Abstract: Disclosed are a sound processing apparatus and a sound processing method. The sound processing method includes extracting a desired voice enhanced signal by a sound source separation and a sound extraction. By using a multi-channel blind source separation method based on independent vector analysis, the desired voice enhanced signal is extracted from a channel having the smallest sum of off-diagonal values of a separation adaptive filter when the power of the desired voice signal is larger than that of other voice signals. According to the present disclosure, a user may build a robust artificial intelligence (AI) speech recognition system by using sound source separation and voice extraction using eMBB, URLLC, and mMTC techniques of 5G mobile communication.

Abstract: Embodiments disclosed herein may include a wearable apparatus including a frame having a memory and processor associated therewith. The apparatus may include a camera associated with the frame and in communication with the processor, the camera configured to track an eye of a wearer. The apparatus may also include at least one microphone associated with the frame. The at least one microphone may be configured to receive a directional instruction from the processor. The directional instruction may be based upon an adaptive beamforming analysis performed in response to a detected eye movement from the infrared camera. The apparatus may also include a speaker associated with the frame configured to provide an audio signal received at the at least one microphone to the wearer.

Abstract: A voice interaction architecture that compiles multiple audio signals captured at different locations within an environment, determines a time offset between a primary audio signal and other captured audio signals and identifies differences between the primary signal and the other signal(s). Thereafter, the architecture may provide the primary audio signal, an indication of the determined time offset(s) and the identified differences to remote computing resources for further processing. For instance, the architecture may send this information to a network-accessible distributed computing platform that performs beamforming and/or automatic speech recognition (ASR) on the received audio. The distributed computing platform may in turn determine a response to provide based upon the beamforming and/or ASR.

Abstract: A telephonic device having a speakerphone function has a loudspeaker and a plurality of microphones. The plurality of microphones are coupled to a plurality of beamformers which produce a first beamform having a spatial null in the direction of the loudspeaker and a second beamform having a spatial null in the direction of near-end signals. The two beamforms are then processed to facilitate echo reduction. By comparing the beamform powers, a state of double-talk can be determined and the determination can be used to enhance echo reduction associated with speakerphone functionality.

Abstract: A full duplex audio communication terminal (100) comprises a noise attenuation module (4) arranged on a transmission path of the terminal. The noise attenuation module comprises two activation thresholds, which are respectively intended to be adjusted above a noise level and above an echo level of transmission signals (TC) produced by the terminal. The first or the second activation threshold is selected as a function of the detection of signals (R) received by said terminal. An improvement to the suppression in the transmission signals of echoes from received signals is thus obtained which is compatible with the principle of full duplex communication. Preferably, the noise attenuation module processes transmission signals which are produced by an echo compensator (2). Residual echo still present in the transmission signals already processed by the echo compensator is thus further reduced.

Abstract: An audio communication system includes a first analog/digital convertor, a second analog/digital convertor, and an echo canceller. The first analog/digital convertor converts an audio signal, which is input from a microphone, to a data signal. The second analog/digital convertor that converts an audio signal, which is input from an auxiliary input terminal, to a data signal. When an audio signal transmitted from the outside is received and output from a speaker, the echo canceller prevents an output of the speaker from being input through the microphone and echoed back to a recipient or remote device. An operating frequency of the echo canceller is relatively lower than a sampling frequency of the second analog/digital convertor.

Abstract: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.

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

Abstract: A system, method, and program product are provided for filtering sound from a selected application on a computer without interrupting voice communications on the computer. The method comprises: monitoring a selected program for an outgoing digital audio signal from a selected application; detecting said digital audio signal; and filtering an analog microphone input with the digital audio signal.

Abstract: At least one exemplary embodiment is directed to a new spatial audio enhancing system including a novel Adaptive Sound Upmixing System (ASUS). In some specific embodiments the ASUS provided converts a two-channel recording into an audio signal including four channels that can be played over four different loudspeakers. In other specific embodiments the ASUS provided converts a two-channel recording into an audio signal including five channels that can be played over five different loudspeakersn even other specific embodiments the ASUS provided converts a five-channel recording (such as those for DVD's) into an audio signal including eight channels that can be played over eight different loudspeakers. More generally, in view of this disclosure those skilled in the art will be able to adapt the ASUS to process and provide an arbitrary number of audio channels both at the input and the output.

