Abstract: Described herein is a method and audio capture system for suppressing noise due to mechanical disturbances in an audio capture system. In one embodiment a method includes the steps of: a) receiving an input audio signal from a microphone; b) receiving an input reference signal separate from the input audio signal; c) processing the input reference signal to generate a control signal for identifying a noise event; d) in response to the control signal, estimating a contribution of the noise event to the input audio signal by applying an adaptive filter to a sliding temporal window of the control signal in the frequency domain; and e) based on the estimated contribution, selectively modifying the input audio signal to generate an output audio signal in which the noise arising from the noise event is at least partially suppressed.

Abstract: 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(fn, A(t1, . . .

Abstract: A split radio access network is provided that efficiently transmits beamforming coefficients from a distributed baseband unit device to a remote radio unit device to facilitate beamforming at the remote radio unit. The beamforming coefficients can be determined at the baseband unit device and transmitted along with the data to be beamformed to the remote radio unit device. Due to the large number of antenna ports however, there can still be a very large number of coefficients to transmit, and the disclosure provides for a compressed set of coefficients that reduces the overhead signaling requirements. Instead of sending coefficients for every kth antenna port, the system can select a subset of the coefficients corresponding to a set of k antenna ports which can be used by the remote radio unit to approximate the full set of beamforming coefficients.

Abstract: Techniques related to active acoustic echo cancellation are discussed. Such techniques may include generating an audio output signal having a portion thereof corresponding to a first audio frequency range to negate a response of an audio input device to an output from a speaker in a second audio frequency range at a response negation rate and decimating an audio input signal based on the response negation rate to generate a resultant audio input signal.

Abstract: Audio pickup systems and methods are provided to enhance an audio signal by removing noise components related to an acoustic environment. The systems and methods receive a primary signal and a reference signal. The reference signal is adaptively filtered and subtracted from the primary signal to minimize an energy content of a resulting output signal.

Abstract: Updating filter coefficients during echo cancellation is disclosed. In some embodiments, a communication device may include a processor configured to operate a background filter configured as an adaptive filter, operate a foreground filter configured as a fixed filter; and update the foreground filter with coefficients from the background filter responsive to monitoring a history of both a near-end signal energy and an estimated far-end signal energy for a pre-determined time window in which conditions of each of the near-end signal energy and the estimated far-end signal energy are satisfied.

Abstract: A direction-switchable transponder of a high speed communications network, e.g., an fiber optic data communications network, is capable of dynamically reversing the data traffic flow of its various communications channels in response to a signal. The signal can specify a number of channels, a channel map, or a required bandwidth. The direction-switchable transponder can receive a signal relating to network bandwidth requirements; select, based on the received signal, one or more fiber optic channels for reversing direction of flow of network traffic; and dynamically and automatically reconfigure the selected fiber optic signal to reverse direction of flow of network traffic. By responding to asymmetric bandwidth requirements, the direction-switchable transponder uses high speed communications network lines more efficiently.

Abstract: Techniques to dynamically engage echo cancellation are described. In one embodiment, an apparatus may comprise a streaming component operative to establish a audio connection between the first client device and a second client device via the network interface controller; and receive a far-end audio stream at the first client device from the second client device via the audio connection; an audio capture component operative to capture a near-end audio stream at the first client device; and an echo processing component operative to compare the near-end audio stream and the far-end audio stream to determine whether a far-end echo is present in the near-end audio stream; and use an echo-cancellation module at the first client device where the far-end echo is present in the near-end audio stream. Other embodiments are described and claimed.

Abstract: An echo cancellation data synchronization control method is disclosed. The method includes: estimating a sound card delay value; initializing and accumulating a near-end audio buffer queue and a reference audio buffer queue until a difference obtained by subtracting a length of the near-end audio buffer queue from a length of the reference audio buffer queue is greater than or equal to an audio data length corresponding to the sound card delay value; extracting, in accordance with audio frames, audio data from the head of the reference audio buffer queue and the head of the near-end audio buffer queue for echo cancellation processing; obtaining a relative delay value generated by performing echo cancellation processing; and adjusting the sound card delay value according to the relative delay value.

