Abstract: A system and method for enhancing communications through a phone. A voice communication is received from a user of the phone. A secondary signal is received from an environment in proximity to the phone. The secondary signal is processed to determine an inverse signal in response to receiving the secondary signal. The inverse signal is combined with the voice communication and the secondary signal to destructively interfere with the secondary signal for allowing a receiving party to more effectively communicate with the user.

Abstract: Apparatus, methods, and machine-readable articles of manufacture enable a telephone system to distinguish between actual events and echoes. Events may include conferees' voice and DTMF commands. In selected embodiments, the system categorizes an event received on one channel as a real event if the energy of the event's waveform is greater than concurrent energy of every other channel of the same conference. Otherwise, the event is stored. When the system receives a new event of the same type on a channel of the same conference during a predetermined period, it compares the energies of the new and stored events. If the energy of the new event exceeds that of the stored event, the new event becomes the stored event, and the old stored event is ignored. At the end of the period, the stored event is categorized as a real event and forwarded to appropriate application code.

Abstract: A technique of operating a communication device includes dividing a frequency band associated with a background noise signal into respective sub-bands. Respective individual level estimates for each of the respective sub-bands are then determined. A total level estimate for the background noise signal is determined. Finally, a comfort noise signal (whose characteristics are based on the respective individual level estimates and the total level estimate) is provided.

Abstract: In an echo canceller including an ADF and a cancellation adder, an ACANC calculator finds an average value of the amount of echo cancellation for a predetermined period to update the average value of the amount of echo cancellation, and detects a convergence state of the ADF to find the amount of echo cancellation in the convergence state of the ADF. A coefficient selector, when receiving a detection signal of the convergence state from the ACANC calculator, finds a power ratio of the sum total of power of all tap coefficients and the partial total of power of tap coefficients not to be assigned to taps to determine the effective number of taps useful for suppressing an echo component on the basis of the amount of echo cancellation and the power ratio. The ADF updates only the effective number of taps to disable the remaining taps.

Abstract: Systems and methods for providing acoustic echo cancellation are provided. Primary and secondary acoustic signals are received by a communication device. The acoustic signals may include loudspeaker leakage. A null coefficient is then adaptively determined for each subband of the secondary acoustic signal. The null coefficient is applied to the secondary acoustic signal to generate a coefficient-modified signal. The coefficient-modified signal is subtracted from the primary acoustic signal to generate a masked acoustic signal with reduced or no echo. The masked acoustic signal may be output.

Abstract: A call directing system receives an incoming call from a caller. The caller is prompted to speak, thus enabling a prosody analyzer to generate an analysis of a prosody of the caller's voice. This analysis provides a basis for generating a caller profile that describes caller preferences of the caller. Based on the analysis of the prosody of the caller's voice and the generated caller profile, the call is directed to a particular call recipient.

Abstract: A device for suppressing ambient sounds from speech received by a microphone array is provided. One embodiment of the device comprises a microphone array, a processor, an analog-to-digital converter, and memory comprising instructions stored therein that are executable by the processor.

Abstract: Disclosed herein are conferencing products implementing an acoustic echo cancellation system that utilizes converging coefficients and a detector of turn-off and/or turn-on events of push-to-talk microphones, and, further, that mitigates against divergence and/or drift of coefficients and other variables of an echo canceller. A push-to-talk detector may be used that includes a high-pass filter, a transient detector, or an adjustable high-pass filter. An echo canceller may be disabled as to a push-to-talk microphone that has been turned off.

Abstract: In a mobile terminal device having a processing unit and a memory unit, an acoustic echo cancellation and a microphone beamforming are provided. Said device includes a plurality of a microphones, a pre-processing stage has an amplifier and a signal converter for an analog-to-digital conversion. In a main signal processing stage, a pre-filter suitable for polynomial beamforming, an acoustic echo cancellation stage and a post-filter for polynomial beamforming are provided. Furthermore, a post-processing stage has an amplifier and a signal converter for a digital-to-analog conversion, and plurality of speakers.