Abstract: A communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

Abstract: An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.

Abstract: A full duplex audio communication terminal (100) comprises a noise attenuation module (4) arranged on a transmission path of the terminal. The noise attenuation module comprises two activation thresholds, which are respectively intended to be adjusted above a noise level and above an echo level of transmission signals (TC) produced by the terminal. The first or the second activation threshold is selected as a function of the detection of signals (R) received by said terminal. An improvement to the suppression in the transmission signals of echoes from received signals is thus obtained which is compatible with the principle of full duplex communication. Preferably, the noise attenuation module processes transmission signals which are produced by an echo compensator (2). Residual echo still present in the transmission signals already processed by the echo compensator is thus further reduced.

Abstract: The method comprises, in the time domain, echo cancellation processing (40) by subtracting from the microphone signal a reference signal delivered by an adaptive circuit for modeling the acoustic coupling between the microphone (18) and the loudspeaker (16). Successive frames of the signal are subjected in the frequency domain to processing (42) for suppressing the residual echo that remains after the echo cancellation, together with noise reduction processing (44). The residual echo suppression processing comprises estimating respective values for the power of the residual echo over a plurality of defined frequency bands for successive signal frames, then calculating and applying a variable gain specific to each frequency band and to each frame, the respective gains being smaller whenever the estimated relative power of the residual echo is high for the frequency band under consideration, and vice versa.

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

Abstract: A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.

Abstract: Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

Abstract: A method and corresponding apparatus for managing voice quality by optimizing the amount of attenuation placed in the echo return path based on the conditions observed during each telephone call. At the beginning of a telephone call, a pre-configured initial amount of attenuation is placed in the echo path of a communications connection that includes an echo canceller. After the call has begun, ERL is measured at a point in the communications connection that includes the attenuation. If the measured ERL is higher than necessary for the echo canceller to accurately perform double-talk detection, the attenuation in the echo path is slowly reduced from the initial amount. As the attenuation is reduced, the measured ERL reduces accordingly. When the measured ERL reduces to a minimum value required to perform double-talk detection, the reduction in attenuation is stopped.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

Abstract: An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.

Abstract: A method and apparatus for adapting dual filters is disclosed. In one aspect, a method may include transforming a signal, adapting a first adaptive filter based on the transformed signal, estimating a delay of an impulse response based on the adaptation of the first filter, delaying a signal based on the estimated delay, and adapting a second adaptive filter based on the delayed signal. In one aspect, an echo or other unwanted signal may be reduced or cancelled based on the adaptation of the second filter.

Abstract: A circuit arrangement for the analogue suppression of echoes, as in particular can be used in a hybrid-circuit for DSL-transmission systems, comprises a replica (8) for emulating the behaviour of the transmission line (17). In addition, a circuit (3, 4) for emulating the behaviour of the transmitter (13) is provided, which comprises at least one lowpass (3, 4). Furthermore, a replica (9, 10) for emulating the behaviour of bridge taps (14) can also be provided, which comprises at least one bandpass (9, 10). Additionally, a replica (19) for emulating the behaviour of the line driver (1) can also be provided.

Abstract: Methods and apparatus compensating for background noise associated with a transmitted signal. In a first embodiment, an adder combines an echo estimate with the the transmitted signal to form a processed signal, a noise reduction circuit uses a background noise estimate to subtract the background noise from the processed signal, and a non-linear processor uses the background noise estimate to combine a compensating background noise signal with the proccssed signal. In another embodiment, the noise reduction circuit divides the processed signal into sub-bands. With variations of an embodiment, the nonlinear processor is a sub-band non-linear processor, the processed signal is divided into sub-bands, and the background estimate includes sub-band noise estimates.

