminals, which can obtain a high-precision sampling rate mismatch in real time, carry out sampling rate correction on transmitting and receiving terminal signals, and send the transmitting terminal signal and the receiving terminal signal that have the same sampling rate after corrected to an echo cancellation system to carry out echo cancellation. The present invention can improve the quality of echo cancellation, simplify the computation and reduce the cost. The method for sampling rate mismatch correction of transmitting and receiving terminals provided in the embodiments of the invention comprises: calculating a transfer function of a receiving terminal signal relative to a transmitting terminal signal at each sampling timing according to the transmitting and receiving terminal signals; obtaining a transmission time delay of the transmitting and receiving terminals at each sampling timing using the transfer function; obtaining a sampling rate mismatch of the transmitting and receiving terminals at each sampling timing by means of parameter fitting using the transmission time delay and the linear relationship between the transmission time delay and the sampling rate mismatch; and adjusting the sampling rate of the transmitting terminal signal or the receiving terminal signal at each sampling timing according to the sampling rate mismatch.

Noisy Environment Communication Enhancement System

A communication system enhances communications in a noisy environment. Multiple input arrays convert a voiced or unvoiced signal into an analog signal. A converter receives the analog signal and generates digital signals. A digital signal processor determines temporal and spatial information from the digital signals. The processed signals are then converted to audible sound.