Abstract: A communication system is disclosed in which a base station communicates with user equipment via an associated cell having a system bandwidth. The system bandwidth comprises a bandwidth part having a set of contiguous physical resource blocks. The base station initiates a random access procedure with the user equipment; provides first information identifying a set of frequency resources for an uplink channel; and provides second information identifying at least one specific frequency resource, within said set of frequency resources, for said uplink communication. The first information is provided prior to completion of the random access procedure.

Abstract: A video conference system including a transmitter device, a first receiver device and a second receiver device is provided. The transmitter device includes a transmitter control unit, a first network interface and a first wireless transmission module. The first receiver device includes a second wireless transmission module, a first receiver input interface and a first receiver video circuit. The second receiver device includes a second network interface and a second receiver video circuit. The first receiver video circuit combines the first video data and the second video data as a first combined video data. The first receiver video circuit transmits the first combined video data to a first display device, and outputs the first combined video data to the transmitter device, which outputs the first combined video data to the second receiver device through the first network interface and the second network interface.

Abstract: A display device provided to a vehicle includes a communication unit which communicates with a mobile device, a control unit which acquires a keyword in a call conversation if the mobile device makes a call, and a display unit which displays additional information corresponding to the keyword. Accordingly, the display device can provide the user with services corresponding to keywords of the call conversation with convenience.

Abstract: A display device provided to a vehicle includes a communication unit which communicates with a mobile device, a control unit which acquires a keyword in a call conversation if the mobile device makes a call, and a display unit which displays additional information corresponding to the keyword. Accordingly, the display device can provide the user with services corresponding to keywords of the call conversation with convenience.

Abstract: Features are disclosed for measuring and correcting clock drift and propagation delay in an audio system through one or more waveforms embedded in an audio signal. A first device in communication with a speaker may be configured to obtain an audio signal and insert one or more waveforms into the audio signal. For example, the waveforms may be inserted during an interval of time. A second device in communication with a microphone may be configured to detect sound as an audio input signal. The second device may obtain a spectral representation of the audio input signal and determine a rotation based on the spectral representation at the frequency of at least one of the inserted waveforms. Clock drift may be determined based on the rotation.

Abstract: In response to a first signal, a first sound wave is output. A second sound wave is received that includes an acoustic echo of the first sound wave. In response to the second sound wave, a second signal is output that cancels an estimate of the acoustic echo. The estimate of the acoustic echo is iteratively adapted to increase a statistical independence between the first and second signals, irrespective of whether a first voice is present in the first sound wave, and irrespective of whether a second voice is present in the second sound wave.

Abstract: An I/O circuit for use with an industrial controller provides an input circuit with low power dissipation through the use of a series connected substantially nonresistance impedance. An AC waveform input activates a light emitting diode (LED) bridge to provide an optical signal which illuminates a photosensitive solid-state switch thereby providing optical isolation between the industrial controller and the controlled process or machine. By using a series connected substantially nonresistance impedance, reduced current demand by the input circuit may be realized, which allows reduced power dissipation. As a result, the same circuit allows receiving input signals over a greater range, such as 120 volts AC and 240 volts AC, and allows significantly smaller components with closer spacing for a more compact design.

Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.

Abstract: System and methods provide acoustic echo monitoring and cancellation for real time media processing in an internet protocol (IP) media server in an IP network. An echo monitor is configured to selectively compare audio streams into and out of the IP media server through a selected port. The comparison determines an occurrence of an echo. An echo canceller in communication with the echo monitor is configured to respond to the determination by the echo monitor so as to remove the echo from at least one of the audio streams. A talk burst detector may be used to detect speech in at least one of the audio streams through the selected port. The echo monitor selectively compares the audio streams in response to a signal from the talk burst detector that indicates detection of speech.

Abstract: The present disclosure relates to systems and methods for noise cancellation of phone conversation. Some of the systems can include a phone having a microphone and a noise cancellation device. In some examples, the noise cancellation device has a microphone for receiving an input sound wave and positioned proximate the phone microphone, a processor for generating a noise cancellation sound wave based on the input sound wave, and a speaker for outputting the noise cancellation sound wave. The phone microphone and the noise cancellation device microphone can be shared or different microphones, and can also be microphone is integral to the phone. The shared or separate microphones can be part of a phone headset.

Abstract: An echo canceler circuit (10) and method attenuates at least post-echo canceler uplink data (90) to produce attenuated uplink data (100) in response to uplink echo return loss based attenuation data (40). The echo canceler circuit (10) includes an echo return loss based attenuation data generator (20) and at least an uplink data attenuator (30). The echo return loss based attenuation data generator (20) produces the uplink echo return loss based attenuation data (40) in response to echo return loss data (70). The echo return loss data (70) is based on at least one of: attenuated downlink data (50), pre-echo canceler uplink data (60), and/or amplifier gain data (80). The uplink data attenuator (30) attenuates the post-echo canceler uplink data (90) to produce attenuated uplink data (100) based on the uplink echo return loss based attenuation data (40).