Abstract: In an echo suppressing apparatus suppressing from a near-end input signal acoustic echo signals caused by a far-end signal, an estimated-echo path characteristic is multiplied by a far-end signal to produce an estimated-echo signal for each frame, and frames of the estimated-echo signals thus obtained are stored in an estimated-echo signal storage. A delay amount estimator calculates the total of differences between the near-end input signal and each estimated-echo signal thus stored, and determines the amount of frame delay having its total of differences minimal. A delay estimated-echo signal calculator uses the amount of frame delay to read out an optimum frame and near frames respectively preceding and following the optimum frame and to calculate a corrected, delay estimated-echo signal. An echo suppressor uses the estimated-echo signal thus corrected to suppress the acoustic echo signals from the near-end input signal.

Abstract: Methods and apparatus for estimating the power spectral density (PSD) of a residual interference having first and second components after adaptive interference cancellation (AIC). The first component can be estimated using a real-valued FIR filter operating on a time series of PSD estimates of a reference signal, and the second component can be estimated using an exponential decay over time corresponding to a reverberation time using the PSD of the reference signal.

Abstract: Disclosed herein are a virtual golf simulation device and method for providing stereophonic sound for weather conditions configured such that it is possible for a user to visually and aurally recognize a weather condition, such as wind, when the user plays a round of virtual golf on a virtual golf course using the virtual golf simulation device, and, in particular, the direction or intensity of the wind is expressed via multi-channel stereophonic sound, whereby it is possible to provide a high level of realism even without actually blowing air.

Abstract: An RF quasi circulator circuit is described herein. In accordance with one example of the disclosure the circuit includes a receive port, a transmit port and an antenna port as well as a differential amplifier stage having a first input, a second input and an output that is coupled to the receive port. The circuit further includes a first phase shifting element and a second phase shifting element. The first phase shifting element is coupled between the transmit port and the first input of the differential amplifier and the second phase shifting element is coupled between the transmit port and the second input of the differential amplifier. A tunable impedance is coupled to the differential amplifier, and the antenna port is coupled to the first input of the differential amplifier. The tunable impedance is controlled to tune the damping in a signal path from the transmit port to the receive port.

Abstract: An echo cancellation device includes an echo signal reduction processing unit that generates a transmitting signal in which an echo signal component is reduced from the acoustic signal; and a filter coefficient update determination unit that instructs the echo signal reduction processing unit to update the filter coefficient when the output voice signal corresponds to a voice section and the acoustic signal includes the echo signal component. The filter coefficient update determination unit calculates a feature value of the acoustic signal and instructs to update the filter coefficient when a difference between a first feature value of a frequency band equal to or lower than the target frequency and a second feature value of a frequency band higher than the target frequency is equal to or greater than a preset update determination threshold.

Abstract: 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.

Abstract: An electronic device comprises an executing module, a display module, and a display controller. The display controller makes the display module display a first display screen containing information about execution of a certain function during execution. The executing module starts execution of a widget for making the display module display information included in the first display screen while the certain function is executed and thereafter when execution of the certain function is finished, the executing module finishes execution of the widget When the display controller makes the display module display a display screen different from the first display screen while the widget is executed, the display controller incorporates widget execution information into a partial region of the different display screen based on execution of the widget. The widget execution information includes at least part of the information on the first display screen.

Abstract: Disclosed is an echo canceller 10 including: mixers 11A and 11B each to mix output signals outputted by speakers 7FL and 7FR with an instructed mixing ratio, and to generate reference signals for microphones 8A and 8B; an adaptive filter 12 to generate two pseudo echo signals from the two reference signals generated by the mixers 11A and 11B; subtractors 13A and 13B to subtract the two pseudo echo signals from input signals from the microphones 8A and 8B, and to generate two echo eliminated signals; residual echo detectors 14A and 14B to detect residual amounts of echo remaining in the echo eliminated signals obtained from the microphones 8A and 8B; and a mixing ratio changer 15 to instruct a change of the mixing ratios of the mixers 11A and 11B in accordance with the residual amounts of echo.

Abstract: The invention provides a wear-resistant steel plate, which has the following chemical composition (wt. %): C: 0.22-0.35%, Si: 0.10-0.40%, Mn: 0.60-1.35%, P: ?0.015%, S: ?0.010%, Nb: 0.010-0.040%, Al: 0.010-0.080%, B: 0.0006-0.0014%, Ti: 0.005-0.050%, Ca: 0.0010-0.0080%, V?0.080%, Cr?0.60%, W?1.00 wt. %, N?0.0080%, O?0.0060%, H?0.0004%, wherein 0.025%?Nb+Ti?0.080%, 0.030%?Al+Ti?0.12%, and the balance of Fe and unavoidable impurities. The method of manufacturing the wear-resistant steel plate comprises the steps of smelting, casting, rolling, post-rolling direct cooling and the like. The wear-resistant steel plate obtained from the above composition and process has high strength, high hardness, good low-temperature toughness, and excellent machinability, and is suitable for quick-wear devices in engineering and mining machinery, such as bucket and scraper transporter, etc.