Abstract: A method enables a participant in a conference call to monitor, as he is speaking, whether his speech is getting through to the other participants. A teleconference bridge receives audio signals from a group of telecommunications endpoints that are involved in a conference call. The bridge generates audio signals to be transmitted, which are based on one or more of the received audio signals. During the ongoing process of minimizing the presence of acoustic echo, the bridge might exclude one or more of the received audio signals from the transmitted audio signals. When this occurs, particularly when an active talker is being excluded, the bridge transmits an indication to one or more of the endpoints as part of one of the transmitted audio signals. The indication can be audible such as a tone or a voice, visual such as a flashing light, or tactile such as vibration.

Abstract: A method, apparatus, system, and signal-bearing medium that in an embodiment determine a degree of correlation between a speaker output signal and a microphone input signal and modulate an adaptive gain of an acoustic echo canceller based on the degree of correlation.

Abstract: Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.

Abstract: The present invention discloses a double-talk detection method and apparatus. The method includes: obtaining a far-end signal frame and a near-end signal frame; and detecting a talk state according to a spectrum difference between the far-end signal frame and the near-end signal frame. The apparatus includes: an obtaining module, adapted to obtain a far-end signal frame and a near-end signal frame; and a spectrum detecting module, adapted to detect a talk state according to a spectrum difference between the far-end signal frame and the near-end signal frame. With the present invention, the talk state is detected according to the spectrum difference between the far-end signal and the near-end signal. Even if nonlinear distortion exists, the precision of detection is not impacted, whereas, the greater the distortion is, the more precise the detection result and the better the detection performance will be.

Abstract: A communications device is presented for providing bi-directional audio communications between a near-end user and a far-end user via a bidirectional communications channel. The communications device includes an adaptive echo canceller receiving a near-end audio signal and a far-end audio signal and providing an echo-canceled near-end audio signal for transmission to the far-end user via the communications channel. The adaptive echo canceller includes a first bank of analysis filters for filtering the near-end audio signal, a second bank of analysis filters for filtering the far-end audio signal, and a bank of synthesis filters for filtering sub-band echo-canceled signals generated within the adaptive echo canceller. The first and second filter banks have a frequency response optimized to reduce echo residual gain.

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

Abstract: 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.

Abstract: An acoustic echo cancellation device (1) for canceling an echo in a microphone signal (z) in response to a far-end signal (x) comprises: an adaptive filter unit (10) arranged for filtering the far-end signal (x) so as to produce an echo cancellation signal (y), —a combination unit (12) arranged for combining the echo cancellation signal (y) with the microphone signal (z) so as to produce a residual signal (r), and a post-processor unit (13) arranged for substantially removing any remaining echoes from the residual signal. In accordance with the invention, the device further comprises: —a near-end pitch estimation unit (18) arranged for estimating the pitch of any near-end signal (s) contained in the microphone signal (z) and for controlling the post-processor unit (13) in dependence of the estimated pitch such that harmonics of the estimated pitch may be substantially preserved.

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 cellular phone accumulates input information, which is inputted by a user using the cellular phone located not only inside but also outside of a vehicle, as parent population data for interest extraction, and extracts interest information, which reflects a present interest of a user, from the parent population data for interest extraction. An in-vehicle information output apparatus internally or externally collects provision information, which matches the interest information extracted in the cellular phone, and provides the user with the collected provision information.

Abstract: An electricity echo cancellation device applied at a terminal includes: an input buffer memory module, a network echo delay computation module and an adaptive filtering module. The adaptive filtering module includes an adaptive filter, a subtracter and a dual-ended voice detection module. An electricity echo cancellation method includes: calculating a network echo delay according to relevant information of an RTCP packet transmitted from the network; and dynamically adjusting a terminal input signal to be adaptively filtered according to the network echo delay. The present invention ensures the electricity echo cancellation effect at the final user end on the whole, and improves the effectiveness of electricity echo cancellation. Meanwhile method of the present invention can be realized with software, thus avoiding influences of hardware memory restricts on the echo cancellation effect. In addition, the present invention only needs a single-point deployment, and thus the cost is saved.

Abstract: A method for providing a context specific speaker adaptation user interface is disclosed. The method allows a specific user of a voice activated system to train a voice recognition system to understand specific uttered commands by receiving a training sequence initiation input, sending at least one command to be trained, receiving a selection input, prompting a verbal input, recording the verbal input, and linking a voice input with an instruction corresponding with the command input.