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

Abstract: A communication terminal includes a decoder which decodes an input bitstream received from another communication terminal, to generate an output audio signal and outputs the generated output audio signal to a speaker; and an echo canceller which obtains an input audio signal representing sound captured by a microphone placed in a space to which the speaker outputs the sound, and removes, for respective subbands, an echo component included in the obtained input audio signal and corresponding to the output audio signal, to generate an audio signal for transmission. An encoder codes the audio signal for transmission to generate an output bitstream and transmits the generated output bitstream to another communication terminal; and a control unit controls, for the respective subbands, echo cancellation processing according to a reproduction band of at least one of the output audio signal and the audio signal for transmission.

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

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 processing an input matrix and a MIMO receiver employing the system or the method. In one embodiment, the system includes: (1) a transformer configured to receive a frame of complex data representing only some elements of an input matrix and perform a fast plane rotation on the complex data to yield rotated data and (2) a matrix updater coupled to the transformer and configured to update a memory configured to contain an output matrix with the rotated data. In one embodiment, the system and method are to estimate and mitigate alien cross-talk experienced in a vectored DSL communication system.

Abstract: A noise suppressing device receives sound signals through a plurality of sound-receiving units and suppresses noise components included in the input sound signals. The noise suppressing device includes a detecting unit which detects a usage pattern of the noise suppressing device from a plurality of usage patterns in which positional relationships of the plurality of sound-receiving units and/or positional relationships between the plurality of sound-receiving units and a target sound source are different from each other, a converting unit which converts using environment information used in a noise suppressing process to each of the sound signals inputted by the plurality of sound-receiving units into using environment information in accordance with a usage pattern detected by the detecting unit and a suppressing unit which performs the noise suppressing process using the using environment information converted by the converting unit to the sound signals.

Abstract: VoIP phones according to the present invention include a microphone, which may be internal or external, and allow the user to communicate unobtrusively, check voice mail and conduct other activities in an environment which can be noisy in general and extremely noisy sometimes. Speech recognition functionally may also be used to generate and send touch tone or DTMF tones such as in response to call trees or voice recognition functionality used by airlines, credit card companies, voice mail systems, and other applications. A system and method of audio processing which provides enhanced speech recognition is provided. Audio input is received at the microphone which is processed by adaptive noise cancellation to generate an enhanced audio signal. The operation of the speech recognition engine and the adaptive noise canceller may be advantageously controlled based on Voice Activity Detection (VAD).

Abstract: A packet voice transceiver adapted to reside at a first end of a communication network and to send an ingress communication signal comprising voice packets to, and receive an egress communication signal comprising voice packets from, a second packet voice transceiver residing at a second end of the communication network. The packet voice transceiver includes a far-end echo canceller that reduces echo that is present in the egress communication signal. The far-end communicates with other functional components of the transceiver system and cancels echo or refrains from canceling echo based on the activity of the other functional components.

Abstract: A system and methods for pre-configuring echo cancellers are shown and described. The system includes a storage device for storing one or more settings of the echo cancellation parameters developed during one or more previously established data connections, and an echo canceller for cancelling echo associated with one or more data connections according to echo cancellation parameters, the echo canceller initially configuring the echo cancellation parameters according to settings developed during previous data connections.

Abstract: Optimal microphone volumes are automatically set for computer applications based on determination of peak volume levels and noise levels from one or more digital audio captures. The peak volume levels and noise levels can be advantageously determined based on distribution curves of sample volume levels in the digital audio captures. Clipping can be automatically compensated for by estimating peak unclipped capture volume levels from the distribution curves.

Abstract: Different sampling rates between a playout unit and a capture unit are compensated for via a system, method and computer program product. The playout unit receives samples from a computational unit, and the capture unit sends samples to the computational unit. A playout FIFO buffer operates in a playout time domain, and a capture FIFO buffer operates in a capture time domain. The computational unit is synchronized to a common clock. A first relationship is calculated between the common clock and a playout fifo buffer read pointer, and a second relationship is calculated between the common clock and a capture FIFO buffer write pointer. For each sample in the playout time domain a corresponding sample in the samples from said computational unit is found and sent to the playout FIFO buffer. For each sample in the common clock time domain the corresponding sample in the capture time domain is found and sent to the computational unit.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.