Abstract: Two-way (full-duplex) wireless links in facilitating network management and improve network performance. Once aspect includes methods for network management using an overlaid network for control signaling. Another aspects include methods to facilitate practical realization and improve performance of some of the network information theoretic configurations, such as Space-Division Multiple Access (SDMA) in uplink and downlink, Interference Channel, and other forms of distributed collaborative signaling schemes. Another aspect includes methods to support cognitive wireless networks.

Abstract: By using the adaptive filter, the reference input signal is processed so as to identify a pseudo-signal of a particular signal to be deleted. The pseudo-signal is subtracted from the mixture containing a target signal inputted from a microphone, the particular signal to be deleted, and a noise so as to obtain an error signal. A stationary noise is estimated to obtain a stationary noise estimated value. A non-stationary noise is estimated to obtain a non-stationary noise estimated value. The stationary noise estimated value is mixed with the non-stationary estimated value to obtain a mixed noise estimated value. An update amount is calculated according to a correlation value between the error signal and the reference input signal, and the mixed noise estimated value. According to the update amount, a coefficient of the adaptive filter is updated.

Abstract: A glasses apparatus and a controlling method thereof, and a display apparatus are provided. The glasses apparatus is configured to operate with the display apparatus, and includes a communicator configured to perform communication with the display apparatus; an osteophony speaker configured to output audio received from the display apparatus; and a controller configured to control the communicator to receive contents audio from the display apparatus if the display apparatus provides a contents image, and to receive call audio from the display apparatus if the display apparatus performs a call function with an external user, and to output one of the received contents audio and call audio to the osteophony speaker.

Abstract: The embodiments of the present disclosure provide a method and an apparatus for reducing an echo, which apply a first stage adaptive filter to a downlink reference signal to obtain a first stage filtered signal and a first error signal; after performing a K-path gain process and a K-path pre-distortion process to the downlink reference signal, apply the at least one second stage adaptive filter to a K-path pre-distorted signal to obtain a second stage filtered signal and a second error signal; and perform a minimum value fusion process to the first error signal and the second error signal so as to obtain a residue signal, which is considered as a final output of adaptive echo cancellation. Accordingly, the present disclosure realizes steadily providing a relatively high echo loss under a circumstance that a speaker suffers a relatively great distortion.

Abstract: The present invention discloses a method and a device for suppressing residual echoes.

Abstract: An audio apparatus including a circuit for applying a dynamic range adjustment processing to an input signal to generate an output signal for rendering by a loudspeaker. The dynamic range adjustment processing is dependent on a set of dynamic range adjustment parameters. A first linear echo-cancellation filter generates a first compensated signal from the input signal and a first adapter determines a set of filter parameters for the first linear echo-cancellation filter in response to the first compensated signal and a microphone signal. An audio circuit generates a second compensated signal by applying the dynamic range adjustment processing and a second echo-cancellation filter to the input signal where the second echo-cancellation filter corresponding to the first echo-cancellation filter. A second adapter determines the set of dynamic range adjustment parameters in response to the second compensated signal and the microphone signal.

Abstract: 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.

Abstract: Audio conferencing systems and methods with mutual adaptation control of adaptive beamforming and adaptive acoustic echo cancellation are disclosed. Mutual adaptation control may be achieved in a system including an adaptive beamforming module, an adaptive acoustic echo cancellation module, and an adaptation control module. The adaptive beamforming module has a controllable beamforming adaptivity and a beamforming adaptation state. The adaptive acoustic echo cancellation module has a controllable AEC adaptivity and an AEC adaptation state. The adaptation control module is configured and/or operates (i) to modify the beamforming adaptivity when the AEC adaptation state is unsettled (adapting to changed conditions), (ii) to modify the AEC adaptivity when the beamforming adaptation state is unsettled (adapting to changed conditions), (iii) to restore the beamforming adaptivity when the AEC adaptation state is settled, and (iv) to restore the AEC adaptivity when the beamforming adaptation state is settled.

Abstract: In an active noise reduction device, in order to solve this problem, a control block determines a level of a reference signal detected by a level detection unit. If determining that the level of the reference signal is small, the control block decreases a level of a cancel signal. This operation suppresses generation of an abnormal sound even if a level of a noise is small.