Abstract: A digit signal processing system may include a near end and a far end connected by a pair of signal transmission paths. A signal processing may be located at the near end through which signal commands are transmitted to the far end. A signal transducing device may be located at the far end to receive the signal commands and to transmit to the near end signals indicating the state of the signal transducing device. A signal canceller may be operatively associated with the signal processor to subtract undesired signals from the received signals. The signal canceller may employ Auxiliary-Vector filtering.

Abstract: Sub-band filtering, adaptive echo cancellation, and residual echo suppression are selectively and separately applied in hierarchical order under the control of a circuit that monitors four separate signals to determine n machine states. The number of machine states is further divided among m levels of noise in determining which, and how much of, sub-band filtering, adaptive echo cancellation, and residual echo suppression to use at any given time. A voice activity detector (VAD) is used to monitor each signal and provides statistical information in addition to whether or not a voice signal is detected. Sub-band filter, echo cancelling, and non-linear processing are not permitted to provide maximum reduction simultaneously. Even so, the invention provides at least 15 dB more echo suppression than systems without a sub-band filter bank series coupled with a non-linear processor in each channel.

Abstract: An adaptive digital filter (10) is adapted to generate a pseudo echo Se which is produced by imitating a noise, using part of receiving signals Si and to output it. An echo produced through an echo path (15) is mixed, as an echo noise, with a sending signal So and this sending signal is attenuated by a level attenuator (6). The level attenuator (6), in order to match the amplitude level of the echo noise of the sending signal So to that of the pseudo echo, attenuates the sending signal So and feeds it to a subtracter (7). The subtracter (7) eliminates echo noises by subtracting the pseudo echo Se from the sending signal So. Even if the echo path is in the state of amplified system, the complete elimination of the echo can be achieved accordingly.

Abstract: There are provided a voice switching system and a voice switching method which are capable of carrying out highly accurate voice detection and performing a switching action properly even in the presence of an unstationary noise or under circumstances where the S/N ratio is low.

Abstract: A bi-directional hands-free communication device includes a microphone for transmitting a signal along a transmit path and a speaker receiving a signal transmitted along a receive path and outputting a corresponding output signal. An echo canceller, positioned in the transmit path and the receive path, cancels echo signals induced by the microphone from the speaker and outputs a corresponding cancelled signal along the transmit path, and a transparency circuit distributes state-dependent additional loss derived from the noise floor margin to the transmit path and the receive path to reduce residual echo signals output from the echo canceller.

Abstract: Echoes caused by discontinuities in a telecommunications transmission lines cause double talk which cause divergence problems with echo cancellers. Divergence problems in echo cancellers cause considerable problems in interactive caller response systems, they cause less problems if all parties to a call are human. One of the biggest problems is the effect of the echoes on automatic speech recognition systems. The echo canceller method and system presented cancels echoes to reduce their interference with ASR and yet provides stability in the presence of double talk.

Abstract: A speaker-phone that is operable to perform system change detection, double-talk detection and if desired, optimal step-size control. The invention differentiates between system change and double-talk and is operable to these two situations differently. The invention does not rely solely on the energy measurements that are employed by conventional speaker-phones. Highly reliable system change detection is offered when compared to conventional speaker-phones. The direction and the amplitude of the system change is measured with very good detection probability in the invention. For double-talk detection, the invention does not rely on assumptions that the far-end echoes are of lower amplitude that the near-end signal in performing double-talk detection. The invention performs double-talk detection within a large diversity of echo-paths and background noise levels. A designer may tune the speaker-phone system to ensure proper double-talk detection near low levels of the near-end signal.

Abstract: An echo canceler for suppressing an echo of a receive signal contained in a transmit signal directed to a telephone circuit and transferring the transmit signal subjected to echo suppression to the telephone circuit. The echo canceler includes a pseudo echo path for attenuating and delaying the receive signal supplied from the telephone circuit and adding the delayed receive signal to the transmit signal, and an echo cancel circuit for estimating a transfer function of the pseudo echo path on the basis of the receive signal supplied from the telephone circuit and the echo of the receive signal, generating a pseudo echo on the basis of a value of the estimation and the receive signal, subtracting the pseudo echo from the transmit signal passed through the pseudo echo path, and thereby transferring the transmit signal subjected to echo suppression to the telephone circuit.