Abstract: Conferencing products for conducting a conference between a local and two or more distant parties through an electronic channel, wherein distant party audio is routed to the distant parties to provide full audio. These conferencing products are capable of open-air interaction with participants in the locality of a product. Acoustic echo cancellation is applied, which may be accomplished through a single set of coefficients or by a plurality of coefficients and/or echo cancelers on different incoming audio streams and speaker combinations. Privacy modes are provided, whereby one distant party cannot hear the communications of another distant party; those modes including a coaching mode where the privacy is one-way and a mediation mode where the privacy is two-way. Detailed information on various example embodiments of the inventions are provided in the Detailed Description below, and the inventions are defined by the appended claims.

Abstract: A mobile communications device contains at least two microphones. One microphone is designated by a selector to provide a voice dominant signal and another microphone is designated to provide a noise or echo dominant signal, for a call or a recording. The selector communicates the designations to a switch that routes the selected microphone signals to the inputs of a processor for voice signal enhancement. The selected voice dominant signal is then enhanced by suppressing ambient noise or canceling echo therein, based on the selected noise or echo dominant signal. The designation of microphones may change at any instant during the call or recording depending on various factors, e.g. based on the quality of the microphone signals. Other embodiments are also described.

Abstract: A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal.

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: An electronic device comprises a microphone, a transceiver circuit, a loudspeaker, a sense element and a processing circuit. The microphone is configured to receive a first audio signal. The transceiver circuit is configured to communicate the first audio signal to a remote device and to receive a second audio signal from the remote device. The loudspeaker is configured to provide an audible signal based on the second audio signal. The sense element is configured to sense the audible signal provided by the loudspeaker. The sense element may be positioned at a distance from the loudspeaker different than a distance between the microphone and the loudspeaker. The processing circuit is configured to process at least one of the first audio signal and the second audio signal based on a sensed signal from the sense element.

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: An oscillation-echo preventing circuit has a microphone/speaker unit and a voltage-canceling circuit for canceling voltages of audio receive signals. The microphone/speaker unit has a main body, at least two microphones, and a speaker or an earphone. The microphone seals a first inside space from an outside space. The microphone seals the first inside space from a second inside space. The speaker or the earphone seals the first inside space from the outside space. The voltage-canceling circuit cancels out the voltages of audio receive signals coming from the microphones, respectively, generating an output of minimum magnitude. Thus, the circuit can sufficiently suppress oscillation and echoing.

Abstract: An echo canceler circuit (10) and method attenuates at least post-echo canceler uplink data (90) to produce attenuated uplink data (100) in response to uplink echo return loss based attenuation data (40). The echo canceler circuit (10) includes an echo return loss based attenuation data generator (20) and at least an uplink data attenuator (30). The echo return loss based attenuation data generator (20) produces the uplink echo return loss based attenuation data (40) in response to echo return loss data (70). The echo return loss data (70) is based on at least one of: attenuated downlink data (50), pre-echo canceler uplink data (60), and/or amplifier gain data (80). The uplink data attenuator (30) attenuates the post-echo canceler uplink data (90) to produce attenuated uplink data (100) based on the uplink echo return loss based attenuation data (40).

Abstract: Technologies are generally described for a system for measuring a quality of experience (QoE). In some examples, a quality of experience (QoE) measuring device may include a background noise detecting unit configured to measure a background noise around a near-end talker, and a decision unit configured to determine whether a double talk event detected by a double talk detector is caused by the background noise around the near-end talker.

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: 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: Disclosed herein are portable teleconferencing products that implement a doubletalk detector in a low frequency range or in a frequency range commensurate with the frequencies through which sound may be efficiently transferred between a speaker and a microphone through an enclosure. Also disclosed herein are teleconferencing systems that implement a secondary doubletalk detector, a non-presumptive doubletalk detector, a confirmatory doubletalk detector, and/or a false doubletalk detector, whereby echo cancellation coefficients may be better adapted after echo path changing events through the use of accelerated coefficient adaptation or half-duplex operation until adaptation is restored.

Abstract: Adaptive filters employing a normalized time domain least mean square algorithm having enhanced convergence rates by virtue of the use of an update gain greater than 2 ? max ? ? or ? ? 2 3 ? ? tr ? [ R ] .

Abstract: A method to design low complexity and low power echo and NEXT cancellers based on wordlength reduction technique is presented. A circuit architecture to implement echo and cancellers is also presented. The low complexity and low power design relies on the fact that a TH precoder can be viewed as an IIR filter with an input equal to the sum of the original input to the TH precoder and a compensation signal. The proposed design also relies on the fact that sum of the original input to the TH precoder and the compensation signal has finite levels, which can be represented in less bits than the original input of the echo and NEXT cancellers. An improved design by exploiting the statistics of the compensation signal is also proposed to further bring down the complexity and power consumption of these cancellers.