Abstract: A method, microphone module and a system for full-duplex audio system implementing full duplex audio, audio echo cancellation and audio codec. The audio signals from the microphones and to the loudspeakers are encoded to reduce the transmission bandwidth requirement during wired or wireless transmission and decoded when needed for further processing. Each microphone has transceivers to receive signals fed to loudspeakers within the microphone module. An audio echo canceller is installed within each microphone module to generate echo free audio signal for further signal processing. Problems due to the combination of lossy codec and echo cancellation are avoided.

Abstract: Methods and apparatus of managing a communication system, wherein a decision regarding a level of activity at a first end is made based at least in part on the level of activity at the second end. In one embodiment, the energy level of a first-end audio signal is measured. The first end is declared voice-active if the first-end energy level is greater than or equal to a first threshold value. The first end is declared voice-inactive if the first-end energy level is less than the first threshold value. To determine the value of the first threshold value, the energy level of a second-end audio signal is measured. If the second-end energy level is greater than or equal to a second threshold value, the second end is declared voice-active, in which case the first threshold is maintained at a relatively high level. If the second-end energy level is less than the second threshold value, the second end is declared voice-inactive, in which case the first threshold is maintained at a relatively lower level.

Abstract: Apparatus, methods, and machine-readable articles of manufacture enable a telephone system to distinguish between actual events and echoes. Events may include conferees' voice and DTMF commands. In selected embodiments, the system categorizes an event received on one channel as a real event if the energy of the event's waveform is greater than concurrent energy of every other channel of the same conference. Otherwise, the event is stored. When the system receives a new event of the same type on a channel of the same conference during a predetermined period, it compares the energies of the new and stored events. If the energy of the new event exceeds that of the stored event, the new event becomes the stored event, and the old stored event is ignored. At the end of the period, the stored event is categorized as a real event and forwarded to appropriate application code.

Abstract: A system and method for improving voice recognition processing at a server system that receives voice input from a remotely located user system. The user system includes a microphone, a processor that performs front-end voice recognition processing of the received user voice input, and a communication component configured to send the front-end processed user voice input to a destination wirelessly over a network. The server system includes a communication component configured to receive the sent front-end processed user voice input, and a processor configured to complete voice recognition processing of the sent front-end processed user voice input.

Abstract: The present invention discloses a double-talk detection method and apparatus. The method includes: obtaining a far-end signal frame and a near-end signal frame; and detecting a talk state according to a spectrum difference between the far-end signal frame and the near-end signal frame. The apparatus includes: an obtaining module, adapted to obtain a far-end signal frame and a near-end signal frame; and a spectrum detecting module, adapted to detect a talk state according to a spectrum difference between the far-end signal frame and the near-end signal frame. With the present invention, the talk state is detected according to the spectrum difference between the far-end signal and the near-end signal. Even if nonlinear distortion exists, the precision of detection is not impacted, whereas, the greater the distortion is, the more precise the detection result and the better the detection performance will be.

Abstract: According to the present invention, a tilt sensor is used to determine the tilt angle of a speakerphone and the surface on which it rests. This information is used to optimize both the receive and transmit signals for the chosen tilt angle. The information can also be used to adjust performance of any beamformer(s) where the speakerphone incorporates a microphone array or loudspeaker array. In one embodiment, vibrational data is provided by the tilt sensor for enhancing the receive signal and acoustic echo cancellation.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

Abstract: A background noise estimate based upon a modified Doblinger noise estimate is used for modulating the output of a pseudo-random phase spectrum generator to produce the comfort noise. The circuit for estimating noise includes a smoothing filter having a slower time constant for updating the noise estimate during noise than during speech. Comfort noise is smoothly inserted by basing the amount of comfort noise on the amount of noise suppression. A discrete inverse Fourier transform converts the comfort noise back to the time domain and overlapping windows eliminate artifacts that may have been produced during processing.

Abstract: A system and method for improving voice recognition processing at a server system that receives voice input from a remotely located user system. The user system includes a microphone, a processor that performs front-end voice recognition processing of the received user voice input, and a communication component configured to send the front-end processed user voice input to a destination wirelessly over a network. The server system includes a communication component configured to receive the sent front-end processed user voice input, and a processor configured to complete voice recognition processing of the sent front-end processed user voice input.