Abstract: The invention relates to a circuit configuration for two-wire/four-wire conversion. The circuit configuration has a receiver which is coupled on an input side to a digital two-wire reception path via which its receives a digital signal. A signal-processor is connected downstream of the receiver and emits a first and second signal. An echo-canceling device receives the first signal and emits a third signal. A digital-to-analog converter is provided which receives the second signal and emits a fourth signal. A hybrid circuit is coupled to an analog four-wire transmission-reception path, receives the fourth signal, and emits a fifth signal. An analog-to-digital converter receives the fifth signal and emits a sixth signal. A transmitter receives the sixth signal and is coupled on an output side to a digital two-wire transmission path and emits a digital transmitted signal to the latter. In the invention, the digital-to-analog converter has a sigma-delta modulator.

Abstract: In an acoustic echo canceler of a loudspeaking telephone apparatus, a sequence of successively delayed tap signals of a remote speech signal from a transmission line are respectively weighted by filter coefficients and summed together to produce an acoustic echo replica of the speech signal. The filter coefficients are updated with a residual echo and restricted to those which correspond to an echo component of the speech signal, or peak impulse response of the acoustic feedback path between the loudspeaker and the microphone. The tap signals are also restricted to those corresponding to the restricted filter coefficients. The output of the microphone is combined in a subtractor with the acoustic echo replica, producing the residual echo. After small amplitude components of the residual echo are suppressed in a first attenuator, the output of this attenuator is supplied through a second attenuator to the line.

Abstract: Methods and apparatus for processing a transmitted voice signal include a centralized frame controller providing at least one boundary control signal to voice processing blocks and controlling the operation of the voice processing blocks on the transmitted voice signal based upon the boundary control signal.

Abstract: Full acoustic coupling and echo cancellation is provided in implementations where there is acoustic coupling between acoustic zones.

Abstract: In handsfree communications devices including a linear echo or noise canceler, both analog and digital gain factors are adjusted in response to user volume control inputs, the analog gain factor being frozen during active communications (e.g., during telephone calls). By providing dual user controls, one of which is disabled when a call is in progress, embodiments of the invention provide the wide dynamic range required in most handsfree applications without degrading performance of the linear echo canceler.

Abstract: A subscriber's telephone installation comprises a plurality of telephone sets coupled in parallel to a subscriber loop so as to provide a voice facility, and a data modern coupled to the subscriber loop so as to provide data facilities at a frequency above that employed for said voice facility. Some or preferably all telephone sets are provided with a respective filter arrangement incorporating series connected inductive and parallel connected capacitive elements for rejecting selected frequencies from the telephone set. The filter also incorporates a switch responsive to current drawn from the line for selectively enabling and disabling the filter arrangement when the telephone set is respectively in its off-hook condition or in its on-hook condition.

Abstract: A communication apparatus includes circuitry operable to facilitate communication over a digital subscriber line. The apparatus also includes a filter operable to be coupled between the circuitry and the digital subscriber line. The filter has a corner frequency and is operable to attenuate transients associated with the circuitry when the apparatus is in use. The filter is tuned to an impedance of the circuitry.

Abstract: In a system including a plurality of telephones coupled to an asymmetric digital subscriber line, a circuit operable to facilitate communication using the asymmetric digital subscriber line includes a filter having a corner frequency and being operable to be coupled between one of the plurality of telephones and the asymmetric digital subscriber line. The filter is also operable to attenuate transients associated with the telephone operating in an off-hook state. The circuit also includes a switch coupled to the filter. The switch is operable to effect a change in the corner frequency of the associated filter depending on whether the telephone coupled to the filter is operating in an on-hook or an off-hook state, the change in the corner frequency operating to attenuate parasitic interference associated with telephones in an on-hook state.