Abstract: A personal audio device that detects speech provides for improved interaction with others. When speech is detected in a microphone output signal of a microphone that measures ambient audio sounds, the audio program being reproduced by the personal audio device may be altered, by attenuating, muting or interrupting the program material. The speech may be provided to a headset that reproduces the program material. The direction of the speech can be used to determine whether the speech is from a person other than the use of the personal audio device.

Abstract: A sound processing apparatus includes: a first sound collecting unit placed in a sound field and configured to collect a sound signal which is speech of a talker; a second sound collecting unit arranged to be movable to a position which is closer to a talker than the first sound collecting unit and configured to collect the sound signal; a transfer function estimating unit configured to estimate a transfer function from a sound signal collected by the first sound collecting unit and a sound signal collected by the second sound collecting unit when a talker is at a predetermined position in the sound field; and a sound signal processing unit configured to perform a process of the sound signal by use of the transfer function estimated by the transfer function estimating unit.

Abstract: According to the embodiment of the present invention, the stereo echo suppressing device that suppresses an acoustic echo signal based on a near-end input signal and far-end signals of two channels as a stereo signal includes: first and second far-end signal amplitude spectrum calculating units for calculating amplitude spectrums of a first far-end signal and a second far-end signal; a near-end input signal amplitude spectrum calculating unit for calculating an amplitude spectrum of the near-end input signal; first and second estimation echo signal estimating units for estimating amplitude spectrums of a first estimation echo signal and a second estimation echo signal; first and second echo suppressing units for suppressing the acoustic echo signal; first and second echo path characteristic updating unit for calculating and updating echo path characteristics of the first far-end signal and the second far-end signal.

Abstract: System of improving sound quality includes loudspeaker, microphone, accelerometer, acoustic-echo-cancellers (AEC), and double-talk detector (DTD). Loudspeaker outputs loudspeaker signal including downlink audio signal from far-end speaker. Microphone generates microphone uplink signal and receives at least one of: near-end speaker, ambient noise, and loudspeaker signals. Accelerometer generates accelerometer-uplink signal and receives at least one of: near-end speaker, ambient noise, and loudspeaker signals. First AEC receives downlink audio, microphone-uplink and double talk control signals, and generates AEC-microphone linear echo estimate and corrected AEC-microphone uplink signal. Second AEC receives downlink audio, accelerometer uplink and double talk control signals, and generates AEC-accelerometer linear echo estimate and corrected AEC-accelerometer uplink signal.

Abstract: An echo cancellation device including a first adaptive filter for producing a first echo cancellation signal, a second adaptive filter for producing a second echo cancellation signal, and a post-processor for suppressing any remaining echo. The first adaptive filter and the second adaptive filter are designed for canceling a first part of the echo impulse response and a second part of the echo impulse response respectively. The device may be utilized in a mobile telephone.

Abstract: A new audio echo cancellation (AEC) approach is disclosed. To facilitate echo cancellation, the method adjusts for errors (called drift) in sampling rates for both capturing audio and playing audio. This ensures that the AEC module receives both the signals at precisely the same sampling frequency. Furthermore, the far-end signal and near-end mixed signal are time aligned to ensure that the alignment is suitable for application of AEC techniques. An additional enhancement to reduce errors utilizes a concept of native frequency. A by-product of drift compensation allows for excellent buffer control for capture/playback and buffer overflow/underflow errors from drift errors are eliminated.

Abstract: Methods and devices for enabling and disabling the updating of an adaptive filter for echo cancellation of a communication device are disclosed. An echo canceller may include an adaptive filter enabled to update responsive to a determination that a first condition is satisfied that there is no double talk detected on the communication device, and a determination that a second condition is satisfied based, at least in part, on a performance characteristic. The adaptive filter may be disabled from updating responsive to at least one of the first condition and the second condition not being satisfied. Described is disabling the adaptive filter from updating responsive to detecting a performance characteristic of the adaptive filter to fail to meet a condition indicating that the adaptive filter is diverging. The condition to be met may be independent of whether double-talk is determined to be present on the communication device.

Abstract: Provided is a signal processing method which receives a plurality of reception signals and subtracts pseudo-echoes generated by a plurality of adaptive filters which input the reception signals, from a plurality of echoes generated by the reception signals, thereby reducing the echoes. The method delays two or more of the reception signals so as to generate delayed reception signals and inputs the reception signals and the delayed reception signals to the adaptive filters so as to generate pseudo-echoes.