Abstract: The invention relates to a real time processing and management method for canceling out the echo between a loudspeaker (HP) and a microphone (M) of a computer terminal. Said method consists of establishing the loudspeaker signal or the microphone signal as a reference signal and synchronizing (A) the other signal with said reference signal, and (B) synchronizing the acquisition and audio reestablishment tasks. The delay that exists (rej) between the other signal and the reference signal is measured (C) and the existing delay value (rej) is validated (D) in order to control the delay applied to the reference signal starting from the current delay (rej). The invention can be used for computer terminals which are equipped with multi-tasks operating systems.

Abstract: A system (200) and method (400) for near-end detection of voice (107) in speakerphone mode is provided. The method can include determining (402) a convergence of an adaptive filter (220), determining (404) a dissimilarity between an autocorrelation (311) of an echo estimate (244) and an autocorrelation (312) of a microphone signal (243) if the adaptive filter has converged, computing (406) a weighting factor (279) based on the dissimilarity, applying the weighting factor to a voice activity level (281) to produce a weighted voice activity level (283), comparing (410) the weighted voice activity level to a constant threshold, and performing (412) a muting operation in accordance with the comparing for providing half-duplex communication.

Abstract: Acoustic echo control for hands-free phone has acoustic echo cancellation and echo suppression with a voice activity detection for the echo suppression based on near-end input power together with an estimate for acoustic echo cancellation gain.

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: There is disclosed an audio coding apparatus which has a wideband encoder and noise canceller. The encoder includes a high-frequency audio coder and low-frequency audio coder. The low-frequency audio coder includes a low-frequency noise canceller. When the high-frequency audio coder is disabled, the noise canceller is disabled, and allows a digital audio signal to pass through it and outputs that signal to the encoder. When the high-frequency audio coder is enabled, the low-frequency noise canceller is disabled, and allows a digital audio signal to pass through it.

Abstract: A non-echo canceling voice band modem capable of detecting subscriber alerting signal (SAS) tones and customer premises equipment alerting signal (CAS) tones is provided. The modem includes a shaping filter, an analog-to-digital converter (ADC), a SAS tone detector, a CAS tone detector and an echo canceler. The shaping filter, which is located in a transmitting path of the modem, filters modem signals to be transmitted from the modem. The ADC, which is located in a receiving path of the modem, converts any incoming analog signal to corresponding digital signals to be used by the modem. The echo canceler, which is coupled to the SAS and CAS tone detectors, cancels any echoes in modem signals before sending said modem signals to the SAS and CAS tone detectors.

Abstract: An apparatus for canceling an echo signal in a mobile terminal of a mobile communication system. A double talk detector (DTD) receives a first signal by canceling an estimated echo signal from a signal received through a microphone, outputs the first signal, and outputs the first and a second signal comprising a background noise signal and a residual echo signal during a non-double talk. An Auto-Regressive (AR) analysis and inverse filtering unit receives the second signal from the DTD, and whitens the second signal. A pitch analysis and inverse filtering unit receives the whitened signal, and cancels a pitch value remaining therein by performing pitch analysis and inverse filtering on the whitened signal. A noise canceller receives the pitch-cancelled whitened signal and the first signal output from the DTD, canceling a residual echo signal and a background noise signal from the first signal using the pitch-cancelled whitened signal.

Abstract: Methods and apparatus for processing a received voice signal from a variable-delay network by a communications system that has at least one echo canceller. An echo canceller determines delay characteristics that are associated with the received voice signal and correspondingly modifies echo cancellation processing. In a variation of the embodiment of the invention, a first echo canceller sends silence descriptor information to a second echo canceller. The second echo canceller reconstructs a voice signal based upon the silence descriptor information and excitation vector information.

Abstract: A noise level calculator for detecting noise in a telephone line echo canceller that utilizes adaptive filters enables silencing of noise switching so that it is not heard on the far end side. The calculator promotes determining whether the reference signal applied to the adaptive filter is a noise or non-noise segment. The calculator operates by calculating the difference in energy of two sample windows, calculating the difference in the energy of the signal within each of the sample windows, updating a variance parameter based on the difference, and performing further alternative operations based on the value of the variance parameter.