Abstract: A method and system for rapid adaptive coupling equalization in beamforming-based communication systems, particularly sector-based beamforming systems, provides smooth transitions for AEC when the look direction of the communication system changes and when the acoustic environment varies with time. The coefficients of inactive beamformers are modified in real-time, using adaptive beamforming techniques based on the real-time loudspeaker-coupling signal, in order to force the outputs of inactive beamformers to have the same response to the loudspeaker coupling signal as the active beamformer does.

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: An echo canceller (106) can include a first multi-band filter (152) which receives a first input signal (108) and generates a first plurality of sub-band signals (110, 111, 112), and a second multi-band filter (154) which receives a second input signal (122) and generates a second plurality of sub-band signals (156, 157, 158). The echo canceller also can include a plurality of double talk detectors (168, 169, 170) that each generate a double talk flag (182, 183, 184) based on at least a respective one of the first sub-band signals and a respective one of the second sub-band signals.

Abstract: A field-mounted process device with multiple isolated connections includes a connection that can be an input or an output. The given input or output can couple to multiple sensors or actuators, respectively. The process device can be wholly powered through its communication I/O port. The process device includes a controller adapted to measure one or more characteristics of sensors coupled to an input connection and to control actuators coupled to an output connection. The controller can be further adapted to execute a user generated control algorithm relating process input information with process output commands.

Abstract: A method for creating a form of a non-linear filter suitable for reducing a computational complexity is proposed. The filter is resolved into polyphase components in such a way that the polyphase components can be interchanged with a conversion of the sampling rate of a signal to be sent to the filter or of a signal to be emitted by the filter. Corresponding filters and filter arrangements are also proposed. In this way, a computational complexity for calculating the signal to be emitted by the filter can be significantly simplified. The invention can be used in echo compensation.

Abstract: Bridge circuit for echo suppression for a reception signal of a communication device connected to a transmission line, to which can be supplied a reception signal received via the transmission line and a transmission signal to be transmitted by the communication device via the transmission line. The bridge circuit is designed such that on balancing the bridge circuit, the reception signal there can be tapped echo-compensated with a transmission signal part generated by the transmission signal being suppressed. To balance the bridge circuit a variable simulation device is provided to simulate at least one circuit section of at least one bridge branch and connect with the at least one bridge branch. By such an arrangement, the signal level in the simulation device may be reduced so far that this can be designed as an integrated circuit and to be programmable.

Abstract: Acoustic echo cancellation, residual echo suppression and sound-source localization/microphone array processes are combined in a two-way voice communication system that uses a microphone array to capture local speech. The processes can be configured according to various alternative architectures and enhancements made to the processes to avoid the adverse effects of non-linear operations in the residual echo suppression on the sound-source localization/microphone array process.

Abstract: A method to reduce acoustic coupling in an audio conferencing system having a loudspeaker and a plurality of microphones, comprising detecting the presence of one of either a source of near-end signal or a source of far-end signal source relative to the audio conferencing system, and in the event of detecting a source of near-end signal then processing the near-end signal with high directivity towards the near-end source, and in the event of detecting a source of far-end signal or double-talk then processing the far-end signal with high coupling characteristics between the loudspeaker and microphones.

Abstract: A system and method for adaptive estimation and compensation of clock drift in echo cancellers is provided. The invention includes an acoustic echo cancellation system with a built in adaptive clock drift compensation system. The acoustic echo cancellation system has an AEC component that performs acoustic echo cancellation on data from a capture buffer, by also using information derived from a render buffer. The clock drift compensation system has access to this capture buffer and render buffer. The clock drift compensation system includes a clock drift compensator that calculates, based on the current location of the capture data being processed by the AEC component as well as additional information, the ideal location in the render buffer from which the AEC component should process data. The clock drift compensator further adjusts the current location in the render buffer from which the AEC component processes data based, at least in part, upon this ideal location.

Abstract: The present invention provides a wireless headset with echo control and noise cancellation. The present invention also provides a method with phase reversion for echo control of a wireless headset wherein the wireless headset comprises closely disposed speakers and acoustic sensors. The present invention further provides a method with beamforming for noise cancellation of a wireless headset wherein the wireless headset comprises two separate units disposed in distance, and wherein each unit comprises an acoustic sensor.

Abstract: An embodiment of the present invention includes an adaptive predictor, a system white noise generator, and a background noise estimator. The adaptive predictor estimates adaptive weights of autoregressive (AR) model of background noise as background samples in an echo canceler. The adaptive predictor generates adaptive error. The system white noise generator generates a white noise using the adaptive error. The noise background estimator estimates the background noise using the white noise and the estimated adaptive weights.