Abstract: Methods, systems, and apparatuses are described for performing acoustic echo cancellation with internal upmixing that allow for a more effective handling of acoustic echo cancellation of audio components that are provided via different channels. In an embodiment in which audio is played back using two loudspeakers, audio components that are panned equally among the loudspeakers form a “phantom center image.” Acoustic echo cancellation is performed by initially upmixing the different channels to internally create modified versions of these channels and a virtual channel representative of the phantom center image. Each of these channels is passed through a respective adaptive filter that is configured to estimate an acoustic echo produced by each respective channel. These estimates are then subtracted from the signal received from one or more microphones (or from a signal obtained by combining multiple microphone signals) to suppress or eliminate the acoustic echo.

Abstract: Methods and devices for updating filter coefficients during echo cancellation are disclosed. A communication device includes a processor configured to operate a background filter configured as an adaptive filter, operate a foreground filter configured as a fixed filter; and update the foreground filter with coefficients from the background filter responsive to monitoring a history of both a near-end signal energy and an estimated far-end signal energy for a pre-determined time window in which conditions of each of the near-end signal energy and the estimated far-end signal energy are satisfied. A related method includes monitoring a near-end signal energy, monitoring an estimated far-end signal energy, performing an echo cancellation, and transferring coefficients from an adaptive filter to a fixed filter responsive to a history of both the near-end signal energy and the estimated far-end signal energy satisfying conditions for a pre-determined time window.

Abstract: Disclosed is a method of processing a signal such as an audio signal. The method includes normalizing a received signal to a level of between ?2 dB and ?10 dB to yield a normalized signal, compressing the normalized signal exceeding a threshold (such as 70% of the maximum volume and above) to yield a compressed signal, performing frequency equalization on the compressed signal to yield an equalized signal; and normalizing the equalized signal.

Abstract: A hearing aid includes: a microphone; a hearing aid processing unit configured to provide a gain to a first signal based on an output signal from the microphone to generate a second signal; a receiver configured to convert the second signal into sound; an adaptive filter configured to adaptively estimate a transfer function corresponding to a pathway from an input side of the receiver to an output side of the microphone; and a feedback removal unit configured to subtract a third signal generated based on the transfer function from the output signal of the microphone to obtain a signal and output the signal as the first signal; and a control unit configured to control the gain setting unit and an adaptive speed of the adaptive filter.

Abstract: An echo canceler includes a signal-to-echo ratio calculating unit 103 and a residual echo suppressing unit 105. The signal-to-echo ratio calculating unit 103 computes a signal-to-echo ratio SE(n) indicating a ratio of an echo component to a received signal x(n) from a first residual signal and a second residual signal. The first residual signal is obtained using a filter coefficient sequence of an update filter 102, which is obtained up to the previous operation. The second residual signal is obtained using an updated filter coefficient sequence that undergoes a coefficient update which is performed, using an arbitrary update step size ?(n), on the filter coefficient sequence of the update filter 102, which is obtained up to the previous operation. The residual echo suppressing unit 105 suppresses the echo component contained in the microphone input signal in accordance with the signal-to-echo ratio the signal-to-echo ratio calculating unit 103 computes.

Abstract: An apparatus for canceling an acoustic echo signal caused by a far-end talker signal is provided. The apparatus for canceling an acoustic echo signal includes: a variance estimating unit configured to estimate a variance of a first audio signal of a near-end talker signal and a first noise signal of the near-end talker signal; a step size determining unit configured to determine a step size by using the variance of the first audio signal and the variance of the first noise signal; an adaptive filter coefficient updating unit configured to update an adaptive filter coefficient of an adaptive filter by using the step size; and an acoustic echo canceling unit configured to estimate an acoustic echo signal by using the adaptive filter coefficient, and cancel the acoustic echo signal from a microphone input signal by using the estimated acoustic echo signal.

Abstract: A multicarrier radio frequency (RF) transceiver implementing dynamic spectrum access (DSA) is capable of sensing a spectral environment and transmitting an RF signal at the same time. The transceiver can cancel an RF transmit signal component within a composite receive signal using a combination of analog and digital cancellation. In one embodiment, analog RF cancellation is performed to reduce the level of a transmit signal component of a composite receive signal before the signal reaches an RF receiver. The RF cancellation may be used to reduce the transmit signal component to a level that maintains the linear operation of the RF receiver. Digital cancellation may be performed to reduce transmit signal remnants within an output signal of the RF receiver. Digital cancellation may be performed on a subcarrier-by-subcarrier basis in some implementations.