Abstract: A computer apparatus is provided, which has a multipath acoustic output system and a multipath acoustic input system, which includes parts for enabling an adaptation of the filter of an acoustic echo canceller incorporated in the computer apparatus that is always precise regardless of the chronological delays of the components of the computer apparatus, wherein the chronological delays are variable for acoustic signals.

Abstract: A method and apparatus for adaptive echo and noise control. A signal can be received at an input 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 based on the background noise in the signal. Adaptively determining the order of noise suppression and echo cancellation 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.

Abstract: A state machine for attenuating the transition into and out of NLP state to reduce voice clipping and to reduce echo leak in a voice over packet signal transmission. The state machine interposes two additional transitional states between the NLP active state and the NLP inactive state to eliminate the sharp transition of NLP activity. An NLP entering state is used to gradually reduce the mixing ratio of echo and voice to attenuate the transition from passage of echo in the NLP inactive state to the suppression of echo in the NLP active state. An NLP exiting state is used to gradually increase (ramping the change) the mixing ratio of echo and voice to attenuate the transition from suppression of echo in the NLP active state to passage of echo in the NLP inactive state.

Abstract: A non-linear processor for managing a speech signal is disclosed. The non-linear processor includes an adaptive lattice filter that generates a prediction error from a residual echo of the speech signal, a signum non-linearity unit that generates an excitation component from the prediction error, an excitation signal mixer that generates an excitation signal from the excitation component and a noise component, an inverse lattice filter that generates a modified residual echo from the excitation signal, and a gain adjuster that adjusts a power of the modified residual echo.

Abstract: A recently received portion of a transmit signal is utilized, and a pitch period of the received signal is determined. Then, the determined pitch period is utilized to search for similarities in the originally transmitted signal. This searching is done over a range of delays. Once the optimal delay is determined, it is then utilized as the delay between the transmitted signal and the received signal. Once this delay is known, the received signal can be time-aligned with the delayed signal.

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: Methods and systems pertaining to preventing the corruption of filter coefficients due to near end speech in an echo canceller that uses an adaptive filter to create a local replica of an echo signal are provided. In accordance with a preferred method, a far end signal is received and the presence of near end speech is indicated. A filter coefficient vector is updated, and copies of the filter coefficient vector are periodically stored. A stored copy of the filter coefficient vector is recovered for use in the echo canceller. Further, an improved echo canceller that prevents the corruption of filter coefficients due to near end speech, having a near end speech indicator, a first buffer, a filter, and a second buffer connected to the filter, is provided. The first buffer receives a far end signal. The filter is responsive to the first buffer and the near end speech indicator.

Abstract: Determining the time delay between a transmitted signal and an echo. Then, the echo canceller is adjusted by the delay amount so that the echo canceller tail length is relatively short.

Abstract: Methods and apparatus for processing a received voice signal from a variable-delay network by a communications system that has at least one echo canceller. An echo canceller determines delay characteristics that are associated with the received voice signal and correspondingly modifies echo cancellation processing. In a variation of the embodiment of the invention, a first echo canceller sends silence descriptor information to a second echo canceller. The second echo canceller reconstructs a voice signal based upon the silence descriptor information and excitation vector information.

Abstract: Techniques for canceling echo and suppressing noise using an array microphone and signal processing. In one system, at least two microphones form an array microphone and provide at least two microphone input signals. Each input signal may be processed by an echo canceller unit to provide a corresponding intermediate signal having some echo removed. An echo cancellation control unit receives the intermediate signals and derives a first gain used for echo cancellation. A noise suppression control unit provides at least one control signal used for noise suppression based on background noise detected in the intermediate signals. An echo cancellation and noise suppression unit derives a second gain based on the control signal(s), cancels echo in a designated intermediate signal based on the first gain, and suppresses noise in this intermediate signal based on the second gain. The signal processing may be performed in the frequency domain.

Abstract: In an acoustic echocanceller (6), an estimate of an echo signal is determined by an adaptive filter (10) and is subtracted from the input signal by a subtracter (14). The spectrum estimator (12) determines the frequency spectrum of the estimate of the echo signal, and the filter (16) filters the output signal of the subtracter (14) with a filter having a transfer function dependent on the spectrum determined by the estimator (12). The use of this combination results in a substantial improvement of the suppression of the echo signal.