Abstract: A receiver architecture and method performs timing recovery in the presence of differential group delay (DGD) (caused, for example, by polarization mode dispersion). A matrix-based linear transformation is applied to the polarization components of a signal received over the optical fiber channel that mitigates or eliminates the effects of the differential group delay. Timing recovery can then be performed on the transformed signal to recover a clock signal. Beneficially, the described technique can recover timing information even in half-baud DGD channels. Furthermore, latency and computational load can be minimized.

Abstract: A beamforming technique for a microphone array is described to attenuate a source of undesired noise that is deemed the most limiting to audio quality in an acoustic environment. Possible sources of undesired noise include echo, background noise (stationary) and other interference signals (non-stationary). The beamforming technique is updated based on long-term evaluations. Once an evaluation occurs and a decision is made, the beamformer adapts with a maximum responsiveness and without intentional delay, and therefore not affecting the beamformer's tracking ability. When fixed beamforming is utilized, one of several fixed beamformers having different attenuation targets are selected to implement noise attenuation. When adaptive beamforming is utilized, the beamformer adapts whenever the selected target is deemed dominant.

Abstract: A system and method for using bi-directional conversation data to improve signal presence detection are disclosed. The detector module is adapted to communicate with a signal enhancement module. The detector module collects data from a transmit direction of the connection and a receive direction of a data connection. The collected data from the transmit and the receive direction is used to classify at least one of data in the transmit direction and data in the receive direction. Responsive to the classification, the signal enhancement module enhances data in one of the transmit direction and the receive direction. Hence, data classification accuracy is improved by using data from both the transmit and receive directions. In one embodiment, the detector module applies a voice activity detection module (VAD) process to detect the presence or absence of voice data in the collected data.

Abstract: An echo path of the echo in a received audio signal is modelled using an adaptive model to determine an adaptive model estimate of the echo. The adaptive model estimate is used to determine an estimate of the echo power of the echo in the received audio signal. The power of the received audio signal is determined. The estimate of the echo power and the determined power of the received audio signal are used to determine echo suppression gains. Temporal smoothing is applied to one or more of the echo suppression gains and the one or more smoothed echo suppression gains are used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal, wherein the amount of smoothing applied to the echo suppression gains is varied according to a non-decreasing function of the frequency of the received audio signal.

Abstract: Systems and methods that reduce or remove idle tones and noise from an audio signal are provided. According to various embodiments, a level of the received input signal is detected and a control signal can be generated based on the detected level. When the detected level is above a pre-determined threshold value, then the input signal (which may have been processed) is output. When the input falls below the pre-determined threshold, then a constant signal is output instead, where the constant signal may be one of a ground signal, or other constant voltage signal.

Abstract: Embodiments are directed towards providing acoustic echo cancellation in a closely-coupled microphone/speaker system. A speaker may produce an audible signal from a reference signal, which may be captured with a microphone. Full band cancellation (FBC) may modify the captured signal to suppress an echo of the reference signal caused by a direct acoustic path between the microphone and speaker. FBC may include a fixed filter and an adaptive filter. The fixed filter may modify the captured signal based on the reference signal. The adaptive filter may automatically adapt based on the captured signal and the reference signal. If a comparison of a performance of the adaptive filter and the fixed filter is above a threshold, then the fixed filter may be updated based on the adaptive filter. Subband acoustic echo cancellation may generate an output signal that suppresses residual echoes of the reference signal based on the modified signal.

Abstract: Systems, methods, and other embodiments associated with echo cancellation are described. According to one embodiment, a method includes generating, in a communication device, an address to a look-up table by using a first signal and a second signal. The look-up table is a two-dimensional look-up table configured to store a plurality of cancellation values arranged in a two-dimensional grid. The method also includes producing, in the communication device, a cancellation signal according to a cancellation value in the look-up table that corresponds to the address.

Abstract: Embodiments of the invention are generally directed to simultaneous transmission of clock and bidirectional data over a communication channel. An embodiment of a transmitting device includes a modulator to generate a modulated signal including a clock signal and a data signal, the clock signal being modulated by a first signal edge of the modulated signal and the data signal being modulated by a position of a second signal edge of the modulated signal; a driver to drive the modulated signal on a communication channel; an echo canceller to subtract reflected signals on the communication channel; and a data recovery module to recover a signal received on the communication channel, the received signal being encoded by Return-to-Zero (RZ) encoding, the signal being received simultaneously with driving the modulated signal on the communication channel.